webrtc_m130/media/engine/webrtc_video_engine_unittest.cc

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/*
* Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "media/engine/webrtc_video_engine.h"
#include <map>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_parameters.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/mock_video_bitrate_allocator.h"
#include "api/test/mock_video_bitrate_allocator_factory.h"
#include "api/test/mock_video_decoder_factory.h"
#include "api/test/mock_video_encoder_factory.h"
#include "api/transport/field_trial_based_config.h"
#include "api/transport/media/media_transport_config.h"
#include "api/units/time_delta.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video/i420_buffer.h"
#include "api/video/video_bitrate_allocation.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "call/flexfec_receive_stream.h"
#include "common_video/h264/profile_level_id.h"
#include "media/base/fake_frame_source.h"
#include "media/base/fake_network_interface.h"
#include "media/base/fake_video_renderer.h"
#include "media/base/media_constants.h"
#include "media/base/rtp_utils.h"
#include "media/base/test_utils.h"
#include "media/engine/constants.h"
#include "media/engine/fake_webrtc_call.h"
#include "media/engine/fake_webrtc_video_engine.h"
#include "media/engine/simulcast.h"
#include "media/engine/webrtc_voice_engine.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/experiments/min_video_bitrate_experiment.h"
#include "rtc_base/fake_clock.h"
#include "rtc_base/gunit.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/time_utils.h"
#include "test/field_trial.h"
#include "test/frame_generator.h"
#include "test/gmock.h"
#include "test/rtp_header_parser.h"
using ::testing::Contains;
using ::testing::Eq;
using ::testing::Field;
using ::testing::IsEmpty;
using ::testing::Pair;
using ::testing::SizeIs;
using webrtc::BitrateConstraints;
using webrtc::RtpExtension;
namespace {
static const int kDefaultQpMax = 56;
static const uint8_t kRedRtxPayloadType = 125;
static const uint32_t kTimeout = 5000U;
static const uint32_t kDefaultReceiveSsrc = 0;
static const uint32_t kSsrc = 1234u;
static const uint32_t kSsrcs4[] = {1, 2, 3, 4};
static const int kVideoWidth = 640;
static const int kVideoHeight = 360;
static const int kFramerate = 30;
static const uint32_t kSsrcs1[] = {1};
static const uint32_t kSsrcs3[] = {1, 2, 3};
static const uint32_t kRtxSsrcs1[] = {4};
static const uint32_t kFlexfecSsrc = 5;
static const uint32_t kIncomingUnsignalledSsrc = 0xC0FFEE;
static const uint32_t kDefaultRecvSsrc = 0;
Reland "Only include payload in bytes sent/received." This is a reland of 74a1b4b1321b426392d4c32e4a02361226ad5358 Original change's description: > Only include payload in bytes sent/received. > > According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and > https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent > statistic should not include headers or padding. > > Similarly, according to > https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes > received are calculated the same way as bytes sent (eg. not including > padding or headers). > > This change stops adding padding and headers to these statistics. > > Bug: webrtc:8516,webrtc:10525 > Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Bjorn Mellem <mellem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28647} Bug: webrtc:8516, webrtc:10525 Change-Id: Iaa1613e5becdfaa0af0f6b9f00e5b871937a719c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147520 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28731}
2019-07-30 08:34:03 -07:00
constexpr uint32_t kRtpHeaderSize = 12;
static const char kUnsupportedExtensionName[] =
"urn:ietf:params:rtp-hdrext:unsupported";
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
cricket::VideoCodec RemoveFeedbackParams(cricket::VideoCodec&& codec) {
codec.feedback_params = cricket::FeedbackParams();
return std::move(codec);
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
}
void VerifyCodecHasDefaultFeedbackParams(const cricket::VideoCodec& codec,
bool lntf_expected) {
EXPECT_EQ(lntf_expected,
codec.HasFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamLntf, cricket::kParamValueEmpty)));
EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)));
EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamNack, cricket::kRtcpFbNackParamPli)));
EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamRemb, cricket::kParamValueEmpty)));
EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamTransportCc, cricket::kParamValueEmpty)));
EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamCcm, cricket::kRtcpFbCcmParamFir)));
}
// Return true if any codec in |codecs| is an RTX codec with associated payload
// type |payload_type|.
bool HasRtxCodec(const std::vector<cricket::VideoCodec>& codecs,
int payload_type) {
for (const cricket::VideoCodec& codec : codecs) {
int associated_payload_type;
if (absl::EqualsIgnoreCase(codec.name.c_str(), "rtx") &&
codec.GetParam(cricket::kCodecParamAssociatedPayloadType,
&associated_payload_type) &&
associated_payload_type == payload_type) {
return true;
}
}
return false;
}
Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ ) Reason for revert: Identified a configuration problem in the video quality tests. Intend to fix and reland. Original issue's description: > Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ ) > > Reason for revert: > This change appears to break ulpfec, with severe regressions, e.g., for webrtc_perf_test FullStackTest.ForemanCifPlr5Ulpfec > > Original issue's description: > > Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ ) > > > > Reason for revert: > > Intend to fix perf failures and reland. > > > > Original issue's description: > > > Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ ) > > > > > > Reason for revert: > > > A few perf tests broken, including > > > > > > RampUpTest.UpDownUpAbsSendTimeSimulcastRedRtx > > > RampUpTest.UpDownUpTransportSequenceNumberRtx > > > RampUpTest.UpDownUpTransportSequenceNumberPacketLoss > > > > > > > > > Original issue's description: > > > > Use RtxReceiveStream. > > > > > > > > This also has the beneficial side-effect that when a media stream > > > > which is protected by FlexFEC receives an RTX retransmission, the > > > > retransmitted media packet is passed into the FlexFEC machinery, > > > > which should improve its ability to recover packets via FEC. > > > > > > > > BUG=webrtc:7135 > > > > > > > > Review-Url: https://codereview.webrtc.org/3008773002 > > > > Cr-Commit-Position: refs/heads/master@{#19649} > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/5c0f6c62ea3b1d2c43f8fc152961af27033475f7 > > > > > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > > > # Skipping CQ checks because original CL landed less than 1 days ago. > > > NOPRESUBMIT=true > > > NOTREECHECKS=true > > > NOTRY=true > > > BUG=webrtc:7135 > > > > > > Review-Url: https://codereview.webrtc.org/3010983002 > > > Cr-Commit-Position: refs/heads/master@{#19653} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/3c39c0137afa274d1d524b150b50304b38a2847b > > > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > > # Skipping CQ checks because original CL landed less than 1 days ago. > > NOPRESUBMIT=true > > NOTREECHECKS=true > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/3006063002 > > Cr-Commit-Position: refs/heads/master@{#19715} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35713eaf565c0fef07c8afc158d7b8fdf7ec3d78 > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/3007303002 > Cr-Commit-Position: refs/heads/master@{#19744} > Committed: https://chromium.googlesource.com/external/webrtc/+/8e7eee035178a7f10e19883681b5eaa4a7523107 TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/3012963002 Cr-Commit-Position: refs/heads/master@{#19765}
2017-09-11 02:32:16 -07:00
// TODO(nisse): Duplicated in call.cc.
const int* FindKeyByValue(const std::map<int, int>& m, int v) {
for (const auto& kv : m) {
if (kv.second == v)
return &kv.first;
}
return nullptr;
}
bool HasRtxReceiveAssociation(const webrtc::VideoReceiveStream::Config& config,
int payload_type) {
return FindKeyByValue(config.rtp.rtx_associated_payload_types,
payload_type) != nullptr;
}
// Check that there's an Rtx payload type for each decoder.
bool VerifyRtxReceiveAssociations(
const webrtc::VideoReceiveStream::Config& config) {
for (const auto& decoder : config.decoders) {
if (!HasRtxReceiveAssociation(config, decoder.payload_type))
return false;
}
return true;
}
rtc::scoped_refptr<webrtc::VideoFrameBuffer> CreateBlackFrameBuffer(
int width,
int height) {
rtc::scoped_refptr<webrtc::I420Buffer> buffer =
webrtc::I420Buffer::Create(width, height);
webrtc::I420Buffer::SetBlack(buffer);
return buffer;
}
void VerifySendStreamHasRtxTypes(const webrtc::VideoSendStream::Config& config,
const std::map<int, int>& rtx_types) {
std::map<int, int>::const_iterator it;
Reland "Reland "Move rtp-specific config out of EncoderSettings."" This reverts commit 6c2c13af06b32778b86950681758a7970d1c5d9e. Reason for revert: Intend to investigate and fix perf problems. Original change's description: > Revert "Reland "Move rtp-specific config out of EncoderSettings."" > > This reverts commit 04dd1768625eb2241d1fb97fd0137897e703e266. > > Reason for revert: Regression in ramp up perf tests. > > Original change's description: > > Reland "Move rtp-specific config out of EncoderSettings." > > > > This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c > > > > Original change's description: > > > Move rtp-specific config out of EncoderSettings. > > > > > > In VideoSendStream::Config, move payload_name and payload_type from > > > EncoderSettings to Rtp. > > > > > > EncoderSettings now contains configuration for VideoStreamEncoder only, > > > and should perhaps be renamed in a follow up cl. It's no longer > > > passed as an argument to VideoCodecInitializer::SetupCodec. > > > > > > The latter then needs a different way to know the codec type, > > > which is provided by a new codec_type member in VideoEncoderConfig. > > > > > > Bug: webrtc:8830 > > > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6 > > > Reviewed-on: https://webrtc-review.googlesource.com/62062 > > > Commit-Queue: Niels Moller <nisse@webrtc.org> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#22532} > > > > Bug: webrtc:8830 > > Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019 > > Reviewed-on: https://webrtc-review.googlesource.com/63721 > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Commit-Queue: Niels Moller <nisse@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22595} > > TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org > > Bug: webrtc:8830,chromium:827080 > Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef > Reviewed-on: https://webrtc-review.googlesource.com/65520 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22677} TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8830, chromium:827080 Change-Id: I9b62987bf5daced90dfeb3ebb6739c80117c487f Reviewed-on: https://webrtc-review.googlesource.com/66862 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22751}
2018-04-05 15:36:51 +02:00
it = rtx_types.find(config.rtp.payload_type);
EXPECT_TRUE(it != rtx_types.end() &&
it->second == config.rtp.rtx.payload_type);
if (config.rtp.ulpfec.red_rtx_payload_type != -1) {
it = rtx_types.find(config.rtp.ulpfec.red_payload_type);
EXPECT_TRUE(it != rtx_types.end() &&
it->second == config.rtp.ulpfec.red_rtx_payload_type);
}
}
Reland of Drop frames until specified bitrate is achieved. (patchset #1 id:1 of https://codereview.webrtc.org/2666303002/ ) Reason for revert: Perf test broke as it made assumptions that quality scaling was turned off. This turns out not to be the case. Fixed by turning quality scaling off for the tests. Original issue's description: > Revert of Drop frames until specified bitrate is achieved. (patchset #12 id:240001 of https://codereview.webrtc.org/2630333002/ ) > > Reason for revert: > due to failures on perf tests (not on perf stats, but fails running due to dcheck failures), see e.g., https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20(K%20Nexus5) > > Original issue's description: > > Drop frames until specified bitrate is achieved. > > > > This CL fixes a regression introduced with the new quality scaler > > where the video would no longer start in a scaled mode. This CL adds > > code that compares incoming captured frames to the target bitrate, > > and if they are found to be too large, they are dropped and sinkWants > > set to a lower resolution. The number of dropped frames should be low > > (0-4 in most cases) and should not introduce a noticeable delay, or > > at least should be preferrable to having the first 2-4 seconds of video > > have very low quality. > > > > BUG=webrtc:6953 > > > > Review-Url: https://codereview.webrtc.org/2630333002 > > Cr-Commit-Position: refs/heads/master@{#16391} > > Committed: https://chromium.googlesource.com/external/webrtc/+/83399caec5762d2dad038b8e9d86163e92c18c9f > > TBR=perkj@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,kthelgason@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6953 > > Review-Url: https://codereview.webrtc.org/2666303002 > Cr-Commit-Position: refs/heads/master@{#16395} > Committed: https://chromium.googlesource.com/external/webrtc/+/35fc2aa82fb5a562f3f76f2b91a55f05ebfd4874 TBR=perkj@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,minyue@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:6953 Review-Url: https://codereview.webrtc.org/2675223002 Cr-Commit-Position: refs/heads/master@{#16473}
2017-02-07 07:02:22 -08:00
cricket::MediaConfig GetMediaConfig() {
cricket::MediaConfig media_config;
media_config.video.enable_cpu_adaptation = false;
Reland of Drop frames until specified bitrate is achieved. (patchset #1 id:1 of https://codereview.webrtc.org/2666303002/ ) Reason for revert: Perf test broke as it made assumptions that quality scaling was turned off. This turns out not to be the case. Fixed by turning quality scaling off for the tests. Original issue's description: > Revert of Drop frames until specified bitrate is achieved. (patchset #12 id:240001 of https://codereview.webrtc.org/2630333002/ ) > > Reason for revert: > due to failures on perf tests (not on perf stats, but fails running due to dcheck failures), see e.g., https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20(K%20Nexus5) > > Original issue's description: > > Drop frames until specified bitrate is achieved. > > > > This CL fixes a regression introduced with the new quality scaler > > where the video would no longer start in a scaled mode. This CL adds > > code that compares incoming captured frames to the target bitrate, > > and if they are found to be too large, they are dropped and sinkWants > > set to a lower resolution. The number of dropped frames should be low > > (0-4 in most cases) and should not introduce a noticeable delay, or > > at least should be preferrable to having the first 2-4 seconds of video > > have very low quality. > > > > BUG=webrtc:6953 > > > > Review-Url: https://codereview.webrtc.org/2630333002 > > Cr-Commit-Position: refs/heads/master@{#16391} > > Committed: https://chromium.googlesource.com/external/webrtc/+/83399caec5762d2dad038b8e9d86163e92c18c9f > > TBR=perkj@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,kthelgason@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6953 > > Review-Url: https://codereview.webrtc.org/2666303002 > Cr-Commit-Position: refs/heads/master@{#16395} > Committed: https://chromium.googlesource.com/external/webrtc/+/35fc2aa82fb5a562f3f76f2b91a55f05ebfd4874 TBR=perkj@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,minyue@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:6953 Review-Url: https://codereview.webrtc.org/2675223002 Cr-Commit-Position: refs/heads/master@{#16473}
2017-02-07 07:02:22 -08:00
return media_config;
}
// Values from GetMaxDefaultVideoBitrateKbps in webrtcvideoengine.cc.
int GetMaxDefaultBitrateBps(size_t width, size_t height) {
if (width * height <= 320 * 240) {
return 600000;
} else if (width * height <= 640 * 480) {
return 1700000;
} else if (width * height <= 960 * 540) {
return 2000000;
} else {
return 2500000;
}
}
class MockVideoSource : public rtc::VideoSourceInterface<webrtc::VideoFrame> {
public:
MOCK_METHOD2(AddOrUpdateSink,
void(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
const rtc::VideoSinkWants& wants));
MOCK_METHOD1(RemoveSink,
void(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink));
};
} // namespace
#define EXPECT_FRAME_WAIT(c, w, h, t) \
EXPECT_EQ_WAIT((c), renderer_.num_rendered_frames(), (t)); \
EXPECT_EQ((w), renderer_.width()); \
EXPECT_EQ((h), renderer_.height()); \
EXPECT_EQ(0, renderer_.errors());
#define EXPECT_FRAME_ON_RENDERER_WAIT(r, c, w, h, t) \
EXPECT_EQ_WAIT((c), (r).num_rendered_frames(), (t)); \
EXPECT_EQ((w), (r).width()); \
EXPECT_EQ((h), (r).height()); \
EXPECT_EQ(0, (r).errors());
namespace cricket {
class WebRtcVideoEngineTest : public ::testing::Test {
public:
WebRtcVideoEngineTest() : WebRtcVideoEngineTest("") {}
explicit WebRtcVideoEngineTest(const std::string& field_trials)
: override_field_trials_(
field_trials.empty()
? nullptr
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
: std::make_unique<webrtc::test::ScopedFieldTrials>(
field_trials)),
task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()),
call_(webrtc::Call::Create([&] {
webrtc::Call::Config call_config(&event_log_);
call_config.task_queue_factory = task_queue_factory_.get();
call_config.trials = &field_trials_;
return call_config;
}())),
encoder_factory_(new cricket::FakeWebRtcVideoEncoderFactory),
decoder_factory_(new cricket::FakeWebRtcVideoDecoderFactory),
video_bitrate_allocator_factory_(
webrtc::CreateBuiltinVideoBitrateAllocatorFactory()),
engine_(std::unique_ptr<cricket::FakeWebRtcVideoEncoderFactory>(
encoder_factory_),
std::unique_ptr<cricket::FakeWebRtcVideoDecoderFactory>(
decoder_factory_)) {
// Ensure fake clock doesn't return 0, which will cause some initializations
// fail inside RTP senders.
fake_clock_.AdvanceTime(webrtc::TimeDelta::us(1));
}
protected:
void AssignDefaultAptRtxTypes();
void AssignDefaultCodec();
// Find the index of the codec in the engine with the given name. The codec
// must be present.
size_t GetEngineCodecIndex(const std::string& name) const;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
// Find the codec in the engine with the given name. The codec must be
// present.
cricket::VideoCodec GetEngineCodec(const std::string& name) const;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
VideoMediaChannel* SetSendParamsWithAllSupportedCodecs();
VideoMediaChannel* SetRecvParamsWithSupportedCodecs(
const std::vector<VideoCodec>& codecs);
void ExpectRtpCapabilitySupport(const char* uri, bool supported) const;
// Has to be the first one, so it is initialized before the call or there is a
// race condition in the clock access.
rtc::ScopedFakeClock fake_clock_;
std::unique_ptr<webrtc::test::ScopedFieldTrials> override_field_trials_;
webrtc::FieldTrialBasedConfig field_trials_;
webrtc::RtcEventLogNull event_log_;
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
// Used in WebRtcVideoEngineVoiceTest, but defined here so it's properly
// initialized when the constructor is called.
std::unique_ptr<webrtc::Call> call_;
cricket::FakeWebRtcVideoEncoderFactory* encoder_factory_;
cricket::FakeWebRtcVideoDecoderFactory* decoder_factory_;
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory_;
WebRtcVideoEngine engine_;
VideoCodec default_codec_;
std::map<int, int> default_apt_rtx_types_;
};
TEST_F(WebRtcVideoEngineTest, DefaultRtxCodecHasAssociatedPayloadTypeSet) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
AssignDefaultCodec();
std::vector<VideoCodec> engine_codecs = engine_.codecs();
for (size_t i = 0; i < engine_codecs.size(); ++i) {
if (engine_codecs[i].name != kRtxCodecName)
continue;
int associated_payload_type;
EXPECT_TRUE(engine_codecs[i].GetParam(kCodecParamAssociatedPayloadType,
&associated_payload_type));
EXPECT_EQ(default_codec_.id, associated_payload_type);
return;
}
FAIL() << "No RTX codec found among default codecs.";
}
TEST_F(WebRtcVideoEngineTest, SupportsTimestampOffsetHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kTimestampOffsetUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsAbsoluteSenderTimeHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kAbsSendTimeUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsTransportSequenceNumberHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kTransportSequenceNumberUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsVideoRotationHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kVideoRotationUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsPlayoutDelayHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kPlayoutDelayUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsVideoContentTypeHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kVideoContentTypeUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsVideoTimingHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kVideoTimingUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsFrameMarkingHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kFrameMarkingUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsColorSpaceHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kColorSpaceUri, true);
}
TEST_F(WebRtcVideoEngineTest, AdvertiseGenericDescriptor00) {
ExpectRtpCapabilitySupport(RtpExtension::kGenericFrameDescriptorUri00, false);
}
TEST_F(WebRtcVideoEngineTest, AdvertiseGenericDescriptor01) {
ExpectRtpCapabilitySupport(RtpExtension::kGenericFrameDescriptorUri01, false);
}
class WebRtcVideoEngineTestWithGenericDescriptor
: public WebRtcVideoEngineTest {
public:
WebRtcVideoEngineTestWithGenericDescriptor()
: WebRtcVideoEngineTest("WebRTC-GenericDescriptorAdvertised/Enabled/") {}
};
TEST_F(WebRtcVideoEngineTestWithGenericDescriptor,
AdvertiseGenericDescriptor00) {
ExpectRtpCapabilitySupport(RtpExtension::kGenericFrameDescriptorUri00, true);
}
TEST_F(WebRtcVideoEngineTestWithGenericDescriptor,
AdvertiseGenericDescriptor01) {
ExpectRtpCapabilitySupport(RtpExtension::kGenericFrameDescriptorUri01, true);
}
TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionBeforeCapturer) {
// Allocate the source first to prevent early destruction before channel's
// dtor is called.
::testing::NiceMock<MockVideoSource> video_source;
encoder_factory_->AddSupportedVideoCodecType("VP8");
std::unique_ptr<VideoMediaChannel> channel(
SetSendParamsWithAllSupportedCodecs());
EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc)));
// Add CVO extension.
const int id = 1;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.extensions.push_back(
RtpExtension(RtpExtension::kVideoRotationUri, id));
EXPECT_TRUE(channel->SetSendParameters(parameters));
EXPECT_CALL(
video_source,
AddOrUpdateSink(::testing::_,
Field(&rtc::VideoSinkWants::rotation_applied, false)));
// Set capturer.
EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &video_source));
// Verify capturer has turned off applying rotation.
::testing::Mock::VerifyAndClear(&video_source);
// Verify removing header extension turns on applying rotation.
parameters.extensions.clear();
EXPECT_CALL(
video_source,
AddOrUpdateSink(::testing::_,
Field(&rtc::VideoSinkWants::rotation_applied, true)));
EXPECT_TRUE(channel->SetSendParameters(parameters));
}
TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionBeforeAddSendStream) {
// Allocate the source first to prevent early destruction before channel's
// dtor is called.
::testing::NiceMock<MockVideoSource> video_source;
encoder_factory_->AddSupportedVideoCodecType("VP8");
std::unique_ptr<VideoMediaChannel> channel(
SetSendParamsWithAllSupportedCodecs());
// Add CVO extension.
const int id = 1;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.extensions.push_back(
RtpExtension(RtpExtension::kVideoRotationUri, id));
EXPECT_TRUE(channel->SetSendParameters(parameters));
EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc)));
// Set source.
EXPECT_CALL(
video_source,
AddOrUpdateSink(::testing::_,
Field(&rtc::VideoSinkWants::rotation_applied, false)));
EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &video_source));
}
TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionAfterCapturer) {
::testing::NiceMock<MockVideoSource> video_source;
encoder_factory_->AddSupportedVideoCodecType("VP8");
encoder_factory_->AddSupportedVideoCodecType("VP9");
std::unique_ptr<VideoMediaChannel> channel(
SetSendParamsWithAllSupportedCodecs());
EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc)));
// Set capturer.
EXPECT_CALL(
video_source,
AddOrUpdateSink(::testing::_,
Field(&rtc::VideoSinkWants::rotation_applied, true)));
EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &video_source));
// Verify capturer has turned on applying rotation.
::testing::Mock::VerifyAndClear(&video_source);
// Add CVO extension.
const int id = 1;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
parameters.extensions.push_back(
RtpExtension(RtpExtension::kVideoRotationUri, id));
// Also remove the first codec to trigger a codec change as well.
parameters.codecs.erase(parameters.codecs.begin());
EXPECT_CALL(
video_source,
AddOrUpdateSink(::testing::_,
Field(&rtc::VideoSinkWants::rotation_applied, false)));
EXPECT_TRUE(channel->SetSendParameters(parameters));
// Verify capturer has turned off applying rotation.
::testing::Mock::VerifyAndClear(&video_source);
// Verify removing header extension turns on applying rotation.
parameters.extensions.clear();
EXPECT_CALL(
video_source,
AddOrUpdateSink(::testing::_,
Field(&rtc::VideoSinkWants::rotation_applied, true)));
EXPECT_TRUE(channel->SetSendParameters(parameters));
}
TEST_F(WebRtcVideoEngineTest, SetSendFailsBeforeSettingCodecs) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(123)));
EXPECT_FALSE(channel->SetSend(true))
<< "Channel should not start without codecs.";
EXPECT_TRUE(channel->SetSend(false))
<< "Channel should be stoppable even without set codecs.";
}
TEST_F(WebRtcVideoEngineTest, GetStatsWithoutSendCodecsSetDoesNotCrash) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(123)));
VideoMediaInfo info;
channel->GetStats(&info);
}
TEST_F(WebRtcVideoEngineTest, UseFactoryForVp8WhenSupported) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
std::unique_ptr<VideoMediaChannel> channel(
SetSendParamsWithAllSupportedCodecs());
channel->OnReadyToSend(true);
EXPECT_TRUE(
channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
EXPECT_EQ(0, encoder_factory_->GetNumCreatedEncoders());
EXPECT_TRUE(channel->SetSend(true));
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
// Sending one frame will have allocate the encoder.
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1));
EXPECT_TRUE_WAIT(encoder_factory_->encoders()[0]->GetNumEncodedFrames() > 0,
kTimeout);
int num_created_encoders = encoder_factory_->GetNumCreatedEncoders();
EXPECT_EQ(num_created_encoders, 1);
// Setting codecs of the same type should not reallocate any encoders
// (expecting a no-op).
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel->SetSendParameters(parameters));
EXPECT_EQ(num_created_encoders, encoder_factory_->GetNumCreatedEncoders());
// Remove stream previously added to free the external encoder instance.
EXPECT_TRUE(channel->RemoveSendStream(kSsrc));
EXPECT_EQ(0u, encoder_factory_->encoders().size());
}
// Test that when an encoder factory supports H264, we add an RTX
// codec for it.
// TODO(deadbeef): This test should be updated if/when we start
// adding RTX codecs for unrecognized codec names.
TEST_F(WebRtcVideoEngineTest, RtxCodecAddedForH264Codec) {
using webrtc::H264::kLevel1;
using webrtc::H264::ProfileLevelId;
using webrtc::H264::ProfileLevelIdToString;
webrtc::SdpVideoFormat h264_constrained_baseline("H264");
h264_constrained_baseline.parameters[kH264FmtpProfileLevelId] =
*ProfileLevelIdToString(
ProfileLevelId(webrtc::H264::kProfileConstrainedBaseline, kLevel1));
webrtc::SdpVideoFormat h264_constrained_high("H264");
h264_constrained_high.parameters[kH264FmtpProfileLevelId] =
*ProfileLevelIdToString(
ProfileLevelId(webrtc::H264::kProfileConstrainedHigh, kLevel1));
webrtc::SdpVideoFormat h264_high("H264");
h264_high.parameters[kH264FmtpProfileLevelId] = *ProfileLevelIdToString(
ProfileLevelId(webrtc::H264::kProfileHigh, kLevel1));
encoder_factory_->AddSupportedVideoCodec(h264_constrained_baseline);
encoder_factory_->AddSupportedVideoCodec(h264_constrained_high);
encoder_factory_->AddSupportedVideoCodec(h264_high);
// First figure out what payload types the test codecs got assigned.
const std::vector<cricket::VideoCodec> codecs = engine_.codecs();
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
// Now search for RTX codecs for them. Expect that they all have associated
// RTX codecs.
EXPECT_TRUE(HasRtxCodec(
codecs,
FindMatchingCodec(codecs, cricket::VideoCodec(h264_constrained_baseline))
->id));
EXPECT_TRUE(HasRtxCodec(
codecs,
FindMatchingCodec(codecs, cricket::VideoCodec(h264_constrained_high))
->id));
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
EXPECT_TRUE(HasRtxCodec(
codecs, FindMatchingCodec(codecs, cricket::VideoCodec(h264_high))->id));
}
#if defined(RTC_ENABLE_VP9)
TEST_F(WebRtcVideoEngineTest, CanConstructDecoderForVp9EncoderFactory) {
encoder_factory_->AddSupportedVideoCodecType("VP9");
std::unique_ptr<VideoMediaChannel> channel(
SetSendParamsWithAllSupportedCodecs());
EXPECT_TRUE(
channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
}
#endif // defined(RTC_ENABLE_VP9)
TEST_F(WebRtcVideoEngineTest, PropagatesInputFrameTimestamp) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
FakeCall* fake_call = new FakeCall();
call_.reset(fake_call);
std::unique_ptr<VideoMediaChannel> channel(
SetSendParamsWithAllSupportedCodecs());
EXPECT_TRUE(
channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 60);
EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &frame_forwarder));
channel->SetSend(true);
FakeVideoSendStream* stream = fake_call->GetVideoSendStreams()[0];
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
int64_t last_timestamp = stream->GetLastTimestamp();
for (int i = 0; i < 10; i++) {
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
int64_t timestamp = stream->GetLastTimestamp();
int64_t interval = timestamp - last_timestamp;
// Precision changes from nanosecond to millisecond.
// Allow error to be no more than 1.
EXPECT_NEAR(cricket::VideoFormat::FpsToInterval(60) / 1E6, interval, 1);
last_timestamp = timestamp;
}
frame_forwarder.IncomingCapturedFrame(
frame_source.GetFrame(1280, 720, webrtc::VideoRotation::kVideoRotation_0,
rtc::kNumMicrosecsPerSec / 30));
last_timestamp = stream->GetLastTimestamp();
for (int i = 0; i < 10; i++) {
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame(
1280, 720, webrtc::VideoRotation::kVideoRotation_0,
rtc::kNumMicrosecsPerSec / 30));
int64_t timestamp = stream->GetLastTimestamp();
int64_t interval = timestamp - last_timestamp;
// Precision changes from nanosecond to millisecond.
// Allow error to be no more than 1.
EXPECT_NEAR(cricket::VideoFormat::FpsToInterval(30) / 1E6, interval, 1);
last_timestamp = timestamp;
}
// Remove stream previously added to free the external encoder instance.
EXPECT_TRUE(channel->RemoveSendStream(kSsrc));
}
void WebRtcVideoEngineTest::AssignDefaultAptRtxTypes() {
std::vector<VideoCodec> engine_codecs = engine_.codecs();
RTC_DCHECK(!engine_codecs.empty());
for (const cricket::VideoCodec& codec : engine_codecs) {
if (codec.name == "rtx") {
int associated_payload_type;
if (codec.GetParam(kCodecParamAssociatedPayloadType,
&associated_payload_type)) {
default_apt_rtx_types_[associated_payload_type] = codec.id;
}
}
}
}
void WebRtcVideoEngineTest::AssignDefaultCodec() {
std::vector<VideoCodec> engine_codecs = engine_.codecs();
RTC_DCHECK(!engine_codecs.empty());
bool codec_set = false;
for (const cricket::VideoCodec& codec : engine_codecs) {
if (!codec_set && codec.name != "rtx" && codec.name != "red" &&
codec.name != "ulpfec") {
default_codec_ = codec;
codec_set = true;
}
}
RTC_DCHECK(codec_set);
}
size_t WebRtcVideoEngineTest::GetEngineCodecIndex(
const std::string& name) const {
const std::vector<cricket::VideoCodec> codecs = engine_.codecs();
for (size_t i = 0; i < codecs.size(); ++i) {
const cricket::VideoCodec engine_codec = codecs[i];
if (!absl::EqualsIgnoreCase(name, engine_codec.name))
continue;
// The tests only use H264 Constrained Baseline. Make sure we don't return
// an internal H264 codec from the engine with a different H264 profile.
if (absl::EqualsIgnoreCase(name.c_str(), kH264CodecName)) {
const absl::optional<webrtc::H264::ProfileLevelId> profile_level_id =
webrtc::H264::ParseSdpProfileLevelId(engine_codec.params);
if (profile_level_id->profile !=
webrtc::H264::kProfileConstrainedBaseline) {
continue;
}
}
return i;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
}
// This point should never be reached.
ADD_FAILURE() << "Unrecognized codec name: " << name;
return -1;
}
cricket::VideoCodec WebRtcVideoEngineTest::GetEngineCodec(
const std::string& name) const {
return engine_.codecs()[GetEngineCodecIndex(name)];
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
}
VideoMediaChannel*
WebRtcVideoEngineTest::SetSendParamsWithAllSupportedCodecs() {
VideoMediaChannel* channel = engine_.CreateMediaChannel(
call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get());
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
// We need to look up the codec in the engine to get the correct payload type.
for (const webrtc::SdpVideoFormat& format :
encoder_factory_->GetSupportedFormats()) {
cricket::VideoCodec engine_codec = GetEngineCodec(format.name);
if (!absl::c_linear_search(parameters.codecs, engine_codec)) {
parameters.codecs.push_back(engine_codec);
}
}
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
EXPECT_TRUE(channel->SetSendParameters(parameters));
return channel;
}
VideoMediaChannel* WebRtcVideoEngineTest::SetRecvParamsWithSupportedCodecs(
const std::vector<VideoCodec>& codecs) {
VideoMediaChannel* channel = engine_.CreateMediaChannel(
call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get());
cricket::VideoRecvParameters parameters;
parameters.codecs = codecs;
EXPECT_TRUE(channel->SetRecvParameters(parameters));
return channel;
}
void WebRtcVideoEngineTest::ExpectRtpCapabilitySupport(const char* uri,
bool supported) const {
const RtpCapabilities capabilities = engine_.GetCapabilities();
if (supported) {
EXPECT_THAT(capabilities.header_extensions,
::testing::Contains(::testing::Field(&RtpExtension::uri, uri)));
} else {
EXPECT_THAT(capabilities.header_extensions,
::testing::Each(::testing::Field(&RtpExtension::uri,
::testing::StrNe(uri))));
}
}
TEST_F(WebRtcVideoEngineTest, UsesSimulcastAdapterForVp8Factories) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
std::unique_ptr<VideoMediaChannel> channel(
SetSendParamsWithAllSupportedCodecs());
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
EXPECT_TRUE(channel->AddSendStream(CreateSimStreamParams("cname", ssrcs)));
EXPECT_TRUE(channel->SetSend(true));
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 60);
EXPECT_TRUE(channel->SetVideoSend(ssrcs.front(), nullptr, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(2));
// Verify that encoders are configured for simulcast through adapter
// (increasing resolution and only configured to send one stream each).
int prev_width = -1;
for (size_t i = 0; i < encoder_factory_->encoders().size(); ++i) {
ASSERT_TRUE(encoder_factory_->encoders()[i]->WaitForInitEncode());
webrtc::VideoCodec codec_settings =
encoder_factory_->encoders()[i]->GetCodecSettings();
EXPECT_EQ(0, codec_settings.numberOfSimulcastStreams);
EXPECT_GT(codec_settings.width, prev_width);
prev_width = codec_settings.width;
}
EXPECT_TRUE(channel->SetVideoSend(ssrcs.front(), nullptr, nullptr));
channel.reset();
ASSERT_EQ(0u, encoder_factory_->encoders().size());
}
TEST_F(WebRtcVideoEngineTest, ChannelWithH264CanChangeToVp8) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
encoder_factory_->AddSupportedVideoCodecType("H264");
// Frame source.
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("H264"));
EXPECT_TRUE(channel->SetSendParameters(parameters));
EXPECT_TRUE(
channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &frame_forwarder));
// Sending one frame will have allocate the encoder.
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_EQ_WAIT(1u, encoder_factory_->encoders().size(), kTimeout);
cricket::VideoSendParameters new_parameters;
new_parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel->SetSendParameters(new_parameters));
// Sending one frame will switch encoder.
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
EXPECT_EQ_WAIT(1u, encoder_factory_->encoders().size(), kTimeout);
}
TEST_F(WebRtcVideoEngineTest,
UsesSimulcastAdapterForVp8WithCombinedVP8AndH264Factory) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
encoder_factory_->AddSupportedVideoCodecType("H264");
std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel->SetSendParameters(parameters));
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
EXPECT_TRUE(channel->AddSendStream(CreateSimStreamParams("cname", ssrcs)));
EXPECT_TRUE(channel->SetSend(true));
// Send a fake frame, or else the media engine will configure the simulcast
// encoder adapter at a low-enough size that it'll only create a single
// encoder layer.
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
EXPECT_TRUE(channel->SetVideoSend(ssrcs.front(), nullptr, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(2));
ASSERT_TRUE(encoder_factory_->encoders()[0]->WaitForInitEncode());
EXPECT_EQ(webrtc::kVideoCodecVP8,
encoder_factory_->encoders()[0]->GetCodecSettings().codecType);
channel.reset();
// Make sure DestroyVideoEncoder was called on the factory.
EXPECT_EQ(0u, encoder_factory_->encoders().size());
}
TEST_F(WebRtcVideoEngineTest,
DestroysNonSimulcastEncoderFromCombinedVP8AndH264Factory) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
encoder_factory_->AddSupportedVideoCodecType("H264");
std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("H264"));
EXPECT_TRUE(channel->SetSendParameters(parameters));
EXPECT_TRUE(
channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
// Send a frame of 720p. This should trigger a "real" encoder initialization.
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1));
ASSERT_EQ(1u, encoder_factory_->encoders().size());
ASSERT_TRUE(encoder_factory_->encoders()[0]->WaitForInitEncode());
EXPECT_EQ(webrtc::kVideoCodecH264,
encoder_factory_->encoders()[0]->GetCodecSettings().codecType);
channel.reset();
// Make sure DestroyVideoEncoder was called on the factory.
ASSERT_EQ(0u, encoder_factory_->encoders().size());
}
TEST_F(WebRtcVideoEngineTest, SimulcastEnabledForH264BehindFieldTrial) {
RTC_DCHECK(!override_field_trials_);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-H264Simulcast/Enabled/");
encoder_factory_->AddSupportedVideoCodecType("H264");
std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("H264"));
EXPECT_TRUE(channel->SetSendParameters(parameters));
const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
EXPECT_TRUE(
channel->AddSendStream(cricket::CreateSimStreamParams("cname", ssrcs)));
// Send a frame of 720p. This should trigger a "real" encoder initialization.
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
EXPECT_TRUE(channel->SetVideoSend(ssrcs[0], nullptr, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1));
ASSERT_EQ(1u, encoder_factory_->encoders().size());
FakeWebRtcVideoEncoder* encoder = encoder_factory_->encoders()[0];
ASSERT_TRUE(encoder_factory_->encoders()[0]->WaitForInitEncode());
EXPECT_EQ(webrtc::kVideoCodecH264, encoder->GetCodecSettings().codecType);
EXPECT_LT(1u, encoder->GetCodecSettings().numberOfSimulcastStreams);
EXPECT_TRUE(channel->SetVideoSend(ssrcs[0], nullptr, nullptr));
}
// Test that the FlexFEC field trial properly alters the output of
// WebRtcVideoEngine::codecs(), for an existing |engine_| object.
//
// TODO(brandtr): Remove this test, when the FlexFEC field trial is gone.
TEST_F(WebRtcVideoEngineTest,
Flexfec03SupportedAsInternalCodecBehindFieldTrial) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
auto flexfec = Field("name", &VideoCodec::name, "flexfec-03");
// FlexFEC is not active without field trial.
EXPECT_THAT(engine_.codecs(), Not(Contains(flexfec)));
// FlexFEC is active with field trial.
RTC_DCHECK(!override_field_trials_);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-FlexFEC-03-Advertised/Enabled/");
EXPECT_THAT(engine_.codecs(), Contains(flexfec));
}
// Test that codecs are added in the order they are reported from the factory.
TEST_F(WebRtcVideoEngineTest, ReportSupportedCodecs) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
const char* kFakeCodecName = "FakeCodec";
encoder_factory_->AddSupportedVideoCodecType(kFakeCodecName);
// The last reported codec should appear after the first codec in the vector.
const size_t vp8_index = GetEngineCodecIndex("VP8");
const size_t fake_codec_index = GetEngineCodecIndex(kFakeCodecName);
EXPECT_LT(vp8_index, fake_codec_index);
}
// Test that a codec that was added after the engine was initialized
// does show up in the codec list after it was added.
TEST_F(WebRtcVideoEngineTest, ReportSupportedAddedCodec) {
const char* kFakeExternalCodecName1 = "FakeExternalCodec1";
const char* kFakeExternalCodecName2 = "FakeExternalCodec2";
// Set up external encoder factory with first codec, and initialize engine.
encoder_factory_->AddSupportedVideoCodecType(kFakeExternalCodecName1);
std::vector<cricket::VideoCodec> codecs_before(engine_.codecs());
// Add second codec.
encoder_factory_->AddSupportedVideoCodecType(kFakeExternalCodecName2);
std::vector<cricket::VideoCodec> codecs_after(engine_.codecs());
// The codec itself and RTX should have been added.
EXPECT_EQ(codecs_before.size() + 2, codecs_after.size());
// Check that both fake codecs are present and that the second fake codec
// appears after the first fake codec.
const size_t fake_codec_index1 = GetEngineCodecIndex(kFakeExternalCodecName1);
const size_t fake_codec_index2 = GetEngineCodecIndex(kFakeExternalCodecName2);
EXPECT_LT(fake_codec_index1, fake_codec_index2);
}
TEST_F(WebRtcVideoEngineTest, ReportRtxForExternalCodec) {
const char* kFakeCodecName = "FakeCodec";
encoder_factory_->AddSupportedVideoCodecType(kFakeCodecName);
const size_t fake_codec_index = GetEngineCodecIndex(kFakeCodecName);
EXPECT_EQ("rtx", engine_.codecs().at(fake_codec_index + 1).name);
}
TEST_F(WebRtcVideoEngineTest, RegisterDecodersIfSupported) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
decoder_factory_->AddSupportedVideoCodecType(webrtc::SdpVideoFormat("VP8"));
cricket::VideoRecvParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
std::unique_ptr<VideoMediaChannel> channel(
SetRecvParamsWithSupportedCodecs(parameters.codecs));
EXPECT_TRUE(
channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
ASSERT_EQ(1u, decoder_factory_->decoders().size());
// Setting codecs of the same type should not reallocate the decoder.
EXPECT_TRUE(channel->SetRecvParameters(parameters));
EXPECT_EQ(1, decoder_factory_->GetNumCreatedDecoders());
// Remove stream previously added to free the external decoder instance.
EXPECT_TRUE(channel->RemoveRecvStream(kSsrc));
EXPECT_EQ(0u, decoder_factory_->decoders().size());
}
// Verifies that we can set up decoders.
TEST_F(WebRtcVideoEngineTest, RegisterH264DecoderIfSupported) {
// TODO(pbos): Do not assume that encoder/decoder support is symmetric. We
// can't even query the WebRtcVideoDecoderFactory for supported codecs.
// For now we add a FakeWebRtcVideoEncoderFactory to add H264 to supported
// codecs.
encoder_factory_->AddSupportedVideoCodecType("H264");
webrtc::SdpVideoFormat supported_h264("H264");
supported_h264.parameters[kH264FmtpPacketizationMode] = "1";
decoder_factory_->AddSupportedVideoCodecType(supported_h264);
std::vector<cricket::VideoCodec> codecs;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
codecs.push_back(GetEngineCodec("H264"));
std::unique_ptr<VideoMediaChannel> channel(
SetRecvParamsWithSupportedCodecs(codecs));
EXPECT_TRUE(
channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
ASSERT_EQ(1u, decoder_factory_->decoders().size());
}
// Tests when GetSources is called with non-existing ssrc, it will return an
// empty list of RtpSource without crashing.
TEST_F(WebRtcVideoEngineTest, GetSourcesWithNonExistingSsrc) {
// Setup an recv stream with |kSsrc|.
encoder_factory_->AddSupportedVideoCodecType("VP8");
decoder_factory_->AddSupportedVideoCodecType(webrtc::SdpVideoFormat("VP8"));
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
std::unique_ptr<VideoMediaChannel> channel(
SetRecvParamsWithSupportedCodecs(parameters.codecs));
EXPECT_TRUE(
channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
// Call GetSources with |kSsrc + 1| which doesn't exist.
std::vector<webrtc::RtpSource> sources = channel->GetSources(kSsrc + 1);
EXPECT_EQ(0u, sources.size());
}
TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, NullFactories) {
std::unique_ptr<webrtc::VideoEncoderFactory> encoder_factory;
std::unique_ptr<webrtc::VideoDecoderFactory> decoder_factory;
WebRtcVideoEngine engine(std::move(encoder_factory),
std::move(decoder_factory));
EXPECT_EQ(0u, engine.codecs().size());
}
TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, EmptyFactories) {
// |engine| take ownership of the factories.
webrtc::MockVideoEncoderFactory* encoder_factory =
new webrtc::MockVideoEncoderFactory();
webrtc::MockVideoDecoderFactory* decoder_factory =
new webrtc::MockVideoDecoderFactory();
WebRtcVideoEngine engine(
(std::unique_ptr<webrtc::VideoEncoderFactory>(encoder_factory)),
(std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)));
EXPECT_CALL(*encoder_factory, GetSupportedFormats());
EXPECT_EQ(0u, engine.codecs().size());
EXPECT_CALL(*encoder_factory, Die());
EXPECT_CALL(*decoder_factory, Die());
}
// Test full behavior in the video engine when video codec factories of the new
// type are injected supporting the single codec Vp8. Check the returned codecs
// from the engine and that we will create a Vp8 encoder and decoder using the
// new factories.
TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) {
// |engine| take ownership of the factories.
webrtc::MockVideoEncoderFactory* encoder_factory =
new webrtc::MockVideoEncoderFactory();
webrtc::MockVideoDecoderFactory* decoder_factory =
new webrtc::MockVideoDecoderFactory();
std::unique_ptr<webrtc::MockVideoBitrateAllocatorFactory>
rate_allocator_factory =
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
std::make_unique<webrtc::MockVideoBitrateAllocatorFactory>();
EXPECT_CALL(*rate_allocator_factory,
CreateVideoBitrateAllocatorProxy(Field(
&webrtc::VideoCodec::codecType, webrtc::kVideoCodecVP8)))
.WillOnce(::testing::Return(new webrtc::MockVideoBitrateAllocator()));
WebRtcVideoEngine engine(
(std::unique_ptr<webrtc::VideoEncoderFactory>(encoder_factory)),
(std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)));
const webrtc::SdpVideoFormat vp8_format("VP8");
const std::vector<webrtc::SdpVideoFormat> supported_formats = {vp8_format};
EXPECT_CALL(*encoder_factory, GetSupportedFormats())
.WillRepeatedly(::testing::Return(supported_formats));
// Verify the codecs from the engine.
const std::vector<VideoCodec> engine_codecs = engine.codecs();
// Verify default codecs has been added correctly.
EXPECT_EQ(5u, engine_codecs.size());
EXPECT_EQ("VP8", engine_codecs.at(0).name);
// RTX codec for VP8.
EXPECT_EQ("rtx", engine_codecs.at(1).name);
int vp8_associated_payload;
EXPECT_TRUE(engine_codecs.at(1).GetParam(kCodecParamAssociatedPayloadType,
&vp8_associated_payload));
EXPECT_EQ(vp8_associated_payload, engine_codecs.at(0).id);
EXPECT_EQ(kRedCodecName, engine_codecs.at(2).name);
// RTX codec for RED.
EXPECT_EQ("rtx", engine_codecs.at(3).name);
int red_associated_payload;
EXPECT_TRUE(engine_codecs.at(3).GetParam(kCodecParamAssociatedPayloadType,
&red_associated_payload));
EXPECT_EQ(red_associated_payload, engine_codecs.at(2).id);
EXPECT_EQ(kUlpfecCodecName, engine_codecs.at(4).name);
int associated_payload_type;
EXPECT_TRUE(engine_codecs.at(1).GetParam(
cricket::kCodecParamAssociatedPayloadType, &associated_payload_type));
EXPECT_EQ(engine_codecs.at(0).id, associated_payload_type);
// Verify default parameters has been added to the VP8 codec.
VerifyCodecHasDefaultFeedbackParams(engine_codecs.at(0),
/*lntf_expected=*/false);
// Mock encoder creation. |engine| take ownership of the encoder.
Revert "Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo" This reverts commit cdc5eb0de179dcc866ef770ea303879c64466879. Reason for revert: Causes wrong CPU adaptation to be used for some HW codecs since GetEncoderInfo() is polled before InitEncode(). Original change's description: > Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo > > Make implementation of VideoEncoderFactory::QueryVideoEncoder optional > until it is removed downstream and remove all implementations of it. > > Bug: webrtc:10065 > Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f > Reviewed-on: https://webrtc-review.googlesource.com/c/113065 > Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org> > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25924} TBR=brandtr@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,mirtad@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:10065 Change-Id: Idaa452e1d8c1c58cdb4ec69b88fce9042589cc3c Reviewed-on: https://webrtc-review.googlesource.com/c/113800 Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25943}
2018-12-10 09:47:34 +00:00
webrtc::VideoEncoderFactory::CodecInfo codec_info;
codec_info.is_hardware_accelerated = false;
codec_info.has_internal_source = false;
const webrtc::SdpVideoFormat format("VP8");
Revert "Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo" This reverts commit cdc5eb0de179dcc866ef770ea303879c64466879. Reason for revert: Causes wrong CPU adaptation to be used for some HW codecs since GetEncoderInfo() is polled before InitEncode(). Original change's description: > Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo > > Make implementation of VideoEncoderFactory::QueryVideoEncoder optional > until it is removed downstream and remove all implementations of it. > > Bug: webrtc:10065 > Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f > Reviewed-on: https://webrtc-review.googlesource.com/c/113065 > Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org> > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25924} TBR=brandtr@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,mirtad@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:10065 Change-Id: Idaa452e1d8c1c58cdb4ec69b88fce9042589cc3c Reviewed-on: https://webrtc-review.googlesource.com/c/113800 Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25943}
2018-12-10 09:47:34 +00:00
EXPECT_CALL(*encoder_factory, QueryVideoEncoder(format))
.WillRepeatedly(::testing::Return(codec_info));
FakeWebRtcVideoEncoder* const encoder = new FakeWebRtcVideoEncoder(nullptr);
rtc::Event encoder_created;
EXPECT_CALL(*encoder_factory, CreateVideoEncoderProxy(format))
.WillOnce(
::testing::DoAll(::testing::InvokeWithoutArgs(
[&encoder_created]() { encoder_created.Set(); }),
::testing::Return(encoder)));
// Mock decoder creation. |engine| take ownership of the decoder.
FakeWebRtcVideoDecoder* const decoder = new FakeWebRtcVideoDecoder(nullptr);
EXPECT_CALL(*decoder_factory, CreateVideoDecoderProxy(format))
.WillOnce(::testing::Return(decoder));
// Create a call.
webrtc::RtcEventLogNull event_log;
auto task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
webrtc::Call::Config call_config(&event_log);
webrtc::FieldTrialBasedConfig field_trials;
call_config.trials = &field_trials;
call_config.task_queue_factory = task_queue_factory.get();
const auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
// Create send channel.
const int send_ssrc = 123;
std::unique_ptr<VideoMediaChannel> send_channel(engine.CreateMediaChannel(
call.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
rate_allocator_factory.get()));
cricket::VideoSendParameters send_parameters;
send_parameters.codecs.push_back(engine_codecs.at(0));
EXPECT_TRUE(send_channel->SetSendParameters(send_parameters));
send_channel->OnReadyToSend(true);
EXPECT_TRUE(
send_channel->AddSendStream(StreamParams::CreateLegacy(send_ssrc)));
EXPECT_TRUE(send_channel->SetSend(true));
// Set capturer.
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
EXPECT_TRUE(send_channel->SetVideoSend(send_ssrc, nullptr, &frame_forwarder));
// Sending one frame will allocate the encoder.
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
encoder_created.Wait(kTimeout);
// Create recv channel.
const int recv_ssrc = 321;
std::unique_ptr<VideoMediaChannel> recv_channel(engine.CreateMediaChannel(
call.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
rate_allocator_factory.get()));
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(engine_codecs.at(0));
EXPECT_TRUE(recv_channel->SetRecvParameters(recv_parameters));
EXPECT_TRUE(recv_channel->AddRecvStream(
cricket::StreamParams::CreateLegacy(recv_ssrc)));
// Remove streams previously added to free the encoder and decoder instance.
EXPECT_CALL(*encoder_factory, Die());
EXPECT_CALL(*decoder_factory, Die());
EXPECT_CALL(*rate_allocator_factory, Die());
EXPECT_TRUE(send_channel->RemoveSendStream(send_ssrc));
EXPECT_TRUE(recv_channel->RemoveRecvStream(recv_ssrc));
}
Reland "Update internal video decoder factory to new interface" This reverts commit 267d84baf0597f89a3d1f66d323db754bc5d9239. Reason for reland: Fix the bug; decoder is not allowed to ever be null and we need to use a NullVideoDecoder that ignores calls instead. Original change's description: > Revert "Update internal video decoder factory to new interface" > > This reverts commit b2fc9b1b104240e68047901309deaee3e8b94bea. > > Reason for revert: Suspected to cause failures on Android bots on webrtc.fyi, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/21051 > > Original change's description: > > Update internal video decoder factory to new interface > > > > We want to move away from cricket::WebRtcVideoDecoderFactory and this CL > > updates the internal factory. Also, VideoDecoderSoftwareFallbackWrapper > > is updated to take a VideoDecoder as argument instead of a factory so it > > can be used with external SW decoders. > > > > Bug: webrtc:7925 > > Change-Id: Ie6dc6c24f8610a2129620c6e2b42e3cebb2ddef7 > > Reviewed-on: https://webrtc-review.googlesource.com/7301 > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org> > > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20597} > > TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org > > Change-Id: I0a12c98fdc30f00d58c85ee7e088f50160d39724 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7925 > Reviewed-on: https://webrtc-review.googlesource.com/21420 > Reviewed-by: Christian Fremerey <chfremer@webrtc.org> > Commit-Queue: Christian Fremerey <chfremer@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20605} TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org,chfremer@webrtc.org,chfremer@google.com Change-Id: I6cf5794dc3fadfa86809a94da80b69dbb4c56f52 No-Try: true Bug: webrtc:7925 Reviewed-on: https://webrtc-review.googlesource.com/21541 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20623}
2017-11-09 13:43:42 +01:00
// Test behavior when decoder factory fails to create a decoder (returns null).
TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, NullDecoder) {
// |engine| take ownership of the factories.
webrtc::MockVideoEncoderFactory* encoder_factory =
new webrtc::MockVideoEncoderFactory();
Reland "Update internal video decoder factory to new interface" This reverts commit 267d84baf0597f89a3d1f66d323db754bc5d9239. Reason for reland: Fix the bug; decoder is not allowed to ever be null and we need to use a NullVideoDecoder that ignores calls instead. Original change's description: > Revert "Update internal video decoder factory to new interface" > > This reverts commit b2fc9b1b104240e68047901309deaee3e8b94bea. > > Reason for revert: Suspected to cause failures on Android bots on webrtc.fyi, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/21051 > > Original change's description: > > Update internal video decoder factory to new interface > > > > We want to move away from cricket::WebRtcVideoDecoderFactory and this CL > > updates the internal factory. Also, VideoDecoderSoftwareFallbackWrapper > > is updated to take a VideoDecoder as argument instead of a factory so it > > can be used with external SW decoders. > > > > Bug: webrtc:7925 > > Change-Id: Ie6dc6c24f8610a2129620c6e2b42e3cebb2ddef7 > > Reviewed-on: https://webrtc-review.googlesource.com/7301 > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org> > > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20597} > > TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org > > Change-Id: I0a12c98fdc30f00d58c85ee7e088f50160d39724 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7925 > Reviewed-on: https://webrtc-review.googlesource.com/21420 > Reviewed-by: Christian Fremerey <chfremer@webrtc.org> > Commit-Queue: Christian Fremerey <chfremer@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20605} TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org,chfremer@webrtc.org,chfremer@google.com Change-Id: I6cf5794dc3fadfa86809a94da80b69dbb4c56f52 No-Try: true Bug: webrtc:7925 Reviewed-on: https://webrtc-review.googlesource.com/21541 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20623}
2017-11-09 13:43:42 +01:00
webrtc::MockVideoDecoderFactory* decoder_factory =
new webrtc::MockVideoDecoderFactory();
std::unique_ptr<webrtc::MockVideoBitrateAllocatorFactory>
rate_allocator_factory =
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
std::make_unique<webrtc::MockVideoBitrateAllocatorFactory>();
Reland "Update internal video decoder factory to new interface" This reverts commit 267d84baf0597f89a3d1f66d323db754bc5d9239. Reason for reland: Fix the bug; decoder is not allowed to ever be null and we need to use a NullVideoDecoder that ignores calls instead. Original change's description: > Revert "Update internal video decoder factory to new interface" > > This reverts commit b2fc9b1b104240e68047901309deaee3e8b94bea. > > Reason for revert: Suspected to cause failures on Android bots on webrtc.fyi, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/21051 > > Original change's description: > > Update internal video decoder factory to new interface > > > > We want to move away from cricket::WebRtcVideoDecoderFactory and this CL > > updates the internal factory. Also, VideoDecoderSoftwareFallbackWrapper > > is updated to take a VideoDecoder as argument instead of a factory so it > > can be used with external SW decoders. > > > > Bug: webrtc:7925 > > Change-Id: Ie6dc6c24f8610a2129620c6e2b42e3cebb2ddef7 > > Reviewed-on: https://webrtc-review.googlesource.com/7301 > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org> > > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20597} > > TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org > > Change-Id: I0a12c98fdc30f00d58c85ee7e088f50160d39724 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7925 > Reviewed-on: https://webrtc-review.googlesource.com/21420 > Reviewed-by: Christian Fremerey <chfremer@webrtc.org> > Commit-Queue: Christian Fremerey <chfremer@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20605} TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org,chfremer@webrtc.org,chfremer@google.com Change-Id: I6cf5794dc3fadfa86809a94da80b69dbb4c56f52 No-Try: true Bug: webrtc:7925 Reviewed-on: https://webrtc-review.googlesource.com/21541 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20623}
2017-11-09 13:43:42 +01:00
WebRtcVideoEngine engine(
(std::unique_ptr<webrtc::VideoEncoderFactory>(encoder_factory)),
(std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)));
Reland "Update internal video decoder factory to new interface" This reverts commit 267d84baf0597f89a3d1f66d323db754bc5d9239. Reason for reland: Fix the bug; decoder is not allowed to ever be null and we need to use a NullVideoDecoder that ignores calls instead. Original change's description: > Revert "Update internal video decoder factory to new interface" > > This reverts commit b2fc9b1b104240e68047901309deaee3e8b94bea. > > Reason for revert: Suspected to cause failures on Android bots on webrtc.fyi, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/21051 > > Original change's description: > > Update internal video decoder factory to new interface > > > > We want to move away from cricket::WebRtcVideoDecoderFactory and this CL > > updates the internal factory. Also, VideoDecoderSoftwareFallbackWrapper > > is updated to take a VideoDecoder as argument instead of a factory so it > > can be used with external SW decoders. > > > > Bug: webrtc:7925 > > Change-Id: Ie6dc6c24f8610a2129620c6e2b42e3cebb2ddef7 > > Reviewed-on: https://webrtc-review.googlesource.com/7301 > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org> > > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20597} > > TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org > > Change-Id: I0a12c98fdc30f00d58c85ee7e088f50160d39724 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7925 > Reviewed-on: https://webrtc-review.googlesource.com/21420 > Reviewed-by: Christian Fremerey <chfremer@webrtc.org> > Commit-Queue: Christian Fremerey <chfremer@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20605} TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org,chfremer@webrtc.org,chfremer@google.com Change-Id: I6cf5794dc3fadfa86809a94da80b69dbb4c56f52 No-Try: true Bug: webrtc:7925 Reviewed-on: https://webrtc-review.googlesource.com/21541 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20623}
2017-11-09 13:43:42 +01:00
const webrtc::SdpVideoFormat vp8_format("VP8");
const std::vector<webrtc::SdpVideoFormat> supported_formats = {vp8_format};
EXPECT_CALL(*encoder_factory, GetSupportedFormats())
.WillRepeatedly(::testing::Return(supported_formats));
Reland "Update internal video decoder factory to new interface" This reverts commit 267d84baf0597f89a3d1f66d323db754bc5d9239. Reason for reland: Fix the bug; decoder is not allowed to ever be null and we need to use a NullVideoDecoder that ignores calls instead. Original change's description: > Revert "Update internal video decoder factory to new interface" > > This reverts commit b2fc9b1b104240e68047901309deaee3e8b94bea. > > Reason for revert: Suspected to cause failures on Android bots on webrtc.fyi, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/21051 > > Original change's description: > > Update internal video decoder factory to new interface > > > > We want to move away from cricket::WebRtcVideoDecoderFactory and this CL > > updates the internal factory. Also, VideoDecoderSoftwareFallbackWrapper > > is updated to take a VideoDecoder as argument instead of a factory so it > > can be used with external SW decoders. > > > > Bug: webrtc:7925 > > Change-Id: Ie6dc6c24f8610a2129620c6e2b42e3cebb2ddef7 > > Reviewed-on: https://webrtc-review.googlesource.com/7301 > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org> > > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20597} > > TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org > > Change-Id: I0a12c98fdc30f00d58c85ee7e088f50160d39724 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7925 > Reviewed-on: https://webrtc-review.googlesource.com/21420 > Reviewed-by: Christian Fremerey <chfremer@webrtc.org> > Commit-Queue: Christian Fremerey <chfremer@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20605} TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org,chfremer@webrtc.org,chfremer@google.com Change-Id: I6cf5794dc3fadfa86809a94da80b69dbb4c56f52 No-Try: true Bug: webrtc:7925 Reviewed-on: https://webrtc-review.googlesource.com/21541 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20623}
2017-11-09 13:43:42 +01:00
// Decoder creation fails.
EXPECT_CALL(*decoder_factory, CreateVideoDecoderProxy(::testing::_))
.WillOnce(::testing::Return(nullptr));
Reland "Update internal video decoder factory to new interface" This reverts commit 267d84baf0597f89a3d1f66d323db754bc5d9239. Reason for reland: Fix the bug; decoder is not allowed to ever be null and we need to use a NullVideoDecoder that ignores calls instead. Original change's description: > Revert "Update internal video decoder factory to new interface" > > This reverts commit b2fc9b1b104240e68047901309deaee3e8b94bea. > > Reason for revert: Suspected to cause failures on Android bots on webrtc.fyi, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/21051 > > Original change's description: > > Update internal video decoder factory to new interface > > > > We want to move away from cricket::WebRtcVideoDecoderFactory and this CL > > updates the internal factory. Also, VideoDecoderSoftwareFallbackWrapper > > is updated to take a VideoDecoder as argument instead of a factory so it > > can be used with external SW decoders. > > > > Bug: webrtc:7925 > > Change-Id: Ie6dc6c24f8610a2129620c6e2b42e3cebb2ddef7 > > Reviewed-on: https://webrtc-review.googlesource.com/7301 > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org> > > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20597} > > TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org > > Change-Id: I0a12c98fdc30f00d58c85ee7e088f50160d39724 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7925 > Reviewed-on: https://webrtc-review.googlesource.com/21420 > Reviewed-by: Christian Fremerey <chfremer@webrtc.org> > Commit-Queue: Christian Fremerey <chfremer@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20605} TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org,chfremer@webrtc.org,chfremer@google.com Change-Id: I6cf5794dc3fadfa86809a94da80b69dbb4c56f52 No-Try: true Bug: webrtc:7925 Reviewed-on: https://webrtc-review.googlesource.com/21541 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20623}
2017-11-09 13:43:42 +01:00
// Create a call.
webrtc::RtcEventLogNull event_log;
auto task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
webrtc::Call::Config call_config(&event_log);
webrtc::FieldTrialBasedConfig field_trials;
call_config.trials = &field_trials;
call_config.task_queue_factory = task_queue_factory.get();
const auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
Reland "Update internal video decoder factory to new interface" This reverts commit 267d84baf0597f89a3d1f66d323db754bc5d9239. Reason for reland: Fix the bug; decoder is not allowed to ever be null and we need to use a NullVideoDecoder that ignores calls instead. Original change's description: > Revert "Update internal video decoder factory to new interface" > > This reverts commit b2fc9b1b104240e68047901309deaee3e8b94bea. > > Reason for revert: Suspected to cause failures on Android bots on webrtc.fyi, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/21051 > > Original change's description: > > Update internal video decoder factory to new interface > > > > We want to move away from cricket::WebRtcVideoDecoderFactory and this CL > > updates the internal factory. Also, VideoDecoderSoftwareFallbackWrapper > > is updated to take a VideoDecoder as argument instead of a factory so it > > can be used with external SW decoders. > > > > Bug: webrtc:7925 > > Change-Id: Ie6dc6c24f8610a2129620c6e2b42e3cebb2ddef7 > > Reviewed-on: https://webrtc-review.googlesource.com/7301 > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org> > > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20597} > > TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org > > Change-Id: I0a12c98fdc30f00d58c85ee7e088f50160d39724 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7925 > Reviewed-on: https://webrtc-review.googlesource.com/21420 > Reviewed-by: Christian Fremerey <chfremer@webrtc.org> > Commit-Queue: Christian Fremerey <chfremer@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20605} TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org,chfremer@webrtc.org,chfremer@google.com Change-Id: I6cf5794dc3fadfa86809a94da80b69dbb4c56f52 No-Try: true Bug: webrtc:7925 Reviewed-on: https://webrtc-review.googlesource.com/21541 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20623}
2017-11-09 13:43:42 +01:00
// Create recv channel.
const int recv_ssrc = 321;
std::unique_ptr<VideoMediaChannel> recv_channel(engine.CreateMediaChannel(
call.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
rate_allocator_factory.get()));
Reland "Update internal video decoder factory to new interface" This reverts commit 267d84baf0597f89a3d1f66d323db754bc5d9239. Reason for reland: Fix the bug; decoder is not allowed to ever be null and we need to use a NullVideoDecoder that ignores calls instead. Original change's description: > Revert "Update internal video decoder factory to new interface" > > This reverts commit b2fc9b1b104240e68047901309deaee3e8b94bea. > > Reason for revert: Suspected to cause failures on Android bots on webrtc.fyi, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/21051 > > Original change's description: > > Update internal video decoder factory to new interface > > > > We want to move away from cricket::WebRtcVideoDecoderFactory and this CL > > updates the internal factory. Also, VideoDecoderSoftwareFallbackWrapper > > is updated to take a VideoDecoder as argument instead of a factory so it > > can be used with external SW decoders. > > > > Bug: webrtc:7925 > > Change-Id: Ie6dc6c24f8610a2129620c6e2b42e3cebb2ddef7 > > Reviewed-on: https://webrtc-review.googlesource.com/7301 > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org> > > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20597} > > TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org > > Change-Id: I0a12c98fdc30f00d58c85ee7e088f50160d39724 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7925 > Reviewed-on: https://webrtc-review.googlesource.com/21420 > Reviewed-by: Christian Fremerey <chfremer@webrtc.org> > Commit-Queue: Christian Fremerey <chfremer@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20605} TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org,chfremer@webrtc.org,chfremer@google.com Change-Id: I6cf5794dc3fadfa86809a94da80b69dbb4c56f52 No-Try: true Bug: webrtc:7925 Reviewed-on: https://webrtc-review.googlesource.com/21541 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20623}
2017-11-09 13:43:42 +01:00
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(engine.codecs().front());
EXPECT_TRUE(recv_channel->SetRecvParameters(recv_parameters));
EXPECT_TRUE(recv_channel->AddRecvStream(
cricket::StreamParams::CreateLegacy(recv_ssrc)));
// Remove streams previously added to free the encoder and decoder instance.
EXPECT_TRUE(recv_channel->RemoveRecvStream(recv_ssrc));
}
TEST_F(WebRtcVideoEngineTest, DISABLED_RecreatesEncoderOnContentTypeChange) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
std::unique_ptr<FakeCall> fake_call(new FakeCall());
std::unique_ptr<VideoMediaChannel> channel(
SetSendParamsWithAllSupportedCodecs());
ASSERT_TRUE(
channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
cricket::VideoCodec codec = GetEngineCodec("VP8");
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(codec);
channel->OnReadyToSend(true);
channel->SetSend(true);
ASSERT_TRUE(channel->SetSendParameters(parameters));
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
VideoOptions options;
EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1));
EXPECT_EQ(webrtc::VideoCodecMode::kRealtimeVideo,
encoder_factory_->encoders().back()->GetCodecSettings().mode);
EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
// No change in content type, keep current encoder.
EXPECT_EQ(1, encoder_factory_->GetNumCreatedEncoders());
options.is_screencast.emplace(true);
EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
// Change to screen content, recreate encoder. For the simulcast encoder
// adapter case, this will result in two calls since InitEncode triggers a
// a new instance.
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(2));
EXPECT_EQ(webrtc::VideoCodecMode::kScreensharing,
encoder_factory_->encoders().back()->GetCodecSettings().mode);
EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
// Still screen content, no need to update encoder.
EXPECT_EQ(2, encoder_factory_->GetNumCreatedEncoders());
options.is_screencast.emplace(false);
options.video_noise_reduction.emplace(false);
EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &frame_forwarder));
// Change back to regular video content, update encoder. Also change
// a non |is_screencast| option just to verify it doesn't affect recreation.
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(3));
EXPECT_EQ(webrtc::VideoCodecMode::kRealtimeVideo,
encoder_factory_->encoders().back()->GetCodecSettings().mode);
// Remove stream previously added to free the external encoder instance.
EXPECT_TRUE(channel->RemoveSendStream(kSsrc));
EXPECT_EQ(0u, encoder_factory_->encoders().size());
}
class WebRtcVideoChannelBaseTest : public ::testing::Test {
protected:
WebRtcVideoChannelBaseTest()
: task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()),
video_bitrate_allocator_factory_(
webrtc::CreateBuiltinVideoBitrateAllocatorFactory()),
engine_(webrtc::CreateBuiltinVideoEncoderFactory(),
webrtc::CreateBuiltinVideoDecoderFactory()) {}
virtual void SetUp() {
// One testcase calls SetUp in a loop, only create call_ once.
if (!call_) {
webrtc::Call::Config call_config(&event_log_);
call_config.task_queue_factory = task_queue_factory_.get();
call_config.trials = &field_trials_;
call_.reset(webrtc::Call::Create(call_config));
}
cricket::MediaConfig media_config;
// Disabling cpu overuse detection actually disables quality scaling too; it
// implies DegradationPreference kMaintainResolution. Automatic scaling
// needs to be disabled, otherwise, tests which check the size of received
// frames become flaky.
media_config.video.enable_cpu_adaptation = false;
channel_.reset(
static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateMediaChannel(
call_.get(), media_config, cricket::VideoOptions(),
webrtc::CryptoOptions(), video_bitrate_allocator_factory_.get())));
channel_->OnReadyToSend(true);
EXPECT_TRUE(channel_.get() != NULL);
network_interface_.SetDestination(channel_.get());
channel_->SetInterface(&network_interface_, webrtc::MediaTransportConfig());
cricket::VideoRecvParameters parameters;
parameters.codecs = engine_.codecs();
channel_->SetRecvParameters(parameters);
EXPECT_TRUE(channel_->AddSendStream(DefaultSendStreamParams()));
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
frame_forwarder_ = std::make_unique<webrtc::test::FrameForwarder>();
frame_source_ = std::make_unique<cricket::FakeFrameSource>(
640, 480, rtc::kNumMicrosecsPerSec / kFramerate);
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, frame_forwarder_.get()));
}
// Utility method to setup an additional stream to send and receive video.
// Used to test send and recv between two streams.
void SetUpSecondStream() {
SetUpSecondStreamWithNoRecv();
// Setup recv for second stream.
EXPECT_TRUE(channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kSsrc + 2)));
// Make the second renderer available for use by a new stream.
EXPECT_TRUE(channel_->SetSink(kSsrc + 2, &renderer2_));
}
// Setup an additional stream just to send video. Defer add recv stream.
// This is required if you want to test unsignalled recv of video rtp packets.
void SetUpSecondStreamWithNoRecv() {
// SetUp() already added kSsrc make sure duplicate SSRCs cant be added.
EXPECT_TRUE(
channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
EXPECT_FALSE(
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
EXPECT_TRUE(channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kSsrc + 2)));
// We dont add recv for the second stream.
// Setup the receive and renderer for second stream after send.
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
frame_forwarder_2_ = std::make_unique<webrtc::test::FrameForwarder>();
EXPECT_TRUE(
channel_->SetVideoSend(kSsrc + 2, nullptr, frame_forwarder_2_.get()));
}
virtual void TearDown() { channel_.reset(); }
bool SetDefaultCodec() { return SetOneCodec(DefaultCodec()); }
bool SetOneCodec(const cricket::VideoCodec& codec) {
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
frame_source_ = std::make_unique<cricket::FakeFrameSource>(
kVideoWidth, kVideoHeight, rtc::kNumMicrosecsPerSec / kFramerate);
bool sending = channel_->sending();
bool success = SetSend(false);
if (success) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(codec);
success = channel_->SetSendParameters(parameters);
}
if (success) {
success = SetSend(sending);
}
return success;
}
bool SetSend(bool send) { return channel_->SetSend(send); }
void SendFrame() {
if (frame_forwarder_2_) {
frame_forwarder_2_->IncomingCapturedFrame(frame_source_->GetFrame());
}
frame_forwarder_->IncomingCapturedFrame(frame_source_->GetFrame());
}
bool WaitAndSendFrame(int wait_ms) {
bool ret = rtc::Thread::Current()->ProcessMessages(wait_ms);
SendFrame();
return ret;
}
int NumRtpBytes() { return network_interface_.NumRtpBytes(); }
int NumRtpBytes(uint32_t ssrc) {
return network_interface_.NumRtpBytes(ssrc);
}
int NumRtpPackets() { return network_interface_.NumRtpPackets(); }
int NumRtpPackets(uint32_t ssrc) {
return network_interface_.NumRtpPackets(ssrc);
}
int NumSentSsrcs() { return network_interface_.NumSentSsrcs(); }
const rtc::CopyOnWriteBuffer* GetRtpPacket(int index) {
return network_interface_.GetRtpPacket(index);
}
static int GetPayloadType(const rtc::CopyOnWriteBuffer* p) {
webrtc::RTPHeader header;
EXPECT_TRUE(ParseRtpPacket(p, &header));
return header.payloadType;
}
static bool ParseRtpPacket(const rtc::CopyOnWriteBuffer* p,
webrtc::RTPHeader* header) {
std::unique_ptr<webrtc::RtpHeaderParser> parser(
webrtc::RtpHeaderParser::CreateForTest());
return parser->Parse(p->cdata(), p->size(), header);
}
// Tests that we can send and receive frames.
void SendAndReceive(const cricket::VideoCodec& codec) {
EXPECT_TRUE(SetOneCodec(codec));
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
EXPECT_EQ(0, renderer_.num_rendered_frames());
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
EXPECT_EQ(codec.id, GetPayloadType(p.get()));
}
void SendReceiveManyAndGetStats(const cricket::VideoCodec& codec,
int duration_sec,
int fps) {
EXPECT_TRUE(SetOneCodec(codec));
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
EXPECT_EQ(0, renderer_.num_rendered_frames());
for (int i = 0; i < duration_sec; ++i) {
for (int frame = 1; frame <= fps; ++frame) {
EXPECT_TRUE(WaitAndSendFrame(1000 / fps));
EXPECT_FRAME_WAIT(frame + i * fps, kVideoWidth, kVideoHeight, kTimeout);
}
}
std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
EXPECT_EQ(codec.id, GetPayloadType(p.get()));
}
cricket::VideoSenderInfo GetSenderStats(size_t i) {
cricket::VideoMediaInfo info;
EXPECT_TRUE(channel_->GetStats(&info));
return info.senders[i];
}
cricket::VideoReceiverInfo GetReceiverStats(size_t i) {
cricket::VideoMediaInfo info;
EXPECT_TRUE(channel_->GetStats(&info));
return info.receivers[i];
}
// Two streams one channel tests.
// Tests that we can send and receive frames.
void TwoStreamsSendAndReceive(const cricket::VideoCodec& codec) {
SetUpSecondStream();
// Test sending and receiving on first stream.
SendAndReceive(codec);
// Test sending and receiving on second stream.
EXPECT_EQ_WAIT(1, renderer2_.num_rendered_frames(), kTimeout);
EXPECT_GT(NumRtpPackets(), 0);
EXPECT_EQ(1, renderer2_.num_rendered_frames());
}
cricket::VideoCodec GetEngineCodec(const std::string& name) {
for (const cricket::VideoCodec& engine_codec : engine_.codecs()) {
if (absl::EqualsIgnoreCase(name, engine_codec.name))
return engine_codec;
}
// This point should never be reached.
ADD_FAILURE() << "Unrecognized codec name: " << name;
return cricket::VideoCodec();
}
cricket::VideoCodec DefaultCodec() { return GetEngineCodec("VP8"); }
cricket::StreamParams DefaultSendStreamParams() {
return cricket::StreamParams::CreateLegacy(kSsrc);
}
webrtc::RtcEventLogNull event_log_;
webrtc::FieldTrialBasedConfig field_trials_;
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
std::unique_ptr<webrtc::Call> call_;
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory_;
WebRtcVideoEngine engine_;
std::unique_ptr<cricket::FakeFrameSource> frame_source_;
std::unique_ptr<webrtc::test::FrameForwarder> frame_forwarder_;
std::unique_ptr<webrtc::test::FrameForwarder> frame_forwarder_2_;
std::unique_ptr<WebRtcVideoChannel> channel_;
cricket::FakeNetworkInterface network_interface_;
cricket::FakeVideoRenderer renderer_;
// Used by test cases where 2 streams are run on the same channel.
cricket::FakeVideoRenderer renderer2_;
};
// Test that SetSend works.
TEST_F(WebRtcVideoChannelBaseTest, SetSend) {
EXPECT_FALSE(channel_->sending());
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, frame_forwarder_.get()));
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_FALSE(channel_->sending());
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(channel_->sending());
SendFrame();
EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
EXPECT_TRUE(SetSend(false));
EXPECT_FALSE(channel_->sending());
}
// Test that SetSend fails without codecs being set.
TEST_F(WebRtcVideoChannelBaseTest, SetSendWithoutCodecs) {
EXPECT_FALSE(channel_->sending());
EXPECT_FALSE(SetSend(true));
EXPECT_FALSE(channel_->sending());
}
// Test that we properly set the send and recv buffer sizes by the time
// SetSend is called.
TEST_F(WebRtcVideoChannelBaseTest, SetSendSetsTransportBufferSizes) {
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
EXPECT_EQ(256 * 1024, network_interface_.recvbuf_size());
}
// Test that we properly set the send and recv buffer sizes when overriding
// via field trials.
TEST_F(WebRtcVideoChannelBaseTest, OverridesRecvBufferSize) {
// Set field trial to override the default recv buffer size, and then re-run
// setup where the interface is created and configured.
const int kCustomRecvBufferSize = 123456;
webrtc::test::ScopedFieldTrials field_trial(
"WebRTC-IncreasedReceivebuffers/123456/");
SetUp();
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
EXPECT_EQ(kCustomRecvBufferSize, network_interface_.recvbuf_size());
}
// Test that we properly set the send and recv buffer sizes when overriding
// via field trials with suffix.
TEST_F(WebRtcVideoChannelBaseTest, OverridesRecvBufferSizeWithSuffix) {
// Set field trial to override the default recv buffer size, and then re-run
// setup where the interface is created and configured.
const int kCustomRecvBufferSize = 123456;
webrtc::test::ScopedFieldTrials field_trial(
"WebRTC-IncreasedReceivebuffers/123456_Dogfood/");
SetUp();
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
EXPECT_EQ(kCustomRecvBufferSize, network_interface_.recvbuf_size());
}
// Test that we properly set the send and recv buffer sizes when overriding
// via field trials that don't make any sense.
TEST_F(WebRtcVideoChannelBaseTest, InvalidRecvBufferSize) {
// Set bogus field trial values to override the default recv buffer size, and
// then re-run setup where the interface is created and configured. The
// default value should still be used.
for (std::string group : {" ", "NotANumber", "-1", "0"}) {
std::string field_trial_string = "WebRTC-IncreasedReceivebuffers/";
field_trial_string += group;
field_trial_string += "/";
webrtc::test::ScopedFieldTrials field_trial(field_trial_string);
SetUp();
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
EXPECT_EQ(256 * 1024, network_interface_.recvbuf_size());
}
}
// Test that stats work properly for a 1-1 call.
TEST_F(WebRtcVideoChannelBaseTest, GetStats) {
Reland "Only include payload in bytes sent/received." This is a reland of 74a1b4b1321b426392d4c32e4a02361226ad5358 Original change's description: > Only include payload in bytes sent/received. > > According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and > https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent > statistic should not include headers or padding. > > Similarly, according to > https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes > received are calculated the same way as bytes sent (eg. not including > padding or headers). > > This change stops adding padding and headers to these statistics. > > Bug: webrtc:8516,webrtc:10525 > Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Bjorn Mellem <mellem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28647} Bug: webrtc:8516, webrtc:10525 Change-Id: Iaa1613e5becdfaa0af0f6b9f00e5b871937a719c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147520 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28731}
2019-07-30 08:34:03 -07:00
SetUp();
const int kDurationSec = 3;
const int kFps = 10;
SendReceiveManyAndGetStats(DefaultCodec(), kDurationSec, kFps);
cricket::VideoMediaInfo info;
EXPECT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(1U, info.senders.size());
// TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload?
// For webrtc, bytes_sent does not include the RTP header length.
EXPECT_EQ(info.senders[0].payload_bytes_sent,
Reland "Only include payload in bytes sent/received." This is a reland of 74a1b4b1321b426392d4c32e4a02361226ad5358 Original change's description: > Only include payload in bytes sent/received. > > According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and > https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent > statistic should not include headers or padding. > > Similarly, according to > https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes > received are calculated the same way as bytes sent (eg. not including > padding or headers). > > This change stops adding padding and headers to these statistics. > > Bug: webrtc:8516,webrtc:10525 > Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Bjorn Mellem <mellem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28647} Bug: webrtc:8516, webrtc:10525 Change-Id: Iaa1613e5becdfaa0af0f6b9f00e5b871937a719c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147520 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28731}
2019-07-30 08:34:03 -07:00
NumRtpBytes() - kRtpHeaderSize * NumRtpPackets());
EXPECT_EQ(NumRtpPackets(), info.senders[0].packets_sent);
EXPECT_EQ(0.0, info.senders[0].fraction_lost);
ASSERT_TRUE(info.senders[0].codec_payload_type);
EXPECT_EQ(DefaultCodec().id, *info.senders[0].codec_payload_type);
EXPECT_EQ(0, info.senders[0].firs_rcvd);
EXPECT_EQ(0, info.senders[0].plis_rcvd);
EXPECT_EQ(0, info.senders[0].nacks_rcvd);
EXPECT_EQ(kVideoWidth, info.senders[0].send_frame_width);
EXPECT_EQ(kVideoHeight, info.senders[0].send_frame_height);
EXPECT_GT(info.senders[0].framerate_input, 0);
EXPECT_GT(info.senders[0].framerate_sent, 0);
EXPECT_EQ(1U, info.send_codecs.count(DefaultCodec().id));
EXPECT_EQ(DefaultCodec().ToCodecParameters(),
info.send_codecs[DefaultCodec().id]);
ASSERT_EQ(1U, info.receivers.size());
EXPECT_EQ(1U, info.senders[0].ssrcs().size());
EXPECT_EQ(1U, info.receivers[0].ssrcs().size());
EXPECT_EQ(info.senders[0].ssrcs()[0], info.receivers[0].ssrcs()[0]);
ASSERT_TRUE(info.receivers[0].codec_payload_type);
EXPECT_EQ(DefaultCodec().id, *info.receivers[0].codec_payload_type);
Reland "Only include payload in bytes sent/received." This is a reland of 74a1b4b1321b426392d4c32e4a02361226ad5358 Original change's description: > Only include payload in bytes sent/received. > > According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and > https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent > statistic should not include headers or padding. > > Similarly, according to > https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes > received are calculated the same way as bytes sent (eg. not including > padding or headers). > > This change stops adding padding and headers to these statistics. > > Bug: webrtc:8516,webrtc:10525 > Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Bjorn Mellem <mellem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28647} Bug: webrtc:8516, webrtc:10525 Change-Id: Iaa1613e5becdfaa0af0f6b9f00e5b871937a719c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147520 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28731}
2019-07-30 08:34:03 -07:00
EXPECT_EQ(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
info.receivers[0].payload_bytes_rcvd);
EXPECT_EQ(NumRtpPackets(), info.receivers[0].packets_rcvd);
EXPECT_EQ(0, info.receivers[0].packets_lost);
// TODO(asapersson): Not set for webrtc. Handle missing stats.
// EXPECT_EQ(0, info.receivers[0].packets_concealed);
EXPECT_EQ(0, info.receivers[0].firs_sent);
EXPECT_EQ(0, info.receivers[0].plis_sent);
EXPECT_EQ(0, info.receivers[0].nacks_sent);
EXPECT_EQ(kVideoWidth, info.receivers[0].frame_width);
EXPECT_EQ(kVideoHeight, info.receivers[0].frame_height);
EXPECT_GT(info.receivers[0].framerate_rcvd, 0);
EXPECT_GT(info.receivers[0].framerate_decoded, 0);
EXPECT_GT(info.receivers[0].framerate_output, 0);
EXPECT_EQ(1U, info.receive_codecs.count(DefaultCodec().id));
EXPECT_EQ(DefaultCodec().ToCodecParameters(),
info.receive_codecs[DefaultCodec().id]);
}
// Test that stats work properly for a conf call with multiple recv streams.
TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleRecvStreams) {
Reland "Only include payload in bytes sent/received." This is a reland of 74a1b4b1321b426392d4c32e4a02361226ad5358 Original change's description: > Only include payload in bytes sent/received. > > According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and > https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent > statistic should not include headers or padding. > > Similarly, according to > https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes > received are calculated the same way as bytes sent (eg. not including > padding or headers). > > This change stops adding padding and headers to these statistics. > > Bug: webrtc:8516,webrtc:10525 > Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Bjorn Mellem <mellem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28647} Bug: webrtc:8516, webrtc:10525 Change-Id: Iaa1613e5becdfaa0af0f6b9f00e5b871937a719c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147520 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28731}
2019-07-30 08:34:03 -07:00
SetUp();
cricket::FakeVideoRenderer renderer1, renderer2;
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(DefaultCodec());
parameters.conference_mode = true;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
EXPECT_TRUE(channel_->SetSink(1, &renderer1));
EXPECT_TRUE(channel_->SetSink(2, &renderer2));
EXPECT_EQ(0, renderer1.num_rendered_frames());
EXPECT_EQ(0, renderer2.num_rendered_frames());
std::vector<uint32_t> ssrcs;
ssrcs.push_back(1);
ssrcs.push_back(2);
network_interface_.SetConferenceMode(true, ssrcs);
SendFrame();
EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kVideoWidth, kVideoHeight,
kTimeout);
EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kVideoWidth, kVideoHeight,
kTimeout);
EXPECT_TRUE(channel_->SetSend(false));
cricket::VideoMediaInfo info;
EXPECT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(1U, info.senders.size());
// TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload?
// For webrtc, bytes_sent does not include the RTP header length.
Reland "Only include payload in bytes sent/received." This is a reland of 74a1b4b1321b426392d4c32e4a02361226ad5358 Original change's description: > Only include payload in bytes sent/received. > > According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and > https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent > statistic should not include headers or padding. > > Similarly, according to > https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes > received are calculated the same way as bytes sent (eg. not including > padding or headers). > > This change stops adding padding and headers to these statistics. > > Bug: webrtc:8516,webrtc:10525 > Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Bjorn Mellem <mellem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28647} Bug: webrtc:8516, webrtc:10525 Change-Id: Iaa1613e5becdfaa0af0f6b9f00e5b871937a719c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147520 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28731}
2019-07-30 08:34:03 -07:00
EXPECT_EQ_WAIT(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
GetSenderStats(0).payload_bytes_sent, kTimeout);
EXPECT_EQ_WAIT(NumRtpPackets(), GetSenderStats(0).packets_sent, kTimeout);
EXPECT_EQ(kVideoWidth, GetSenderStats(0).send_frame_width);
EXPECT_EQ(kVideoHeight, GetSenderStats(0).send_frame_height);
ASSERT_EQ(2U, info.receivers.size());
for (size_t i = 0; i < info.receivers.size(); ++i) {
EXPECT_EQ(1U, GetReceiverStats(i).ssrcs().size());
EXPECT_EQ(i + 1, GetReceiverStats(i).ssrcs()[0]);
Reland "Only include payload in bytes sent/received." This is a reland of 74a1b4b1321b426392d4c32e4a02361226ad5358 Original change's description: > Only include payload in bytes sent/received. > > According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and > https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent > statistic should not include headers or padding. > > Similarly, according to > https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes > received are calculated the same way as bytes sent (eg. not including > padding or headers). > > This change stops adding padding and headers to these statistics. > > Bug: webrtc:8516,webrtc:10525 > Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Bjorn Mellem <mellem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28647} Bug: webrtc:8516, webrtc:10525 Change-Id: Iaa1613e5becdfaa0af0f6b9f00e5b871937a719c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147520 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28731}
2019-07-30 08:34:03 -07:00
EXPECT_EQ_WAIT(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
GetReceiverStats(i).payload_bytes_rcvd, kTimeout);
EXPECT_EQ_WAIT(NumRtpPackets(), GetReceiverStats(i).packets_rcvd, kTimeout);
EXPECT_EQ_WAIT(kVideoWidth, GetReceiverStats(i).frame_width, kTimeout);
EXPECT_EQ_WAIT(kVideoHeight, GetReceiverStats(i).frame_height, kTimeout);
}
}
// Test that stats work properly for a conf call with multiple send streams.
TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleSendStreams) {
// Normal setup; note that we set the SSRC explicitly to ensure that
// it will come first in the senders map.
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(DefaultCodec());
parameters.conference_mode = true;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(
channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
EXPECT_TRUE(SetSend(true));
SendFrame();
EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
// Add an additional capturer, and hook up a renderer to receive it.
cricket::FakeVideoRenderer renderer2;
webrtc::test::FrameForwarder frame_forwarder;
const int kTestWidth = 160;
const int kTestHeight = 120;
cricket::FakeFrameSource frame_source(kTestWidth, kTestHeight,
rtc::kNumMicrosecsPerSec / 5);
EXPECT_TRUE(
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(5678)));
EXPECT_TRUE(channel_->SetVideoSend(5678, nullptr, &frame_forwarder));
EXPECT_TRUE(
channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(5678)));
EXPECT_TRUE(channel_->SetSink(5678, &renderer2));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kTestWidth, kTestHeight,
kTimeout);
// Get stats, and make sure they are correct for two senders. We wait until
// the number of expected packets have been sent to avoid races where we
// check stats before it has been updated.
cricket::VideoMediaInfo info;
for (uint32_t i = 0; i < kTimeout; ++i) {
rtc::Thread::Current()->ProcessMessages(1);
EXPECT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(2U, info.senders.size());
if (info.senders[0].packets_sent + info.senders[1].packets_sent ==
NumRtpPackets()) {
// Stats have been updated for both sent frames, expectations can be
// checked now.
break;
}
}
EXPECT_EQ(NumRtpPackets(),
info.senders[0].packets_sent + info.senders[1].packets_sent)
<< "Timed out while waiting for packet counts for all sent packets.";
EXPECT_EQ(1U, info.senders[0].ssrcs().size());
EXPECT_EQ(1234U, info.senders[0].ssrcs()[0]);
EXPECT_EQ(kVideoWidth, info.senders[0].send_frame_width);
EXPECT_EQ(kVideoHeight, info.senders[0].send_frame_height);
EXPECT_EQ(1U, info.senders[1].ssrcs().size());
EXPECT_EQ(5678U, info.senders[1].ssrcs()[0]);
EXPECT_EQ(kTestWidth, info.senders[1].send_frame_width);
EXPECT_EQ(kTestHeight, info.senders[1].send_frame_height);
// The capturer must be unregistered here as it runs out of it's scope next.
channel_->SetVideoSend(5678, nullptr, nullptr);
}
// Test that we can set the bandwidth.
TEST_F(WebRtcVideoChannelBaseTest, SetSendBandwidth) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(DefaultCodec());
parameters.max_bandwidth_bps = -1; // <= 0 means unlimited.
EXPECT_TRUE(channel_->SetSendParameters(parameters));
parameters.max_bandwidth_bps = 128 * 1024;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
}
// Test that we can set the SSRC for the default send source.
TEST_F(WebRtcVideoChannelBaseTest, SetSendSsrc) {
EXPECT_TRUE(SetDefaultCodec());
EXPECT_TRUE(SetSend(true));
SendFrame();
EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
webrtc::RTPHeader header;
std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
EXPECT_TRUE(ParseRtpPacket(p.get(), &header));
EXPECT_EQ(kSsrc, header.ssrc);
// Packets are being paced out, so these can mismatch between the first and
// second call to NumRtpPackets until pending packets are paced out.
EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(header.ssrc), kTimeout);
EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(header.ssrc), kTimeout);
EXPECT_EQ(1, NumSentSsrcs());
EXPECT_EQ(0, NumRtpPackets(kSsrc - 1));
EXPECT_EQ(0, NumRtpBytes(kSsrc - 1));
}
// Test that we can set the SSRC even after codecs are set.
TEST_F(WebRtcVideoChannelBaseTest, SetSendSsrcAfterSetCodecs) {
// Remove stream added in Setup.
EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
EXPECT_TRUE(SetDefaultCodec());
EXPECT_TRUE(
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(999)));
EXPECT_TRUE(channel_->SetVideoSend(999u, nullptr, frame_forwarder_.get()));
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(WaitAndSendFrame(0));
EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
webrtc::RTPHeader header;
std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
EXPECT_TRUE(ParseRtpPacket(p.get(), &header));
EXPECT_EQ(999u, header.ssrc);
// Packets are being paced out, so these can mismatch between the first and
// second call to NumRtpPackets until pending packets are paced out.
EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(header.ssrc), kTimeout);
EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(header.ssrc), kTimeout);
EXPECT_EQ(1, NumSentSsrcs());
EXPECT_EQ(0, NumRtpPackets(kSsrc));
EXPECT_EQ(0, NumRtpBytes(kSsrc));
}
// Test that we can set the default video renderer before and after
// media is received.
TEST_F(WebRtcVideoChannelBaseTest, SetSink) {
uint8_t data1[] = {0x80, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
rtc::CopyOnWriteBuffer packet1(data1, sizeof(data1));
rtc::SetBE32(packet1.data() + 8, kSsrc);
channel_->SetSink(kDefaultReceiveSsrc, NULL);
EXPECT_TRUE(SetDefaultCodec());
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(0, renderer_.num_rendered_frames());
channel_->OnPacketReceived(packet1, /* packet_time_us */ -1);
EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
}
// Tests setting up and configuring a send stream.
TEST_F(WebRtcVideoChannelBaseTest, AddRemoveSendStreams) {
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_GT(NumRtpPackets(), 0);
webrtc::RTPHeader header;
size_t last_packet = NumRtpPackets() - 1;
std::unique_ptr<const rtc::CopyOnWriteBuffer> p(
GetRtpPacket(static_cast<int>(last_packet)));
EXPECT_TRUE(ParseRtpPacket(p.get(), &header));
EXPECT_EQ(kSsrc, header.ssrc);
// Remove the send stream that was added during Setup.
EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
int rtp_packets = NumRtpPackets();
EXPECT_TRUE(
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(789u)));
EXPECT_TRUE(channel_->SetVideoSend(789u, nullptr, frame_forwarder_.get()));
EXPECT_EQ(rtp_packets, NumRtpPackets());
// Wait 30ms to guarantee the engine does not drop the frame.
EXPECT_TRUE(WaitAndSendFrame(30));
EXPECT_TRUE_WAIT(NumRtpPackets() > rtp_packets, kTimeout);
last_packet = NumRtpPackets() - 1;
p.reset(GetRtpPacket(static_cast<int>(last_packet)));
EXPECT_TRUE(ParseRtpPacket(p.get(), &header));
EXPECT_EQ(789u, header.ssrc);
}
// Tests the behavior of incoming streams in a conference scenario.
TEST_F(WebRtcVideoChannelBaseTest, SimulateConference) {
cricket::FakeVideoRenderer renderer1, renderer2;
EXPECT_TRUE(SetDefaultCodec());
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(DefaultCodec());
parameters.conference_mode = true;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
EXPECT_TRUE(channel_->SetSink(1, &renderer1));
EXPECT_TRUE(channel_->SetSink(2, &renderer2));
EXPECT_EQ(0, renderer1.num_rendered_frames());
EXPECT_EQ(0, renderer2.num_rendered_frames());
std::vector<uint32_t> ssrcs;
ssrcs.push_back(1);
ssrcs.push_back(2);
network_interface_.SetConferenceMode(true, ssrcs);
SendFrame();
EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kVideoWidth, kVideoHeight,
kTimeout);
EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kVideoWidth, kVideoHeight,
kTimeout);
std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
EXPECT_EQ(DefaultCodec().id, GetPayloadType(p.get()));
EXPECT_EQ(kVideoWidth, renderer1.width());
EXPECT_EQ(kVideoHeight, renderer1.height());
EXPECT_EQ(kVideoWidth, renderer2.width());
EXPECT_EQ(kVideoHeight, renderer2.height());
EXPECT_TRUE(channel_->RemoveRecvStream(2));
EXPECT_TRUE(channel_->RemoveRecvStream(1));
}
// Tests that we can add and remove capturers and frames are sent out properly
TEST_F(WebRtcVideoChannelBaseTest, DISABLED_AddRemoveCapturer) {
using cricket::FOURCC_I420;
using cricket::VideoCodec;
using cricket::VideoFormat;
using cricket::VideoOptions;
VideoCodec codec = DefaultCodec();
const int time_between_send_ms = VideoFormat::FpsToInterval(kFramerate);
EXPECT_TRUE(SetOneCodec(codec));
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
EXPECT_EQ(0, renderer_.num_rendered_frames());
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(480, 360, rtc::kNumMicrosecsPerSec / 30,
rtc::kNumMicrosecsPerSec / 30);
// TODO(nisse): This testcase fails if we don't configure
// screencast. It's unclear why, I see nothing obvious in this
// test which is related to screencast logic.
VideoOptions video_options;
video_options.is_screencast = true;
channel_->SetVideoSend(kSsrc, &video_options, nullptr);
int captured_frames = 1;
for (int iterations = 0; iterations < 2; ++iterations) {
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, &frame_forwarder));
rtc::Thread::Current()->ProcessMessages(time_between_send_ms);
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
++captured_frames;
// Wait until frame of right size is captured.
EXPECT_TRUE_WAIT(renderer_.num_rendered_frames() >= captured_frames &&
480 == renderer_.width() &&
360 == renderer_.height() && !renderer_.black_frame(),
kTimeout);
EXPECT_GE(renderer_.num_rendered_frames(), captured_frames);
EXPECT_EQ(480, renderer_.width());
EXPECT_EQ(360, renderer_.height());
captured_frames = renderer_.num_rendered_frames() + 1;
EXPECT_FALSE(renderer_.black_frame());
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr));
// Make sure a black frame is generated within the specified timeout.
// The black frame should be the resolution of the previous frame to
// prevent expensive encoder reconfigurations.
EXPECT_TRUE_WAIT(renderer_.num_rendered_frames() >= captured_frames &&
480 == renderer_.width() &&
360 == renderer_.height() && renderer_.black_frame(),
kTimeout);
EXPECT_GE(renderer_.num_rendered_frames(), captured_frames);
EXPECT_EQ(480, renderer_.width());
EXPECT_EQ(360, renderer_.height());
EXPECT_TRUE(renderer_.black_frame());
// The black frame has the same timestamp as the next frame since it's
// timestamp is set to the last frame's timestamp + interval. WebRTC will
// not render a frame with the same timestamp so capture another frame
// with the frame capturer to increment the next frame's timestamp.
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
}
}
// Tests that if SetVideoSend is called with a NULL capturer after the
// capturer was already removed, the application doesn't crash (and no black
// frame is sent).
TEST_F(WebRtcVideoChannelBaseTest, RemoveCapturerWithoutAdd) {
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
EXPECT_EQ(0, renderer_.num_rendered_frames());
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
// Wait for one frame so they don't get dropped because we send frames too
// tightly.
rtc::Thread::Current()->ProcessMessages(30);
// Remove the capturer.
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr));
// No capturer was added, so this SetVideoSend shouldn't do anything.
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr));
rtc::Thread::Current()->ProcessMessages(300);
// Verify no more frames were sent.
EXPECT_EQ(1, renderer_.num_rendered_frames());
}
// Tests that we can add and remove capturer as unique sources.
TEST_F(WebRtcVideoChannelBaseTest, AddRemoveCapturerMultipleSources) {
// WebRTC implementation will drop frames if pushed to quickly. Wait the
// interval time to avoid that.
// WebRTC implementation will drop frames if pushed to quickly. Wait the
// interval time to avoid that.
// Set up the stream associated with the engine.
EXPECT_TRUE(
channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
cricket::VideoFormat capture_format(
kVideoWidth, kVideoHeight,
cricket::VideoFormat::FpsToInterval(kFramerate), cricket::FOURCC_I420);
// Set up additional stream 1.
cricket::FakeVideoRenderer renderer1;
EXPECT_FALSE(channel_->SetSink(1, &renderer1));
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
EXPECT_TRUE(channel_->SetSink(1, &renderer1));
EXPECT_TRUE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(1)));
webrtc::test::FrameForwarder frame_forwarder1;
cricket::FakeFrameSource frame_source(kVideoWidth, kVideoHeight,
rtc::kNumMicrosecsPerSec / kFramerate);
// Set up additional stream 2.
cricket::FakeVideoRenderer renderer2;
EXPECT_FALSE(channel_->SetSink(2, &renderer2));
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
EXPECT_TRUE(channel_->SetSink(2, &renderer2));
EXPECT_TRUE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(2)));
webrtc::test::FrameForwarder frame_forwarder2;
// State for all the streams.
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
// A limitation in the lmi implementation requires that SetVideoSend() is
// called after SetOneCodec().
// TODO(hellner): this seems like an unnecessary constraint, fix it.
EXPECT_TRUE(channel_->SetVideoSend(1, nullptr, &frame_forwarder1));
EXPECT_TRUE(channel_->SetVideoSend(2, nullptr, &frame_forwarder2));
EXPECT_TRUE(SetSend(true));
// Test capturer associated with engine.
const int kTestWidth = 160;
const int kTestHeight = 120;
frame_forwarder1.IncomingCapturedFrame(frame_source.GetFrame(
kTestWidth, kTestHeight, webrtc::VideoRotation::kVideoRotation_0,
rtc::kNumMicrosecsPerSec / kFramerate));
EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kTestWidth, kTestHeight,
kTimeout);
// Capture a frame with additional capturer2, frames should be received
frame_forwarder2.IncomingCapturedFrame(frame_source.GetFrame(
kTestWidth, kTestHeight, webrtc::VideoRotation::kVideoRotation_0,
rtc::kNumMicrosecsPerSec / kFramerate));
EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kTestWidth, kTestHeight,
kTimeout);
// Successfully remove the capturer.
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr));
// The capturers must be unregistered here as it runs out of it's scope
// next.
EXPECT_TRUE(channel_->SetVideoSend(1, nullptr, nullptr));
EXPECT_TRUE(channel_->SetVideoSend(2, nullptr, nullptr));
}
// Tests empty StreamParams is rejected.
TEST_F(WebRtcVideoChannelBaseTest, RejectEmptyStreamParams) {
// Remove the send stream that was added during Setup.
EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
cricket::StreamParams empty;
EXPECT_FALSE(channel_->AddSendStream(empty));
EXPECT_TRUE(
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(789u)));
}
// Test that multiple send streams can be created and deleted properly.
TEST_F(WebRtcVideoChannelBaseTest, MultipleSendStreams) {
// Remove stream added in Setup. I.e. remove stream corresponding to default
// channel.
EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
const unsigned int kSsrcsSize = sizeof(kSsrcs4) / sizeof(kSsrcs4[0]);
for (unsigned int i = 0; i < kSsrcsSize; ++i) {
EXPECT_TRUE(channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kSsrcs4[i])));
}
// Delete one of the non default channel streams, let the destructor delete
// the remaining ones.
EXPECT_TRUE(channel_->RemoveSendStream(kSsrcs4[kSsrcsSize - 1]));
// Stream should already be deleted.
EXPECT_FALSE(channel_->RemoveSendStream(kSsrcs4[kSsrcsSize - 1]));
}
TEST_F(WebRtcVideoChannelBaseTest, SendAndReceiveVp8Vga) {
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
SendAndReceive(GetEngineCodec("VP8"));
}
TEST_F(WebRtcVideoChannelBaseTest, SendAndReceiveVp8Qvga) {
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
SendAndReceive(GetEngineCodec("VP8"));
}
TEST_F(WebRtcVideoChannelBaseTest, SendAndReceiveVp8SvcQqvga) {
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
SendAndReceive(GetEngineCodec("VP8"));
}
TEST_F(WebRtcVideoChannelBaseTest, TwoStreamsSendAndReceive) {
// Set a high bitrate to not be downscaled by VP8 due to low initial start
// bitrates. This currently happens at <250k, and two streams sharing 300k
// initially will use QVGA instead of VGA.
// TODO(pbos): Set up the quality scaler so that both senders reliably start
// at QVGA, then verify that instead.
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
cricket::VideoCodec codec = GetEngineCodec("VP8");
codec.params[kCodecParamStartBitrate] = "1000000";
TwoStreamsSendAndReceive(codec);
}
TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderFallback) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP9"));
parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel_->SetSendParameters(parameters));
VideoCodec codec;
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_EQ("VP9", codec.name);
// RequestEncoderFallback will post a task to the worker thread (which is also
// the current thread), hence the ProcessMessages call.
channel_->RequestEncoderFallback();
rtc::Thread::Current()->ProcessMessages(30);
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_EQ("VP8", codec.name);
// No other codec to fall back to, keep using VP8.
channel_->RequestEncoderFallback();
rtc::Thread::Current()->ProcessMessages(30);
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_EQ("VP8", codec.name);
}
TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderSwitchWithConfig) {
const std::string kParam = "the-param";
const std::string kPing = "ping";
const std::string kPong = "pong";
cricket::VideoSendParameters parameters;
VideoCodec vp9 = GetEngineCodec("VP9");
vp9.params[kParam] = kPong;
parameters.codecs.push_back(vp9);
VideoCodec vp8 = GetEngineCodec("VP8");
vp8.params[kParam] = kPing;
parameters.codecs.push_back(vp8);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
channel_->SetVideoCodecSwitchingEnabled(true);
VideoCodec codec;
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_THAT(codec.name, Eq("VP9"));
// RequestEncoderSwitch will post a task to the worker thread (which is also
// the current thread), hence the ProcessMessages call.
webrtc::EncoderSwitchRequestCallback::Config conf1{"VP8", kParam, kPing};
channel_->RequestEncoderSwitch(conf1);
rtc::Thread::Current()->ProcessMessages(30);
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_THAT(codec.name, Eq("VP8"));
EXPECT_THAT(codec.params, Contains(Pair(kParam, kPing)));
webrtc::EncoderSwitchRequestCallback::Config conf2{"VP9", kParam, kPong};
channel_->RequestEncoderSwitch(conf2);
rtc::Thread::Current()->ProcessMessages(30);
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_THAT(codec.name, Eq("VP9"));
EXPECT_THAT(codec.params, Contains(Pair(kParam, kPong)));
}
TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderSwitchIncorrectParam) {
const std::string kParam = "the-param";
const std::string kPing = "ping";
const std::string kPong = "pong";
cricket::VideoSendParameters parameters;
VideoCodec vp9 = GetEngineCodec("VP9");
vp9.params[kParam] = kPong;
parameters.codecs.push_back(vp9);
VideoCodec vp8 = GetEngineCodec("VP8");
vp8.params[kParam] = kPing;
parameters.codecs.push_back(vp8);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
channel_->SetVideoCodecSwitchingEnabled(true);
VideoCodec codec;
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_THAT(codec.name, Eq("VP9"));
// RequestEncoderSwitch will post a task to the worker thread (which is also
// the current thread), hence the ProcessMessages call.
webrtc::EncoderSwitchRequestCallback::Config conf1{"VP8", kParam, kPing};
channel_->RequestEncoderSwitch(conf1);
rtc::Thread::Current()->ProcessMessages(30);
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_THAT(codec.name, Eq("VP8"));
EXPECT_THAT(codec.params, Contains(Pair(kParam, kPing)));
// Incorrect conf2.value, expect no codec switch.
webrtc::EncoderSwitchRequestCallback::Config conf2{"VP9", kParam, kPing};
channel_->RequestEncoderSwitch(conf2);
rtc::Thread::Current()->ProcessMessages(30);
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_THAT(codec.name, Eq("VP8"));
EXPECT_THAT(codec.params, Contains(Pair(kParam, kPing)));
}
class WebRtcVideoChannelTest : public WebRtcVideoEngineTest {
public:
WebRtcVideoChannelTest() : WebRtcVideoChannelTest("") {}
explicit WebRtcVideoChannelTest(const char* field_trials)
: WebRtcVideoEngineTest(field_trials),
frame_source_(1280, 720, rtc::kNumMicrosecsPerSec / 30),
last_ssrc_(0) {}
void SetUp() override {
encoder_factory_->AddSupportedVideoCodecType("VP8");
encoder_factory_->AddSupportedVideoCodecType("VP9");
#if defined(WEBRTC_USE_H264)
encoder_factory_->AddSupportedVideoCodecType("H264");
#endif
fake_call_.reset(new FakeCall());
channel_.reset(engine_.CreateMediaChannel(
fake_call_.get(), GetMediaConfig(), VideoOptions(),
webrtc::CryptoOptions(), video_bitrate_allocator_factory_.get()));
channel_->OnReadyToSend(true);
last_ssrc_ = 123;
send_parameters_.codecs = engine_.codecs();
recv_parameters_.codecs = engine_.codecs();
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
}
protected:
FakeVideoSendStream* AddSendStream() {
return AddSendStream(StreamParams::CreateLegacy(++last_ssrc_));
}
FakeVideoSendStream* AddSendStream(const StreamParams& sp) {
size_t num_streams = fake_call_->GetVideoSendStreams().size();
EXPECT_TRUE(channel_->AddSendStream(sp));
std::vector<FakeVideoSendStream*> streams =
fake_call_->GetVideoSendStreams();
EXPECT_EQ(num_streams + 1, streams.size());
return streams[streams.size() - 1];
}
std::vector<FakeVideoSendStream*> GetFakeSendStreams() {
return fake_call_->GetVideoSendStreams();
}
FakeVideoReceiveStream* AddRecvStream() {
return AddRecvStream(StreamParams::CreateLegacy(++last_ssrc_));
}
FakeVideoReceiveStream* AddRecvStream(const StreamParams& sp) {
size_t num_streams = fake_call_->GetVideoReceiveStreams().size();
EXPECT_TRUE(channel_->AddRecvStream(sp));
std::vector<FakeVideoReceiveStream*> streams =
fake_call_->GetVideoReceiveStreams();
EXPECT_EQ(num_streams + 1, streams.size());
return streams[streams.size() - 1];
}
void SetSendCodecsShouldWorkForBitrates(const char* min_bitrate_kbps,
int expected_min_bitrate_bps,
const char* start_bitrate_kbps,
int expected_start_bitrate_bps,
const char* max_bitrate_kbps,
int expected_max_bitrate_bps) {
ExpectSetBitrateParameters(expected_min_bitrate_bps,
expected_start_bitrate_bps,
expected_max_bitrate_bps);
auto& codecs = send_parameters_.codecs;
codecs.clear();
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
codecs.push_back(GetEngineCodec("VP8"));
codecs[0].params[kCodecParamMinBitrate] = min_bitrate_kbps;
codecs[0].params[kCodecParamStartBitrate] = start_bitrate_kbps;
codecs[0].params[kCodecParamMaxBitrate] = max_bitrate_kbps;
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
}
void ExpectSetBitrateParameters(int min_bitrate_bps,
int start_bitrate_bps,
int max_bitrate_bps) {
EXPECT_CALL(
*fake_call_->GetMockTransportControllerSend(),
SetSdpBitrateParameters(AllOf(
Field(&BitrateConstraints::min_bitrate_bps, min_bitrate_bps),
Field(&BitrateConstraints::start_bitrate_bps, start_bitrate_bps),
Field(&BitrateConstraints::max_bitrate_bps, max_bitrate_bps))));
}
void ExpectSetMaxBitrate(int max_bitrate_bps) {
EXPECT_CALL(*fake_call_->GetMockTransportControllerSend(),
SetSdpBitrateParameters(Field(
&BitrateConstraints::max_bitrate_bps, max_bitrate_bps)));
}
void TestExtmapAllowMixedCaller(bool extmap_allow_mixed) {
// For a caller, the answer will be applied in set remote description
// where SetSendParameters() is called.
EXPECT_TRUE(
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
send_parameters_.extmap_allow_mixed = extmap_allow_mixed;
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
const webrtc::VideoSendStream::Config& config =
fake_call_->GetVideoSendStreams()[0]->GetConfig();
EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed);
}
void TestExtmapAllowMixedCallee(bool extmap_allow_mixed) {
// For a callee, the answer will be applied in set local description
// where SetExtmapAllowMixed() and AddSendStream() are called.
channel_->SetExtmapAllowMixed(extmap_allow_mixed);
EXPECT_TRUE(
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
const webrtc::VideoSendStream::Config& config =
fake_call_->GetVideoSendStreams()[0]->GetConfig();
EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed);
}
void TestSetSendRtpHeaderExtensions(const std::string& ext_uri) {
// Enable extension.
const int id = 1;
cricket::VideoSendParameters parameters = send_parameters_;
parameters.extensions.push_back(RtpExtension(ext_uri, id));
EXPECT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* send_stream =
AddSendStream(cricket::StreamParams::CreateLegacy(123));
// Verify the send extension id.
ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size());
EXPECT_EQ(id, send_stream->GetConfig().rtp.extensions[0].id);
EXPECT_EQ(ext_uri, send_stream->GetConfig().rtp.extensions[0].uri);
// Verify call with same set of extensions returns true.
EXPECT_TRUE(channel_->SetSendParameters(parameters));
// Verify that SetSendRtpHeaderExtensions doesn't implicitly add them for
// receivers.
EXPECT_TRUE(AddRecvStream(cricket::StreamParams::CreateLegacy(123))
->GetConfig()
.rtp.extensions.empty());
// Verify that existing RTP header extensions can be removed.
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size());
send_stream = fake_call_->GetVideoSendStreams()[0];
EXPECT_TRUE(send_stream->GetConfig().rtp.extensions.empty());
// Verify that adding receive RTP header extensions adds them for existing
// streams.
EXPECT_TRUE(channel_->SetSendParameters(parameters));
send_stream = fake_call_->GetVideoSendStreams()[0];
ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size());
EXPECT_EQ(id, send_stream->GetConfig().rtp.extensions[0].id);
EXPECT_EQ(ext_uri, send_stream->GetConfig().rtp.extensions[0].uri);
}
void TestSetRecvRtpHeaderExtensions(const std::string& ext_uri) {
// Enable extension.
const int id = 1;
cricket::VideoRecvParameters parameters = recv_parameters_;
parameters.extensions.push_back(RtpExtension(ext_uri, id));
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
FakeVideoReceiveStream* recv_stream =
AddRecvStream(cricket::StreamParams::CreateLegacy(123));
// Verify the recv extension id.
ASSERT_EQ(1u, recv_stream->GetConfig().rtp.extensions.size());
EXPECT_EQ(id, recv_stream->GetConfig().rtp.extensions[0].id);
EXPECT_EQ(ext_uri, recv_stream->GetConfig().rtp.extensions[0].uri);
// Verify call with same set of extensions returns true.
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
// Verify that SetRecvRtpHeaderExtensions doesn't implicitly add them for
// senders.
EXPECT_TRUE(AddSendStream(cricket::StreamParams::CreateLegacy(123))
->GetConfig()
.rtp.extensions.empty());
// Verify that existing RTP header extensions can be removed.
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_TRUE(recv_stream->GetConfig().rtp.extensions.empty());
// Verify that adding receive RTP header extensions adds them for existing
// streams.
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
ASSERT_EQ(1u, recv_stream->GetConfig().rtp.extensions.size());
EXPECT_EQ(id, recv_stream->GetConfig().rtp.extensions[0].id);
EXPECT_EQ(ext_uri, recv_stream->GetConfig().rtp.extensions[0].uri);
}
void TestLossNotificationState(bool expect_lntf_enabled) {
AssignDefaultCodec();
VerifyCodecHasDefaultFeedbackParams(default_codec_, expect_lntf_enabled);
cricket::VideoSendParameters parameters;
parameters.codecs = engine_.codecs();
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(channel_->SetSend(true));
// Send side.
FakeVideoSendStream* send_stream =
AddSendStream(cricket::StreamParams::CreateLegacy(1));
EXPECT_EQ(send_stream->GetConfig().rtp.lntf.enabled, expect_lntf_enabled);
// Receiver side.
FakeVideoReceiveStream* recv_stream =
AddRecvStream(cricket::StreamParams::CreateLegacy(1));
EXPECT_EQ(recv_stream->GetConfig().rtp.lntf.enabled, expect_lntf_enabled);
}
void TestExtensionFilter(const std::vector<std::string>& extensions,
const std::string& expected_extension) {
cricket::VideoSendParameters parameters = send_parameters_;
int expected_id = -1;
int id = 1;
for (const std::string& extension : extensions) {
if (extension == expected_extension)
expected_id = id;
parameters.extensions.push_back(RtpExtension(extension, id++));
}
EXPECT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* send_stream =
AddSendStream(cricket::StreamParams::CreateLegacy(123));
// Verify that only one of them has been set, and that it is the one with
// highest priority (transport sequence number).
ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size());
EXPECT_EQ(expected_id, send_stream->GetConfig().rtp.extensions[0].id);
EXPECT_EQ(expected_extension,
send_stream->GetConfig().rtp.extensions[0].uri);
}
void TestDegradationPreference(bool resolution_scaling_enabled,
bool fps_scaling_enabled);
void TestCpuAdaptation(bool enable_overuse, bool is_screenshare);
void TestReceiverLocalSsrcConfiguration(bool receiver_first);
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
void TestReceiveUnsignaledSsrcPacket(uint8_t payload_type,
bool expect_created_receive_stream);
FakeVideoSendStream* SetDenoisingOption(
uint32_t ssrc,
webrtc::test::FrameForwarder* frame_forwarder,
bool enabled) {
cricket::VideoOptions options;
options.video_noise_reduction = enabled;
EXPECT_TRUE(channel_->SetVideoSend(ssrc, &options, frame_forwarder));
// Options only take effect on the next frame.
frame_forwarder->IncomingCapturedFrame(frame_source_.GetFrame());
return fake_call_->GetVideoSendStreams().back();
}
FakeVideoSendStream* SetUpSimulcast(bool enabled, bool with_rtx) {
const int kRtxSsrcOffset = 0xDEADBEEF;
last_ssrc_ += 3;
std::vector<uint32_t> ssrcs;
std::vector<uint32_t> rtx_ssrcs;
uint32_t num_streams = enabled ? 3 : 1;
for (uint32_t i = 0; i < num_streams; ++i) {
uint32_t ssrc = last_ssrc_ + i;
ssrcs.push_back(ssrc);
if (with_rtx) {
rtx_ssrcs.push_back(ssrc + kRtxSsrcOffset);
}
}
if (with_rtx) {
return AddSendStream(
cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs));
}
return AddSendStream(CreateSimStreamParams("cname", ssrcs));
}
int GetMaxEncoderBitrate() {
std::vector<FakeVideoSendStream*> streams =
fake_call_->GetVideoSendStreams();
EXPECT_EQ(1u, streams.size());
FakeVideoSendStream* stream = streams[streams.size() - 1];
EXPECT_EQ(1u, stream->GetEncoderConfig().number_of_streams);
return stream->GetVideoStreams()[0].max_bitrate_bps;
}
void SetAndExpectMaxBitrate(int global_max,
int stream_max,
int expected_encoder_bitrate) {
VideoSendParameters limited_send_params = send_parameters_;
limited_send_params.max_bandwidth_bps = global_max;
EXPECT_TRUE(channel_->SetSendParameters(limited_send_params));
webrtc::RtpParameters parameters =
channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
parameters.encodings[0].max_bitrate_bps = stream_max;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Read back the parameteres and verify they have the correct value
parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_EQ(stream_max, parameters.encodings[0].max_bitrate_bps);
// Verify that the new value propagated down to the encoder
EXPECT_EQ(expected_encoder_bitrate, GetMaxEncoderBitrate());
}
// Values from kSimulcastConfigs in simulcast.cc.
const std::vector<webrtc::VideoStream> GetSimulcastBitrates720p() const {
std::vector<webrtc::VideoStream> layers(3);
layers[0].min_bitrate_bps = 30000;
layers[0].target_bitrate_bps = 150000;
layers[0].max_bitrate_bps = 200000;
layers[1].min_bitrate_bps = 150000;
layers[1].target_bitrate_bps = 500000;
layers[1].max_bitrate_bps = 700000;
layers[2].min_bitrate_bps = 600000;
layers[2].target_bitrate_bps = 2500000;
layers[2].max_bitrate_bps = 2500000;
return layers;
}
cricket::FakeFrameSource frame_source_;
std::unique_ptr<FakeCall> fake_call_;
std::unique_ptr<VideoMediaChannel> channel_;
cricket::VideoSendParameters send_parameters_;
cricket::VideoRecvParameters recv_parameters_;
uint32_t last_ssrc_;
};
TEST_F(WebRtcVideoChannelTest, SetsSyncGroupFromSyncLabel) {
const uint32_t kVideoSsrc = 123;
const std::string kSyncLabel = "AvSyncLabel";
cricket::StreamParams sp = cricket::StreamParams::CreateLegacy(kVideoSsrc);
sp.set_stream_ids({kSyncLabel});
EXPECT_TRUE(channel_->AddRecvStream(sp));
EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
EXPECT_EQ(kSyncLabel,
fake_call_->GetVideoReceiveStreams()[0]->GetConfig().sync_group)
<< "SyncGroup should be set based on sync_label";
}
TEST_F(WebRtcVideoChannelTest, RecvStreamWithSimAndRtx) {
cricket::VideoSendParameters parameters;
parameters.codecs = engine_.codecs();
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(channel_->SetSend(true));
parameters.conference_mode = true;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
// Send side.
const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs1);
const std::vector<uint32_t> rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1);
FakeVideoSendStream* send_stream = AddSendStream(
cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs));
ASSERT_EQ(rtx_ssrcs.size(), send_stream->GetConfig().rtp.rtx.ssrcs.size());
for (size_t i = 0; i < rtx_ssrcs.size(); ++i)
EXPECT_EQ(rtx_ssrcs[i], send_stream->GetConfig().rtp.rtx.ssrcs[i]);
// Receiver side.
FakeVideoReceiveStream* recv_stream = AddRecvStream(
cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs));
EXPECT_FALSE(
recv_stream->GetConfig().rtp.rtx_associated_payload_types.empty());
Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ ) Reason for revert: Identified a configuration problem in the video quality tests. Intend to fix and reland. Original issue's description: > Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ ) > > Reason for revert: > This change appears to break ulpfec, with severe regressions, e.g., for webrtc_perf_test FullStackTest.ForemanCifPlr5Ulpfec > > Original issue's description: > > Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ ) > > > > Reason for revert: > > Intend to fix perf failures and reland. > > > > Original issue's description: > > > Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ ) > > > > > > Reason for revert: > > > A few perf tests broken, including > > > > > > RampUpTest.UpDownUpAbsSendTimeSimulcastRedRtx > > > RampUpTest.UpDownUpTransportSequenceNumberRtx > > > RampUpTest.UpDownUpTransportSequenceNumberPacketLoss > > > > > > > > > Original issue's description: > > > > Use RtxReceiveStream. > > > > > > > > This also has the beneficial side-effect that when a media stream > > > > which is protected by FlexFEC receives an RTX retransmission, the > > > > retransmitted media packet is passed into the FlexFEC machinery, > > > > which should improve its ability to recover packets via FEC. > > > > > > > > BUG=webrtc:7135 > > > > > > > > Review-Url: https://codereview.webrtc.org/3008773002 > > > > Cr-Commit-Position: refs/heads/master@{#19649} > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/5c0f6c62ea3b1d2c43f8fc152961af27033475f7 > > > > > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > > > # Skipping CQ checks because original CL landed less than 1 days ago. > > > NOPRESUBMIT=true > > > NOTREECHECKS=true > > > NOTRY=true > > > BUG=webrtc:7135 > > > > > > Review-Url: https://codereview.webrtc.org/3010983002 > > > Cr-Commit-Position: refs/heads/master@{#19653} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/3c39c0137afa274d1d524b150b50304b38a2847b > > > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > > # Skipping CQ checks because original CL landed less than 1 days ago. > > NOPRESUBMIT=true > > NOTREECHECKS=true > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/3006063002 > > Cr-Commit-Position: refs/heads/master@{#19715} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35713eaf565c0fef07c8afc158d7b8fdf7ec3d78 > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/3007303002 > Cr-Commit-Position: refs/heads/master@{#19744} > Committed: https://chromium.googlesource.com/external/webrtc/+/8e7eee035178a7f10e19883681b5eaa4a7523107 TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/3012963002 Cr-Commit-Position: refs/heads/master@{#19765}
2017-09-11 02:32:16 -07:00
EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig()))
<< "RTX should be mapped for all decoders/payload types.";
Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ ) Reason for revert: Identified a configuration problem in the video quality tests. Intend to fix and reland. Original issue's description: > Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ ) > > Reason for revert: > This change appears to break ulpfec, with severe regressions, e.g., for webrtc_perf_test FullStackTest.ForemanCifPlr5Ulpfec > > Original issue's description: > > Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ ) > > > > Reason for revert: > > Intend to fix perf failures and reland. > > > > Original issue's description: > > > Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ ) > > > > > > Reason for revert: > > > A few perf tests broken, including > > > > > > RampUpTest.UpDownUpAbsSendTimeSimulcastRedRtx > > > RampUpTest.UpDownUpTransportSequenceNumberRtx > > > RampUpTest.UpDownUpTransportSequenceNumberPacketLoss > > > > > > > > > Original issue's description: > > > > Use RtxReceiveStream. > > > > > > > > This also has the beneficial side-effect that when a media stream > > > > which is protected by FlexFEC receives an RTX retransmission, the > > > > retransmitted media packet is passed into the FlexFEC machinery, > > > > which should improve its ability to recover packets via FEC. > > > > > > > > BUG=webrtc:7135 > > > > > > > > Review-Url: https://codereview.webrtc.org/3008773002 > > > > Cr-Commit-Position: refs/heads/master@{#19649} > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/5c0f6c62ea3b1d2c43f8fc152961af27033475f7 > > > > > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > > > # Skipping CQ checks because original CL landed less than 1 days ago. > > > NOPRESUBMIT=true > > > NOTREECHECKS=true > > > NOTRY=true > > > BUG=webrtc:7135 > > > > > > Review-Url: https://codereview.webrtc.org/3010983002 > > > Cr-Commit-Position: refs/heads/master@{#19653} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/3c39c0137afa274d1d524b150b50304b38a2847b > > > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > > # Skipping CQ checks because original CL landed less than 1 days ago. > > NOPRESUBMIT=true > > NOTREECHECKS=true > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/3006063002 > > Cr-Commit-Position: refs/heads/master@{#19715} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35713eaf565c0fef07c8afc158d7b8fdf7ec3d78 > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/3007303002 > Cr-Commit-Position: refs/heads/master@{#19744} > Committed: https://chromium.googlesource.com/external/webrtc/+/8e7eee035178a7f10e19883681b5eaa4a7523107 TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/3012963002 Cr-Commit-Position: refs/heads/master@{#19765}
2017-09-11 02:32:16 -07:00
EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(),
GetEngineCodec("red").id))
<< "RTX should be mapped for the RED payload type";
Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ ) Reason for revert: Downstream project relied on changed struct. Transition made possible by https://codereview.webrtc.org/2655243006/. Original issue's description: > Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ ) > > Reason for revert: > Breaks internal downstream project. > > Original issue's description: > > Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. > > > > Prior to this CL, received RTX (associated) payload types were only configured > > when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX > > SSRC was set up. > > > > After this CL, the RTX (associated) payload types are set in > > WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set > > them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since > > that is the code path that sets other SSRCs. > > > > As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed. > > We remove the possibility for each video payload type to have an associated > > specific RTX SSRC. Although the config previously allowed for this, all payload > > types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry > > did not support multiple SSRCs. This change to the config struct should thus not > > have any functional impact. The change does however affect the RtcEventLog, since > > that is used for storing the VideoReceiveStream::Configs. For simplicity, > > this CL does not change the event log proto definitions, instead duplicating > > the serialized RTX SSRCs such that they fit in the existing proto definition. > > > > BUG=webrtc:7011 > > > > Review-Url: https://codereview.webrtc.org/2646073004 > > Cr-Commit-Position: refs/heads/master@{#16302} > > Committed: https://chromium.googlesource.com/external/webrtc/+/fe2bef39cd2a5c891a49f7320514fb04324dc66c > > TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7011 > > Review-Url: https://codereview.webrtc.org/2649323010 > Cr-Commit-Position: refs/heads/master@{#16307} > Committed: https://chromium.googlesource.com/external/webrtc/+/e4974953ce0d03a60fae7659b199a6a62a79fa30 TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true # NOTREECHECKS=true # NOTRY=true BUG=webrtc:7011 Review-Url: https://codereview.webrtc.org/2654163006 Cr-Commit-Position: refs/heads/master@{#16322}
2017-01-27 04:53:07 -08:00
EXPECT_EQ(rtx_ssrcs[0], recv_stream->GetConfig().rtp.rtx_ssrc);
}
TEST_F(WebRtcVideoChannelTest, RecvStreamWithRtx) {
// Setup one channel with an associated RTX stream.
cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
params.AddFidSsrc(kSsrcs1[0], kRtxSsrcs1[0]);
FakeVideoReceiveStream* recv_stream = AddRecvStream(params);
Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ ) Reason for revert: Downstream project relied on changed struct. Transition made possible by https://codereview.webrtc.org/2655243006/. Original issue's description: > Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ ) > > Reason for revert: > Breaks internal downstream project. > > Original issue's description: > > Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. > > > > Prior to this CL, received RTX (associated) payload types were only configured > > when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX > > SSRC was set up. > > > > After this CL, the RTX (associated) payload types are set in > > WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set > > them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since > > that is the code path that sets other SSRCs. > > > > As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed. > > We remove the possibility for each video payload type to have an associated > > specific RTX SSRC. Although the config previously allowed for this, all payload > > types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry > > did not support multiple SSRCs. This change to the config struct should thus not > > have any functional impact. The change does however affect the RtcEventLog, since > > that is used for storing the VideoReceiveStream::Configs. For simplicity, > > this CL does not change the event log proto definitions, instead duplicating > > the serialized RTX SSRCs such that they fit in the existing proto definition. > > > > BUG=webrtc:7011 > > > > Review-Url: https://codereview.webrtc.org/2646073004 > > Cr-Commit-Position: refs/heads/master@{#16302} > > Committed: https://chromium.googlesource.com/external/webrtc/+/fe2bef39cd2a5c891a49f7320514fb04324dc66c > > TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7011 > > Review-Url: https://codereview.webrtc.org/2649323010 > Cr-Commit-Position: refs/heads/master@{#16307} > Committed: https://chromium.googlesource.com/external/webrtc/+/e4974953ce0d03a60fae7659b199a6a62a79fa30 TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true # NOTREECHECKS=true # NOTRY=true BUG=webrtc:7011 Review-Url: https://codereview.webrtc.org/2654163006 Cr-Commit-Position: refs/heads/master@{#16322}
2017-01-27 04:53:07 -08:00
EXPECT_EQ(kRtxSsrcs1[0], recv_stream->GetConfig().rtp.rtx_ssrc);
Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ ) Reason for revert: Identified a configuration problem in the video quality tests. Intend to fix and reland. Original issue's description: > Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ ) > > Reason for revert: > This change appears to break ulpfec, with severe regressions, e.g., for webrtc_perf_test FullStackTest.ForemanCifPlr5Ulpfec > > Original issue's description: > > Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ ) > > > > Reason for revert: > > Intend to fix perf failures and reland. > > > > Original issue's description: > > > Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ ) > > > > > > Reason for revert: > > > A few perf tests broken, including > > > > > > RampUpTest.UpDownUpAbsSendTimeSimulcastRedRtx > > > RampUpTest.UpDownUpTransportSequenceNumberRtx > > > RampUpTest.UpDownUpTransportSequenceNumberPacketLoss > > > > > > > > > Original issue's description: > > > > Use RtxReceiveStream. > > > > > > > > This also has the beneficial side-effect that when a media stream > > > > which is protected by FlexFEC receives an RTX retransmission, the > > > > retransmitted media packet is passed into the FlexFEC machinery, > > > > which should improve its ability to recover packets via FEC. > > > > > > > > BUG=webrtc:7135 > > > > > > > > Review-Url: https://codereview.webrtc.org/3008773002 > > > > Cr-Commit-Position: refs/heads/master@{#19649} > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/5c0f6c62ea3b1d2c43f8fc152961af27033475f7 > > > > > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > > > # Skipping CQ checks because original CL landed less than 1 days ago. > > > NOPRESUBMIT=true > > > NOTREECHECKS=true > > > NOTRY=true > > > BUG=webrtc:7135 > > > > > > Review-Url: https://codereview.webrtc.org/3010983002 > > > Cr-Commit-Position: refs/heads/master@{#19653} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/3c39c0137afa274d1d524b150b50304b38a2847b > > > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > > # Skipping CQ checks because original CL landed less than 1 days ago. > > NOPRESUBMIT=true > > NOTREECHECKS=true > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/3006063002 > > Cr-Commit-Position: refs/heads/master@{#19715} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35713eaf565c0fef07c8afc158d7b8fdf7ec3d78 > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/3007303002 > Cr-Commit-Position: refs/heads/master@{#19744} > Committed: https://chromium.googlesource.com/external/webrtc/+/8e7eee035178a7f10e19883681b5eaa4a7523107 TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/3012963002 Cr-Commit-Position: refs/heads/master@{#19765}
2017-09-11 02:32:16 -07:00
EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig()))
<< "RTX should be mapped for all decoders/payload types.";
EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(),
GetEngineCodec("red").id))
<< "RTX should be mapped for the RED payload type";
}
TEST_F(WebRtcVideoChannelTest, RecvStreamNoRtx) {
// Setup one channel without an associated RTX stream.
cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
FakeVideoReceiveStream* recv_stream = AddRecvStream(params);
Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ ) Reason for revert: Downstream project relied on changed struct. Transition made possible by https://codereview.webrtc.org/2655243006/. Original issue's description: > Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ ) > > Reason for revert: > Breaks internal downstream project. > > Original issue's description: > > Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. > > > > Prior to this CL, received RTX (associated) payload types were only configured > > when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX > > SSRC was set up. > > > > After this CL, the RTX (associated) payload types are set in > > WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set > > them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since > > that is the code path that sets other SSRCs. > > > > As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed. > > We remove the possibility for each video payload type to have an associated > > specific RTX SSRC. Although the config previously allowed for this, all payload > > types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry > > did not support multiple SSRCs. This change to the config struct should thus not > > have any functional impact. The change does however affect the RtcEventLog, since > > that is used for storing the VideoReceiveStream::Configs. For simplicity, > > this CL does not change the event log proto definitions, instead duplicating > > the serialized RTX SSRCs such that they fit in the existing proto definition. > > > > BUG=webrtc:7011 > > > > Review-Url: https://codereview.webrtc.org/2646073004 > > Cr-Commit-Position: refs/heads/master@{#16302} > > Committed: https://chromium.googlesource.com/external/webrtc/+/fe2bef39cd2a5c891a49f7320514fb04324dc66c > > TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7011 > > Review-Url: https://codereview.webrtc.org/2649323010 > Cr-Commit-Position: refs/heads/master@{#16307} > Committed: https://chromium.googlesource.com/external/webrtc/+/e4974953ce0d03a60fae7659b199a6a62a79fa30 TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true # NOTREECHECKS=true # NOTRY=true BUG=webrtc:7011 Review-Url: https://codereview.webrtc.org/2654163006 Cr-Commit-Position: refs/heads/master@{#16322}
2017-01-27 04:53:07 -08:00
ASSERT_EQ(0U, recv_stream->GetConfig().rtp.rtx_ssrc);
}
// Test propagation of extmap allow mixed setting.
TEST_F(WebRtcVideoChannelTest, SetExtmapAllowMixedAsCaller) {
TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/true);
}
TEST_F(WebRtcVideoChannelTest, SetExtmapAllowMixedDisabledAsCaller) {
TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/false);
}
TEST_F(WebRtcVideoChannelTest, SetExtmapAllowMixedAsCallee) {
TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/true);
}
TEST_F(WebRtcVideoChannelTest, SetExtmapAllowMixedDisabledAsCallee) {
TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/false);
}
TEST_F(WebRtcVideoChannelTest, NoHeaderExtesionsByDefault) {
FakeVideoSendStream* send_stream =
AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcs1[0]));
ASSERT_TRUE(send_stream->GetConfig().rtp.extensions.empty());
FakeVideoReceiveStream* recv_stream =
AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrcs1[0]));
ASSERT_TRUE(recv_stream->GetConfig().rtp.extensions.empty());
}
// Test support for RTP timestamp offset header extension.
TEST_F(WebRtcVideoChannelTest, SendRtpTimestampOffsetHeaderExtensions) {
TestSetSendRtpHeaderExtensions(RtpExtension::kTimestampOffsetUri);
}
TEST_F(WebRtcVideoChannelTest, RecvRtpTimestampOffsetHeaderExtensions) {
TestSetRecvRtpHeaderExtensions(RtpExtension::kTimestampOffsetUri);
}
// Test support for absolute send time header extension.
TEST_F(WebRtcVideoChannelTest, SendAbsoluteSendTimeHeaderExtensions) {
TestSetSendRtpHeaderExtensions(RtpExtension::kAbsSendTimeUri);
}
TEST_F(WebRtcVideoChannelTest, RecvAbsoluteSendTimeHeaderExtensions) {
TestSetRecvRtpHeaderExtensions(RtpExtension::kAbsSendTimeUri);
}
TEST_F(WebRtcVideoChannelTest, FiltersExtensionsPicksTransportSeqNum) {
// Enable three redundant extensions.
std::vector<std::string> extensions;
extensions.push_back(RtpExtension::kAbsSendTimeUri);
extensions.push_back(RtpExtension::kTimestampOffsetUri);
extensions.push_back(RtpExtension::kTransportSequenceNumberUri);
TestExtensionFilter(extensions, RtpExtension::kTransportSequenceNumberUri);
}
TEST_F(WebRtcVideoChannelTest, FiltersExtensionsPicksAbsSendTime) {
// Enable two redundant extensions.
std::vector<std::string> extensions;
extensions.push_back(RtpExtension::kAbsSendTimeUri);
extensions.push_back(RtpExtension::kTimestampOffsetUri);
TestExtensionFilter(extensions, RtpExtension::kAbsSendTimeUri);
}
// Test support for transport sequence number header extension.
TEST_F(WebRtcVideoChannelTest, SendTransportSequenceNumberHeaderExtensions) {
TestSetSendRtpHeaderExtensions(RtpExtension::kTransportSequenceNumberUri);
}
TEST_F(WebRtcVideoChannelTest, RecvTransportSequenceNumberHeaderExtensions) {
TestSetRecvRtpHeaderExtensions(RtpExtension::kTransportSequenceNumberUri);
}
// Test support for video rotation header extension.
TEST_F(WebRtcVideoChannelTest, SendVideoRotationHeaderExtensions) {
TestSetSendRtpHeaderExtensions(RtpExtension::kVideoRotationUri);
}
TEST_F(WebRtcVideoChannelTest, RecvVideoRotationHeaderExtensions) {
TestSetRecvRtpHeaderExtensions(RtpExtension::kVideoRotationUri);
}
TEST_F(WebRtcVideoChannelTest, IdenticalSendExtensionsDoesntRecreateStream) {
const int kAbsSendTimeId = 1;
const int kVideoRotationId = 2;
send_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
send_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kVideoRotationUri, kVideoRotationId));
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
FakeVideoSendStream* send_stream =
AddSendStream(cricket::StreamParams::CreateLegacy(123));
EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams());
ASSERT_EQ(2u, send_stream->GetConfig().rtp.extensions.size());
// Setting the same extensions (even if in different order) shouldn't
// reallocate the stream.
absl::c_reverse(send_parameters_.extensions);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams());
// Setting different extensions should recreate the stream.
send_parameters_.extensions.resize(1);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_EQ(2, fake_call_->GetNumCreatedSendStreams());
}
TEST_F(WebRtcVideoChannelTest, IdenticalRecvExtensionsDoesntRecreateStream) {
const int kTOffsetId = 1;
const int kAbsSendTimeId = 2;
const int kVideoRotationId = 3;
recv_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
recv_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOffsetId));
recv_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kVideoRotationUri, kVideoRotationId));
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
FakeVideoReceiveStream* recv_stream =
AddRecvStream(cricket::StreamParams::CreateLegacy(123));
EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams());
ASSERT_EQ(3u, recv_stream->GetConfig().rtp.extensions.size());
// Setting the same extensions (even if in different order) shouldn't
// reallocate the stream.
absl::c_reverse(recv_parameters_.extensions);
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams());
// Setting different extensions should recreate the stream.
recv_parameters_.extensions.resize(1);
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
EXPECT_EQ(2, fake_call_->GetNumCreatedReceiveStreams());
}
TEST_F(WebRtcVideoChannelTest,
SetSendRtpHeaderExtensionsExcludeUnsupportedExtensions) {
const int kUnsupportedId = 1;
const int kTOffsetId = 2;
send_parameters_.extensions.push_back(
RtpExtension(kUnsupportedExtensionName, kUnsupportedId));
send_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOffsetId));
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
FakeVideoSendStream* send_stream =
AddSendStream(cricket::StreamParams::CreateLegacy(123));
// Only timestamp offset extension is set to send stream,
// unsupported rtp extension is ignored.
ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size());
EXPECT_STREQ(RtpExtension::kTimestampOffsetUri,
send_stream->GetConfig().rtp.extensions[0].uri.c_str());
}
TEST_F(WebRtcVideoChannelTest,
SetRecvRtpHeaderExtensionsExcludeUnsupportedExtensions) {
const int kUnsupportedId = 1;
const int kTOffsetId = 2;
recv_parameters_.extensions.push_back(
RtpExtension(kUnsupportedExtensionName, kUnsupportedId));
recv_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOffsetId));
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
FakeVideoReceiveStream* recv_stream =
AddRecvStream(cricket::StreamParams::CreateLegacy(123));
// Only timestamp offset extension is set to receive stream,
// unsupported rtp extension is ignored.
ASSERT_EQ(1u, recv_stream->GetConfig().rtp.extensions.size());
EXPECT_STREQ(RtpExtension::kTimestampOffsetUri,
recv_stream->GetConfig().rtp.extensions[0].uri.c_str());
}
TEST_F(WebRtcVideoChannelTest, SetSendRtpHeaderExtensionsRejectsIncorrectIds) {
const int kIncorrectIds[] = {-2, -1, 0, 15, 16};
for (size_t i = 0; i < arraysize(kIncorrectIds); ++i) {
send_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kIncorrectIds[i]));
EXPECT_FALSE(channel_->SetSendParameters(send_parameters_))
<< "Bad extension id '" << kIncorrectIds[i] << "' accepted.";
}
}
TEST_F(WebRtcVideoChannelTest, SetRecvRtpHeaderExtensionsRejectsIncorrectIds) {
const int kIncorrectIds[] = {-2, -1, 0, 15, 16};
for (size_t i = 0; i < arraysize(kIncorrectIds); ++i) {
recv_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kIncorrectIds[i]));
EXPECT_FALSE(channel_->SetRecvParameters(recv_parameters_))
<< "Bad extension id '" << kIncorrectIds[i] << "' accepted.";
}
}
TEST_F(WebRtcVideoChannelTest, SetSendRtpHeaderExtensionsRejectsDuplicateIds) {
const int id = 1;
send_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, id));
send_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, id));
EXPECT_FALSE(channel_->SetSendParameters(send_parameters_));
// Duplicate entries are also not supported.
send_parameters_.extensions.clear();
send_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, id));
send_parameters_.extensions.push_back(send_parameters_.extensions.back());
EXPECT_FALSE(channel_->SetSendParameters(send_parameters_));
}
TEST_F(WebRtcVideoChannelTest, SetRecvRtpHeaderExtensionsRejectsDuplicateIds) {
const int id = 1;
recv_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, id));
recv_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, id));
EXPECT_FALSE(channel_->SetRecvParameters(recv_parameters_));
// Duplicate entries are also not supported.
recv_parameters_.extensions.clear();
recv_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, id));
recv_parameters_.extensions.push_back(recv_parameters_.extensions.back());
EXPECT_FALSE(channel_->SetRecvParameters(recv_parameters_));
}
TEST_F(WebRtcVideoChannelTest, AddRecvStreamOnlyUsesOneReceiveStream) {
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
}
TEST_F(WebRtcVideoChannelTest, RtcpIsCompoundByDefault) {
FakeVideoReceiveStream* stream = AddRecvStream();
EXPECT_EQ(webrtc::RtcpMode::kCompound, stream->GetConfig().rtp.rtcp_mode);
}
TEST_F(WebRtcVideoChannelTest, TransportCcIsEnabledByDefault) {
FakeVideoReceiveStream* stream = AddRecvStream();
EXPECT_TRUE(stream->GetConfig().rtp.transport_cc);
}
TEST_F(WebRtcVideoChannelTest, TransportCcCanBeEnabledAndDisabled) {
FakeVideoReceiveStream* stream = AddRecvStream();
EXPECT_TRUE(stream->GetConfig().rtp.transport_cc);
// Verify that transport cc feedback is turned off when send(!) codecs without
// transport cc feedback are set.
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(RemoveFeedbackParams(GetEngineCodec("VP8")));
EXPECT_TRUE(parameters.codecs[0].feedback_params.params().empty());
EXPECT_TRUE(channel_->SetSendParameters(parameters));
stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_FALSE(stream->GetConfig().rtp.transport_cc);
// Verify that transport cc feedback is turned on when setting default codecs
// since the default codecs have transport cc feedback enabled.
parameters.codecs = engine_.codecs();
EXPECT_TRUE(channel_->SetSendParameters(parameters));
stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_TRUE(stream->GetConfig().rtp.transport_cc);
}
TEST_F(WebRtcVideoChannelTest, LossNotificationIsDisabledByDefault) {
TestLossNotificationState(false);
}
TEST_F(WebRtcVideoChannelTest, LossNotificationIsEnabledByFieldTrial) {
RTC_DCHECK(!override_field_trials_);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-RtcpLossNotification/Enabled/");
SetUp();
TestLossNotificationState(true);
}
TEST_F(WebRtcVideoChannelTest, LossNotificationCanBeEnabledAndDisabled) {
RTC_DCHECK(!override_field_trials_);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-RtcpLossNotification/Enabled/");
SetUp();
AssignDefaultCodec();
VerifyCodecHasDefaultFeedbackParams(default_codec_, true);
{
cricket::VideoSendParameters parameters;
parameters.codecs = engine_.codecs();
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(channel_->SetSend(true));
}
// Start with LNTF enabled.
FakeVideoSendStream* send_stream =
AddSendStream(cricket::StreamParams::CreateLegacy(1));
ASSERT_TRUE(send_stream->GetConfig().rtp.lntf.enabled);
FakeVideoReceiveStream* recv_stream =
AddRecvStream(cricket::StreamParams::CreateLegacy(1));
ASSERT_TRUE(recv_stream->GetConfig().rtp.lntf.enabled);
// Verify that LNTF is turned off when send(!) codecs without LNTF are set.
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(RemoveFeedbackParams(GetEngineCodec("VP8")));
EXPECT_TRUE(parameters.codecs[0].feedback_params.params().empty());
EXPECT_TRUE(channel_->SetSendParameters(parameters));
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_FALSE(recv_stream->GetConfig().rtp.lntf.enabled);
send_stream = fake_call_->GetVideoSendStreams()[0];
EXPECT_FALSE(send_stream->GetConfig().rtp.lntf.enabled);
// Setting the default codecs again, including VP8, turns LNTF back on.
parameters.codecs = engine_.codecs();
EXPECT_TRUE(channel_->SetSendParameters(parameters));
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_TRUE(recv_stream->GetConfig().rtp.lntf.enabled);
send_stream = fake_call_->GetVideoSendStreams()[0];
EXPECT_TRUE(send_stream->GetConfig().rtp.lntf.enabled);
}
TEST_F(WebRtcVideoChannelTest, NackIsEnabledByDefault) {
AssignDefaultCodec();
VerifyCodecHasDefaultFeedbackParams(default_codec_, false);
cricket::VideoSendParameters parameters;
parameters.codecs = engine_.codecs();
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(channel_->SetSend(true));
// Send side.
FakeVideoSendStream* send_stream =
AddSendStream(cricket::StreamParams::CreateLegacy(1));
EXPECT_GT(send_stream->GetConfig().rtp.nack.rtp_history_ms, 0);
// Receiver side.
FakeVideoReceiveStream* recv_stream =
AddRecvStream(cricket::StreamParams::CreateLegacy(1));
EXPECT_GT(recv_stream->GetConfig().rtp.nack.rtp_history_ms, 0);
// Nack history size should match between sender and receiver.
EXPECT_EQ(send_stream->GetConfig().rtp.nack.rtp_history_ms,
recv_stream->GetConfig().rtp.nack.rtp_history_ms);
}
TEST_F(WebRtcVideoChannelTest, NackCanBeEnabledAndDisabled) {
FakeVideoSendStream* send_stream = AddSendStream();
FakeVideoReceiveStream* recv_stream = AddRecvStream();
EXPECT_GT(recv_stream->GetConfig().rtp.nack.rtp_history_ms, 0);
EXPECT_GT(send_stream->GetConfig().rtp.nack.rtp_history_ms, 0);
// Verify that NACK is turned off when send(!) codecs without NACK are set.
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(RemoveFeedbackParams(GetEngineCodec("VP8")));
EXPECT_TRUE(parameters.codecs[0].feedback_params.params().empty());
EXPECT_TRUE(channel_->SetSendParameters(parameters));
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_EQ(0, recv_stream->GetConfig().rtp.nack.rtp_history_ms);
send_stream = fake_call_->GetVideoSendStreams()[0];
EXPECT_EQ(0, send_stream->GetConfig().rtp.nack.rtp_history_ms);
// Verify that NACK is turned on when setting default codecs since the
// default codecs have NACK enabled.
parameters.codecs = engine_.codecs();
EXPECT_TRUE(channel_->SetSendParameters(parameters));
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_GT(recv_stream->GetConfig().rtp.nack.rtp_history_ms, 0);
send_stream = fake_call_->GetVideoSendStreams()[0];
EXPECT_GT(send_stream->GetConfig().rtp.nack.rtp_history_ms, 0);
}
// This test verifies that new frame sizes reconfigures encoders even though not
// (yet) sending. The purpose of this is to permit encoding as quickly as
// possible once we start sending. Likely the frames being input are from the
// same source that will be sent later, which just means that we're ready
// earlier.
TEST_F(WebRtcVideoChannelTest, ReconfiguresEncodersWhenNotSending) {
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
channel_->SetSend(false);
FakeVideoSendStream* stream = AddSendStream();
// No frames entered.
std::vector<webrtc::VideoStream> streams = stream->GetVideoStreams();
EXPECT_EQ(0u, streams[0].width);
EXPECT_EQ(0u, streams[0].height);
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
// Frame entered, should be reconfigured to new dimensions.
streams = stream->GetVideoStreams();
EXPECT_EQ(rtc::checked_cast<size_t>(1280), streams[0].width);
EXPECT_EQ(rtc::checked_cast<size_t>(720), streams[0].height);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, UsesCorrectSettingsForScreencast) {
static const int kScreenshareMinBitrateKbps = 800;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
cricket::VideoCodec codec = GetEngineCodec("VP8");
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(codec);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
AddSendStream();
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
VideoOptions min_bitrate_options;
min_bitrate_options.screencast_min_bitrate_kbps = kScreenshareMinBitrateKbps;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &min_bitrate_options,
&frame_forwarder));
EXPECT_TRUE(channel_->SetSend(true));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size());
FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames());
// Verify non-screencast settings.
webrtc::VideoEncoderConfig encoder_config =
send_stream->GetEncoderConfig().Copy();
EXPECT_EQ(webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo,
encoder_config.content_type);
std::vector<webrtc::VideoStream> streams = send_stream->GetVideoStreams();
EXPECT_EQ(rtc::checked_cast<size_t>(1280), streams.front().width);
EXPECT_EQ(rtc::checked_cast<size_t>(720), streams.front().height);
EXPECT_EQ(0, encoder_config.min_transmit_bitrate_bps)
<< "Non-screenshare shouldn't use min-transmit bitrate.";
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames());
VideoOptions screencast_options;
screencast_options.is_screencast = true;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &screencast_options,
&frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
// Send stream recreated after option change.
ASSERT_EQ(2, fake_call_->GetNumCreatedSendStreams());
send_stream = fake_call_->GetVideoSendStreams().front();
EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames());
// Verify screencast settings.
encoder_config = send_stream->GetEncoderConfig().Copy();
EXPECT_EQ(webrtc::VideoEncoderConfig::ContentType::kScreen,
encoder_config.content_type);
EXPECT_EQ(kScreenshareMinBitrateKbps * 1000,
encoder_config.min_transmit_bitrate_bps);
streams = send_stream->GetVideoStreams();
EXPECT_EQ(rtc::checked_cast<size_t>(1280), streams.front().width);
EXPECT_EQ(rtc::checked_cast<size_t>(720), streams.front().height);
Reland "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers." This reverts commit e27e0aca9411b6990fcdf56d8a3475569ee5fd2f. Reason for revert: <INSERT REASONING HERE> Original change's description: > Revert "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers." > > This reverts commit d2ed0a4c9e7f04060d8e3358eb0006c31579bb86. > > Reason for revert: Breaks downstream projects. > > Original change's description: > > Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers. > > > > temporal_layer_thresholds_bps served only one purpose: its size was used > > to infer number of temporal layers. I replaced it with num_temporal_layers, > > which does what is says. > > > > The practical reason for this change is the need to have possibility to > > distinguish between cases when VP9 SVC temporal layering was/not set > > through field trial. That was not possible with > > temporal_layer_thresholds_bps[] because empty vector means 1 temporal > > layer. > > > > Bug: webrtc:8518 > > Change-Id: I275ec3a8c74e8ba409eb049878199f132a20ec51 > > Reviewed-on: https://webrtc-review.googlesource.com/58084 > > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22230} > > TBR=sprang@webrtc.org,stefan@webrtc.org,ssilkin@webrtc.org > > Change-Id: Ic2940f7f78a74312170940d51ad8967cde8ad42f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8518 > Reviewed-on: https://webrtc-review.googlesource.com/58902 > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22234} TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,ssilkin@webrtc.org Change-Id: I1900c6b845b9baa9430fb72c3f4e7f2a44b3a8b1 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8518 Reviewed-on: https://webrtc-review.googlesource.com/59160 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22256}
2018-03-01 15:11:29 +01:00
EXPECT_FALSE(streams[0].num_temporal_layers.has_value());
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
ConferenceModeScreencastConfiguresTemporalLayer) {
static const int kConferenceScreencastTemporalBitrateBps = 200 * 1000;
send_parameters_.conference_mode = true;
channel_->SetSendParameters(send_parameters_);
AddSendStream();
VideoOptions options;
options.is_screencast = true;
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
EXPECT_TRUE(channel_->SetSend(true));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size());
FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
webrtc::VideoEncoderConfig encoder_config =
send_stream->GetEncoderConfig().Copy();
// Verify screencast settings.
encoder_config = send_stream->GetEncoderConfig().Copy();
EXPECT_EQ(webrtc::VideoEncoderConfig::ContentType::kScreen,
encoder_config.content_type);
std::vector<webrtc::VideoStream> streams = send_stream->GetVideoStreams();
ASSERT_EQ(1u, streams.size());
Reland "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers." This reverts commit e27e0aca9411b6990fcdf56d8a3475569ee5fd2f. Reason for revert: <INSERT REASONING HERE> Original change's description: > Revert "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers." > > This reverts commit d2ed0a4c9e7f04060d8e3358eb0006c31579bb86. > > Reason for revert: Breaks downstream projects. > > Original change's description: > > Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers. > > > > temporal_layer_thresholds_bps served only one purpose: its size was used > > to infer number of temporal layers. I replaced it with num_temporal_layers, > > which does what is says. > > > > The practical reason for this change is the need to have possibility to > > distinguish between cases when VP9 SVC temporal layering was/not set > > through field trial. That was not possible with > > temporal_layer_thresholds_bps[] because empty vector means 1 temporal > > layer. > > > > Bug: webrtc:8518 > > Change-Id: I275ec3a8c74e8ba409eb049878199f132a20ec51 > > Reviewed-on: https://webrtc-review.googlesource.com/58084 > > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22230} > > TBR=sprang@webrtc.org,stefan@webrtc.org,ssilkin@webrtc.org > > Change-Id: Ic2940f7f78a74312170940d51ad8967cde8ad42f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8518 > Reviewed-on: https://webrtc-review.googlesource.com/58902 > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22234} TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,ssilkin@webrtc.org Change-Id: I1900c6b845b9baa9430fb72c3f4e7f2a44b3a8b1 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8518 Reviewed-on: https://webrtc-review.googlesource.com/59160 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22256}
2018-03-01 15:11:29 +01:00
ASSERT_EQ(2u, streams[0].num_temporal_layers);
EXPECT_EQ(kConferenceScreencastTemporalBitrateBps,
Reland "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers." This reverts commit e27e0aca9411b6990fcdf56d8a3475569ee5fd2f. Reason for revert: <INSERT REASONING HERE> Original change's description: > Revert "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers." > > This reverts commit d2ed0a4c9e7f04060d8e3358eb0006c31579bb86. > > Reason for revert: Breaks downstream projects. > > Original change's description: > > Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers. > > > > temporal_layer_thresholds_bps served only one purpose: its size was used > > to infer number of temporal layers. I replaced it with num_temporal_layers, > > which does what is says. > > > > The practical reason for this change is the need to have possibility to > > distinguish between cases when VP9 SVC temporal layering was/not set > > through field trial. That was not possible with > > temporal_layer_thresholds_bps[] because empty vector means 1 temporal > > layer. > > > > Bug: webrtc:8518 > > Change-Id: I275ec3a8c74e8ba409eb049878199f132a20ec51 > > Reviewed-on: https://webrtc-review.googlesource.com/58084 > > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22230} > > TBR=sprang@webrtc.org,stefan@webrtc.org,ssilkin@webrtc.org > > Change-Id: Ic2940f7f78a74312170940d51ad8967cde8ad42f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8518 > Reviewed-on: https://webrtc-review.googlesource.com/58902 > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22234} TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,ssilkin@webrtc.org Change-Id: I1900c6b845b9baa9430fb72c3f4e7f2a44b3a8b1 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8518 Reviewed-on: https://webrtc-review.googlesource.com/59160 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22256}
2018-03-01 15:11:29 +01:00
streams[0].target_bitrate_bps);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, SuspendBelowMinBitrateDisabledByDefault) {
FakeVideoSendStream* stream = AddSendStream();
EXPECT_FALSE(stream->GetConfig().suspend_below_min_bitrate);
}
TEST_F(WebRtcVideoChannelTest, SetMediaConfigSuspendBelowMinBitrate) {
Reland of Drop frames until specified bitrate is achieved. (patchset #1 id:1 of https://codereview.webrtc.org/2666303002/ ) Reason for revert: Perf test broke as it made assumptions that quality scaling was turned off. This turns out not to be the case. Fixed by turning quality scaling off for the tests. Original issue's description: > Revert of Drop frames until specified bitrate is achieved. (patchset #12 id:240001 of https://codereview.webrtc.org/2630333002/ ) > > Reason for revert: > due to failures on perf tests (not on perf stats, but fails running due to dcheck failures), see e.g., https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20(K%20Nexus5) > > Original issue's description: > > Drop frames until specified bitrate is achieved. > > > > This CL fixes a regression introduced with the new quality scaler > > where the video would no longer start in a scaled mode. This CL adds > > code that compares incoming captured frames to the target bitrate, > > and if they are found to be too large, they are dropped and sinkWants > > set to a lower resolution. The number of dropped frames should be low > > (0-4 in most cases) and should not introduce a noticeable delay, or > > at least should be preferrable to having the first 2-4 seconds of video > > have very low quality. > > > > BUG=webrtc:6953 > > > > Review-Url: https://codereview.webrtc.org/2630333002 > > Cr-Commit-Position: refs/heads/master@{#16391} > > Committed: https://chromium.googlesource.com/external/webrtc/+/83399caec5762d2dad038b8e9d86163e92c18c9f > > TBR=perkj@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,kthelgason@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6953 > > Review-Url: https://codereview.webrtc.org/2666303002 > Cr-Commit-Position: refs/heads/master@{#16395} > Committed: https://chromium.googlesource.com/external/webrtc/+/35fc2aa82fb5a562f3f76f2b91a55f05ebfd4874 TBR=perkj@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,minyue@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:6953 Review-Url: https://codereview.webrtc.org/2675223002 Cr-Commit-Position: refs/heads/master@{#16473}
2017-02-07 07:02:22 -08:00
MediaConfig media_config = GetMediaConfig();
media_config.video.suspend_below_min_bitrate = true;
channel_.reset(engine_.CreateMediaChannel(
fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
channel_->OnReadyToSend(true);
channel_->SetSendParameters(send_parameters_);
FakeVideoSendStream* stream = AddSendStream();
EXPECT_TRUE(stream->GetConfig().suspend_below_min_bitrate);
media_config.video.suspend_below_min_bitrate = false;
channel_.reset(engine_.CreateMediaChannel(
fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
channel_->OnReadyToSend(true);
channel_->SetSendParameters(send_parameters_);
stream = AddSendStream();
EXPECT_FALSE(stream->GetConfig().suspend_below_min_bitrate);
}
TEST_F(WebRtcVideoChannelTest, Vp8DenoisingEnabledByDefault) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoCodecVP8 vp8_settings;
ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set.";
EXPECT_TRUE(vp8_settings.denoisingOn);
}
TEST_F(WebRtcVideoChannelTest, VerifyVp8SpecificSettings) {
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
// Single-stream settings should apply with RTX as well (verifies that we
// check number of regular SSRCs and not StreamParams::ssrcs which contains
// both RTX and regular SSRCs).
FakeVideoSendStream* stream = SetUpSimulcast(false, true);
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
webrtc::VideoCodecVP8 vp8_settings;
ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set.";
EXPECT_TRUE(vp8_settings.denoisingOn)
<< "VP8 denoising should be on by default.";
stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, false);
ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set.";
EXPECT_FALSE(vp8_settings.denoisingOn);
EXPECT_TRUE(vp8_settings.automaticResizeOn);
EXPECT_TRUE(vp8_settings.frameDroppingOn);
stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, true);
ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set.";
EXPECT_TRUE(vp8_settings.denoisingOn);
EXPECT_TRUE(vp8_settings.automaticResizeOn);
EXPECT_TRUE(vp8_settings.frameDroppingOn);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
stream = SetUpSimulcast(true, false);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
EXPECT_EQ(3u, stream->GetVideoStreams().size());
ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set.";
// Autmatic resize off when using simulcast.
EXPECT_FALSE(vp8_settings.automaticResizeOn);
EXPECT_TRUE(vp8_settings.frameDroppingOn);
// In screen-share mode, denoising is forced off.
VideoOptions options;
options.is_screencast = true;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
New flag is_screencast in VideoOptions. This cl copies the value of cricket::VideoCapturer::IsScreencast into a flag in VideoOptions. It is passed on via the chain VideortpSender::SetVideoSend WebRtcVideoChannel2::SetVideoSend WebRtcVideoChannel2::SetOptions WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions Where it's used, in WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame, we can look it up in parameters_, instead of calling capturer_->IsScreencast(). Doesn't touch screencast logic related to cpu adaptation, since that code is in flux in a different cl. Also drop the is_screencast flag from the Dimensions struct, and drop separate options argument from ConfigureVideoEncoderSettings and SetCodecAndOptions, instead always using the options recorded in VideoSendStreamParameters::options. In the tests, changed FakeVideoCapturer::is_screencast to be a construction time flag. Generally, unittests of screencast have to both use a capturer configured for screencast, and set the screencast flag using SetSendParameters. Since the automatic connection via VideoSource and VideoRtpSender isn't involved in the unit tests. Note that using SetSendParameters to set the screencast flag doesn't make sense, since it's not per-stream. SetVideoSend would be more appropriate. That should be fixed if/when we drop VideoOptions from SetSendParameters. BUG=webrtc:5426 R=pbos@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1711763003 . Cr-Commit-Position: refs/heads/master@{#11837}
2016-03-02 11:41:36 +01:00
stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, false);
EXPECT_EQ(3u, stream->GetVideoStreams().size());
ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set.";
EXPECT_FALSE(vp8_settings.denoisingOn);
// Resizing and frame dropping always off for screen sharing.
EXPECT_FALSE(vp8_settings.automaticResizeOn);
EXPECT_FALSE(vp8_settings.frameDroppingOn);
stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, true);
ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set.";
EXPECT_FALSE(vp8_settings.denoisingOn);
EXPECT_FALSE(vp8_settings.automaticResizeOn);
EXPECT_FALSE(vp8_settings.frameDroppingOn);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
// Test that setting the same options doesn't result in the encoder being
// reconfigured.
TEST_F(WebRtcVideoChannelTest, SetIdenticalOptionsDoesntReconfigureEncoder) {
VideoOptions options;
webrtc::test::FrameForwarder frame_forwarder;
AddSendStream();
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Expect 1 reconfigurations at this point from the initial configuration.
EXPECT_EQ(1, send_stream->num_encoder_reconfigurations());
// Set the options one more time and expect no additional reconfigurations.
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
EXPECT_EQ(1, send_stream->num_encoder_reconfigurations());
// Change |options| and expect 2 reconfigurations.
options.video_noise_reduction = true;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
EXPECT_EQ(2, send_stream->num_encoder_reconfigurations());
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
class Vp9SettingsTest : public WebRtcVideoChannelTest {
public:
Vp9SettingsTest() : Vp9SettingsTest("") {}
explicit Vp9SettingsTest(const char* field_trials)
: WebRtcVideoChannelTest(field_trials) {
encoder_factory_->AddSupportedVideoCodecType("VP9");
}
virtual ~Vp9SettingsTest() {}
protected:
void TearDown() override {
// Remove references to encoder_factory_ since this will be destroyed
// before channel_ and engine_.
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
}
};
TEST_F(Vp9SettingsTest, VerifyVp9SpecificSettings) {
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP9"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = SetUpSimulcast(false, false);
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
webrtc::VideoCodecVP9 vp9_settings;
ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
EXPECT_TRUE(vp9_settings.denoisingOn)
<< "VP9 denoising should be on by default.";
stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, false);
ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
EXPECT_FALSE(vp9_settings.denoisingOn);
// Frame dropping always on for real time video.
EXPECT_TRUE(vp9_settings.frameDroppingOn);
stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, true);
ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
EXPECT_TRUE(vp9_settings.denoisingOn);
EXPECT_TRUE(vp9_settings.frameDroppingOn);
// In screen-share mode, denoising is forced off.
VideoOptions options;
options.is_screencast = true;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, false);
ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
EXPECT_FALSE(vp9_settings.denoisingOn);
// Frame dropping always on for screen sharing.
EXPECT_TRUE(vp9_settings.frameDroppingOn);
stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, false);
ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
EXPECT_FALSE(vp9_settings.denoisingOn);
EXPECT_TRUE(vp9_settings.frameDroppingOn);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(Vp9SettingsTest, MultipleSsrcsEnablesSvc) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP9"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
FakeVideoSendStream* stream =
AddSendStream(CreateSimStreamParams("cname", ssrcs));
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(ssrcs[0], nullptr, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
webrtc::VideoCodecVP9 vp9_settings;
ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
const size_t kNumSpatialLayers = ssrcs.size();
const size_t kNumTemporalLayers = 3;
EXPECT_EQ(vp9_settings.numberOfSpatialLayers, kNumSpatialLayers);
EXPECT_EQ(vp9_settings.numberOfTemporalLayers, kNumTemporalLayers);
EXPECT_TRUE(channel_->SetVideoSend(ssrcs[0], nullptr, nullptr));
}
TEST_F(Vp9SettingsTest, SvcModeCreatesSingleRtpStream) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP9"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
FakeVideoSendStream* stream =
AddSendStream(CreateSimStreamParams("cname", ssrcs));
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
// Despite 3 ssrcs provided, single layer is used.
EXPECT_EQ(1u, config.rtp.ssrcs.size());
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(ssrcs[0], nullptr, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
webrtc::VideoCodecVP9 vp9_settings;
ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
const size_t kNumSpatialLayers = ssrcs.size();
EXPECT_EQ(vp9_settings.numberOfSpatialLayers, kNumSpatialLayers);
EXPECT_TRUE(channel_->SetVideoSend(ssrcs[0], nullptr, nullptr));
}
TEST_F(Vp9SettingsTest, AllEncodingParametersCopied) {
cricket::VideoSendParameters send_parameters;
send_parameters.codecs.push_back(GetEngineCodec("VP9"));
ASSERT_TRUE(channel_->SetSendParameters(send_parameters));
const size_t kNumSpatialLayers = 3;
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
FakeVideoSendStream* stream =
AddSendStream(CreateSimStreamParams("cname", ssrcs));
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(ssrcs[0]);
ASSERT_EQ(kNumSpatialLayers, parameters.encodings.size());
ASSERT_TRUE(parameters.encodings[0].active);
ASSERT_TRUE(parameters.encodings[1].active);
ASSERT_TRUE(parameters.encodings[2].active);
// Invert value to verify copying.
parameters.encodings[1].active = false;
EXPECT_TRUE(channel_->SetRtpSendParameters(ssrcs[0], parameters).ok());
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
// number_of_streams should be 1 since all spatial layers are sent on the
// same SSRC. But encoding parameters of all layers is supposed to be copied
// and stored in simulcast_layers[].
EXPECT_EQ(1u, encoder_config.number_of_streams);
EXPECT_EQ(encoder_config.simulcast_layers.size(), kNumSpatialLayers);
EXPECT_TRUE(encoder_config.simulcast_layers[0].active);
EXPECT_FALSE(encoder_config.simulcast_layers[1].active);
EXPECT_TRUE(encoder_config.simulcast_layers[2].active);
}
class Vp9SettingsTestWithFieldTrial
: public Vp9SettingsTest,
public ::testing::WithParamInterface<
::testing::tuple<const char*, int, int, webrtc::InterLayerPredMode>> {
protected:
Vp9SettingsTestWithFieldTrial()
: Vp9SettingsTest(::testing::get<0>(GetParam())),
num_spatial_layers_(::testing::get<1>(GetParam())),
num_temporal_layers_(::testing::get<2>(GetParam())),
inter_layer_pred_mode_(::testing::get<3>(GetParam())) {}
void VerifySettings(int num_spatial_layers,
int num_temporal_layers,
webrtc::InterLayerPredMode interLayerPred) {
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP9"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = SetUpSimulcast(false, false);
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
webrtc::VideoCodecVP9 vp9_settings;
ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
EXPECT_EQ(num_spatial_layers, vp9_settings.numberOfSpatialLayers);
EXPECT_EQ(num_temporal_layers, vp9_settings.numberOfTemporalLayers);
EXPECT_EQ(inter_layer_pred_mode_, vp9_settings.interLayerPred);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
const uint8_t num_spatial_layers_;
const uint8_t num_temporal_layers_;
const webrtc::InterLayerPredMode inter_layer_pred_mode_;
};
TEST_P(Vp9SettingsTestWithFieldTrial, VerifyCodecSettings) {
VerifySettings(num_spatial_layers_, num_temporal_layers_,
inter_layer_pred_mode_);
}
INSTANTIATE_TEST_SUITE_P(
All,
Vp9SettingsTestWithFieldTrial,
::testing::Values(
std::make_tuple("", 1, 1, webrtc::InterLayerPredMode::kOnKeyPic),
std::make_tuple("WebRTC-SupportVP9SVC/Default/",
1,
1,
webrtc::InterLayerPredMode::kOnKeyPic),
std::make_tuple("WebRTC-SupportVP9SVC/EnabledByFlag_2SL3TL/",
2,
3,
webrtc::InterLayerPredMode::kOnKeyPic),
std::make_tuple("WebRTC-Vp9InterLayerPred/Default/",
1,
1,
webrtc::InterLayerPredMode::kOnKeyPic),
std::make_tuple("WebRTC-Vp9InterLayerPred/Disabled/",
1,
1,
webrtc::InterLayerPredMode::kOnKeyPic),
std::make_tuple(
"WebRTC-Vp9InterLayerPred/Enabled,inter_layer_pred_mode:off/",
1,
1,
webrtc::InterLayerPredMode::kOff),
std::make_tuple(
"WebRTC-Vp9InterLayerPred/Enabled,inter_layer_pred_mode:on/",
1,
1,
webrtc::InterLayerPredMode::kOn),
std::make_tuple(
"WebRTC-Vp9InterLayerPred/Enabled,inter_layer_pred_mode:onkeypic/",
1,
1,
webrtc::InterLayerPredMode::kOnKeyPic)));
TEST_F(WebRtcVideoChannelTest, VerifyMinBitrate) {
std::vector<webrtc::VideoStream> streams = AddSendStream()->GetVideoStreams();
ASSERT_EQ(1u, streams.size());
EXPECT_EQ(webrtc::kDefaultMinVideoBitrateBps, streams[0].min_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest, VerifyMinBitrateWithForcedFallbackFieldTrial) {
RTC_DCHECK(!override_field_trials_);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-VP8-Forced-Fallback-Encoder-v2/Enabled-1,2,34567/");
std::vector<webrtc::VideoStream> streams = AddSendStream()->GetVideoStreams();
ASSERT_EQ(1u, streams.size());
EXPECT_EQ(34567, streams[0].min_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest,
BalancedDegradationPreferenceNotSupportedWithoutFieldtrial) {
RTC_DCHECK(!override_field_trials_);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-Video-BalancedDegradation/Disabled/");
const bool kResolutionScalingEnabled = true;
const bool kFpsScalingEnabled = false;
TestDegradationPreference(kResolutionScalingEnabled, kFpsScalingEnabled);
}
TEST_F(WebRtcVideoChannelTest,
BalancedDegradationPreferenceSupportedBehindFieldtrial) {
RTC_DCHECK(!override_field_trials_);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-Video-BalancedDegradation/Enabled/");
const bool kResolutionScalingEnabled = true;
const bool kFpsScalingEnabled = true;
TestDegradationPreference(kResolutionScalingEnabled, kFpsScalingEnabled);
}
TEST_F(WebRtcVideoChannelTest, AdaptsOnOveruse) {
TestCpuAdaptation(true, false);
}
TEST_F(WebRtcVideoChannelTest, DoesNotAdaptOnOveruseWhenDisabled) {
TestCpuAdaptation(false, false);
}
TEST_F(WebRtcVideoChannelTest, DoesNotAdaptWhenScreeensharing) {
TestCpuAdaptation(false, true);
}
TEST_F(WebRtcVideoChannelTest, DoesNotAdaptOnOveruseWhenScreensharing) {
TestCpuAdaptation(true, true);
}
TEST_F(WebRtcVideoChannelTest, PreviousAdaptationDoesNotApplyToScreenshare) {
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
cricket::VideoCodec codec = GetEngineCodec("VP8");
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(codec);
Reland of Drop frames until specified bitrate is achieved. (patchset #1 id:1 of https://codereview.webrtc.org/2666303002/ ) Reason for revert: Perf test broke as it made assumptions that quality scaling was turned off. This turns out not to be the case. Fixed by turning quality scaling off for the tests. Original issue's description: > Revert of Drop frames until specified bitrate is achieved. (patchset #12 id:240001 of https://codereview.webrtc.org/2630333002/ ) > > Reason for revert: > due to failures on perf tests (not on perf stats, but fails running due to dcheck failures), see e.g., https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20(K%20Nexus5) > > Original issue's description: > > Drop frames until specified bitrate is achieved. > > > > This CL fixes a regression introduced with the new quality scaler > > where the video would no longer start in a scaled mode. This CL adds > > code that compares incoming captured frames to the target bitrate, > > and if they are found to be too large, they are dropped and sinkWants > > set to a lower resolution. The number of dropped frames should be low > > (0-4 in most cases) and should not introduce a noticeable delay, or > > at least should be preferrable to having the first 2-4 seconds of video > > have very low quality. > > > > BUG=webrtc:6953 > > > > Review-Url: https://codereview.webrtc.org/2630333002 > > Cr-Commit-Position: refs/heads/master@{#16391} > > Committed: https://chromium.googlesource.com/external/webrtc/+/83399caec5762d2dad038b8e9d86163e92c18c9f > > TBR=perkj@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,kthelgason@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6953 > > Review-Url: https://codereview.webrtc.org/2666303002 > Cr-Commit-Position: refs/heads/master@{#16395} > Committed: https://chromium.googlesource.com/external/webrtc/+/35fc2aa82fb5a562f3f76f2b91a55f05ebfd4874 TBR=perkj@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,minyue@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:6953 Review-Url: https://codereview.webrtc.org/2675223002 Cr-Commit-Position: refs/heads/master@{#16473}
2017-02-07 07:02:22 -08:00
MediaConfig media_config = GetMediaConfig();
media_config.video.enable_cpu_adaptation = true;
channel_.reset(engine_.CreateMediaChannel(
fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
channel_->OnReadyToSend(true);
ASSERT_TRUE(channel_->SetSendParameters(parameters));
AddSendStream();
webrtc::test::FrameForwarder frame_forwarder;
ASSERT_TRUE(channel_->SetSend(true));
cricket::VideoOptions camera_options;
camera_options.is_screencast = false;
channel_->SetVideoSend(last_ssrc_, &camera_options, &frame_forwarder);
ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size());
FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
EXPECT_TRUE(send_stream->resolution_scaling_enabled());
// Dont' expect anything on framerate_scaling_enabled, since the default is
// transitioning from MAINTAIN_FRAMERATE to BALANCED.
// Switch to screen share. Expect no resolution scaling.
cricket::VideoOptions screenshare_options;
screenshare_options.is_screencast = true;
channel_->SetVideoSend(last_ssrc_, &screenshare_options, &frame_forwarder);
ASSERT_EQ(2, fake_call_->GetNumCreatedSendStreams());
send_stream = fake_call_->GetVideoSendStreams().front();
EXPECT_FALSE(send_stream->resolution_scaling_enabled());
// Switch back to the normal capturer. Expect resolution scaling to be
// reenabled.
channel_->SetVideoSend(last_ssrc_, &camera_options, &frame_forwarder);
send_stream = fake_call_->GetVideoSendStreams().front();
ASSERT_EQ(3, fake_call_->GetNumCreatedSendStreams());
send_stream = fake_call_->GetVideoSendStreams().front();
EXPECT_TRUE(send_stream->resolution_scaling_enabled());
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
// TODO(asapersson): Remove this test when the balanced field trial is removed.
void WebRtcVideoChannelTest::TestDegradationPreference(
bool resolution_scaling_enabled,
bool fps_scaling_enabled) {
cricket::VideoCodec codec = GetEngineCodec("VP8");
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(codec);
MediaConfig media_config = GetMediaConfig();
media_config.video.enable_cpu_adaptation = true;
channel_.reset(engine_.CreateMediaChannel(
fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
channel_->OnReadyToSend(true);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
AddSendStream();
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
EXPECT_TRUE(channel_->SetSend(true));
FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
EXPECT_EQ(resolution_scaling_enabled,
send_stream->resolution_scaling_enabled());
EXPECT_EQ(fps_scaling_enabled, send_stream->framerate_scaling_enabled());
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
void WebRtcVideoChannelTest::TestCpuAdaptation(bool enable_overuse,
bool is_screenshare) {
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
cricket::VideoCodec codec = GetEngineCodec("VP8");
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(codec);
Reland of Drop frames until specified bitrate is achieved. (patchset #1 id:1 of https://codereview.webrtc.org/2666303002/ ) Reason for revert: Perf test broke as it made assumptions that quality scaling was turned off. This turns out not to be the case. Fixed by turning quality scaling off for the tests. Original issue's description: > Revert of Drop frames until specified bitrate is achieved. (patchset #12 id:240001 of https://codereview.webrtc.org/2630333002/ ) > > Reason for revert: > due to failures on perf tests (not on perf stats, but fails running due to dcheck failures), see e.g., https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20(K%20Nexus5) > > Original issue's description: > > Drop frames until specified bitrate is achieved. > > > > This CL fixes a regression introduced with the new quality scaler > > where the video would no longer start in a scaled mode. This CL adds > > code that compares incoming captured frames to the target bitrate, > > and if they are found to be too large, they are dropped and sinkWants > > set to a lower resolution. The number of dropped frames should be low > > (0-4 in most cases) and should not introduce a noticeable delay, or > > at least should be preferrable to having the first 2-4 seconds of video > > have very low quality. > > > > BUG=webrtc:6953 > > > > Review-Url: https://codereview.webrtc.org/2630333002 > > Cr-Commit-Position: refs/heads/master@{#16391} > > Committed: https://chromium.googlesource.com/external/webrtc/+/83399caec5762d2dad038b8e9d86163e92c18c9f > > TBR=perkj@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,kthelgason@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6953 > > Review-Url: https://codereview.webrtc.org/2666303002 > Cr-Commit-Position: refs/heads/master@{#16395} > Committed: https://chromium.googlesource.com/external/webrtc/+/35fc2aa82fb5a562f3f76f2b91a55f05ebfd4874 TBR=perkj@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,minyue@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:6953 Review-Url: https://codereview.webrtc.org/2675223002 Cr-Commit-Position: refs/heads/master@{#16473}
2017-02-07 07:02:22 -08:00
MediaConfig media_config = GetMediaConfig();
if (enable_overuse) {
media_config.video.enable_cpu_adaptation = true;
}
channel_.reset(engine_.CreateMediaChannel(
fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
channel_->OnReadyToSend(true);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
AddSendStream();
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
options.is_screencast = is_screenshare;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
EXPECT_TRUE(channel_->SetSend(true));
FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ ) Reason for revert: Seem to be a flaky test rather than an issue with this cl. Creating reland, will add code to reduce flakiness to that test. Original issue's description: > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ ) > > Reason for revert: > This has resulted in failure of CallPerfTest.ReceivesCpuOveruseAndUnderuse test on the Win7 build bot https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1780 > > Original issue's description: > > Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ ) > > > > Reason for revert: > > Found issue with test case, will add fix to reland cl. > > > > Original issue's description: > > > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ ) > > > > > > Reason for revert: > > > Breaks perf tests: > > > https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679 > > > https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325 > > > > > > Original issue's description: > > > > Add framerate to VideoSinkWants and ability to signal on overuse > > > > > > > > In ViEEncoder, try to reduce framerate instead of resolution if the > > > > current degradation preference is maintain-resolution rather than > > > > balanced. > > > > > > > > BUG=webrtc:4172 > > > > > > > > Review-Url: https://codereview.webrtc.org/2716643002 > > > > Cr-Commit-Position: refs/heads/master@{#17327} > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/72acf2526177bb4dbb5103cd6e165eb4361a5ae6 > > > > > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org > > > # Skipping CQ checks because original CL landed less than 1 days ago. > > > NOPRESUBMIT=true > > > NOTREECHECKS=true > > > NOTRY=true > > > BUG=webrtc:4172 > > > > > > Review-Url: https://codereview.webrtc.org/2764133002 > > > Cr-Commit-Position: refs/heads/master@{#17331} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/8b45b11144c968b4173215c76f78c710c9a2ed0b > > > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org > > # Not skipping CQ checks because original CL landed more than 1 days ago. > > BUG=webrtc:4172 > > > > Review-Url: https://codereview.webrtc.org/2781433002 > > Cr-Commit-Position: refs/heads/master@{#17474} > > Committed: https://chromium.googlesource.com/external/webrtc/+/3ea3c77e93121b1ab9d5e46641e6764f2cca0d51 > > TBR=ilnik@webrtc.org,stefan@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:4172 > > Review-Url: https://codereview.webrtc.org/2783183003 > Cr-Commit-Position: refs/heads/master@{#17477} > Committed: https://chromium.googlesource.com/external/webrtc/+/f9ed235c9b7248694edcb46feb1f29ce7456ab59 R=ilnik@webrtc.org,stefan@webrtc.org BUG=webrtc:4172 Review-Url: https://codereview.webrtc.org/2789823002 Cr-Commit-Position: refs/heads/master@{#17498}
2017-04-02 23:53:04 -07:00
if (!enable_overuse) {
EXPECT_FALSE(send_stream->resolution_scaling_enabled());
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ ) Reason for revert: Seem to be a flaky test rather than an issue with this cl. Creating reland, will add code to reduce flakiness to that test. Original issue's description: > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ ) > > Reason for revert: > This has resulted in failure of CallPerfTest.ReceivesCpuOveruseAndUnderuse test on the Win7 build bot https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1780 > > Original issue's description: > > Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ ) > > > > Reason for revert: > > Found issue with test case, will add fix to reland cl. > > > > Original issue's description: > > > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ ) > > > > > > Reason for revert: > > > Breaks perf tests: > > > https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679 > > > https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325 > > > > > > Original issue's description: > > > > Add framerate to VideoSinkWants and ability to signal on overuse > > > > > > > > In ViEEncoder, try to reduce framerate instead of resolution if the > > > > current degradation preference is maintain-resolution rather than > > > > balanced. > > > > > > > > BUG=webrtc:4172 > > > > > > > > Review-Url: https://codereview.webrtc.org/2716643002 > > > > Cr-Commit-Position: refs/heads/master@{#17327} > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/72acf2526177bb4dbb5103cd6e165eb4361a5ae6 > > > > > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org > > > # Skipping CQ checks because original CL landed less than 1 days ago. > > > NOPRESUBMIT=true > > > NOTREECHECKS=true > > > NOTRY=true > > > BUG=webrtc:4172 > > > > > > Review-Url: https://codereview.webrtc.org/2764133002 > > > Cr-Commit-Position: refs/heads/master@{#17331} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/8b45b11144c968b4173215c76f78c710c9a2ed0b > > > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org > > # Not skipping CQ checks because original CL landed more than 1 days ago. > > BUG=webrtc:4172 > > > > Review-Url: https://codereview.webrtc.org/2781433002 > > Cr-Commit-Position: refs/heads/master@{#17474} > > Committed: https://chromium.googlesource.com/external/webrtc/+/3ea3c77e93121b1ab9d5e46641e6764f2cca0d51 > > TBR=ilnik@webrtc.org,stefan@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:4172 > > Review-Url: https://codereview.webrtc.org/2783183003 > Cr-Commit-Position: refs/heads/master@{#17477} > Committed: https://chromium.googlesource.com/external/webrtc/+/f9ed235c9b7248694edcb46feb1f29ce7456ab59 R=ilnik@webrtc.org,stefan@webrtc.org BUG=webrtc:4172 Review-Url: https://codereview.webrtc.org/2789823002 Cr-Commit-Position: refs/heads/master@{#17498}
2017-04-02 23:53:04 -07:00
EXPECT_FALSE(send_stream->framerate_scaling_enabled());
} else if (is_screenshare) {
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ ) Reason for revert: Seem to be a flaky test rather than an issue with this cl. Creating reland, will add code to reduce flakiness to that test. Original issue's description: > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ ) > > Reason for revert: > This has resulted in failure of CallPerfTest.ReceivesCpuOveruseAndUnderuse test on the Win7 build bot https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1780 > > Original issue's description: > > Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ ) > > > > Reason for revert: > > Found issue with test case, will add fix to reland cl. > > > > Original issue's description: > > > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ ) > > > > > > Reason for revert: > > > Breaks perf tests: > > > https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679 > > > https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325 > > > > > > Original issue's description: > > > > Add framerate to VideoSinkWants and ability to signal on overuse > > > > > > > > In ViEEncoder, try to reduce framerate instead of resolution if the > > > > current degradation preference is maintain-resolution rather than > > > > balanced. > > > > > > > > BUG=webrtc:4172 > > > > > > > > Review-Url: https://codereview.webrtc.org/2716643002 > > > > Cr-Commit-Position: refs/heads/master@{#17327} > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/72acf2526177bb4dbb5103cd6e165eb4361a5ae6 > > > > > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org > > > # Skipping CQ checks because original CL landed less than 1 days ago. > > > NOPRESUBMIT=true > > > NOTREECHECKS=true > > > NOTRY=true > > > BUG=webrtc:4172 > > > > > > Review-Url: https://codereview.webrtc.org/2764133002 > > > Cr-Commit-Position: refs/heads/master@{#17331} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/8b45b11144c968b4173215c76f78c710c9a2ed0b > > > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org > > # Not skipping CQ checks because original CL landed more than 1 days ago. > > BUG=webrtc:4172 > > > > Review-Url: https://codereview.webrtc.org/2781433002 > > Cr-Commit-Position: refs/heads/master@{#17474} > > Committed: https://chromium.googlesource.com/external/webrtc/+/3ea3c77e93121b1ab9d5e46641e6764f2cca0d51 > > TBR=ilnik@webrtc.org,stefan@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:4172 > > Review-Url: https://codereview.webrtc.org/2783183003 > Cr-Commit-Position: refs/heads/master@{#17477} > Committed: https://chromium.googlesource.com/external/webrtc/+/f9ed235c9b7248694edcb46feb1f29ce7456ab59 R=ilnik@webrtc.org,stefan@webrtc.org BUG=webrtc:4172 Review-Url: https://codereview.webrtc.org/2789823002 Cr-Commit-Position: refs/heads/master@{#17498}
2017-04-02 23:53:04 -07:00
EXPECT_FALSE(send_stream->resolution_scaling_enabled());
EXPECT_TRUE(send_stream->framerate_scaling_enabled());
} else {
EXPECT_TRUE(send_stream->resolution_scaling_enabled());
EXPECT_FALSE(send_stream->framerate_scaling_enabled());
}
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, EstimatesNtpStartTimeCorrectly) {
// Start at last timestamp to verify that wraparounds are estimated correctly.
static const uint32_t kInitialTimestamp = 0xFFFFFFFFu;
static const int64_t kInitialNtpTimeMs = 1247891230;
static const int kFrameOffsetMs = 20;
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
FakeVideoReceiveStream* stream = AddRecvStream();
cricket::FakeVideoRenderer renderer;
EXPECT_TRUE(channel_->SetSink(last_ssrc_, &renderer));
webrtc::VideoFrame video_frame =
webrtc::VideoFrame::Builder()
.set_video_frame_buffer(CreateBlackFrameBuffer(4, 4))
.set_timestamp_rtp(kInitialTimestamp)
.set_timestamp_us(0)
.set_rotation(webrtc::kVideoRotation_0)
.build();
// Initial NTP time is not available on the first frame, but should still be
// able to be estimated.
stream->InjectFrame(video_frame);
EXPECT_EQ(1, renderer.num_rendered_frames());
// This timestamp is kInitialTimestamp (-1) + kFrameOffsetMs * 90, which
// triggers a constant-overflow warning, hence we're calculating it explicitly
// here.
fake_clock_.AdvanceTime(webrtc::TimeDelta::ms(kFrameOffsetMs));
video_frame.set_timestamp(kFrameOffsetMs * 90 - 1);
video_frame.set_ntp_time_ms(kInitialNtpTimeMs + kFrameOffsetMs);
stream->InjectFrame(video_frame);
EXPECT_EQ(2, renderer.num_rendered_frames());
// Verify that NTP time has been correctly deduced.
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(1u, info.receivers.size());
EXPECT_EQ(kInitialNtpTimeMs, info.receivers[0].capture_start_ntp_time_ms);
}
TEST_F(WebRtcVideoChannelTest, SetDefaultSendCodecs) {
AssignDefaultAptRtxTypes();
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
VideoCodec codec;
EXPECT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_TRUE(codec.Matches(engine_.codecs()[0]));
// Using a RTX setup to verify that the default RTX payload type is good.
const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs1);
const std::vector<uint32_t> rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1);
FakeVideoSendStream* stream = AddSendStream(
cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs));
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
// Make sure NACK and FEC are enabled on the correct payload types.
EXPECT_EQ(1000, config.rtp.nack.rtp_history_ms);
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
EXPECT_EQ(GetEngineCodec("ulpfec").id, config.rtp.ulpfec.ulpfec_payload_type);
EXPECT_EQ(GetEngineCodec("red").id, config.rtp.ulpfec.red_payload_type);
EXPECT_EQ(1u, config.rtp.rtx.ssrcs.size());
EXPECT_EQ(kRtxSsrcs1[0], config.rtp.rtx.ssrcs[0]);
VerifySendStreamHasRtxTypes(config, default_apt_rtx_types_);
// TODO(juberti): Check RTCP, PLI, TMMBR.
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithoutPacketization) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = AddSendStream();
const webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_FALSE(config.rtp.raw_payload);
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithPacketization) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.back().packetization = kPacketizationParamRaw;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = AddSendStream();
const webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_TRUE(config.rtp.raw_payload);
}
// The following four tests ensures that FlexFEC is not activated by default
// when the field trials are not enabled.
// TODO(brandtr): Remove or update these tests when FlexFEC _is_ enabled by
// default.
TEST_F(WebRtcVideoChannelTest, FlexfecSendCodecWithoutSsrcNotExposedByDefault) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
EXPECT_EQ(0U, config.rtp.flexfec.ssrc);
EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty());
}
TEST_F(WebRtcVideoChannelTest, FlexfecSendCodecWithSsrcNotExposedByDefault) {
FakeVideoSendStream* stream = AddSendStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
EXPECT_EQ(0U, config.rtp.flexfec.ssrc);
EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty());
}
TEST_F(WebRtcVideoChannelTest, FlexfecRecvCodecWithoutSsrcNotExposedByDefault) {
AddRecvStream();
const std::vector<FakeFlexfecReceiveStream*>& streams =
fake_call_->GetFlexfecReceiveStreams();
EXPECT_TRUE(streams.empty());
}
TEST_F(WebRtcVideoChannelTest, FlexfecRecvCodecWithSsrcNotExposedByDefault) {
AddRecvStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
const std::vector<FakeFlexfecReceiveStream*>& streams =
fake_call_->GetFlexfecReceiveStreams();
EXPECT_TRUE(streams.empty());
}
// TODO(brandtr): When FlexFEC is no longer behind a field trial, merge all
// tests that use this test fixture into the corresponding "non-field trial"
// tests.
class WebRtcVideoChannelFlexfecRecvTest : public WebRtcVideoChannelTest {
public:
WebRtcVideoChannelFlexfecRecvTest()
: WebRtcVideoChannelTest("WebRTC-FlexFEC-03-Advertised/Enabled/") {}
};
TEST_F(WebRtcVideoChannelFlexfecRecvTest,
DefaultFlexfecCodecHasTransportCcAndRembFeedbackParam) {
EXPECT_TRUE(cricket::HasTransportCc(GetEngineCodec("flexfec-03")));
EXPECT_TRUE(cricket::HasRemb(GetEngineCodec("flexfec-03")));
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetDefaultRecvCodecsWithoutSsrc) {
AddRecvStream();
const std::vector<FakeFlexfecReceiveStream*>& streams =
fake_call_->GetFlexfecReceiveStreams();
EXPECT_TRUE(streams.empty());
const std::vector<FakeVideoReceiveStream*>& video_streams =
fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(1U, video_streams.size());
const FakeVideoReceiveStream& video_stream = *video_streams.front();
EXPECT_EQ(0, video_stream.GetNumAddedSecondarySinks());
EXPECT_EQ(0, video_stream.GetNumRemovedSecondarySinks());
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetDefaultRecvCodecsWithSsrc) {
AddRecvStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
const std::vector<FakeFlexfecReceiveStream*>& streams =
fake_call_->GetFlexfecReceiveStreams();
ASSERT_EQ(1U, streams.size());
const FakeFlexfecReceiveStream* stream = streams.front();
const webrtc::FlexfecReceiveStream::Config& config = stream->GetConfig();
EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.payload_type);
EXPECT_EQ(kFlexfecSsrc, config.remote_ssrc);
ASSERT_EQ(1U, config.protected_media_ssrcs.size());
EXPECT_EQ(kSsrcs1[0], config.protected_media_ssrcs[0]);
const std::vector<FakeVideoReceiveStream*>& video_streams =
fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(1U, video_streams.size());
const FakeVideoReceiveStream& video_stream = *video_streams.front();
EXPECT_EQ(1, video_stream.GetNumAddedSecondarySinks());
const webrtc::VideoReceiveStream::Config& video_config =
video_stream.GetConfig();
EXPECT_TRUE(video_config.rtp.protected_by_flexfec);
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest,
EnablingFlexfecDoesNotRecreateVideoReceiveStream) {
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
AddRecvStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams());
const std::vector<FakeVideoReceiveStream*>& video_streams =
fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(1U, video_streams.size());
const FakeVideoReceiveStream& video_stream = *video_streams.front();
EXPECT_EQ(0, video_stream.GetNumAddedSecondarySinks());
EXPECT_EQ(0, video_stream.GetNumRemovedSecondarySinks());
// Enable FlexFEC.
recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
EXPECT_EQ(2, fake_call_->GetNumCreatedReceiveStreams())
<< "Enabling FlexFEC should create FlexfecReceiveStream.";
EXPECT_EQ(1U, fake_call_->GetVideoReceiveStreams().size())
<< "Enabling FlexFEC should not create VideoReceiveStream.";
EXPECT_EQ(1U, fake_call_->GetFlexfecReceiveStreams().size())
<< "Enabling FlexFEC should create a single FlexfecReceiveStream.";
EXPECT_EQ(1, video_stream.GetNumAddedSecondarySinks());
EXPECT_EQ(0, video_stream.GetNumRemovedSecondarySinks());
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest,
DisablingFlexfecDoesNotRecreateVideoReceiveStream) {
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
AddRecvStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
EXPECT_EQ(2, fake_call_->GetNumCreatedReceiveStreams());
EXPECT_EQ(1U, fake_call_->GetFlexfecReceiveStreams().size());
const std::vector<FakeVideoReceiveStream*>& video_streams =
fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(1U, video_streams.size());
const FakeVideoReceiveStream& video_stream = *video_streams.front();
EXPECT_EQ(1, video_stream.GetNumAddedSecondarySinks());
EXPECT_EQ(0, video_stream.GetNumRemovedSecondarySinks());
// Disable FlexFEC.
recv_parameters.codecs.clear();
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
EXPECT_EQ(2, fake_call_->GetNumCreatedReceiveStreams())
<< "Disabling FlexFEC should not recreate VideoReceiveStream.";
EXPECT_EQ(1U, fake_call_->GetVideoReceiveStreams().size())
<< "Disabling FlexFEC should not destroy VideoReceiveStream.";
EXPECT_TRUE(fake_call_->GetFlexfecReceiveStreams().empty())
<< "Disabling FlexFEC should destroy FlexfecReceiveStream.";
EXPECT_EQ(1, video_stream.GetNumAddedSecondarySinks());
EXPECT_EQ(1, video_stream.GetNumRemovedSecondarySinks());
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest, DuplicateFlexfecCodecIsDropped) {
constexpr int kUnusedPayloadType1 = 127;
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
cricket::VideoCodec duplicate = GetEngineCodec("flexfec-03");
duplicate.id = kUnusedPayloadType1;
recv_parameters.codecs.push_back(duplicate);
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
AddRecvStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
const std::vector<FakeFlexfecReceiveStream*>& streams =
fake_call_->GetFlexfecReceiveStreams();
ASSERT_EQ(1U, streams.size());
const FakeFlexfecReceiveStream* stream = streams.front();
const webrtc::FlexfecReceiveStream::Config& config = stream->GetConfig();
EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.payload_type);
}
// TODO(brandtr): When FlexFEC is no longer behind a field trial, merge all
// tests that use this test fixture into the corresponding "non-field trial"
// tests.
class WebRtcVideoChannelFlexfecSendRecvTest : public WebRtcVideoChannelTest {
public:
WebRtcVideoChannelFlexfecSendRecvTest()
: WebRtcVideoChannelTest(
"WebRTC-FlexFEC-03-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/") {
}
};
TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetDefaultSendCodecsWithoutSsrc) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.rtp.flexfec.payload_type);
EXPECT_EQ(0U, config.rtp.flexfec.ssrc);
EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty());
}
TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetDefaultSendCodecsWithSsrc) {
FakeVideoSendStream* stream = AddSendStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.rtp.flexfec.payload_type);
EXPECT_EQ(kFlexfecSsrc, config.rtp.flexfec.ssrc);
ASSERT_EQ(1U, config.rtp.flexfec.protected_media_ssrcs.size());
EXPECT_EQ(kSsrcs1[0], config.rtp.flexfec.protected_media_ssrcs[0]);
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithoutFec) {
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(-1, config.rtp.ulpfec.ulpfec_payload_type);
EXPECT_EQ(-1, config.rtp.ulpfec.red_payload_type);
}
TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetSendCodecsWithoutFec) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetRecvCodecsWithFec) {
AddRecvStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
const std::vector<FakeFlexfecReceiveStream*>& flexfec_streams =
fake_call_->GetFlexfecReceiveStreams();
ASSERT_EQ(1U, flexfec_streams.size());
const FakeFlexfecReceiveStream* flexfec_stream = flexfec_streams.front();
const webrtc::FlexfecReceiveStream::Config& flexfec_stream_config =
flexfec_stream->GetConfig();
EXPECT_EQ(GetEngineCodec("flexfec-03").id,
flexfec_stream_config.payload_type);
EXPECT_EQ(kFlexfecSsrc, flexfec_stream_config.remote_ssrc);
ASSERT_EQ(1U, flexfec_stream_config.protected_media_ssrcs.size());
EXPECT_EQ(kSsrcs1[0], flexfec_stream_config.protected_media_ssrcs[0]);
const std::vector<FakeVideoReceiveStream*>& video_streams =
fake_call_->GetVideoReceiveStreams();
const FakeVideoReceiveStream* video_stream = video_streams.front();
const webrtc::VideoReceiveStream::Config& video_stream_config =
video_stream->GetConfig();
EXPECT_EQ(video_stream_config.rtp.local_ssrc,
flexfec_stream_config.local_ssrc);
EXPECT_EQ(video_stream_config.rtp.rtcp_mode, flexfec_stream_config.rtcp_mode);
EXPECT_EQ(video_stream_config.rtcp_send_transport,
flexfec_stream_config.rtcp_send_transport);
// TODO(brandtr): Update this EXPECT when we set |transport_cc| in a
// spec-compliant way.
EXPECT_EQ(video_stream_config.rtp.transport_cc,
flexfec_stream_config.transport_cc);
EXPECT_EQ(video_stream_config.rtp.rtcp_mode, flexfec_stream_config.rtcp_mode);
EXPECT_EQ(video_stream_config.rtp.extensions,
flexfec_stream_config.rtp_header_extensions);
}
// We should not send FlexFEC, even if we advertise it, unless the right
// field trial is set.
// TODO(brandtr): Remove when FlexFEC is enabled by default.
TEST_F(WebRtcVideoChannelFlexfecRecvTest,
SetSendCodecsWithoutSsrcWithFecDoesNotEnableFec) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
EXPECT_EQ(0u, config.rtp.flexfec.ssrc);
EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty());
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest,
SetSendCodecsWithSsrcWithFecDoesNotEnableFec) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = AddSendStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
EXPECT_EQ(0u, config.rtp.flexfec.ssrc);
EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty());
}
TEST_F(WebRtcVideoChannelTest,
SetSendCodecRejectsRtxWithoutAssociatedPayloadType) {
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
const int kUnusedPayloadType = 127;
EXPECT_FALSE(FindCodecById(engine_.codecs(), kUnusedPayloadType));
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
cricket::VideoCodec rtx_codec(kUnusedPayloadType, "rtx");
parameters.codecs.push_back(rtx_codec);
EXPECT_FALSE(channel_->SetSendParameters(parameters))
<< "RTX codec without associated payload type should be rejected.";
}
TEST_F(WebRtcVideoChannelTest,
SetSendCodecRejectsRtxWithoutMatchingVideoCodec) {
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
const int kUnusedPayloadType1 = 126;
const int kUnusedPayloadType2 = 127;
EXPECT_FALSE(FindCodecById(engine_.codecs(), kUnusedPayloadType1));
EXPECT_FALSE(FindCodecById(engine_.codecs(), kUnusedPayloadType2));
{
cricket::VideoCodec rtx_codec = cricket::VideoCodec::CreateRtxCodec(
kUnusedPayloadType1, GetEngineCodec("VP8").id);
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(rtx_codec);
ASSERT_TRUE(channel_->SetSendParameters(parameters));
}
{
cricket::VideoCodec rtx_codec = cricket::VideoCodec::CreateRtxCodec(
kUnusedPayloadType1, kUnusedPayloadType2);
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(rtx_codec);
EXPECT_FALSE(channel_->SetSendParameters(parameters))
<< "RTX without matching video codec should be rejected.";
}
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithChangedRtxPayloadType) {
Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ ) Reason for revert: Downstream project relied on changed struct. Transition made possible by https://codereview.webrtc.org/2655243006/. Original issue's description: > Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ ) > > Reason for revert: > Breaks internal downstream project. > > Original issue's description: > > Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. > > > > Prior to this CL, received RTX (associated) payload types were only configured > > when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX > > SSRC was set up. > > > > After this CL, the RTX (associated) payload types are set in > > WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set > > them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since > > that is the code path that sets other SSRCs. > > > > As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed. > > We remove the possibility for each video payload type to have an associated > > specific RTX SSRC. Although the config previously allowed for this, all payload > > types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry > > did not support multiple SSRCs. This change to the config struct should thus not > > have any functional impact. The change does however affect the RtcEventLog, since > > that is used for storing the VideoReceiveStream::Configs. For simplicity, > > this CL does not change the event log proto definitions, instead duplicating > > the serialized RTX SSRCs such that they fit in the existing proto definition. > > > > BUG=webrtc:7011 > > > > Review-Url: https://codereview.webrtc.org/2646073004 > > Cr-Commit-Position: refs/heads/master@{#16302} > > Committed: https://chromium.googlesource.com/external/webrtc/+/fe2bef39cd2a5c891a49f7320514fb04324dc66c > > TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7011 > > Review-Url: https://codereview.webrtc.org/2649323010 > Cr-Commit-Position: refs/heads/master@{#16307} > Committed: https://chromium.googlesource.com/external/webrtc/+/e4974953ce0d03a60fae7659b199a6a62a79fa30 TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true # NOTREECHECKS=true # NOTRY=true BUG=webrtc:7011 Review-Url: https://codereview.webrtc.org/2654163006 Cr-Commit-Position: refs/heads/master@{#16322}
2017-01-27 04:53:07 -08:00
const int kUnusedPayloadType1 = 126;
const int kUnusedPayloadType2 = 127;
EXPECT_FALSE(FindCodecById(engine_.codecs(), kUnusedPayloadType1));
EXPECT_FALSE(FindCodecById(engine_.codecs(), kUnusedPayloadType2));
// SSRCs for RTX.
cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
params.AddFidSsrc(kSsrcs1[0], kRtxSsrcs1[0]);
AddSendStream(params);
// Original payload type for RTX.
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
cricket::VideoCodec rtx_codec(kUnusedPayloadType1, "rtx");
rtx_codec.SetParam("apt", GetEngineCodec("VP8").id);
parameters.codecs.push_back(rtx_codec);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
ASSERT_EQ(1U, fake_call_->GetVideoSendStreams().size());
const webrtc::VideoSendStream::Config& config_before =
fake_call_->GetVideoSendStreams()[0]->GetConfig();
EXPECT_EQ(kUnusedPayloadType1, config_before.rtp.rtx.payload_type);
ASSERT_EQ(1U, config_before.rtp.rtx.ssrcs.size());
EXPECT_EQ(kRtxSsrcs1[0], config_before.rtp.rtx.ssrcs[0]);
// Change payload type for RTX.
parameters.codecs[1].id = kUnusedPayloadType2;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
ASSERT_EQ(1U, fake_call_->GetVideoSendStreams().size());
const webrtc::VideoSendStream::Config& config_after =
fake_call_->GetVideoSendStreams()[0]->GetConfig();
EXPECT_EQ(kUnusedPayloadType2, config_after.rtp.rtx.payload_type);
ASSERT_EQ(1U, config_after.rtp.rtx.ssrcs.size());
EXPECT_EQ(kRtxSsrcs1[0], config_after.rtp.rtx.ssrcs[0]);
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithoutFecDisablesFec) {
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("ulpfec"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
EXPECT_EQ(GetEngineCodec("ulpfec").id, config.rtp.ulpfec.ulpfec_payload_type);
parameters.codecs.pop_back();
ASSERT_TRUE(channel_->SetSendParameters(parameters));
stream = fake_call_->GetVideoSendStreams()[0];
ASSERT_TRUE(stream != nullptr);
config = stream->GetConfig().Copy();
EXPECT_EQ(-1, config.rtp.ulpfec.ulpfec_payload_type)
<< "SetSendCodec without ULPFEC should disable current ULPFEC.";
}
TEST_F(WebRtcVideoChannelFlexfecSendRecvTest,
SetSendCodecsWithoutFecDisablesFec) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = AddSendStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.rtp.flexfec.payload_type);
EXPECT_EQ(kFlexfecSsrc, config.rtp.flexfec.ssrc);
ASSERT_EQ(1U, config.rtp.flexfec.protected_media_ssrcs.size());
EXPECT_EQ(kSsrcs1[0], config.rtp.flexfec.protected_media_ssrcs[0]);
parameters.codecs.pop_back();
ASSERT_TRUE(channel_->SetSendParameters(parameters));
stream = fake_call_->GetVideoSendStreams()[0];
ASSERT_TRUE(stream != nullptr);
config = stream->GetConfig().Copy();
EXPECT_EQ(-1, config.rtp.flexfec.payload_type)
<< "SetSendCodec without FlexFEC should disable current FlexFEC.";
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsChangesExistingStreams) {
cricket::VideoSendParameters parameters;
cricket::VideoCodec codec(100, "VP8");
codec.SetParam(kCodecParamMaxQuantization, kDefaultQpMax);
parameters.codecs.push_back(codec);
ASSERT_TRUE(channel_->SetSendParameters(parameters));
channel_->SetSend(true);
FakeVideoSendStream* stream = AddSendStream();
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
std::vector<webrtc::VideoStream> streams = stream->GetVideoStreams();
EXPECT_EQ(kDefaultQpMax, streams[0].max_qp);
parameters.codecs.clear();
codec.SetParam(kCodecParamMaxQuantization, kDefaultQpMax + 1);
parameters.codecs.push_back(codec);
ASSERT_TRUE(channel_->SetSendParameters(parameters));
streams = fake_call_->GetVideoSendStreams()[0]->GetVideoStreams();
EXPECT_EQ(kDefaultQpMax + 1, streams[0].max_qp);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithBitrates) {
SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200",
200000);
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithHighMaxBitrate) {
SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "10000", 10000000);
std::vector<webrtc::VideoStream> streams = AddSendStream()->GetVideoStreams();
ASSERT_EQ(1u, streams.size());
EXPECT_EQ(10000000, streams[0].max_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest,
SetSendCodecsWithoutBitratesUsesCorrectDefaults) {
SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "", -1);
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsCapsMinAndStartBitrate) {
SetSendCodecsShouldWorkForBitrates("-1", 0, "-100", -1, "", -1);
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsRejectsMaxLessThanMinBitrate) {
send_parameters_.codecs[0].params[kCodecParamMinBitrate] = "300";
send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "200";
EXPECT_FALSE(channel_->SetSendParameters(send_parameters_));
}
// Test that when both the codec-specific bitrate params and max_bandwidth_bps
// are present in the same send parameters, the settings are combined correctly.
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithBitratesAndMaxSendBandwidth) {
send_parameters_.codecs[0].params[kCodecParamMinBitrate] = "100";
send_parameters_.codecs[0].params[kCodecParamStartBitrate] = "200";
send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "300";
send_parameters_.max_bandwidth_bps = 400000;
// We expect max_bandwidth_bps to take priority, if set.
ExpectSetBitrateParameters(100000, 200000, 400000);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
// Since the codec isn't changing, start_bitrate_bps should be -1.
ExpectSetBitrateParameters(100000, -1, 350000);
// Decrease max_bandwidth_bps.
send_parameters_.max_bandwidth_bps = 350000;
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
// Now try again with the values flipped around.
send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "400";
send_parameters_.max_bandwidth_bps = 300000;
ExpectSetBitrateParameters(100000, 200000, 300000);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
// If we change the codec max, max_bandwidth_bps should still apply.
send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "350";
ExpectSetBitrateParameters(100000, 200000, 300000);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
}
TEST_F(WebRtcVideoChannelTest, SetMaxSendBandwidthShouldPreserveOtherBitrates) {
SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200",
200000);
send_parameters_.max_bandwidth_bps = 300000;
// Setting max bitrate should keep previous min bitrate.
// Setting max bitrate should not reset start bitrate.
ExpectSetBitrateParameters(100000, -1, 300000);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
}
TEST_F(WebRtcVideoChannelTest, SetMaxSendBandwidthShouldBeRemovable) {
send_parameters_.max_bandwidth_bps = 300000;
ExpectSetMaxBitrate(300000);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
// -1 means to disable max bitrate (set infinite).
send_parameters_.max_bandwidth_bps = -1;
ExpectSetMaxBitrate(-1);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
}
TEST_F(WebRtcVideoChannelTest, SetMaxSendBandwidthAndAddSendStream) {
send_parameters_.max_bandwidth_bps = 99999;
FakeVideoSendStream* stream = AddSendStream();
ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
ASSERT_EQ(1u, stream->GetVideoStreams().size());
EXPECT_EQ(send_parameters_.max_bandwidth_bps,
stream->GetVideoStreams()[0].max_bitrate_bps);
send_parameters_.max_bandwidth_bps = 77777;
ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_EQ(send_parameters_.max_bandwidth_bps,
stream->GetVideoStreams()[0].max_bitrate_bps);
}
// Tests that when the codec specific max bitrate and VideoSendParameters
// max_bandwidth_bps are used, that it sets the VideoStream's max bitrate
// appropriately.
TEST_F(WebRtcVideoChannelTest,
MaxBitratePrioritizesVideoSendParametersOverCodecMaxBitrate) {
send_parameters_.codecs[0].params[kCodecParamMinBitrate] = "100";
send_parameters_.codecs[0].params[kCodecParamStartBitrate] = "200";
send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "300";
send_parameters_.max_bandwidth_bps = -1;
AddSendStream();
ExpectSetMaxBitrate(300000);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
std::vector<FakeVideoSendStream*> video_send_streams = GetFakeSendStreams();
ASSERT_EQ(1u, video_send_streams.size());
FakeVideoSendStream* video_send_stream = video_send_streams[0];
ASSERT_EQ(1u, video_send_streams[0]->GetVideoStreams().size());
// First the max bitrate is set based upon the codec param.
EXPECT_EQ(300000,
video_send_streams[0]->GetVideoStreams()[0].max_bitrate_bps);
// The VideoSendParameters max bitrate overrides the codec's.
send_parameters_.max_bandwidth_bps = 500000;
ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
ASSERT_EQ(1u, video_send_stream->GetVideoStreams().size());
EXPECT_EQ(500000, video_send_stream->GetVideoStreams()[0].max_bitrate_bps);
}
// Tests that when the codec specific max bitrate and RtpParameters
// max_bitrate_bps are used, that it sets the VideoStream's max bitrate
// appropriately.
TEST_F(WebRtcVideoChannelTest,
MaxBitratePrioritizesRtpParametersOverCodecMaxBitrate) {
send_parameters_.codecs[0].params[kCodecParamMinBitrate] = "100";
send_parameters_.codecs[0].params[kCodecParamStartBitrate] = "200";
send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "300";
send_parameters_.max_bandwidth_bps = -1;
AddSendStream();
ExpectSetMaxBitrate(300000);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
std::vector<FakeVideoSendStream*> video_send_streams = GetFakeSendStreams();
ASSERT_EQ(1u, video_send_streams.size());
FakeVideoSendStream* video_send_stream = video_send_streams[0];
ASSERT_EQ(1u, video_send_stream->GetVideoStreams().size());
// First the max bitrate is set based upon the codec param.
EXPECT_EQ(300000, video_send_stream->GetVideoStreams()[0].max_bitrate_bps);
// The RtpParameter max bitrate overrides the codec's.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(1u, parameters.encodings.size());
parameters.encodings[0].max_bitrate_bps = 500000;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
ASSERT_EQ(1u, video_send_stream->GetVideoStreams().size());
EXPECT_EQ(parameters.encodings[0].max_bitrate_bps,
video_send_stream->GetVideoStreams()[0].max_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest,
MaxBitrateIsMinimumOfMaxSendBandwidthAndMaxEncodingBitrate) {
send_parameters_.max_bandwidth_bps = 99999;
FakeVideoSendStream* stream = AddSendStream();
ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
ASSERT_EQ(1u, stream->GetVideoStreams().size());
EXPECT_EQ(send_parameters_.max_bandwidth_bps,
stream->GetVideoStreams()[0].max_bitrate_bps);
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1u, parameters.encodings.size());
parameters.encodings[0].max_bitrate_bps = 99999 - 1;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
EXPECT_EQ(parameters.encodings[0].max_bitrate_bps,
stream->GetVideoStreams()[0].max_bitrate_bps);
parameters.encodings[0].max_bitrate_bps = 99999 + 1;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
EXPECT_EQ(send_parameters_.max_bandwidth_bps,
stream->GetVideoStreams()[0].max_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest, SetMaxSendBitrateCanIncreaseSenderBitrate) {
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
channel_->SetSend(true);
FakeVideoSendStream* stream = AddSendStream();
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
std::vector<webrtc::VideoStream> streams = stream->GetVideoStreams();
int initial_max_bitrate_bps = streams[0].max_bitrate_bps;
EXPECT_GT(initial_max_bitrate_bps, 0);
parameters.max_bandwidth_bps = initial_max_bitrate_bps * 2;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
// Insert a frame to update the encoder config.
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
streams = stream->GetVideoStreams();
EXPECT_EQ(initial_max_bitrate_bps * 2, streams[0].max_bitrate_bps);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
SetMaxSendBitrateCanIncreaseSimulcastSenderBitrate) {
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
channel_->SetSend(true);
FakeVideoSendStream* stream = AddSendStream(
cricket::CreateSimStreamParams("cname", MAKE_VECTOR(kSsrcs3)));
// Send a frame to make sure this scales up to >1 stream (simulcast).
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(kSsrcs3[0], nullptr, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
std::vector<webrtc::VideoStream> streams = stream->GetVideoStreams();
ASSERT_GT(streams.size(), 1u)
<< "Without simulcast this test doesn't make sense.";
int initial_max_bitrate_bps = GetTotalMaxBitrateBps(streams);
EXPECT_GT(initial_max_bitrate_bps, 0);
parameters.max_bandwidth_bps = initial_max_bitrate_bps * 2;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
// Insert a frame to update the encoder config.
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
streams = stream->GetVideoStreams();
int increased_max_bitrate_bps = GetTotalMaxBitrateBps(streams);
EXPECT_EQ(initial_max_bitrate_bps * 2, increased_max_bitrate_bps);
EXPECT_TRUE(channel_->SetVideoSend(kSsrcs3[0], nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithMaxQuantization) {
static const char* kMaxQuantization = "21";
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs[0].params[kCodecParamMaxQuantization] = kMaxQuantization;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_EQ(atoi(kMaxQuantization),
AddSendStream()->GetVideoStreams().back().max_qp);
VideoCodec codec;
EXPECT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_EQ(kMaxQuantization, codec.params[kCodecParamMaxQuantization]);
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsRejectBadPayloadTypes) {
// TODO(pbos): Should we only allow the dynamic range?
static const int kIncorrectPayloads[] = {-2, -1, 128, 129};
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
for (size_t i = 0; i < arraysize(kIncorrectPayloads); ++i) {
parameters.codecs[0].id = kIncorrectPayloads[i];
EXPECT_FALSE(channel_->SetSendParameters(parameters))
<< "Bad payload type '" << kIncorrectPayloads[i] << "' accepted.";
}
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsAcceptAllValidPayloadTypes) {
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
for (int payload_type = 96; payload_type <= 127; ++payload_type) {
parameters.codecs[0].id = payload_type;
EXPECT_TRUE(channel_->SetSendParameters(parameters))
<< "Payload type '" << payload_type << "' rejected.";
}
}
// Test that setting the a different set of codecs but with an identical front
// codec doesn't result in the stream being recreated.
// This may happen when a subsequent negotiation includes fewer codecs, as a
// result of one of the codecs being rejected.
TEST_F(WebRtcVideoChannelTest,
SetSendCodecsIdenticalFirstCodecDoesntRecreateStream) {
cricket::VideoSendParameters parameters1;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters1.codecs.push_back(GetEngineCodec("VP8"));
parameters1.codecs.push_back(GetEngineCodec("VP9"));
EXPECT_TRUE(channel_->SetSendParameters(parameters1));
AddSendStream();
EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams());
cricket::VideoSendParameters parameters2;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters2.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel_->SetSendParameters(parameters2));
EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams());
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithOnlyVp8) {
cricket::VideoRecvParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
}
// Test that we set our inbound RTX codecs properly.
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithRtx) {
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
const int kUnusedPayloadType1 = 126;
const int kUnusedPayloadType2 = 127;
EXPECT_FALSE(FindCodecById(engine_.codecs(), kUnusedPayloadType1));
EXPECT_FALSE(FindCodecById(engine_.codecs(), kUnusedPayloadType2));
cricket::VideoRecvParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
cricket::VideoCodec rtx_codec(kUnusedPayloadType1, "rtx");
parameters.codecs.push_back(rtx_codec);
EXPECT_FALSE(channel_->SetRecvParameters(parameters))
<< "RTX codec without associated payload should be rejected.";
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs[1].SetParam("apt", kUnusedPayloadType2);
EXPECT_FALSE(channel_->SetRecvParameters(parameters))
<< "RTX codec with invalid associated payload type should be rejected.";
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs[1].SetParam("apt", GetEngineCodec("VP8").id);
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
cricket::VideoCodec rtx_codec2(kUnusedPayloadType2, "rtx");
rtx_codec2.SetParam("apt", rtx_codec.id);
parameters.codecs.push_back(rtx_codec2);
EXPECT_FALSE(channel_->SetRecvParameters(parameters))
<< "RTX codec with another RTX as associated payload type should be "
"rejected.";
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithPacketization) {
cricket::VideoCodec vp8_codec = GetEngineCodec("VP8");
vp8_codec.packetization = kPacketizationParamRaw;
cricket::VideoRecvParameters parameters;
parameters.codecs = {vp8_codec, GetEngineCodec("VP9")};
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
const cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
AddRecvStream(params);
ASSERT_THAT(fake_call_->GetVideoReceiveStreams(), testing::SizeIs(1));
const webrtc::VideoReceiveStream::Config& config =
fake_call_->GetVideoReceiveStreams()[0]->GetConfig();
ASSERT_THAT(config.rtp.raw_payload_types, testing::SizeIs(1));
EXPECT_EQ(config.rtp.raw_payload_types.count(vp8_codec.id), 1U);
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithPacketizationRecreatesStream) {
cricket::VideoRecvParameters parameters;
parameters.codecs = {GetEngineCodec("VP8"), GetEngineCodec("VP9")};
parameters.codecs.back().packetization = kPacketizationParamRaw;
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
const cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
AddRecvStream(params);
ASSERT_THAT(fake_call_->GetVideoReceiveStreams(), testing::SizeIs(1));
EXPECT_EQ(fake_call_->GetNumCreatedReceiveStreams(), 1);
parameters.codecs.back().packetization.reset();
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_EQ(fake_call_->GetNumCreatedReceiveStreams(), 2);
}
TEST_F(WebRtcVideoChannelTest, DuplicateUlpfecCodecIsDropped) {
constexpr int kFirstUlpfecPayloadType = 126;
constexpr int kSecondUlpfecPayloadType = 127;
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(
cricket::VideoCodec(kFirstUlpfecPayloadType, cricket::kUlpfecCodecName));
parameters.codecs.push_back(
cricket::VideoCodec(kSecondUlpfecPayloadType, cricket::kUlpfecCodecName));
ASSERT_TRUE(channel_->SetRecvParameters(parameters));
FakeVideoReceiveStream* recv_stream = AddRecvStream();
EXPECT_EQ(kFirstUlpfecPayloadType,
recv_stream->GetConfig().rtp.ulpfec_payload_type);
}
TEST_F(WebRtcVideoChannelTest, DuplicateRedCodecIsDropped) {
constexpr int kFirstRedPayloadType = 126;
constexpr int kSecondRedPayloadType = 127;
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(
cricket::VideoCodec(kFirstRedPayloadType, cricket::kRedCodecName));
parameters.codecs.push_back(
cricket::VideoCodec(kSecondRedPayloadType, cricket::kRedCodecName));
ASSERT_TRUE(channel_->SetRecvParameters(parameters));
FakeVideoReceiveStream* recv_stream = AddRecvStream();
EXPECT_EQ(kFirstRedPayloadType,
recv_stream->GetConfig().rtp.red_payload_type);
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithChangedRtxPayloadType) {
Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ ) Reason for revert: Downstream project relied on changed struct. Transition made possible by https://codereview.webrtc.org/2655243006/. Original issue's description: > Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ ) > > Reason for revert: > Breaks internal downstream project. > > Original issue's description: > > Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. > > > > Prior to this CL, received RTX (associated) payload types were only configured > > when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX > > SSRC was set up. > > > > After this CL, the RTX (associated) payload types are set in > > WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set > > them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since > > that is the code path that sets other SSRCs. > > > > As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed. > > We remove the possibility for each video payload type to have an associated > > specific RTX SSRC. Although the config previously allowed for this, all payload > > types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry > > did not support multiple SSRCs. This change to the config struct should thus not > > have any functional impact. The change does however affect the RtcEventLog, since > > that is used for storing the VideoReceiveStream::Configs. For simplicity, > > this CL does not change the event log proto definitions, instead duplicating > > the serialized RTX SSRCs such that they fit in the existing proto definition. > > > > BUG=webrtc:7011 > > > > Review-Url: https://codereview.webrtc.org/2646073004 > > Cr-Commit-Position: refs/heads/master@{#16302} > > Committed: https://chromium.googlesource.com/external/webrtc/+/fe2bef39cd2a5c891a49f7320514fb04324dc66c > > TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7011 > > Review-Url: https://codereview.webrtc.org/2649323010 > Cr-Commit-Position: refs/heads/master@{#16307} > Committed: https://chromium.googlesource.com/external/webrtc/+/e4974953ce0d03a60fae7659b199a6a62a79fa30 TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true # NOTREECHECKS=true # NOTRY=true BUG=webrtc:7011 Review-Url: https://codereview.webrtc.org/2654163006 Cr-Commit-Position: refs/heads/master@{#16322}
2017-01-27 04:53:07 -08:00
const int kUnusedPayloadType1 = 126;
const int kUnusedPayloadType2 = 127;
EXPECT_FALSE(FindCodecById(engine_.codecs(), kUnusedPayloadType1));
EXPECT_FALSE(FindCodecById(engine_.codecs(), kUnusedPayloadType2));
// SSRCs for RTX.
cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
params.AddFidSsrc(kSsrcs1[0], kRtxSsrcs1[0]);
AddRecvStream(params);
// Original payload type for RTX.
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
cricket::VideoCodec rtx_codec(kUnusedPayloadType1, "rtx");
rtx_codec.SetParam("apt", GetEngineCodec("VP8").id);
parameters.codecs.push_back(rtx_codec);
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
ASSERT_EQ(1U, fake_call_->GetVideoReceiveStreams().size());
const webrtc::VideoReceiveStream::Config& config_before =
fake_call_->GetVideoReceiveStreams()[0]->GetConfig();
EXPECT_EQ(1U, config_before.rtp.rtx_associated_payload_types.size());
const int* payload_type_before = FindKeyByValue(
config_before.rtp.rtx_associated_payload_types, GetEngineCodec("VP8").id);
ASSERT_NE(payload_type_before, nullptr);
EXPECT_EQ(kUnusedPayloadType1, *payload_type_before);
Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ ) Reason for revert: Downstream project relied on changed struct. Transition made possible by https://codereview.webrtc.org/2655243006/. Original issue's description: > Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ ) > > Reason for revert: > Breaks internal downstream project. > > Original issue's description: > > Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. > > > > Prior to this CL, received RTX (associated) payload types were only configured > > when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX > > SSRC was set up. > > > > After this CL, the RTX (associated) payload types are set in > > WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set > > them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since > > that is the code path that sets other SSRCs. > > > > As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed. > > We remove the possibility for each video payload type to have an associated > > specific RTX SSRC. Although the config previously allowed for this, all payload > > types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry > > did not support multiple SSRCs. This change to the config struct should thus not > > have any functional impact. The change does however affect the RtcEventLog, since > > that is used for storing the VideoReceiveStream::Configs. For simplicity, > > this CL does not change the event log proto definitions, instead duplicating > > the serialized RTX SSRCs such that they fit in the existing proto definition. > > > > BUG=webrtc:7011 > > > > Review-Url: https://codereview.webrtc.org/2646073004 > > Cr-Commit-Position: refs/heads/master@{#16302} > > Committed: https://chromium.googlesource.com/external/webrtc/+/fe2bef39cd2a5c891a49f7320514fb04324dc66c > > TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7011 > > Review-Url: https://codereview.webrtc.org/2649323010 > Cr-Commit-Position: refs/heads/master@{#16307} > Committed: https://chromium.googlesource.com/external/webrtc/+/e4974953ce0d03a60fae7659b199a6a62a79fa30 TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true # NOTREECHECKS=true # NOTRY=true BUG=webrtc:7011 Review-Url: https://codereview.webrtc.org/2654163006 Cr-Commit-Position: refs/heads/master@{#16322}
2017-01-27 04:53:07 -08:00
EXPECT_EQ(kRtxSsrcs1[0], config_before.rtp.rtx_ssrc);
// Change payload type for RTX.
parameters.codecs[1].id = kUnusedPayloadType2;
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
ASSERT_EQ(1U, fake_call_->GetVideoReceiveStreams().size());
const webrtc::VideoReceiveStream::Config& config_after =
fake_call_->GetVideoReceiveStreams()[0]->GetConfig();
EXPECT_EQ(1U, config_after.rtp.rtx_associated_payload_types.size());
const int* payload_type_after = FindKeyByValue(
config_after.rtp.rtx_associated_payload_types, GetEngineCodec("VP8").id);
ASSERT_NE(payload_type_after, nullptr);
EXPECT_EQ(kUnusedPayloadType2, *payload_type_after);
Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ ) Reason for revert: Downstream project relied on changed struct. Transition made possible by https://codereview.webrtc.org/2655243006/. Original issue's description: > Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ ) > > Reason for revert: > Breaks internal downstream project. > > Original issue's description: > > Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. > > > > Prior to this CL, received RTX (associated) payload types were only configured > > when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX > > SSRC was set up. > > > > After this CL, the RTX (associated) payload types are set in > > WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set > > them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since > > that is the code path that sets other SSRCs. > > > > As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed. > > We remove the possibility for each video payload type to have an associated > > specific RTX SSRC. Although the config previously allowed for this, all payload > > types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry > > did not support multiple SSRCs. This change to the config struct should thus not > > have any functional impact. The change does however affect the RtcEventLog, since > > that is used for storing the VideoReceiveStream::Configs. For simplicity, > > this CL does not change the event log proto definitions, instead duplicating > > the serialized RTX SSRCs such that they fit in the existing proto definition. > > > > BUG=webrtc:7011 > > > > Review-Url: https://codereview.webrtc.org/2646073004 > > Cr-Commit-Position: refs/heads/master@{#16302} > > Committed: https://chromium.googlesource.com/external/webrtc/+/fe2bef39cd2a5c891a49f7320514fb04324dc66c > > TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7011 > > Review-Url: https://codereview.webrtc.org/2649323010 > Cr-Commit-Position: refs/heads/master@{#16307} > Committed: https://chromium.googlesource.com/external/webrtc/+/e4974953ce0d03a60fae7659b199a6a62a79fa30 TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true # NOTREECHECKS=true # NOTRY=true BUG=webrtc:7011 Review-Url: https://codereview.webrtc.org/2654163006 Cr-Commit-Position: refs/heads/master@{#16322}
2017-01-27 04:53:07 -08:00
EXPECT_EQ(kRtxSsrcs1[0], config_after.rtp.rtx_ssrc);
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsDifferentPayloadType) {
cricket::VideoRecvParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs[0].id = 99;
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsAcceptDefaultCodecs) {
cricket::VideoRecvParameters parameters;
parameters.codecs = engine_.codecs();
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
FakeVideoReceiveStream* stream = AddRecvStream();
const webrtc::VideoReceiveStream::Config& config = stream->GetConfig();
EXPECT_EQ(engine_.codecs()[0].name, config.decoders[0].video_format.name);
EXPECT_EQ(engine_.codecs()[0].id, config.decoders[0].payload_type);
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsRejectUnsupportedCodec) {
cricket::VideoRecvParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(VideoCodec(101, "WTF3"));
EXPECT_FALSE(channel_->SetRecvParameters(parameters));
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsAcceptsMultipleVideoCodecs) {
cricket::VideoRecvParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithoutFecDisablesFec) {
cricket::VideoSendParameters send_parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
send_parameters.codecs.push_back(GetEngineCodec("VP8"));
send_parameters.codecs.push_back(GetEngineCodec("red"));
send_parameters.codecs.push_back(GetEngineCodec("ulpfec"));
ASSERT_TRUE(channel_->SetSendParameters(send_parameters));
FakeVideoReceiveStream* stream = AddRecvStream();
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
EXPECT_EQ(GetEngineCodec("ulpfec").id,
stream->GetConfig().rtp.ulpfec_payload_type);
cricket::VideoRecvParameters recv_parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
stream = fake_call_->GetVideoReceiveStreams()[0];
ASSERT_TRUE(stream != nullptr);
EXPECT_EQ(-1, stream->GetConfig().rtp.ulpfec_payload_type)
<< "SetSendCodec without ULPFEC should disable current ULPFEC.";
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetRecvParamsWithoutFecDisablesFec) {
AddRecvStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
const std::vector<FakeFlexfecReceiveStream*>& streams =
fake_call_->GetFlexfecReceiveStreams();
ASSERT_EQ(1U, streams.size());
const FakeFlexfecReceiveStream* stream = streams.front();
EXPECT_EQ(GetEngineCodec("flexfec-03").id, stream->GetConfig().payload_type);
EXPECT_EQ(kFlexfecSsrc, stream->GetConfig().remote_ssrc);
ASSERT_EQ(1U, stream->GetConfig().protected_media_ssrcs.size());
EXPECT_EQ(kSsrcs1[0], stream->GetConfig().protected_media_ssrcs[0]);
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
EXPECT_TRUE(streams.empty())
<< "SetSendCodec without FlexFEC should disable current FlexFEC.";
}
TEST_F(WebRtcVideoChannelTest, SetSendParamsWithFecEnablesFec) {
FakeVideoReceiveStream* stream = AddRecvStream();
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
EXPECT_EQ(GetEngineCodec("ulpfec").id,
stream->GetConfig().rtp.ulpfec_payload_type);
cricket::VideoRecvParameters recv_parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
recv_parameters.codecs.push_back(GetEngineCodec("red"));
recv_parameters.codecs.push_back(GetEngineCodec("ulpfec"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
stream = fake_call_->GetVideoReceiveStreams()[0];
ASSERT_TRUE(stream != nullptr);
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
EXPECT_EQ(GetEngineCodec("ulpfec").id,
stream->GetConfig().rtp.ulpfec_payload_type)
<< "ULPFEC should be enabled on the receive stream.";
cricket::VideoSendParameters send_parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
send_parameters.codecs.push_back(GetEngineCodec("VP8"));
send_parameters.codecs.push_back(GetEngineCodec("red"));
send_parameters.codecs.push_back(GetEngineCodec("ulpfec"));
ASSERT_TRUE(channel_->SetSendParameters(send_parameters));
stream = fake_call_->GetVideoReceiveStreams()[0];
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
EXPECT_EQ(GetEngineCodec("ulpfec").id,
stream->GetConfig().rtp.ulpfec_payload_type)
<< "ULPFEC should be enabled on the receive stream.";
}
TEST_F(WebRtcVideoChannelFlexfecSendRecvTest,
SetSendRecvParamsWithFecEnablesFec) {
AddRecvStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
const std::vector<FakeFlexfecReceiveStream*>& streams =
fake_call_->GetFlexfecReceiveStreams();
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
ASSERT_EQ(1U, streams.size());
const FakeFlexfecReceiveStream* stream_with_recv_params = streams.front();
EXPECT_EQ(GetEngineCodec("flexfec-03").id,
stream_with_recv_params->GetConfig().payload_type);
EXPECT_EQ(kFlexfecSsrc, stream_with_recv_params->GetConfig().remote_ssrc);
EXPECT_EQ(1U,
stream_with_recv_params->GetConfig().protected_media_ssrcs.size());
EXPECT_EQ(kSsrcs1[0],
stream_with_recv_params->GetConfig().protected_media_ssrcs[0]);
cricket::VideoSendParameters send_parameters;
send_parameters.codecs.push_back(GetEngineCodec("VP8"));
send_parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
ASSERT_TRUE(channel_->SetSendParameters(send_parameters));
ASSERT_EQ(1U, streams.size());
const FakeFlexfecReceiveStream* stream_with_send_params = streams.front();
EXPECT_EQ(GetEngineCodec("flexfec-03").id,
stream_with_send_params->GetConfig().payload_type);
EXPECT_EQ(kFlexfecSsrc, stream_with_send_params->GetConfig().remote_ssrc);
EXPECT_EQ(1U,
stream_with_send_params->GetConfig().protected_media_ssrcs.size());
EXPECT_EQ(kSsrcs1[0],
stream_with_send_params->GetConfig().protected_media_ssrcs[0]);
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsRejectDuplicateFecPayloads) {
cricket::VideoRecvParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("red"));
parameters.codecs[1].id = parameters.codecs[0].id;
EXPECT_FALSE(channel_->SetRecvParameters(parameters));
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest,
SetRecvCodecsRejectDuplicateFecPayloads) {
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
parameters.codecs[1].id = parameters.codecs[0].id;
EXPECT_FALSE(channel_->SetRecvParameters(parameters));
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsRejectDuplicateCodecPayloads) {
cricket::VideoRecvParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
parameters.codecs[1].id = parameters.codecs[0].id;
EXPECT_FALSE(channel_->SetRecvParameters(parameters));
}
TEST_F(WebRtcVideoChannelTest,
SetRecvCodecsAcceptSameCodecOnMultiplePayloadTypes) {
cricket::VideoRecvParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs[1].id += 1;
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
}
// Test that setting the same codecs but with a different order
// doesn't result in the stream being recreated.
TEST_F(WebRtcVideoChannelTest,
SetRecvCodecsDifferentOrderDoesntRecreateStream) {
cricket::VideoRecvParameters parameters1;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters1.codecs.push_back(GetEngineCodec("VP8"));
parameters1.codecs.push_back(GetEngineCodec("red"));
EXPECT_TRUE(channel_->SetRecvParameters(parameters1));
AddRecvStream(cricket::StreamParams::CreateLegacy(123));
EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams());
cricket::VideoRecvParameters parameters2;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters2.codecs.push_back(GetEngineCodec("red"));
parameters2.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel_->SetRecvParameters(parameters2));
EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams());
}
TEST_F(WebRtcVideoChannelTest, SendStreamNotSendingByDefault) {
EXPECT_FALSE(AddSendStream()->IsSending());
}
TEST_F(WebRtcVideoChannelTest, ReceiveStreamReceivingByDefault) {
EXPECT_TRUE(AddRecvStream()->IsReceiving());
}
TEST_F(WebRtcVideoChannelTest, SetSend) {
FakeVideoSendStream* stream = AddSendStream();
EXPECT_FALSE(stream->IsSending());
// false->true
EXPECT_TRUE(channel_->SetSend(true));
EXPECT_TRUE(stream->IsSending());
// true->true
EXPECT_TRUE(channel_->SetSend(true));
EXPECT_TRUE(stream->IsSending());
// true->false
EXPECT_TRUE(channel_->SetSend(false));
EXPECT_FALSE(stream->IsSending());
// false->false
EXPECT_TRUE(channel_->SetSend(false));
EXPECT_FALSE(stream->IsSending());
EXPECT_TRUE(channel_->SetSend(true));
FakeVideoSendStream* new_stream = AddSendStream();
EXPECT_TRUE(new_stream->IsSending())
<< "Send stream created after SetSend(true) not sending initially.";
}
// This test verifies DSCP settings are properly applied on video media channel.
TEST_F(WebRtcVideoChannelTest, TestSetDscpOptions) {
std::unique_ptr<cricket::FakeNetworkInterface> network_interface(
new cricket::FakeNetworkInterface);
MediaConfig config;
std::unique_ptr<cricket::WebRtcVideoChannel> channel;
webrtc::RtpParameters parameters;
channel.reset(
static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateMediaChannel(
call_.get(), config, VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get())));
channel->SetInterface(network_interface.get(),
webrtc::MediaTransportConfig());
// Default value when DSCP is disabled should be DSCP_DEFAULT.
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
// Default value when DSCP is enabled is also DSCP_DEFAULT, until it is set
// through rtp parameters.
config.enable_dscp = true;
channel.reset(
static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateMediaChannel(
call_.get(), config, VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get())));
channel->SetInterface(network_interface.get(),
webrtc::MediaTransportConfig());
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
// Create a send stream to configure
EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc)));
parameters = channel->GetRtpSendParameters(kSsrc);
ASSERT_FALSE(parameters.encodings.empty());
// Various priorities map to various dscp values.
parameters.encodings[0].network_priority = 4.0;
ASSERT_TRUE(channel->SetRtpSendParameters(kSsrc, parameters).ok());
EXPECT_EQ(rtc::DSCP_AF41, network_interface->dscp());
parameters.encodings[0].network_priority = 0.5;
ASSERT_TRUE(channel->SetRtpSendParameters(kSsrc, parameters).ok());
EXPECT_EQ(rtc::DSCP_CS1, network_interface->dscp());
// A bad priority does not change the dscp value.
parameters.encodings[0].network_priority = 0.0;
ASSERT_FALSE(channel->SetRtpSendParameters(kSsrc, parameters).ok());
EXPECT_EQ(rtc::DSCP_CS1, network_interface->dscp());
// Packets should also self-identify their dscp in PacketOptions.
const uint8_t kData[10] = {0};
EXPECT_TRUE(static_cast<webrtc::Transport*>(channel.get())
->SendRtcp(kData, sizeof(kData)));
EXPECT_EQ(rtc::DSCP_CS1, network_interface->options().dscp);
// Verify that setting the option to false resets the
// DiffServCodePoint.
config.enable_dscp = false;
channel.reset(
static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateMediaChannel(
call_.get(), config, VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get())));
channel->SetInterface(network_interface.get(),
webrtc::MediaTransportConfig());
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
}
// This test verifies that the RTCP reduced size mode is properly applied to
// send video streams.
TEST_F(WebRtcVideoChannelTest, TestSetSendRtcpReducedSize) {
// Create stream, expecting that default mode is "compound".
FakeVideoSendStream* stream1 = AddSendStream();
EXPECT_EQ(webrtc::RtcpMode::kCompound, stream1->GetConfig().rtp.rtcp_mode);
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_FALSE(rtp_parameters.rtcp.reduced_size);
// Now enable reduced size mode.
send_parameters_.rtcp.reduced_size = true;
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
stream1 = fake_call_->GetVideoSendStreams()[0];
EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream1->GetConfig().rtp.rtcp_mode);
rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_TRUE(rtp_parameters.rtcp.reduced_size);
// Create a new stream and ensure it picks up the reduced size mode.
FakeVideoSendStream* stream2 = AddSendStream();
EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream2->GetConfig().rtp.rtcp_mode);
}
// This test verifies that the RTCP reduced size mode is properly applied to
// receive video streams.
TEST_F(WebRtcVideoChannelTest, TestSetRecvRtcpReducedSize) {
// Create stream, expecting that default mode is "compound".
FakeVideoReceiveStream* stream1 = AddRecvStream();
EXPECT_EQ(webrtc::RtcpMode::kCompound, stream1->GetConfig().rtp.rtcp_mode);
// Now enable reduced size mode.
// TODO(deadbeef): Once "recv_parameters" becomes "receiver_parameters",
// the reduced_size flag should come from that.
send_parameters_.rtcp.reduced_size = true;
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
stream1 = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream1->GetConfig().rtp.rtcp_mode);
// Create a new stream and ensure it picks up the reduced size mode.
FakeVideoReceiveStream* stream2 = AddRecvStream();
EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream2->GetConfig().rtp.rtcp_mode);
}
TEST_F(WebRtcVideoChannelTest, OnReadyToSendSignalsNetworkState) {
EXPECT_EQ(webrtc::kNetworkUp,
fake_call_->GetNetworkState(webrtc::MediaType::VIDEO));
EXPECT_EQ(webrtc::kNetworkUp,
fake_call_->GetNetworkState(webrtc::MediaType::AUDIO));
channel_->OnReadyToSend(false);
EXPECT_EQ(webrtc::kNetworkDown,
fake_call_->GetNetworkState(webrtc::MediaType::VIDEO));
EXPECT_EQ(webrtc::kNetworkUp,
fake_call_->GetNetworkState(webrtc::MediaType::AUDIO));
channel_->OnReadyToSend(true);
EXPECT_EQ(webrtc::kNetworkUp,
fake_call_->GetNetworkState(webrtc::MediaType::VIDEO));
EXPECT_EQ(webrtc::kNetworkUp,
fake_call_->GetNetworkState(webrtc::MediaType::AUDIO));
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsSentCodecName) {
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel_->SetSendParameters(parameters));
AddSendStream();
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
EXPECT_EQ("VP8", info.senders[0].codec_name);
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsEncoderImplementationName) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.encoder_implementation_name = "encoder_implementation_name";
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(stats.encoder_implementation_name,
info.senders[0].encoder_implementation_name);
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsCpuOveruseMetrics) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.avg_encode_time_ms = 13;
stats.encode_usage_percent = 42;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(stats.avg_encode_time_ms, info.senders[0].avg_encode_ms);
EXPECT_EQ(stats.encode_usage_percent, info.senders[0].encode_usage_percent);
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsFramesEncoded) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.frames_encoded = 13;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(stats.frames_encoded, info.senders[0].frames_encoded);
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsKeyFramesEncoded) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.substreams[123].frame_counts.key_frames = 10;
stats.substreams[456].frame_counts.key_frames = 87;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
// TODO(bugs.webrtc.org/9547): Populate individual outbound-rtp stats objects
// for each simulcast stream, instead of accumulating all keyframes encoded
// over all simulcast streams in the same outbound-rtp stats object.
EXPECT_EQ(97u, info.senders[0].key_frames_encoded);
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsQpSum) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.qp_sum = 13;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(stats.qp_sum, info.senders[0].qp_sum);
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsUpperResolution) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.substreams[17].width = 123;
stats.substreams[17].height = 40;
stats.substreams[42].width = 80;
stats.substreams[42].height = 31;
stats.substreams[11].width = 20;
stats.substreams[11].height = 90;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(1u, info.senders.size());
EXPECT_EQ(123, info.senders[0].send_frame_width);
EXPECT_EQ(90, info.senders[0].send_frame_height);
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsCpuAdaptationStats) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.number_of_cpu_adapt_changes = 2;
stats.cpu_limited_resolution = true;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
EXPECT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(1U, info.senders.size());
EXPECT_EQ(WebRtcVideoChannel::ADAPTREASON_CPU, info.senders[0].adapt_reason);
EXPECT_EQ(stats.number_of_cpu_adapt_changes, info.senders[0].adapt_changes);
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsAdaptationAndBandwidthStats) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.number_of_cpu_adapt_changes = 2;
stats.cpu_limited_resolution = true;
stats.bw_limited_resolution = true;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
EXPECT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(1U, info.senders.size());
EXPECT_EQ(WebRtcVideoChannel::ADAPTREASON_CPU |
WebRtcVideoChannel::ADAPTREASON_BANDWIDTH,
info.senders[0].adapt_reason);
EXPECT_EQ(stats.number_of_cpu_adapt_changes, info.senders[0].adapt_changes);
}
TEST_F(WebRtcVideoChannelTest,
GetStatsTranslatesBandwidthLimitedResolutionCorrectly) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.bw_limited_resolution = true;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
EXPECT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(1U, info.senders.size());
EXPECT_EQ(WebRtcVideoChannel::ADAPTREASON_BANDWIDTH,
info.senders[0].adapt_reason);
}
TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesSendRtcpPacketTypesCorrectly) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.substreams[17].rtcp_packet_type_counts.fir_packets = 2;
stats.substreams[17].rtcp_packet_type_counts.nack_packets = 3;
stats.substreams[17].rtcp_packet_type_counts.pli_packets = 4;
stats.substreams[42].rtcp_packet_type_counts.fir_packets = 5;
stats.substreams[42].rtcp_packet_type_counts.nack_packets = 7;
stats.substreams[42].rtcp_packet_type_counts.pli_packets = 9;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(7, info.senders[0].firs_rcvd);
EXPECT_EQ(10, info.senders[0].nacks_rcvd);
EXPECT_EQ(13, info.senders[0].plis_rcvd);
}
TEST_F(WebRtcVideoChannelTest,
GetStatsTranslatesReceiveRtcpPacketTypesCorrectly) {
FakeVideoReceiveStream* stream = AddRecvStream();
webrtc::VideoReceiveStream::Stats stats;
stats.rtcp_packet_type_counts.fir_packets = 2;
stats.rtcp_packet_type_counts.nack_packets = 3;
stats.rtcp_packet_type_counts.pli_packets = 4;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(stats.rtcp_packet_type_counts.fir_packets,
rtc::checked_cast<unsigned int>(info.receivers[0].firs_sent));
EXPECT_EQ(stats.rtcp_packet_type_counts.nack_packets,
rtc::checked_cast<unsigned int>(info.receivers[0].nacks_sent));
EXPECT_EQ(stats.rtcp_packet_type_counts.pli_packets,
rtc::checked_cast<unsigned int>(info.receivers[0].plis_sent));
}
TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesDecodeStatsCorrectly) {
FakeVideoReceiveStream* stream = AddRecvStream();
webrtc::VideoReceiveStream::Stats stats;
stats.decoder_implementation_name = "decoder_implementation_name";
stats.decode_ms = 2;
stats.max_decode_ms = 3;
stats.current_delay_ms = 4;
stats.target_delay_ms = 5;
stats.jitter_buffer_ms = 6;
stats.jitter_buffer_delay_seconds = 60;
stats.jitter_buffer_emitted_count = 6;
stats.min_playout_delay_ms = 7;
stats.render_delay_ms = 8;
stats.width = 9;
stats.height = 10;
stats.frame_counts.key_frames = 11;
stats.frame_counts.delta_frames = 12;
stats.frames_rendered = 13;
stats.frames_decoded = 14;
stats.qp_sum = 15;
stats.total_decode_time_ms = 16;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(stats.decoder_implementation_name,
info.receivers[0].decoder_implementation_name);
EXPECT_EQ(stats.decode_ms, info.receivers[0].decode_ms);
EXPECT_EQ(stats.max_decode_ms, info.receivers[0].max_decode_ms);
EXPECT_EQ(stats.current_delay_ms, info.receivers[0].current_delay_ms);
EXPECT_EQ(stats.target_delay_ms, info.receivers[0].target_delay_ms);
EXPECT_EQ(stats.jitter_buffer_ms, info.receivers[0].jitter_buffer_ms);
EXPECT_EQ(stats.jitter_buffer_delay_seconds,
info.receivers[0].jitter_buffer_delay_seconds);
EXPECT_EQ(stats.jitter_buffer_emitted_count,
info.receivers[0].jitter_buffer_emitted_count);
EXPECT_EQ(stats.min_playout_delay_ms, info.receivers[0].min_playout_delay_ms);
EXPECT_EQ(stats.render_delay_ms, info.receivers[0].render_delay_ms);
EXPECT_EQ(stats.width, info.receivers[0].frame_width);
EXPECT_EQ(stats.height, info.receivers[0].frame_height);
EXPECT_EQ(rtc::checked_cast<unsigned int>(stats.frame_counts.key_frames +
stats.frame_counts.delta_frames),
info.receivers[0].frames_received);
EXPECT_EQ(stats.frames_rendered, info.receivers[0].frames_rendered);
EXPECT_EQ(stats.frames_decoded, info.receivers[0].frames_decoded);
EXPECT_EQ(rtc::checked_cast<unsigned int>(stats.frame_counts.key_frames),
info.receivers[0].key_frames_decoded);
EXPECT_EQ(stats.qp_sum, info.receivers[0].qp_sum);
EXPECT_EQ(stats.total_decode_time_ms, info.receivers[0].total_decode_time_ms);
}
TEST_F(WebRtcVideoChannelTest,
GetStatsTranslatesInterFrameDelayStatsCorrectly) {
FakeVideoReceiveStream* stream = AddRecvStream();
webrtc::VideoReceiveStream::Stats stats;
stats.total_inter_frame_delay = 0.123;
stats.total_squared_inter_frame_delay = 0.00456;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(stats.total_inter_frame_delay,
info.receivers[0].total_inter_frame_delay);
EXPECT_EQ(stats.total_squared_inter_frame_delay,
info.receivers[0].total_squared_inter_frame_delay);
}
TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesReceivePacketStatsCorrectly) {
FakeVideoReceiveStream* stream = AddRecvStream();
webrtc::VideoReceiveStream::Stats stats;
stats.rtp_stats.packet_counter.payload_bytes = 2;
stats.rtp_stats.packet_counter.header_bytes = 3;
stats.rtp_stats.packet_counter.padding_bytes = 4;
stats.rtp_stats.packet_counter.packets = 5;
stats.rtp_stats.packets_lost = 6;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(stats.rtp_stats.packet_counter.payload_bytes,
rtc::checked_cast<size_t>(info.receivers[0].payload_bytes_rcvd));
EXPECT_EQ(stats.rtp_stats.packet_counter.packets,
rtc::checked_cast<unsigned int>(info.receivers[0].packets_rcvd));
EXPECT_EQ(stats.rtp_stats.packets_lost, info.receivers[0].packets_lost);
}
TEST_F(WebRtcVideoChannelTest, TranslatesCallStatsCorrectly) {
AddSendStream();
AddSendStream();
webrtc::Call::Stats stats;
stats.rtt_ms = 123;
fake_call_->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(2u, info.senders.size());
EXPECT_EQ(stats.rtt_ms, info.senders[0].rtt_ms);
EXPECT_EQ(stats.rtt_ms, info.senders[1].rtt_ms);
}
TEST_F(WebRtcVideoChannelTest, TranslatesSenderBitrateStatsCorrectly) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.target_media_bitrate_bps = 156;
stats.media_bitrate_bps = 123;
stats.substreams[17].total_bitrate_bps = 1;
stats.substreams[17].retransmit_bitrate_bps = 2;
stats.substreams[42].total_bitrate_bps = 3;
stats.substreams[42].retransmit_bitrate_bps = 4;
stream->SetStats(stats);
FakeVideoSendStream* stream2 = AddSendStream();
webrtc::VideoSendStream::Stats stats2;
stats2.target_media_bitrate_bps = 200;
stats2.media_bitrate_bps = 321;
stats2.substreams[13].total_bitrate_bps = 5;
stats2.substreams[13].retransmit_bitrate_bps = 6;
stats2.substreams[21].total_bitrate_bps = 7;
stats2.substreams[21].retransmit_bitrate_bps = 8;
stream2->SetStats(stats2);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(2u, info.senders.size());
BandwidthEstimationInfo bwe_info;
channel_->FillBitrateInfo(&bwe_info);
// Assuming stream and stream2 corresponds to senders[0] and [1] respectively
// is OK as std::maps are sorted and AddSendStream() gives increasing SSRCs.
EXPECT_EQ(stats.media_bitrate_bps, info.senders[0].nominal_bitrate);
EXPECT_EQ(stats2.media_bitrate_bps, info.senders[1].nominal_bitrate);
EXPECT_EQ(stats.target_media_bitrate_bps + stats2.target_media_bitrate_bps,
bwe_info.target_enc_bitrate);
EXPECT_EQ(stats.media_bitrate_bps + stats2.media_bitrate_bps,
bwe_info.actual_enc_bitrate);
EXPECT_EQ(1 + 3 + 5 + 7, bwe_info.transmit_bitrate)
<< "Bandwidth stats should take all streams into account.";
EXPECT_EQ(2 + 4 + 6 + 8, bwe_info.retransmit_bitrate)
<< "Bandwidth stats should take all streams into account.";
}
TEST_F(WebRtcVideoChannelTest, DefaultReceiveStreamReconfiguresToUseRtx) {
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs1);
const std::vector<uint32_t> rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1);
ASSERT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
const size_t kDataLength = 12;
uint8_t data[kDataLength];
memset(data, 0, sizeof(data));
rtc::SetBE32(&data[8], ssrcs[0]);
rtc::CopyOnWriteBuffer packet(data, kDataLength);
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size())
<< "No default receive stream created.";
FakeVideoReceiveStream* recv_stream = fake_call_->GetVideoReceiveStreams()[0];
Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ ) Reason for revert: Downstream project relied on changed struct. Transition made possible by https://codereview.webrtc.org/2655243006/. Original issue's description: > Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ ) > > Reason for revert: > Breaks internal downstream project. > > Original issue's description: > > Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. > > > > Prior to this CL, received RTX (associated) payload types were only configured > > when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX > > SSRC was set up. > > > > After this CL, the RTX (associated) payload types are set in > > WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set > > them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since > > that is the code path that sets other SSRCs. > > > > As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed. > > We remove the possibility for each video payload type to have an associated > > specific RTX SSRC. Although the config previously allowed for this, all payload > > types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry > > did not support multiple SSRCs. This change to the config struct should thus not > > have any functional impact. The change does however affect the RtcEventLog, since > > that is used for storing the VideoReceiveStream::Configs. For simplicity, > > this CL does not change the event log proto definitions, instead duplicating > > the serialized RTX SSRCs such that they fit in the existing proto definition. > > > > BUG=webrtc:7011 > > > > Review-Url: https://codereview.webrtc.org/2646073004 > > Cr-Commit-Position: refs/heads/master@{#16302} > > Committed: https://chromium.googlesource.com/external/webrtc/+/fe2bef39cd2a5c891a49f7320514fb04324dc66c > > TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7011 > > Review-Url: https://codereview.webrtc.org/2649323010 > Cr-Commit-Position: refs/heads/master@{#16307} > Committed: https://chromium.googlesource.com/external/webrtc/+/e4974953ce0d03a60fae7659b199a6a62a79fa30 TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true # NOTREECHECKS=true # NOTRY=true BUG=webrtc:7011 Review-Url: https://codereview.webrtc.org/2654163006 Cr-Commit-Position: refs/heads/master@{#16322}
2017-01-27 04:53:07 -08:00
EXPECT_EQ(0u, recv_stream->GetConfig().rtp.rtx_ssrc)
<< "Default receive stream should not have configured RTX";
EXPECT_TRUE(channel_->AddRecvStream(
cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs)));
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size())
Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ ) Reason for revert: Downstream project relied on changed struct. Transition made possible by https://codereview.webrtc.org/2655243006/. Original issue's description: > Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ ) > > Reason for revert: > Breaks internal downstream project. > > Original issue's description: > > Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. > > > > Prior to this CL, received RTX (associated) payload types were only configured > > when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX > > SSRC was set up. > > > > After this CL, the RTX (associated) payload types are set in > > WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set > > them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since > > that is the code path that sets other SSRCs. > > > > As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed. > > We remove the possibility for each video payload type to have an associated > > specific RTX SSRC. Although the config previously allowed for this, all payload > > types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry > > did not support multiple SSRCs. This change to the config struct should thus not > > have any functional impact. The change does however affect the RtcEventLog, since > > that is used for storing the VideoReceiveStream::Configs. For simplicity, > > this CL does not change the event log proto definitions, instead duplicating > > the serialized RTX SSRCs such that they fit in the existing proto definition. > > > > BUG=webrtc:7011 > > > > Review-Url: https://codereview.webrtc.org/2646073004 > > Cr-Commit-Position: refs/heads/master@{#16302} > > Committed: https://chromium.googlesource.com/external/webrtc/+/fe2bef39cd2a5c891a49f7320514fb04324dc66c > > TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7011 > > Review-Url: https://codereview.webrtc.org/2649323010 > Cr-Commit-Position: refs/heads/master@{#16307} > Committed: https://chromium.googlesource.com/external/webrtc/+/e4974953ce0d03a60fae7659b199a6a62a79fa30 TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true # NOTREECHECKS=true # NOTRY=true BUG=webrtc:7011 Review-Url: https://codereview.webrtc.org/2654163006 Cr-Commit-Position: refs/heads/master@{#16322}
2017-01-27 04:53:07 -08:00
<< "AddRecvStream should have reconfigured, not added a new receiver.";
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_FALSE(
recv_stream->GetConfig().rtp.rtx_associated_payload_types.empty());
Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ ) Reason for revert: Identified a configuration problem in the video quality tests. Intend to fix and reland. Original issue's description: > Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ ) > > Reason for revert: > This change appears to break ulpfec, with severe regressions, e.g., for webrtc_perf_test FullStackTest.ForemanCifPlr5Ulpfec > > Original issue's description: > > Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ ) > > > > Reason for revert: > > Intend to fix perf failures and reland. > > > > Original issue's description: > > > Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ ) > > > > > > Reason for revert: > > > A few perf tests broken, including > > > > > > RampUpTest.UpDownUpAbsSendTimeSimulcastRedRtx > > > RampUpTest.UpDownUpTransportSequenceNumberRtx > > > RampUpTest.UpDownUpTransportSequenceNumberPacketLoss > > > > > > > > > Original issue's description: > > > > Use RtxReceiveStream. > > > > > > > > This also has the beneficial side-effect that when a media stream > > > > which is protected by FlexFEC receives an RTX retransmission, the > > > > retransmitted media packet is passed into the FlexFEC machinery, > > > > which should improve its ability to recover packets via FEC. > > > > > > > > BUG=webrtc:7135 > > > > > > > > Review-Url: https://codereview.webrtc.org/3008773002 > > > > Cr-Commit-Position: refs/heads/master@{#19649} > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/5c0f6c62ea3b1d2c43f8fc152961af27033475f7 > > > > > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > > > # Skipping CQ checks because original CL landed less than 1 days ago. > > > NOPRESUBMIT=true > > > NOTREECHECKS=true > > > NOTRY=true > > > BUG=webrtc:7135 > > > > > > Review-Url: https://codereview.webrtc.org/3010983002 > > > Cr-Commit-Position: refs/heads/master@{#19653} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/3c39c0137afa274d1d524b150b50304b38a2847b > > > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > > # Skipping CQ checks because original CL landed less than 1 days ago. > > NOPRESUBMIT=true > > NOTREECHECKS=true > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/3006063002 > > Cr-Commit-Position: refs/heads/master@{#19715} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35713eaf565c0fef07c8afc158d7b8fdf7ec3d78 > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/3007303002 > Cr-Commit-Position: refs/heads/master@{#19744} > Committed: https://chromium.googlesource.com/external/webrtc/+/8e7eee035178a7f10e19883681b5eaa4a7523107 TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/3012963002 Cr-Commit-Position: refs/heads/master@{#19765}
2017-09-11 02:32:16 -07:00
EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig()))
<< "RTX should be mapped for all decoders/payload types.";
Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ ) Reason for revert: Identified a configuration problem in the video quality tests. Intend to fix and reland. Original issue's description: > Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ ) > > Reason for revert: > This change appears to break ulpfec, with severe regressions, e.g., for webrtc_perf_test FullStackTest.ForemanCifPlr5Ulpfec > > Original issue's description: > > Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ ) > > > > Reason for revert: > > Intend to fix perf failures and reland. > > > > Original issue's description: > > > Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ ) > > > > > > Reason for revert: > > > A few perf tests broken, including > > > > > > RampUpTest.UpDownUpAbsSendTimeSimulcastRedRtx > > > RampUpTest.UpDownUpTransportSequenceNumberRtx > > > RampUpTest.UpDownUpTransportSequenceNumberPacketLoss > > > > > > > > > Original issue's description: > > > > Use RtxReceiveStream. > > > > > > > > This also has the beneficial side-effect that when a media stream > > > > which is protected by FlexFEC receives an RTX retransmission, the > > > > retransmitted media packet is passed into the FlexFEC machinery, > > > > which should improve its ability to recover packets via FEC. > > > > > > > > BUG=webrtc:7135 > > > > > > > > Review-Url: https://codereview.webrtc.org/3008773002 > > > > Cr-Commit-Position: refs/heads/master@{#19649} > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/5c0f6c62ea3b1d2c43f8fc152961af27033475f7 > > > > > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > > > # Skipping CQ checks because original CL landed less than 1 days ago. > > > NOPRESUBMIT=true > > > NOTREECHECKS=true > > > NOTRY=true > > > BUG=webrtc:7135 > > > > > > Review-Url: https://codereview.webrtc.org/3010983002 > > > Cr-Commit-Position: refs/heads/master@{#19653} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/3c39c0137afa274d1d524b150b50304b38a2847b > > > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > > # Skipping CQ checks because original CL landed less than 1 days ago. > > NOPRESUBMIT=true > > NOTREECHECKS=true > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/3006063002 > > Cr-Commit-Position: refs/heads/master@{#19715} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35713eaf565c0fef07c8afc158d7b8fdf7ec3d78 > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/3007303002 > Cr-Commit-Position: refs/heads/master@{#19744} > Committed: https://chromium.googlesource.com/external/webrtc/+/8e7eee035178a7f10e19883681b5eaa4a7523107 TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/3012963002 Cr-Commit-Position: refs/heads/master@{#19765}
2017-09-11 02:32:16 -07:00
EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(),
GetEngineCodec("red").id))
<< "RTX should be mapped also for the RED payload type";
Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ ) Reason for revert: Downstream project relied on changed struct. Transition made possible by https://codereview.webrtc.org/2655243006/. Original issue's description: > Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ ) > > Reason for revert: > Breaks internal downstream project. > > Original issue's description: > > Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. > > > > Prior to this CL, received RTX (associated) payload types were only configured > > when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX > > SSRC was set up. > > > > After this CL, the RTX (associated) payload types are set in > > WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set > > them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since > > that is the code path that sets other SSRCs. > > > > As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed. > > We remove the possibility for each video payload type to have an associated > > specific RTX SSRC. Although the config previously allowed for this, all payload > > types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry > > did not support multiple SSRCs. This change to the config struct should thus not > > have any functional impact. The change does however affect the RtcEventLog, since > > that is used for storing the VideoReceiveStream::Configs. For simplicity, > > this CL does not change the event log proto definitions, instead duplicating > > the serialized RTX SSRCs such that they fit in the existing proto definition. > > > > BUG=webrtc:7011 > > > > Review-Url: https://codereview.webrtc.org/2646073004 > > Cr-Commit-Position: refs/heads/master@{#16302} > > Committed: https://chromium.googlesource.com/external/webrtc/+/fe2bef39cd2a5c891a49f7320514fb04324dc66c > > TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7011 > > Review-Url: https://codereview.webrtc.org/2649323010 > Cr-Commit-Position: refs/heads/master@{#16307} > Committed: https://chromium.googlesource.com/external/webrtc/+/e4974953ce0d03a60fae7659b199a6a62a79fa30 TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true # NOTREECHECKS=true # NOTRY=true BUG=webrtc:7011 Review-Url: https://codereview.webrtc.org/2654163006 Cr-Commit-Position: refs/heads/master@{#16322}
2017-01-27 04:53:07 -08:00
EXPECT_EQ(rtx_ssrcs[0], recv_stream->GetConfig().rtp.rtx_ssrc);
}
TEST_F(WebRtcVideoChannelTest, RejectsAddingStreamsWithMissingSsrcsForRtx) {
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs1);
const std::vector<uint32_t> rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1);
StreamParams sp =
cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs);
sp.ssrcs = ssrcs; // Without RTXs, this is the important part.
EXPECT_FALSE(channel_->AddSendStream(sp));
EXPECT_FALSE(channel_->AddRecvStream(sp));
}
TEST_F(WebRtcVideoChannelTest, RejectsAddingStreamsWithOverlappingRtxSsrcs) {
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs1);
const std::vector<uint32_t> rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1);
StreamParams sp =
cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs);
EXPECT_TRUE(channel_->AddSendStream(sp));
EXPECT_TRUE(channel_->AddRecvStream(sp));
// The RTX SSRC is already used in previous streams, using it should fail.
sp = cricket::StreamParams::CreateLegacy(rtx_ssrcs[0]);
EXPECT_FALSE(channel_->AddSendStream(sp));
EXPECT_FALSE(channel_->AddRecvStream(sp));
// After removing the original stream this should be fine to add (makes sure
// that RTX ssrcs are not forever taken).
EXPECT_TRUE(channel_->RemoveSendStream(ssrcs[0]));
EXPECT_TRUE(channel_->RemoveRecvStream(ssrcs[0]));
EXPECT_TRUE(channel_->AddSendStream(sp));
EXPECT_TRUE(channel_->AddRecvStream(sp));
}
TEST_F(WebRtcVideoChannelTest,
RejectsAddingStreamsWithOverlappingSimulcastSsrcs) {
static const uint32_t kFirstStreamSsrcs[] = {1, 2, 3};
static const uint32_t kOverlappingStreamSsrcs[] = {4, 3, 5};
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
StreamParams sp =
cricket::CreateSimStreamParams("cname", MAKE_VECTOR(kFirstStreamSsrcs));
EXPECT_TRUE(channel_->AddSendStream(sp));
EXPECT_TRUE(channel_->AddRecvStream(sp));
// One of the SSRCs is already used in previous streams, using it should fail.
sp = cricket::CreateSimStreamParams("cname",
MAKE_VECTOR(kOverlappingStreamSsrcs));
EXPECT_FALSE(channel_->AddSendStream(sp));
EXPECT_FALSE(channel_->AddRecvStream(sp));
// After removing the original stream this should be fine to add (makes sure
// that RTX ssrcs are not forever taken).
EXPECT_TRUE(channel_->RemoveSendStream(kFirstStreamSsrcs[0]));
EXPECT_TRUE(channel_->RemoveRecvStream(kFirstStreamSsrcs[0]));
EXPECT_TRUE(channel_->AddSendStream(sp));
EXPECT_TRUE(channel_->AddRecvStream(sp));
}
TEST_F(WebRtcVideoChannelTest, ReportsSsrcGroupsInStats) {
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
static const uint32_t kSenderSsrcs[] = {4, 7, 10};
static const uint32_t kSenderRtxSsrcs[] = {5, 8, 11};
StreamParams sender_sp = cricket::CreateSimWithRtxStreamParams(
"cname", MAKE_VECTOR(kSenderSsrcs), MAKE_VECTOR(kSenderRtxSsrcs));
EXPECT_TRUE(channel_->AddSendStream(sender_sp));
static const uint32_t kReceiverSsrcs[] = {3};
static const uint32_t kReceiverRtxSsrcs[] = {2};
StreamParams receiver_sp = cricket::CreateSimWithRtxStreamParams(
"cname", MAKE_VECTOR(kReceiverSsrcs), MAKE_VECTOR(kReceiverRtxSsrcs));
EXPECT_TRUE(channel_->AddRecvStream(receiver_sp));
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(1u, info.senders.size());
ASSERT_EQ(1u, info.receivers.size());
EXPECT_NE(sender_sp.ssrc_groups, receiver_sp.ssrc_groups);
EXPECT_EQ(sender_sp.ssrc_groups, info.senders[0].ssrc_groups);
EXPECT_EQ(receiver_sp.ssrc_groups, info.receivers[0].ssrc_groups);
}
TEST_F(WebRtcVideoChannelTest, MapsReceivedPayloadTypeToCodecName) {
FakeVideoReceiveStream* stream = AddRecvStream();
webrtc::VideoReceiveStream::Stats stats;
cricket::VideoMediaInfo info;
// Report no codec name before receiving.
stream->SetStats(stats);
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_STREQ("", info.receivers[0].codec_name.c_str());
// Report VP8 if we're receiving it.
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
stats.current_payload_type = GetEngineCodec("VP8").id;
stream->SetStats(stats);
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_STREQ(kVp8CodecName, info.receivers[0].codec_name.c_str());
// Report no codec name for unknown playload types.
stats.current_payload_type = 3;
stream->SetStats(stats);
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_STREQ("", info.receivers[0].codec_name.c_str());
}
// Tests that when we add a stream without SSRCs, but contains a stream_id
// that it is stored and its stream id is later used when the first packet
// arrives to properly create a receive stream with a sync label.
TEST_F(WebRtcVideoChannelTest, RecvUnsignaledSsrcWithSignaledStreamId) {
const char kSyncLabel[] = "sync_label";
cricket::StreamParams unsignaled_stream;
unsignaled_stream.set_stream_ids({kSyncLabel});
ASSERT_TRUE(channel_->AddRecvStream(unsignaled_stream));
// The stream shouldn't have been created at this point because it doesn't
// have any SSRCs.
EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
// Create and deliver packet.
const size_t kDataLength = 12;
uint8_t data[kDataLength];
memset(data, 0, sizeof(data));
rtc::SetBE32(&data[8], kIncomingUnsignalledSsrc);
rtc::CopyOnWriteBuffer packet(data, kDataLength);
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
// The stream should now be created with the appropriate sync label.
EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
EXPECT_EQ(kSyncLabel,
fake_call_->GetVideoReceiveStreams()[0]->GetConfig().sync_group);
// Reset the unsignaled stream to clear the cache. This time when
// a default video receive stream is created it won't have a sync_group.
channel_->ResetUnsignaledRecvStream();
ASSERT_TRUE(channel_->RemoveRecvStream(kIncomingUnsignalledSsrc));
EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
EXPECT_TRUE(
fake_call_->GetVideoReceiveStreams()[0]->GetConfig().sync_group.empty());
}
// Test BaseMinimumPlayoutDelayMs on receive streams.
TEST_F(WebRtcVideoChannelTest, BaseMinimumPlayoutDelayMs) {
// Test that set won't work for non-existing receive streams.
EXPECT_FALSE(channel_->SetBaseMinimumPlayoutDelayMs(kSsrc + 2, 200));
// Test that get won't work for non-existing receive streams.
EXPECT_FALSE(channel_->GetBaseMinimumPlayoutDelayMs(kSsrc + 2));
EXPECT_TRUE(AddRecvStream());
// Test that set works for the existing receive stream.
EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(last_ssrc_, 200));
auto* recv_stream = fake_call_->GetVideoReceiveStream(last_ssrc_);
EXPECT_TRUE(recv_stream);
EXPECT_EQ(recv_stream->base_mininum_playout_delay_ms(), 200);
EXPECT_EQ(channel_->GetBaseMinimumPlayoutDelayMs(last_ssrc_).value_or(0),
200);
}
// Test BaseMinimumPlayoutDelayMs on unsignaled receive streams.
TEST_F(WebRtcVideoChannelTest, BaseMinimumPlayoutDelayMsUnsignaledRecvStream) {
absl::optional<int> delay_ms;
const FakeVideoReceiveStream* recv_stream;
// Set default stream with SSRC 0
EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(0, 200));
EXPECT_EQ(200, channel_->GetBaseMinimumPlayoutDelayMs(0).value_or(0));
// Spawn an unsignaled stream by sending a packet, it should inherit
// default delay 200.
const size_t kDataLength = 12;
uint8_t data[kDataLength];
memset(data, 0, sizeof(data));
rtc::SetBE32(&data[8], kIncomingUnsignalledSsrc);
rtc::CopyOnWriteBuffer packet(data, kDataLength);
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
recv_stream = fake_call_->GetVideoReceiveStream(kIncomingUnsignalledSsrc);
EXPECT_EQ(recv_stream->base_mininum_playout_delay_ms(), 200);
delay_ms = channel_->GetBaseMinimumPlayoutDelayMs(kIncomingUnsignalledSsrc);
EXPECT_EQ(200, delay_ms.value_or(0));
// Check that now if we change delay for SSRC 0 it will change delay for the
// default receiving stream as well.
EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(0, 300));
EXPECT_EQ(300, channel_->GetBaseMinimumPlayoutDelayMs(0).value_or(0));
delay_ms = channel_->GetBaseMinimumPlayoutDelayMs(kIncomingUnsignalledSsrc);
EXPECT_EQ(300, delay_ms.value_or(0));
recv_stream = fake_call_->GetVideoReceiveStream(kIncomingUnsignalledSsrc);
EXPECT_EQ(recv_stream->base_mininum_playout_delay_ms(), 300);
}
void WebRtcVideoChannelTest::TestReceiveUnsignaledSsrcPacket(
uint8_t payload_type,
bool expect_created_receive_stream) {
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
// kRedRtxPayloadType must currently be unused.
EXPECT_FALSE(FindCodecById(engine_.codecs(), kRedRtxPayloadType));
// Add a RED RTX codec.
VideoCodec red_rtx_codec =
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
VideoCodec::CreateRtxCodec(kRedRtxPayloadType, GetEngineCodec("red").id);
recv_parameters_.codecs.push_back(red_rtx_codec);
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
ASSERT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
const size_t kDataLength = 12;
uint8_t data[kDataLength];
memset(data, 0, sizeof(data));
rtc::Set8(data, 1, payload_type);
rtc::SetBE32(&data[8], kIncomingUnsignalledSsrc);
rtc::CopyOnWriteBuffer packet(data, kDataLength);
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
if (expect_created_receive_stream) {
EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size())
<< "Should have created a receive stream for payload type: "
<< payload_type;
} else {
EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size())
<< "Shouldn't have created a receive stream for payload type: "
<< payload_type;
}
}
class WebRtcVideoChannelDiscardUnknownSsrcTest : public WebRtcVideoChannelTest {
public:
WebRtcVideoChannelDiscardUnknownSsrcTest()
: WebRtcVideoChannelTest(
"WebRTC-Video-DiscardPacketsWithUnknownSsrc/Enabled/") {}
};
TEST_F(WebRtcVideoChannelDiscardUnknownSsrcTest, NoUnsignalledStreamCreated) {
TestReceiveUnsignaledSsrcPacket(GetEngineCodec("VP8").id,
false /* expect_created_receive_stream */);
}
TEST_F(WebRtcVideoChannelTest, Vp8PacketCreatesUnsignalledStream) {
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
TestReceiveUnsignaledSsrcPacket(GetEngineCodec("VP8").id,
true /* expect_created_receive_stream */);
}
TEST_F(WebRtcVideoChannelTest, Vp9PacketCreatesUnsignalledStream) {
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
TestReceiveUnsignaledSsrcPacket(GetEngineCodec("VP9").id,
true /* expect_created_receive_stream */);
}
TEST_F(WebRtcVideoChannelTest, RtxPacketDoesntCreateUnsignalledStream) {
AssignDefaultAptRtxTypes();
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
const cricket::VideoCodec vp8 = GetEngineCodec("VP8");
const int rtx_vp8_payload_type = default_apt_rtx_types_[vp8.id];
TestReceiveUnsignaledSsrcPacket(rtx_vp8_payload_type,
false /* expect_created_receive_stream */);
}
TEST_F(WebRtcVideoChannelTest, UlpfecPacketDoesntCreateUnsignalledStream) {
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
TestReceiveUnsignaledSsrcPacket(GetEngineCodec("ulpfec").id,
false /* expect_created_receive_stream */);
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest,
FlexfecPacketDoesntCreateUnsignalledStream) {
TestReceiveUnsignaledSsrcPacket(GetEngineCodec("flexfec-03").id,
false /* expect_created_receive_stream */);
}
TEST_F(WebRtcVideoChannelTest, RedRtxPacketDoesntCreateUnsignalledStream) {
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
TestReceiveUnsignaledSsrcPacket(kRedRtxPayloadType,
false /* expect_created_receive_stream */);
}
// Test that receiving any unsignalled SSRC works even if it changes.
// The first unsignalled SSRC received will create a default receive stream.
// Any different unsignalled SSRC received will replace the default.
TEST_F(WebRtcVideoChannelTest, ReceiveDifferentUnsignaledSsrc) {
// Allow receiving VP8, VP9, H264 (if enabled).
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
#if defined(WEBRTC_USE_H264)
cricket::VideoCodec H264codec(126, "H264");
parameters.codecs.push_back(H264codec);
#endif
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
// No receive streams yet.
ASSERT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
cricket::FakeVideoRenderer renderer;
EXPECT_TRUE(channel_->SetSink(kDefaultRecvSsrc, &renderer));
// Receive VP8 packet on first SSRC.
uint8_t data[kMinRtpPacketLen];
cricket::RtpHeader rtpHeader;
rtpHeader.payload_type = GetEngineCodec("VP8").id;
rtpHeader.seq_num = rtpHeader.timestamp = 0;
rtpHeader.ssrc = kIncomingUnsignalledSsrc + 1;
cricket::SetRtpHeader(data, sizeof(data), rtpHeader);
rtc::CopyOnWriteBuffer packet(data, sizeof(data));
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
// VP8 packet should create default receive stream.
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
FakeVideoReceiveStream* recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_EQ(rtpHeader.ssrc, recv_stream->GetConfig().rtp.remote_ssrc);
// Verify that the receive stream sinks to a renderer.
webrtc::VideoFrame video_frame =
webrtc::VideoFrame::Builder()
.set_video_frame_buffer(CreateBlackFrameBuffer(4, 4))
.set_timestamp_rtp(100)
.set_timestamp_us(0)
.set_rotation(webrtc::kVideoRotation_0)
.build();
recv_stream->InjectFrame(video_frame);
EXPECT_EQ(1, renderer.num_rendered_frames());
// Receive VP9 packet on second SSRC.
rtpHeader.payload_type = GetEngineCodec("VP9").id;
rtpHeader.ssrc = kIncomingUnsignalledSsrc + 2;
cricket::SetRtpHeader(data, sizeof(data), rtpHeader);
rtc::CopyOnWriteBuffer packet2(data, sizeof(data));
channel_->OnPacketReceived(packet2, /* packet_time_us */ -1);
// VP9 packet should replace the default receive SSRC.
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_EQ(rtpHeader.ssrc, recv_stream->GetConfig().rtp.remote_ssrc);
// Verify that the receive stream sinks to a renderer.
webrtc::VideoFrame video_frame2 =
webrtc::VideoFrame::Builder()
.set_video_frame_buffer(CreateBlackFrameBuffer(4, 4))
.set_timestamp_rtp(200)
.set_timestamp_us(0)
.set_rotation(webrtc::kVideoRotation_0)
.build();
recv_stream->InjectFrame(video_frame2);
EXPECT_EQ(2, renderer.num_rendered_frames());
#if defined(WEBRTC_USE_H264)
// Receive H264 packet on third SSRC.
rtpHeader.payload_type = 126;
rtpHeader.ssrc = kIncomingUnsignalledSsrc + 3;
cricket::SetRtpHeader(data, sizeof(data), rtpHeader);
rtc::CopyOnWriteBuffer packet3(data, sizeof(data));
channel_->OnPacketReceived(packet3, /* packet_time_us */ -1);
// H264 packet should replace the default receive SSRC.
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_EQ(rtpHeader.ssrc, recv_stream->GetConfig().rtp.remote_ssrc);
// Verify that the receive stream sinks to a renderer.
webrtc::VideoFrame video_frame3 =
webrtc::VideoFrame::Builder()
.set_video_frame_buffer(CreateBlackFrameBuffer(4, 4))
.set_timestamp_rtp(300)
.set_timestamp_us(0)
.set_rotation(webrtc::kVideoRotation_0)
.build();
recv_stream->InjectFrame(video_frame3);
EXPECT_EQ(3, renderer.num_rendered_frames());
#endif
}
// This test verifies that when a new default stream is created for a new
// unsignaled SSRC, the new stream does not overwrite any old stream that had
// been the default receive stream before being properly signaled.
TEST_F(WebRtcVideoChannelTest,
NewUnsignaledStreamDoesNotDestroyPreviouslyUnsignaledStream) {
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetRecvParameters(parameters));
// No streams signaled and no packets received, so we should not have any
// stream objects created yet.
EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
// Receive packet on an unsignaled SSRC.
uint8_t data[kMinRtpPacketLen];
cricket::RtpHeader rtp_header;
rtp_header.payload_type = GetEngineCodec("VP8").id;
rtp_header.seq_num = rtp_header.timestamp = 0;
rtp_header.ssrc = kSsrcs3[0];
cricket::SetRtpHeader(data, sizeof(data), rtp_header);
rtc::CopyOnWriteBuffer packet(data, sizeof(data));
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
// Default receive stream should be created.
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
FakeVideoReceiveStream* recv_stream0 =
fake_call_->GetVideoReceiveStreams()[0];
EXPECT_EQ(kSsrcs3[0], recv_stream0->GetConfig().rtp.remote_ssrc);
// Signal the SSRC.
EXPECT_TRUE(
channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrcs3[0])));
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
recv_stream0 = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_EQ(kSsrcs3[0], recv_stream0->GetConfig().rtp.remote_ssrc);
// Receive packet on a different unsignaled SSRC.
rtp_header.ssrc = kSsrcs3[1];
cricket::SetRtpHeader(data, sizeof(data), rtp_header);
packet.SetData(data, sizeof(data));
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
// New default receive stream should be created, but old stream should remain.
ASSERT_EQ(2u, fake_call_->GetVideoReceiveStreams().size());
EXPECT_EQ(recv_stream0, fake_call_->GetVideoReceiveStreams()[0]);
FakeVideoReceiveStream* recv_stream1 =
fake_call_->GetVideoReceiveStreams()[1];
EXPECT_EQ(kSsrcs3[1], recv_stream1->GetConfig().rtp.remote_ssrc);
}
TEST_F(WebRtcVideoChannelTest, CanSetMaxBitrateForExistingStream) {
AddSendStream();
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
EXPECT_TRUE(channel_->SetSend(true));
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
int default_encoder_bitrate = GetMaxEncoderBitrate();
EXPECT_GT(default_encoder_bitrate, 1000);
// TODO(skvlad): Resolve the inconsistency between the interpretation
// of the global bitrate limit for audio and video:
// - Audio: max_bandwidth_bps = 0 - fail the operation,
// max_bandwidth_bps = -1 - remove the bandwidth limit
// - Video: max_bandwidth_bps = 0 - remove the bandwidth limit,
// max_bandwidth_bps = -1 - remove the bandwidth limit
SetAndExpectMaxBitrate(1000, 0, 1000);
SetAndExpectMaxBitrate(1000, 800, 800);
SetAndExpectMaxBitrate(600, 800, 600);
SetAndExpectMaxBitrate(0, 800, 800);
SetAndExpectMaxBitrate(0, 0, default_encoder_bitrate);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, CannotSetMaxBitrateForNonexistentStream) {
webrtc::RtpParameters nonexistent_parameters =
channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(0u, nonexistent_parameters.encodings.size());
nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters());
EXPECT_FALSE(
channel_->SetRtpSendParameters(last_ssrc_, nonexistent_parameters).ok());
}
TEST_F(WebRtcVideoChannelTest,
SetLowMaxBitrateOverwritesVideoStreamMinBitrate) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_FALSE(parameters.encodings[0].max_bitrate_bps.has_value());
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Note that this is testing the behavior of the FakeVideoSendStream, which
// also calls to CreateEncoderStreams to get the VideoStreams, so essentially
// we are just testing the behavior of
// EncoderStreamFactory::CreateEncoderStreams.
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_EQ(webrtc::kDefaultMinVideoBitrateBps,
stream->GetVideoStreams()[0].min_bitrate_bps);
// Set a low max bitrate & check that VideoStream.min_bitrate_bps is limited
// by this amount.
parameters = channel_->GetRtpSendParameters(last_ssrc_);
int low_max_bitrate_bps = webrtc::kDefaultMinVideoBitrateBps - 1000;
parameters.encodings[0].max_bitrate_bps = low_max_bitrate_bps;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_EQ(low_max_bitrate_bps, stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(low_max_bitrate_bps, stream->GetVideoStreams()[0].max_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest,
SetHighMinBitrateOverwritesVideoStreamMaxBitrate) {
FakeVideoSendStream* stream = AddSendStream();
// Note that this is testing the behavior of the FakeVideoSendStream, which
// also calls to CreateEncoderStreams to get the VideoStreams, so essentially
// we are just testing the behavior of
// EncoderStreamFactory::CreateEncoderStreams.
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
int high_min_bitrate_bps = stream->GetVideoStreams()[0].max_bitrate_bps + 1;
// Set a high min bitrate and check that max_bitrate_bps is adjusted up.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
parameters.encodings[0].min_bitrate_bps = high_min_bitrate_bps;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_EQ(high_min_bitrate_bps, stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(high_min_bitrate_bps, stream->GetVideoStreams()[0].max_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest,
SetMinBitrateAboveMaxBitrateLimitAdjustsMinBitrateDown) {
send_parameters_.max_bandwidth_bps = 99999;
FakeVideoSendStream* stream = AddSendStream();
ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_EQ(webrtc::kDefaultMinVideoBitrateBps,
stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(send_parameters_.max_bandwidth_bps,
stream->GetVideoStreams()[0].max_bitrate_bps);
// Set min bitrate above global max bitrate and check that min_bitrate_bps is
// adjusted down.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
parameters.encodings[0].min_bitrate_bps = 99999 + 1;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_EQ(send_parameters_.max_bandwidth_bps,
stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(send_parameters_.max_bandwidth_bps,
stream->GetVideoStreams()[0].max_bitrate_bps);
}
Reland "Add initial support for RtpEncodingParameters max_framerate." This reverts commit 948b7e37557af68b3bc9b81b29ae2daffb2784ad. Reason for revert: downstream project fixed. Original change's description: > Revert "Add initial support for RtpEncodingParameters max_framerate." > > This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f. > > Reason for revert: Breaks downstream project. > > Original change's description: > > Add initial support for RtpEncodingParameters max_framerate. > > > > Add support to set the framerate to the maximum of |max_framerate|. > > Different framerates are currently not supported per stream for video. > > > > Bug: webrtc:9597 > > Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe > > Reviewed-on: https://webrtc-review.googlesource.com/92392 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Commit-Queue: Åsa Persson <asapersson@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#24270} > > TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org > > Change-Id: I508fe48e0c53996654f657357913ac307dc256bd > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9597 > Reviewed-on: https://webrtc-review.googlesource.com/94060 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24277} TBR=steveanton@webrtc.org,mbonadei@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org Bug: webrtc:9597 Change-Id: Ieed9d62787f3e9dcb439399bfe7529012292381e Reviewed-on: https://webrtc-review.googlesource.com/100080 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24720}
2018-09-13 10:42:19 +02:00
TEST_F(WebRtcVideoChannelTest, SetMaxFramerateOneStream) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_FALSE(parameters.encodings[0].max_framerate.has_value());
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Note that this is testing the behavior of the FakeVideoSendStream, which
// also calls to CreateEncoderStreams to get the VideoStreams, so essentially
// we are just testing the behavior of
// EncoderStreamFactory::CreateEncoderStreams.
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_EQ(kDefaultVideoMaxFramerate,
stream->GetVideoStreams()[0].max_framerate);
// Set max framerate and check that VideoStream.max_framerate is set.
const int kNewMaxFramerate = kDefaultVideoMaxFramerate - 1;
parameters = channel_->GetRtpSendParameters(last_ssrc_);
parameters.encodings[0].max_framerate = kNewMaxFramerate;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_EQ(kNewMaxFramerate, stream->GetVideoStreams()[0].max_framerate);
}
TEST_F(WebRtcVideoChannelTest, SetNumTemporalLayersForSingleStream) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_FALSE(parameters.encodings[0].num_temporal_layers.has_value());
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Note that this is testing the behavior of the FakeVideoSendStream, which
// also calls to CreateEncoderStreams to get the VideoStreams, so essentially
// we are just testing the behavior of
// EncoderStreamFactory::CreateEncoderStreams.
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_FALSE(stream->GetVideoStreams()[0].num_temporal_layers.has_value());
// Set temporal layers and check that VideoStream.num_temporal_layers is set.
parameters = channel_->GetRtpSendParameters(last_ssrc_);
parameters.encodings[0].num_temporal_layers = 2;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_EQ(2UL, stream->GetVideoStreams()[0].num_temporal_layers);
}
TEST_F(WebRtcVideoChannelTest,
CannotSetRtpSendParametersWithIncorrectNumberOfEncodings) {
AddSendStream();
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
// Two or more encodings should result in failure.
parameters.encodings.push_back(webrtc::RtpEncodingParameters());
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Zero encodings should also fail.
parameters.encodings.clear();
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
}
TEST_F(WebRtcVideoChannelTest,
CannotSetSimulcastRtpSendParametersWithIncorrectNumberOfEncodings) {
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
StreamParams sp = CreateSimStreamParams("cname", ssrcs);
AddSendStream(sp);
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
// Additional encodings should result in failure.
parameters.encodings.push_back(webrtc::RtpEncodingParameters());
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Zero encodings should also fail.
parameters.encodings.clear();
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
}
// Changing the SSRC through RtpParameters is not allowed.
TEST_F(WebRtcVideoChannelTest, CannotSetSsrcInRtpSendParameters) {
AddSendStream();
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
parameters.encodings[0].ssrc = 0xdeadbeef;
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
}
// Tests that when RTCRtpEncodingParameters.bitrate_priority gets set to
// a value <= 0, setting the parameters returns false.
TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersInvalidBitratePriority) {
AddSendStream();
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_EQ(webrtc::kDefaultBitratePriority,
parameters.encodings[0].bitrate_priority);
parameters.encodings[0].bitrate_priority = 0;
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
parameters.encodings[0].bitrate_priority = -2;
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
}
// Tests when the the RTCRtpEncodingParameters.bitrate_priority gets set
// properly on the VideoChannel and propogates down to the video encoder.
TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPriorityOneStream) {
AddSendStream();
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_EQ(webrtc::kDefaultBitratePriority,
parameters.encodings[0].bitrate_priority);
// Change the value and set it on the VideoChannel.
double new_bitrate_priority = 2.0;
parameters.encodings[0].bitrate_priority = new_bitrate_priority;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the encoding parameters bitrate_priority is set for the
// VideoChannel.
parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_EQ(new_bitrate_priority, parameters.encodings[0].bitrate_priority);
// Verify that the new value propagated down to the encoder.
std::vector<FakeVideoSendStream*> video_send_streams =
fake_call_->GetVideoSendStreams();
EXPECT_EQ(1UL, video_send_streams.size());
FakeVideoSendStream* video_send_stream = video_send_streams.front();
// Check that the WebRtcVideoSendStream updated the VideoEncoderConfig
// appropriately.
EXPECT_EQ(new_bitrate_priority,
video_send_stream->GetEncoderConfig().bitrate_priority);
// Check that the vector of VideoStreams also was propagated correctly. Note
// that this is testing the behavior of the FakeVideoSendStream, which mimics
// the calls to CreateEncoderStreams to get the VideoStreams.
EXPECT_EQ(absl::optional<double>(new_bitrate_priority),
video_send_stream->GetVideoStreams()[0].bitrate_priority);
}
// Tests that the RTCRtpEncodingParameters.bitrate_priority is set for the
// VideoChannel and the value propogates to the video encoder with all simulcast
// streams.
TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPrioritySimulcastStreams) {
// Create the stream params with multiple ssrcs for simulcast.
const size_t kNumSimulcastStreams = 3;
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
StreamParams stream_params = CreateSimStreamParams("cname", ssrcs);
AddSendStream(stream_params);
uint32_t primary_ssrc = stream_params.first_ssrc();
// Using the FrameForwarder, we manually send a full size
// frame. This creates multiple VideoStreams for all simulcast layers when
// reconfiguring, and allows us to test this behavior.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame(
1920, 1080, webrtc::VideoRotation::kVideoRotation_0,
rtc::kNumMicrosecsPerSec / 30));
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters =
channel_->GetRtpSendParameters(primary_ssrc);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_EQ(webrtc::kDefaultBitratePriority,
parameters.encodings[0].bitrate_priority);
// Change the value and set it on the VideoChannel.
double new_bitrate_priority = 2.0;
parameters.encodings[0].bitrate_priority = new_bitrate_priority;
EXPECT_TRUE(channel_->SetRtpSendParameters(primary_ssrc, parameters).ok());
// Verify that the encoding parameters priority is set on the VideoChannel.
parameters = channel_->GetRtpSendParameters(primary_ssrc);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_EQ(new_bitrate_priority, parameters.encodings[0].bitrate_priority);
// Verify that the new value propagated down to the encoder.
std::vector<FakeVideoSendStream*> video_send_streams =
fake_call_->GetVideoSendStreams();
EXPECT_EQ(1UL, video_send_streams.size());
FakeVideoSendStream* video_send_stream = video_send_streams.front();
// Check that the WebRtcVideoSendStream updated the VideoEncoderConfig
// appropriately.
EXPECT_EQ(kNumSimulcastStreams,
video_send_stream->GetEncoderConfig().number_of_streams);
EXPECT_EQ(new_bitrate_priority,
video_send_stream->GetEncoderConfig().bitrate_priority);
// Check that the vector of VideoStreams also propagated correctly. The
// FakeVideoSendStream calls CreateEncoderStreams, and we are testing that
// these are created appropriately for the simulcast case.
EXPECT_EQ(kNumSimulcastStreams, video_send_stream->GetVideoStreams().size());
EXPECT_EQ(absl::optional<double>(new_bitrate_priority),
video_send_stream->GetVideoStreams()[0].bitrate_priority);
// Since we are only setting bitrate priority per-sender, the other
// VideoStreams should have a bitrate priority of 0.
EXPECT_EQ(absl::nullopt,
video_send_stream->GetVideoStreams()[1].bitrate_priority);
EXPECT_EQ(absl::nullopt,
video_send_stream->GetVideoStreams()[2].bitrate_priority);
EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, nullptr, nullptr));
}
// RTCRtpEncodingParameters.network_priority must be one of a few values
// derived from the default priority, corresponding to very-low, low, medium,
// or high.
TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersInvalidNetworkPriority) {
AddSendStream();
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_EQ(webrtc::kDefaultBitratePriority,
parameters.encodings[0].network_priority);
double good_values[] = {0.5, 1.0, 2.0, 4.0};
double bad_values[] = {-1.0, 0.0, 0.49, 0.51, 1.1, 3.99, 4.1, 5.0};
for (auto it : good_values) {
parameters.encodings[0].network_priority = it;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
}
for (auto it : bad_values) {
parameters.encodings[0].network_priority = it;
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
}
}
TEST_F(WebRtcVideoChannelTest,
GetAndSetRtpSendParametersScaleResolutionDownByVP8) {
VideoSendParameters parameters;
parameters.codecs.push_back(VideoCodec(kVp8CodecName));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
webrtc::test::FrameForwarder frame_forwarder;
FakeFrameSource frame_source(1280, 720, rtc::kNumMicrosecsPerSec / 30);
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
// Try layers in natural order (smallest to largest).
{
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
rtp_parameters.encodings[0].scale_resolution_down_by = 4.0;
rtp_parameters.encodings[1].scale_resolution_down_by = 2.0;
rtp_parameters.encodings[2].scale_resolution_down_by = 1.0;
auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(3u, video_streams.size());
EXPECT_EQ(320u, video_streams[0].width);
EXPECT_EQ(180u, video_streams[0].height);
EXPECT_EQ(640u, video_streams[1].width);
EXPECT_EQ(360u, video_streams[1].height);
EXPECT_EQ(1280u, video_streams[2].width);
EXPECT_EQ(720u, video_streams[2].height);
}
// Try layers in reverse natural order (largest to smallest).
{
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
rtp_parameters.encodings[0].scale_resolution_down_by = 1.0;
rtp_parameters.encodings[1].scale_resolution_down_by = 2.0;
rtp_parameters.encodings[2].scale_resolution_down_by = 4.0;
auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(3u, video_streams.size());
EXPECT_EQ(1280u, video_streams[0].width);
EXPECT_EQ(720u, video_streams[0].height);
EXPECT_EQ(640u, video_streams[1].width);
EXPECT_EQ(360u, video_streams[1].height);
EXPECT_EQ(320u, video_streams[2].width);
EXPECT_EQ(180u, video_streams[2].height);
}
// Try layers in mixed order.
{
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
rtp_parameters.encodings[0].scale_resolution_down_by = 10.0;
rtp_parameters.encodings[1].scale_resolution_down_by = 2.0;
rtp_parameters.encodings[2].scale_resolution_down_by = 4.0;
auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(3u, video_streams.size());
EXPECT_EQ(128u, video_streams[0].width);
EXPECT_EQ(72u, video_streams[0].height);
EXPECT_EQ(640u, video_streams[1].width);
EXPECT_EQ(360u, video_streams[1].height);
EXPECT_EQ(320u, video_streams[2].width);
EXPECT_EQ(180u, video_streams[2].height);
}
// Try with a missing scale setting, defaults to 1.0 if any other is set.
{
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
rtp_parameters.encodings[0].scale_resolution_down_by = 1.0;
rtp_parameters.encodings[1].scale_resolution_down_by.reset();
rtp_parameters.encodings[2].scale_resolution_down_by = 4.0;
auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(3u, video_streams.size());
EXPECT_EQ(1280u, video_streams[0].width);
EXPECT_EQ(720u, video_streams[0].height);
EXPECT_EQ(1280u, video_streams[1].width);
EXPECT_EQ(720u, video_streams[1].height);
EXPECT_EQ(320u, video_streams[2].width);
EXPECT_EQ(180u, video_streams[2].height);
}
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
GetAndSetRtpSendParametersScaleResolutionDownByVP8WithOddResolution) {
// Ensure that the top layer has width and height divisible by 2^3,
// so that the bottom layer has width and height divisible by 2.
// TODO(bugs.webrtc.org/8785): Remove this field trial when we fully trust
// the number of simulcast layers set by the app.
webrtc::test::ScopedFieldTrials field_trial(
"WebRTC-NormalizeSimulcastResolution/Enabled-3/");
// Set up WebRtcVideoChannel for 3-layer VP8 simulcast.
VideoSendParameters parameters;
parameters.codecs.push_back(VideoCodec(kVp8CodecName));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, /*options=*/nullptr,
&frame_forwarder));
channel_->SetSend(true);
// Set |scale_resolution_down_by|'s.
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(rtp_parameters.encodings.size(), 3u);
rtp_parameters.encodings[0].scale_resolution_down_by = 1.0;
rtp_parameters.encodings[1].scale_resolution_down_by = 2.0;
rtp_parameters.encodings[2].scale_resolution_down_by = 4.0;
const auto result =
channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
// Use a capture resolution whose width and height are not divisible by 2^3.
// (See field trial set at the top of the test.)
FakeFrameSource frame_source(2007, 1207, rtc::kNumMicrosecsPerSec / 30);
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
// Ensure the scaling is correct.
const auto video_streams = stream->GetVideoStreams();
ASSERT_EQ(video_streams.size(), 3u);
// Ensure that we round the capture resolution down for the top layer...
EXPECT_EQ(video_streams[0].width, 2000u);
EXPECT_EQ(video_streams[0].height, 1200u);
EXPECT_EQ(video_streams[1].width, 1000u);
EXPECT_EQ(video_streams[1].height, 600u);
// ...and that the bottom layer has a width/height divisible by 2.
EXPECT_EQ(video_streams[2].width, 500u);
EXPECT_EQ(video_streams[2].height, 300u);
// Tear down.
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
GetAndSetRtpSendParametersScaleResolutionDownByH264) {
encoder_factory_->AddSupportedVideoCodecType(kH264CodecName);
VideoSendParameters parameters;
parameters.codecs.push_back(VideoCodec(kH264CodecName));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
webrtc::test::FrameForwarder frame_forwarder;
FakeFrameSource frame_source(1280, 720, rtc::kNumMicrosecsPerSec / 30);
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
// Try layers in natural order (smallest to largest).
{
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
rtp_parameters.encodings[0].scale_resolution_down_by = 4.0;
rtp_parameters.encodings[1].scale_resolution_down_by = 2.0;
rtp_parameters.encodings[2].scale_resolution_down_by = 1.0;
auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(3u, video_streams.size());
EXPECT_EQ(320u, video_streams[0].width);
EXPECT_EQ(180u, video_streams[0].height);
EXPECT_EQ(640u, video_streams[1].width);
EXPECT_EQ(360u, video_streams[1].height);
EXPECT_EQ(1280u, video_streams[2].width);
EXPECT_EQ(720u, video_streams[2].height);
}
// Try layers in reverse natural order (largest to smallest).
{
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
rtp_parameters.encodings[0].scale_resolution_down_by = 1.0;
rtp_parameters.encodings[1].scale_resolution_down_by = 2.0;
rtp_parameters.encodings[2].scale_resolution_down_by = 4.0;
auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(3u, video_streams.size());
EXPECT_EQ(1280u, video_streams[0].width);
EXPECT_EQ(720u, video_streams[0].height);
EXPECT_EQ(640u, video_streams[1].width);
EXPECT_EQ(360u, video_streams[1].height);
EXPECT_EQ(320u, video_streams[2].width);
EXPECT_EQ(180u, video_streams[2].height);
}
// Try layers in mixed order.
{
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
rtp_parameters.encodings[0].scale_resolution_down_by = 10.0;
rtp_parameters.encodings[1].scale_resolution_down_by = 2.0;
rtp_parameters.encodings[2].scale_resolution_down_by = 4.0;
auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(3u, video_streams.size());
EXPECT_EQ(128u, video_streams[0].width);
EXPECT_EQ(72u, video_streams[0].height);
EXPECT_EQ(640u, video_streams[1].width);
EXPECT_EQ(360u, video_streams[1].height);
EXPECT_EQ(320u, video_streams[2].width);
EXPECT_EQ(180u, video_streams[2].height);
}
// Try with a missing scale setting, defaults to 1.0 if any other is set.
{
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
rtp_parameters.encodings[0].scale_resolution_down_by = 1.0;
rtp_parameters.encodings[1].scale_resolution_down_by.reset();
rtp_parameters.encodings[2].scale_resolution_down_by = 4.0;
auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(3u, video_streams.size());
EXPECT_EQ(1280u, video_streams[0].width);
EXPECT_EQ(720u, video_streams[0].height);
EXPECT_EQ(1280u, video_streams[1].width);
EXPECT_EQ(720u, video_streams[1].height);
EXPECT_EQ(320u, video_streams[2].width);
EXPECT_EQ(180u, video_streams[2].height);
}
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
GetAndSetRtpSendParametersScaleResolutionDownByH264WithOddResolution) {
// Ensure that the top layer has width and height divisible by 2^3,
// so that the bottom layer has width and height divisible by 2.
// TODO(bugs.webrtc.org/8785): Remove this field trial when we fully trust
// the number of simulcast layers set by the app.
webrtc::test::ScopedFieldTrials field_trial(
"WebRTC-NormalizeSimulcastResolution/Enabled-3/");
// Set up WebRtcVideoChannel for 3-layer H264 simulcast.
encoder_factory_->AddSupportedVideoCodecType(kH264CodecName);
VideoSendParameters parameters;
parameters.codecs.push_back(VideoCodec(kH264CodecName));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, /*options=*/nullptr,
&frame_forwarder));
channel_->SetSend(true);
// Set |scale_resolution_down_by|'s.
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(rtp_parameters.encodings.size(), 3u);
rtp_parameters.encodings[0].scale_resolution_down_by = 1.0;
rtp_parameters.encodings[1].scale_resolution_down_by = 2.0;
rtp_parameters.encodings[2].scale_resolution_down_by = 4.0;
const auto result =
channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
// Use a capture resolution whose width and height are not divisible by 2^3.
// (See field trial set at the top of the test.)
FakeFrameSource frame_source(2007, 1207, rtc::kNumMicrosecsPerSec / 30);
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
// Ensure the scaling is correct.
const auto video_streams = stream->GetVideoStreams();
ASSERT_EQ(video_streams.size(), 3u);
// Ensure that we round the capture resolution down for the top layer...
EXPECT_EQ(video_streams[0].width, 2000u);
EXPECT_EQ(video_streams[0].height, 1200u);
EXPECT_EQ(video_streams[1].width, 1000u);
EXPECT_EQ(video_streams[1].height, 600u);
// ...and that the bottom layer has a width/height divisible by 2.
EXPECT_EQ(video_streams[2].width, 500u);
EXPECT_EQ(video_streams[2].height, 300u);
// Tear down.
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
Reland "Add initial support for RtpEncodingParameters max_framerate." This reverts commit 948b7e37557af68b3bc9b81b29ae2daffb2784ad. Reason for revert: downstream project fixed. Original change's description: > Revert "Add initial support for RtpEncodingParameters max_framerate." > > This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f. > > Reason for revert: Breaks downstream project. > > Original change's description: > > Add initial support for RtpEncodingParameters max_framerate. > > > > Add support to set the framerate to the maximum of |max_framerate|. > > Different framerates are currently not supported per stream for video. > > > > Bug: webrtc:9597 > > Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe > > Reviewed-on: https://webrtc-review.googlesource.com/92392 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Commit-Queue: Åsa Persson <asapersson@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#24270} > > TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org > > Change-Id: I508fe48e0c53996654f657357913ac307dc256bd > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9597 > Reviewed-on: https://webrtc-review.googlesource.com/94060 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24277} TBR=steveanton@webrtc.org,mbonadei@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org Bug: webrtc:9597 Change-Id: Ieed9d62787f3e9dcb439399bfe7529012292381e Reviewed-on: https://webrtc-review.googlesource.com/100080 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24720}
2018-09-13 10:42:19 +02:00
TEST_F(WebRtcVideoChannelTest, GetAndSetRtpSendParametersMaxFramerate) {
const size_t kNumSimulcastStreams = 3;
SetUpSimulcast(true, false);
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
for (const auto& encoding : parameters.encodings) {
EXPECT_FALSE(encoding.max_framerate);
}
// Change the value and set it on the VideoChannel.
parameters.encodings[0].max_framerate = 10;
parameters.encodings[1].max_framerate = 20;
parameters.encodings[2].max_framerate = 25;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the bitrates are set on the VideoChannel.
parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_EQ(10, parameters.encodings[0].max_framerate);
EXPECT_EQ(20, parameters.encodings[1].max_framerate);
EXPECT_EQ(25, parameters.encodings[2].max_framerate);
}
TEST_F(WebRtcVideoChannelTest,
SetRtpSendParametersNumTemporalLayersFailsForInvalidRange) {
const size_t kNumSimulcastStreams = 3;
SetUpSimulcast(true, false);
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
// Num temporal layers should be in the range [1, kMaxTemporalStreams].
parameters.encodings[0].num_temporal_layers = 0;
EXPECT_EQ(webrtc::RTCErrorType::INVALID_RANGE,
channel_->SetRtpSendParameters(last_ssrc_, parameters).type());
parameters.encodings[0].num_temporal_layers = webrtc::kMaxTemporalStreams + 1;
EXPECT_EQ(webrtc::RTCErrorType::INVALID_RANGE,
channel_->SetRtpSendParameters(last_ssrc_, parameters).type());
}
TEST_F(WebRtcVideoChannelTest,
SetRtpSendParametersNumTemporalLayersFailsForInvalidModification) {
const size_t kNumSimulcastStreams = 3;
SetUpSimulcast(true, false);
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
// No/all layers should be set.
parameters.encodings[0].num_temporal_layers = 1;
EXPECT_EQ(webrtc::RTCErrorType::INVALID_MODIFICATION,
channel_->SetRtpSendParameters(last_ssrc_, parameters).type());
// Different values not supported.
parameters.encodings[0].num_temporal_layers = 1;
parameters.encodings[1].num_temporal_layers = 2;
parameters.encodings[2].num_temporal_layers = 2;
EXPECT_EQ(webrtc::RTCErrorType::INVALID_MODIFICATION,
channel_->SetRtpSendParameters(last_ssrc_, parameters).type());
}
TEST_F(WebRtcVideoChannelTest, GetAndSetRtpSendParametersNumTemporalLayers) {
const size_t kNumSimulcastStreams = 3;
SetUpSimulcast(true, false);
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
for (const auto& encoding : parameters.encodings)
EXPECT_FALSE(encoding.num_temporal_layers);
// Change the value and set it on the VideoChannel.
parameters.encodings[0].num_temporal_layers = 3;
parameters.encodings[1].num_temporal_layers = 3;
parameters.encodings[2].num_temporal_layers = 3;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the number of temporal layers are set on the VideoChannel.
parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_EQ(3, parameters.encodings[0].num_temporal_layers);
EXPECT_EQ(3, parameters.encodings[1].num_temporal_layers);
EXPECT_EQ(3, parameters.encodings[2].num_temporal_layers);
}
TEST_F(WebRtcVideoChannelTest, NumTemporalLayersPropagatedToEncoder) {
const size_t kNumSimulcastStreams = 3;
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Get and set the rtp encoding parameters.
// Change the value and set it on the VideoChannel.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
parameters.encodings[0].num_temporal_layers = 2;
parameters.encodings[1].num_temporal_layers = 2;
parameters.encodings[2].num_temporal_layers = 2;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the new value is propagated down to the encoder.
// Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
EXPECT_EQ(2, stream->num_encoder_reconfigurations());
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams);
EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size());
EXPECT_EQ(2UL, encoder_config.simulcast_layers[0].num_temporal_layers);
EXPECT_EQ(2UL, encoder_config.simulcast_layers[1].num_temporal_layers);
EXPECT_EQ(2UL, encoder_config.simulcast_layers[2].num_temporal_layers);
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately for the simulcast case.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
EXPECT_EQ(2UL, stream->GetVideoStreams()[0].num_temporal_layers);
EXPECT_EQ(2UL, stream->GetVideoStreams()[1].num_temporal_layers);
EXPECT_EQ(2UL, stream->GetVideoStreams()[2].num_temporal_layers);
// No parameter changed, encoder should not be reconfigured.
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
EXPECT_EQ(2, stream->num_encoder_reconfigurations());
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
DefaultValuePropagatedToEncoderForUnsetNumTemporalLayers) {
const size_t kDefaultNumTemporalLayers = 3;
const size_t kNumSimulcastStreams = 3;
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Change rtp encoding parameters, num_temporal_layers not changed.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
parameters.encodings[0].min_bitrate_bps = 33000;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that no value is propagated down to the encoder.
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams);
EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size());
EXPECT_FALSE(encoder_config.simulcast_layers[0].num_temporal_layers);
EXPECT_FALSE(encoder_config.simulcast_layers[1].num_temporal_layers);
EXPECT_FALSE(encoder_config.simulcast_layers[2].num_temporal_layers);
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately for the simulcast case.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
EXPECT_EQ(kDefaultNumTemporalLayers,
stream->GetVideoStreams()[0].num_temporal_layers);
EXPECT_EQ(kDefaultNumTemporalLayers,
stream->GetVideoStreams()[1].num_temporal_layers);
EXPECT_EQ(kDefaultNumTemporalLayers,
stream->GetVideoStreams()[2].num_temporal_layers);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, MaxSimulcastFrameratePropagatedToEncoder) {
const size_t kNumSimulcastStreams = 3;
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Get and set the rtp encoding parameters.
// Change the value and set it on the VideoChannel.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
parameters.encodings[0].max_framerate = 15;
parameters.encodings[1].max_framerate = 25;
parameters.encodings[2].max_framerate = 20;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the new value propagated down to the encoder.
// Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
EXPECT_EQ(2, stream->num_encoder_reconfigurations());
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams);
EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size());
EXPECT_EQ(15, encoder_config.simulcast_layers[0].max_framerate);
EXPECT_EQ(25, encoder_config.simulcast_layers[1].max_framerate);
EXPECT_EQ(20, encoder_config.simulcast_layers[2].max_framerate);
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately for the simulcast case.
// Currently the maximum |max_framerate| is used.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
EXPECT_EQ(25, stream->GetVideoStreams()[0].max_framerate);
EXPECT_EQ(25, stream->GetVideoStreams()[1].max_framerate);
EXPECT_EQ(25, stream->GetVideoStreams()[2].max_framerate);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
Reland "Add initial support for RtpEncodingParameters max_framerate." This reverts commit 948b7e37557af68b3bc9b81b29ae2daffb2784ad. Reason for revert: downstream project fixed. Original change's description: > Revert "Add initial support for RtpEncodingParameters max_framerate." > > This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f. > > Reason for revert: Breaks downstream project. > > Original change's description: > > Add initial support for RtpEncodingParameters max_framerate. > > > > Add support to set the framerate to the maximum of |max_framerate|. > > Different framerates are currently not supported per stream for video. > > > > Bug: webrtc:9597 > > Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe > > Reviewed-on: https://webrtc-review.googlesource.com/92392 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Commit-Queue: Åsa Persson <asapersson@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#24270} > > TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org > > Change-Id: I508fe48e0c53996654f657357913ac307dc256bd > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9597 > Reviewed-on: https://webrtc-review.googlesource.com/94060 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24277} TBR=steveanton@webrtc.org,mbonadei@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org Bug: webrtc:9597 Change-Id: Ieed9d62787f3e9dcb439399bfe7529012292381e Reviewed-on: https://webrtc-review.googlesource.com/100080 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24720}
2018-09-13 10:42:19 +02:00
TEST_F(WebRtcVideoChannelTest,
DefaultValuePropagatedToEncoderForUnsetFramerate) {
const size_t kNumSimulcastStreams = 3;
const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p();
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
Reland "Add initial support for RtpEncodingParameters max_framerate." This reverts commit 948b7e37557af68b3bc9b81b29ae2daffb2784ad. Reason for revert: downstream project fixed. Original change's description: > Revert "Add initial support for RtpEncodingParameters max_framerate." > > This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f. > > Reason for revert: Breaks downstream project. > > Original change's description: > > Add initial support for RtpEncodingParameters max_framerate. > > > > Add support to set the framerate to the maximum of |max_framerate|. > > Different framerates are currently not supported per stream for video. > > > > Bug: webrtc:9597 > > Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe > > Reviewed-on: https://webrtc-review.googlesource.com/92392 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Commit-Queue: Åsa Persson <asapersson@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#24270} > > TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org > > Change-Id: I508fe48e0c53996654f657357913ac307dc256bd > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9597 > Reviewed-on: https://webrtc-review.googlesource.com/94060 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24277} TBR=steveanton@webrtc.org,mbonadei@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org Bug: webrtc:9597 Change-Id: Ieed9d62787f3e9dcb439399bfe7529012292381e Reviewed-on: https://webrtc-review.googlesource.com/100080 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24720}
2018-09-13 10:42:19 +02:00
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
Reland "Add initial support for RtpEncodingParameters max_framerate." This reverts commit 948b7e37557af68b3bc9b81b29ae2daffb2784ad. Reason for revert: downstream project fixed. Original change's description: > Revert "Add initial support for RtpEncodingParameters max_framerate." > > This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f. > > Reason for revert: Breaks downstream project. > > Original change's description: > > Add initial support for RtpEncodingParameters max_framerate. > > > > Add support to set the framerate to the maximum of |max_framerate|. > > Different framerates are currently not supported per stream for video. > > > > Bug: webrtc:9597 > > Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe > > Reviewed-on: https://webrtc-review.googlesource.com/92392 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Commit-Queue: Åsa Persson <asapersson@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#24270} > > TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org > > Change-Id: I508fe48e0c53996654f657357913ac307dc256bd > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9597 > Reviewed-on: https://webrtc-review.googlesource.com/94060 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24277} TBR=steveanton@webrtc.org,mbonadei@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org Bug: webrtc:9597 Change-Id: Ieed9d62787f3e9dcb439399bfe7529012292381e Reviewed-on: https://webrtc-review.googlesource.com/100080 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24720}
2018-09-13 10:42:19 +02:00
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
Reland "Add initial support for RtpEncodingParameters max_framerate." This reverts commit 948b7e37557af68b3bc9b81b29ae2daffb2784ad. Reason for revert: downstream project fixed. Original change's description: > Revert "Add initial support for RtpEncodingParameters max_framerate." > > This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f. > > Reason for revert: Breaks downstream project. > > Original change's description: > > Add initial support for RtpEncodingParameters max_framerate. > > > > Add support to set the framerate to the maximum of |max_framerate|. > > Different framerates are currently not supported per stream for video. > > > > Bug: webrtc:9597 > > Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe > > Reviewed-on: https://webrtc-review.googlesource.com/92392 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Commit-Queue: Åsa Persson <asapersson@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#24270} > > TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org > > Change-Id: I508fe48e0c53996654f657357913ac307dc256bd > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9597 > Reviewed-on: https://webrtc-review.googlesource.com/94060 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24277} TBR=steveanton@webrtc.org,mbonadei@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org Bug: webrtc:9597 Change-Id: Ieed9d62787f3e9dcb439399bfe7529012292381e Reviewed-on: https://webrtc-review.googlesource.com/100080 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24720}
2018-09-13 10:42:19 +02:00
// Get and set the rtp encoding parameters.
// Change the value and set it on the VideoChannel.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
parameters.encodings[0].max_framerate = 15;
parameters.encodings[2].max_framerate = 20;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the new value propagated down to the encoder.
// Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams);
EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size());
EXPECT_EQ(15, encoder_config.simulcast_layers[0].max_framerate);
EXPECT_EQ(-1, encoder_config.simulcast_layers[1].max_framerate);
EXPECT_EQ(20, encoder_config.simulcast_layers[2].max_framerate);
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately for the simulcast case.
// The maximum |max_framerate| is used, kDefaultVideoMaxFramerate: 60.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
EXPECT_EQ(kDefaultVideoMaxFramerate,
stream->GetVideoStreams()[0].max_framerate);
Reland "Add initial support for RtpEncodingParameters max_framerate." This reverts commit 948b7e37557af68b3bc9b81b29ae2daffb2784ad. Reason for revert: downstream project fixed. Original change's description: > Revert "Add initial support for RtpEncodingParameters max_framerate." > > This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f. > > Reason for revert: Breaks downstream project. > > Original change's description: > > Add initial support for RtpEncodingParameters max_framerate. > > > > Add support to set the framerate to the maximum of |max_framerate|. > > Different framerates are currently not supported per stream for video. > > > > Bug: webrtc:9597 > > Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe > > Reviewed-on: https://webrtc-review.googlesource.com/92392 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Commit-Queue: Åsa Persson <asapersson@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#24270} > > TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org > > Change-Id: I508fe48e0c53996654f657357913ac307dc256bd > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9597 > Reviewed-on: https://webrtc-review.googlesource.com/94060 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24277} TBR=steveanton@webrtc.org,mbonadei@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org Bug: webrtc:9597 Change-Id: Ieed9d62787f3e9dcb439399bfe7529012292381e Reviewed-on: https://webrtc-review.googlesource.com/100080 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24720}
2018-09-13 10:42:19 +02:00
EXPECT_EQ(kDefaultVideoMaxFramerate,
stream->GetVideoStreams()[1].max_framerate);
EXPECT_EQ(kDefaultVideoMaxFramerate,
stream->GetVideoStreams()[2].max_framerate);
Reland "Add initial support for RtpEncodingParameters max_framerate." This reverts commit 948b7e37557af68b3bc9b81b29ae2daffb2784ad. Reason for revert: downstream project fixed. Original change's description: > Revert "Add initial support for RtpEncodingParameters max_framerate." > > This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f. > > Reason for revert: Breaks downstream project. > > Original change's description: > > Add initial support for RtpEncodingParameters max_framerate. > > > > Add support to set the framerate to the maximum of |max_framerate|. > > Different framerates are currently not supported per stream for video. > > > > Bug: webrtc:9597 > > Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe > > Reviewed-on: https://webrtc-review.googlesource.com/92392 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Commit-Queue: Åsa Persson <asapersson@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#24270} > > TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org > > Change-Id: I508fe48e0c53996654f657357913ac307dc256bd > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9597 > Reviewed-on: https://webrtc-review.googlesource.com/94060 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24277} TBR=steveanton@webrtc.org,mbonadei@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org Bug: webrtc:9597 Change-Id: Ieed9d62787f3e9dcb439399bfe7529012292381e Reviewed-on: https://webrtc-review.googlesource.com/100080 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24720}
2018-09-13 10:42:19 +02:00
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, GetAndSetRtpSendParametersMinAndMaxBitrate) {
const size_t kNumSimulcastStreams = 3;
SetUpSimulcast(true, false);
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
for (const auto& encoding : parameters.encodings) {
EXPECT_FALSE(encoding.min_bitrate_bps);
EXPECT_FALSE(encoding.max_bitrate_bps);
}
// Change the value and set it on the VideoChannel.
parameters.encodings[0].min_bitrate_bps = 100000;
parameters.encodings[0].max_bitrate_bps = 200000;
parameters.encodings[1].min_bitrate_bps = 300000;
parameters.encodings[1].max_bitrate_bps = 400000;
parameters.encodings[2].min_bitrate_bps = 500000;
parameters.encodings[2].max_bitrate_bps = 600000;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the bitrates are set on the VideoChannel.
parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_EQ(100000, parameters.encodings[0].min_bitrate_bps);
EXPECT_EQ(200000, parameters.encodings[0].max_bitrate_bps);
EXPECT_EQ(300000, parameters.encodings[1].min_bitrate_bps);
EXPECT_EQ(400000, parameters.encodings[1].max_bitrate_bps);
EXPECT_EQ(500000, parameters.encodings[2].min_bitrate_bps);
EXPECT_EQ(600000, parameters.encodings[2].max_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersFailsWithIncorrectBitrate) {
const size_t kNumSimulcastStreams = 3;
SetUpSimulcast(true, false);
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
// Max bitrate lower than min bitrate should fail.
parameters.encodings[2].min_bitrate_bps = 100000;
parameters.encodings[2].max_bitrate_bps = 100000 - 1;
EXPECT_EQ(webrtc::RTCErrorType::INVALID_RANGE,
channel_->SetRtpSendParameters(last_ssrc_, parameters).type());
}
// Test that min and max bitrate values set via RtpParameters are correctly
// propagated to the underlying encoder, and that the target is set to 3/4 of
// the maximum (3/4 was chosen because it's similar to the simulcast defaults
// that are used if no min/max are specified).
TEST_F(WebRtcVideoChannelTest, MinAndMaxSimulcastBitratePropagatedToEncoder) {
const size_t kNumSimulcastStreams = 3;
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Get and set the rtp encoding parameters.
// Change the value and set it on the VideoChannel.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
parameters.encodings[0].min_bitrate_bps = 100000;
parameters.encodings[0].max_bitrate_bps = 200000;
parameters.encodings[1].min_bitrate_bps = 300000;
parameters.encodings[1].max_bitrate_bps = 400000;
parameters.encodings[2].min_bitrate_bps = 500000;
parameters.encodings[2].max_bitrate_bps = 600000;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the new value propagated down to the encoder.
// Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
EXPECT_EQ(2, stream->num_encoder_reconfigurations());
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams);
EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size());
EXPECT_EQ(100000, encoder_config.simulcast_layers[0].min_bitrate_bps);
EXPECT_EQ(200000, encoder_config.simulcast_layers[0].max_bitrate_bps);
EXPECT_EQ(300000, encoder_config.simulcast_layers[1].min_bitrate_bps);
EXPECT_EQ(400000, encoder_config.simulcast_layers[1].max_bitrate_bps);
EXPECT_EQ(500000, encoder_config.simulcast_layers[2].min_bitrate_bps);
EXPECT_EQ(600000, encoder_config.simulcast_layers[2].max_bitrate_bps);
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately for the simulcast case.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
// Target bitrate: 200000 * 3 / 4 = 150000.
EXPECT_EQ(100000, stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(150000, stream->GetVideoStreams()[0].target_bitrate_bps);
EXPECT_EQ(200000, stream->GetVideoStreams()[0].max_bitrate_bps);
// Target bitrate: 400000 * 3 / 4 = 300000.
EXPECT_EQ(300000, stream->GetVideoStreams()[1].min_bitrate_bps);
EXPECT_EQ(300000, stream->GetVideoStreams()[1].target_bitrate_bps);
EXPECT_EQ(400000, stream->GetVideoStreams()[1].max_bitrate_bps);
// Target bitrate: 600000 * 3 / 4 = 450000, less than min -> max.
EXPECT_EQ(500000, stream->GetVideoStreams()[2].min_bitrate_bps);
EXPECT_EQ(600000, stream->GetVideoStreams()[2].target_bitrate_bps);
EXPECT_EQ(600000, stream->GetVideoStreams()[2].max_bitrate_bps);
// No parameter changed, encoder should not be reconfigured.
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
EXPECT_EQ(2, stream->num_encoder_reconfigurations());
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
// Test to only specify the min or max bitrate value for a layer via
// RtpParameters. The unspecified min/max and target value should be set to the
// simulcast default that is used if no min/max are specified.
TEST_F(WebRtcVideoChannelTest, MinOrMaxSimulcastBitratePropagatedToEncoder) {
const size_t kNumSimulcastStreams = 3;
const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p();
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
// Change the value and set it on the VideoChannel.
// Layer 0: only configure min bitrate.
const int kMinBpsLayer0 = kDefault[0].min_bitrate_bps + 1;
parameters.encodings[0].min_bitrate_bps = kMinBpsLayer0;
// Layer 1: only configure max bitrate.
const int kMaxBpsLayer1 = kDefault[1].max_bitrate_bps - 1;
parameters.encodings[1].max_bitrate_bps = kMaxBpsLayer1;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the new value propagated down to the encoder.
// Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams);
EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size());
EXPECT_EQ(kMinBpsLayer0, encoder_config.simulcast_layers[0].min_bitrate_bps);
EXPECT_EQ(-1, encoder_config.simulcast_layers[0].max_bitrate_bps);
EXPECT_EQ(-1, encoder_config.simulcast_layers[1].min_bitrate_bps);
EXPECT_EQ(kMaxBpsLayer1, encoder_config.simulcast_layers[1].max_bitrate_bps);
EXPECT_EQ(-1, encoder_config.simulcast_layers[2].min_bitrate_bps);
EXPECT_EQ(-1, encoder_config.simulcast_layers[2].max_bitrate_bps);
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately for the simulcast case.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
// Layer 0: min configured bitrate should overwrite min default.
EXPECT_EQ(kMinBpsLayer0, stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(kDefault[0].target_bitrate_bps,
stream->GetVideoStreams()[0].target_bitrate_bps);
EXPECT_EQ(kDefault[0].max_bitrate_bps,
stream->GetVideoStreams()[0].max_bitrate_bps);
// Layer 1: max configured bitrate should overwrite max default.
EXPECT_EQ(kDefault[1].min_bitrate_bps,
stream->GetVideoStreams()[1].min_bitrate_bps);
EXPECT_EQ(kDefault[1].target_bitrate_bps,
stream->GetVideoStreams()[1].target_bitrate_bps);
EXPECT_EQ(kMaxBpsLayer1, stream->GetVideoStreams()[1].max_bitrate_bps);
// Layer 2: min and max bitrate not configured, default expected.
EXPECT_EQ(kDefault[2].min_bitrate_bps,
stream->GetVideoStreams()[2].min_bitrate_bps);
EXPECT_EQ(kDefault[2].target_bitrate_bps,
stream->GetVideoStreams()[2].target_bitrate_bps);
EXPECT_EQ(kDefault[2].max_bitrate_bps,
stream->GetVideoStreams()[2].max_bitrate_bps);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
// Test that specifying the min (or max) bitrate value for a layer via
// RtpParameters above (or below) the simulcast default max (or min) adjusts the
// unspecified values accordingly.
TEST_F(WebRtcVideoChannelTest, SetMinAndMaxSimulcastBitrateAboveBelowDefault) {
const size_t kNumSimulcastStreams = 3;
const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p();
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
// Change the value and set it on the VideoChannel.
// For layer 0, set the min bitrate above the default max.
const int kMinBpsLayer0 = kDefault[0].max_bitrate_bps + 1;
parameters.encodings[0].min_bitrate_bps = kMinBpsLayer0;
// For layer 1, set the max bitrate below the default min.
const int kMaxBpsLayer1 = kDefault[1].min_bitrate_bps - 1;
parameters.encodings[1].max_bitrate_bps = kMaxBpsLayer1;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the new value propagated down to the encoder.
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately for the simulcast case.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
// Layer 0: Min bitrate above default max (target/max should be adjusted).
EXPECT_EQ(kMinBpsLayer0, stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(kMinBpsLayer0, stream->GetVideoStreams()[0].target_bitrate_bps);
EXPECT_EQ(kMinBpsLayer0, stream->GetVideoStreams()[0].max_bitrate_bps);
// Layer 1: Max bitrate below default min (min/target should be adjusted).
EXPECT_EQ(kMaxBpsLayer1, stream->GetVideoStreams()[1].min_bitrate_bps);
EXPECT_EQ(kMaxBpsLayer1, stream->GetVideoStreams()[1].target_bitrate_bps);
EXPECT_EQ(kMaxBpsLayer1, stream->GetVideoStreams()[1].max_bitrate_bps);
// Layer 2: min and max bitrate not configured, default expected.
EXPECT_EQ(kDefault[2].min_bitrate_bps,
stream->GetVideoStreams()[2].min_bitrate_bps);
EXPECT_EQ(kDefault[2].target_bitrate_bps,
stream->GetVideoStreams()[2].target_bitrate_bps);
EXPECT_EQ(kDefault[2].max_bitrate_bps,
stream->GetVideoStreams()[2].max_bitrate_bps);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, BandwidthAboveTotalMaxBitrateGivenToMaxLayer) {
const size_t kNumSimulcastStreams = 3;
const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p();
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Set max bitrate for all but the highest layer.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
parameters.encodings[0].max_bitrate_bps = kDefault[0].max_bitrate_bps;
parameters.encodings[1].max_bitrate_bps = kDefault[1].max_bitrate_bps;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Set max bandwidth equal to total max bitrate.
send_parameters_.max_bandwidth_bps =
GetTotalMaxBitrateBps(stream->GetVideoStreams());
ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
// No bitrate above the total max to give to the highest layer.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
EXPECT_EQ(kDefault[2].max_bitrate_bps,
stream->GetVideoStreams()[2].max_bitrate_bps);
// Set max bandwidth above the total max bitrate.
send_parameters_.max_bandwidth_bps =
GetTotalMaxBitrateBps(stream->GetVideoStreams()) + 1;
ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
// The highest layer has no max bitrate set -> the bitrate above the total
// max should be given to the highest layer.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
EXPECT_EQ(send_parameters_.max_bandwidth_bps,
GetTotalMaxBitrateBps(stream->GetVideoStreams()));
EXPECT_EQ(kDefault[2].max_bitrate_bps + 1,
stream->GetVideoStreams()[2].max_bitrate_bps);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
BandwidthAboveTotalMaxBitrateNotGivenToMaxLayerIfMaxBitrateSet) {
const size_t kNumSimulcastStreams = 3;
const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p();
EXPECT_EQ(kNumSimulcastStreams, kDefault.size());
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Set max bitrate for the highest layer.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
parameters.encodings[2].max_bitrate_bps = kDefault[2].max_bitrate_bps;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Set max bandwidth above the total max bitrate.
send_parameters_.max_bandwidth_bps =
GetTotalMaxBitrateBps(stream->GetVideoStreams()) + 1;
ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
// The highest layer has the max bitrate set -> the bitrate above the total
// max should not be given to the highest layer.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
EXPECT_EQ(*parameters.encodings[2].max_bitrate_bps,
stream->GetVideoStreams()[2].max_bitrate_bps);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
// Test that min and max bitrate values set via RtpParameters are correctly
// propagated to the underlying encoder for a single stream.
TEST_F(WebRtcVideoChannelTest, MinAndMaxBitratePropagatedToEncoder) {
FakeVideoSendStream* stream = AddSendStream();
EXPECT_TRUE(channel_->SetSend(true));
EXPECT_TRUE(stream->IsSending());
// Set min and max bitrate.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1u, parameters.encodings.size());
parameters.encodings[0].min_bitrate_bps = 80000;
parameters.encodings[0].max_bitrate_bps = 150000;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
EXPECT_EQ(1u, encoder_config.number_of_streams);
EXPECT_EQ(1u, encoder_config.simulcast_layers.size());
EXPECT_EQ(80000, encoder_config.simulcast_layers[0].min_bitrate_bps);
EXPECT_EQ(150000, encoder_config.simulcast_layers[0].max_bitrate_bps);
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately.
EXPECT_EQ(1u, stream->GetVideoStreams().size());
EXPECT_EQ(80000, stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(150000, stream->GetVideoStreams()[0].target_bitrate_bps);
EXPECT_EQ(150000, stream->GetVideoStreams()[0].max_bitrate_bps);
}
// Test the default min and max bitrate value are correctly propagated to the
// underlying encoder for a single stream (when the values are not set via
// RtpParameters).
TEST_F(WebRtcVideoChannelTest, DefaultMinAndMaxBitratePropagatedToEncoder) {
FakeVideoSendStream* stream = AddSendStream();
EXPECT_TRUE(channel_->SetSend(true));
EXPECT_TRUE(stream->IsSending());
// Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
EXPECT_EQ(1u, encoder_config.number_of_streams);
EXPECT_EQ(1u, encoder_config.simulcast_layers.size());
EXPECT_EQ(-1, encoder_config.simulcast_layers[0].min_bitrate_bps);
EXPECT_EQ(-1, encoder_config.simulcast_layers[0].max_bitrate_bps);
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately.
EXPECT_EQ(1u, stream->GetVideoStreams().size());
EXPECT_EQ(webrtc::kDefaultMinVideoBitrateBps,
stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_GT(stream->GetVideoStreams()[0].max_bitrate_bps,
stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(stream->GetVideoStreams()[0].max_bitrate_bps,
stream->GetVideoStreams()[0].target_bitrate_bps);
}
// Test that a stream will not be sending if its encoding is made inactive
// through SetRtpSendParameters.
TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersOneEncodingActive) {
FakeVideoSendStream* stream = AddSendStream();
EXPECT_TRUE(channel_->SetSend(true));
EXPECT_TRUE(stream->IsSending());
// Get current parameters and change "active" to false.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(1u, parameters.encodings.size());
ASSERT_TRUE(parameters.encodings[0].active);
parameters.encodings[0].active = false;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
EXPECT_FALSE(stream->IsSending());
// Now change it back to active and verify we resume sending.
parameters.encodings[0].active = true;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
EXPECT_TRUE(stream->IsSending());
}
// Tests that when active is updated for any simulcast layer then the send
// stream's sending state will be updated and it will be reconfigured with the
// new appropriate active simulcast streams.
TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersMultipleEncodingsActive) {
// Create the stream params with multiple ssrcs for simulcast.
const size_t kNumSimulcastStreams = 3;
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
StreamParams stream_params = CreateSimStreamParams("cname", ssrcs);
FakeVideoSendStream* fake_video_send_stream = AddSendStream(stream_params);
uint32_t primary_ssrc = stream_params.first_ssrc();
// Using the FrameForwarder, we manually send a full size
// frame. This allows us to test that ReconfigureEncoder is called
// appropriately.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame(
1920, 1080, webrtc::VideoRotation::kVideoRotation_0,
rtc::kNumMicrosecsPerSec / 30));
// Check that all encodings are initially active.
webrtc::RtpParameters parameters =
channel_->GetRtpSendParameters(primary_ssrc);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_TRUE(parameters.encodings[0].active);
EXPECT_TRUE(parameters.encodings[1].active);
EXPECT_TRUE(parameters.encodings[2].active);
EXPECT_TRUE(fake_video_send_stream->IsSending());
// Only turn on only the middle stream.
parameters.encodings[0].active = false;
parameters.encodings[1].active = true;
parameters.encodings[2].active = false;
EXPECT_TRUE(channel_->SetRtpSendParameters(primary_ssrc, parameters).ok());
// Verify that the active fields are set on the VideoChannel.
parameters = channel_->GetRtpSendParameters(primary_ssrc);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_FALSE(parameters.encodings[0].active);
EXPECT_TRUE(parameters.encodings[1].active);
EXPECT_FALSE(parameters.encodings[2].active);
// Check that the VideoSendStream is updated appropriately. This means its
// send state was updated and it was reconfigured.
EXPECT_TRUE(fake_video_send_stream->IsSending());
std::vector<webrtc::VideoStream> simulcast_streams =
fake_video_send_stream->GetVideoStreams();
EXPECT_EQ(kNumSimulcastStreams, simulcast_streams.size());
EXPECT_FALSE(simulcast_streams[0].active);
EXPECT_TRUE(simulcast_streams[1].active);
EXPECT_FALSE(simulcast_streams[2].active);
// Turn off all streams.
parameters.encodings[0].active = false;
parameters.encodings[1].active = false;
parameters.encodings[2].active = false;
EXPECT_TRUE(channel_->SetRtpSendParameters(primary_ssrc, parameters).ok());
// Verify that the active fields are set on the VideoChannel.
parameters = channel_->GetRtpSendParameters(primary_ssrc);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_FALSE(parameters.encodings[0].active);
EXPECT_FALSE(parameters.encodings[1].active);
EXPECT_FALSE(parameters.encodings[2].active);
// Check that the VideoSendStream is off.
EXPECT_FALSE(fake_video_send_stream->IsSending());
simulcast_streams = fake_video_send_stream->GetVideoStreams();
EXPECT_EQ(kNumSimulcastStreams, simulcast_streams.size());
EXPECT_FALSE(simulcast_streams[0].active);
EXPECT_FALSE(simulcast_streams[1].active);
EXPECT_FALSE(simulcast_streams[2].active);
EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, nullptr, nullptr));
}
// Test that if a stream is reconfigured (due to a codec change or other
// change) while its encoding is still inactive, it doesn't start sending.
TEST_F(WebRtcVideoChannelTest,
InactiveStreamDoesntStartSendingWhenReconfigured) {
// Set an initial codec list, which will be modified later.
cricket::VideoSendParameters parameters1;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters1.codecs.push_back(GetEngineCodec("VP8"));
parameters1.codecs.push_back(GetEngineCodec("VP9"));
EXPECT_TRUE(channel_->SetSendParameters(parameters1));
FakeVideoSendStream* stream = AddSendStream();
EXPECT_TRUE(channel_->SetSend(true));
EXPECT_TRUE(stream->IsSending());
// Get current parameters and change "active" to false.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(1u, parameters.encodings.size());
ASSERT_TRUE(parameters.encodings[0].active);
parameters.encodings[0].active = false;
EXPECT_EQ(1u, GetFakeSendStreams().size());
EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams());
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
EXPECT_FALSE(stream->IsSending());
// Reorder the codec list, causing the stream to be reconfigured.
cricket::VideoSendParameters parameters2;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters2.codecs.push_back(GetEngineCodec("VP9"));
parameters2.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel_->SetSendParameters(parameters2));
auto new_streams = GetFakeSendStreams();
// Assert that a new underlying stream was created due to the codec change.
// Otherwise, this test isn't testing what it set out to test.
EXPECT_EQ(1u, GetFakeSendStreams().size());
EXPECT_EQ(2, fake_call_->GetNumCreatedSendStreams());
// Verify that we still are not sending anything, due to the inactive
// encoding.
EXPECT_FALSE(new_streams[0]->IsSending());
}
// Test that GetRtpSendParameters returns the currently configured codecs.
TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersCodecs) {
AddSendStream();
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
EXPECT_TRUE(channel_->SetSendParameters(parameters));
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(2u, rtp_parameters.codecs.size());
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
EXPECT_EQ(GetEngineCodec("VP8").ToCodecParameters(),
rtp_parameters.codecs[0]);
EXPECT_EQ(GetEngineCodec("VP9").ToCodecParameters(),
rtp_parameters.codecs[1]);
}
// Test that GetRtpSendParameters returns the currently configured RTCP CNAME.
TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersRtcpCname) {
StreamParams params = StreamParams::CreateLegacy(kSsrc);
params.cname = "rtcpcname";
AddSendStream(params);
webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrc);
EXPECT_STREQ("rtcpcname", rtp_parameters.rtcp.cname.c_str());
}
// Test that RtpParameters for send stream has one encoding and it has
// the correct SSRC.
TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersSsrc) {
AddSendStream();
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(1u, rtp_parameters.encodings.size());
EXPECT_EQ(last_ssrc_, rtp_parameters.encodings[0].ssrc);
}
TEST_F(WebRtcVideoChannelTest, DetectRtpSendParameterHeaderExtensionsChange) {
AddSendStream();
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpSendParameters(last_ssrc_);
rtp_parameters.header_extensions.emplace_back();
EXPECT_NE(0u, rtp_parameters.header_extensions.size());
webrtc::RTCError result =
channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
EXPECT_EQ(webrtc::RTCErrorType::INVALID_MODIFICATION, result.type());
}
TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersDegradationPreference) {
AddSendStream();
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(rtp_parameters.degradation_preference,
webrtc::DegradationPreference::BALANCED);
rtp_parameters.degradation_preference =
webrtc::DegradationPreference::MAINTAIN_FRAMERATE;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters).ok());
webrtc::RtpParameters updated_rtp_parameters =
channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(updated_rtp_parameters.degradation_preference,
webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
// Remove the source since it will be destroyed before the channel
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
// Test that if we set/get parameters multiple times, we get the same results.
TEST_F(WebRtcVideoChannelTest, SetAndGetRtpSendParameters) {
AddSendStream();
cricket::VideoSendParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
EXPECT_TRUE(channel_->SetSendParameters(parameters));
webrtc::RtpParameters initial_params =
channel_->GetRtpSendParameters(last_ssrc_);
// We should be able to set the params we just got.
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, initial_params).ok());
// ... And this shouldn't change the params returned by GetRtpSendParameters.
EXPECT_EQ(initial_params, channel_->GetRtpSendParameters(last_ssrc_));
}
// Test that GetRtpReceiveParameters returns the currently configured codecs.
TEST_F(WebRtcVideoChannelTest, GetRtpReceiveParametersCodecs) {
AddRecvStream();
cricket::VideoRecvParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpReceiveParameters(last_ssrc_);
ASSERT_EQ(2u, rtp_parameters.codecs.size());
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
EXPECT_EQ(GetEngineCodec("VP8").ToCodecParameters(),
rtp_parameters.codecs[0]);
EXPECT_EQ(GetEngineCodec("VP9").ToCodecParameters(),
rtp_parameters.codecs[1]);
}
#if defined(WEBRTC_USE_H264)
TEST_F(WebRtcVideoChannelTest, GetRtpReceiveFmtpSprop) {
#else
TEST_F(WebRtcVideoChannelTest, DISABLED_GetRtpReceiveFmtpSprop) {
#endif
cricket::VideoRecvParameters parameters;
cricket::VideoCodec kH264sprop1(101, "H264");
kH264sprop1.SetParam(kH264FmtpSpropParameterSets, "uvw");
parameters.codecs.push_back(kH264sprop1);
cricket::VideoCodec kH264sprop2(102, "H264");
kH264sprop2.SetParam(kH264FmtpSpropParameterSets, "xyz");
parameters.codecs.push_back(kH264sprop2);
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
FakeVideoReceiveStream* recv_stream = AddRecvStream();
const webrtc::VideoReceiveStream::Config& cfg = recv_stream->GetConfig();
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpReceiveParameters(last_ssrc_);
ASSERT_EQ(2u, rtp_parameters.codecs.size());
EXPECT_EQ(kH264sprop1.ToCodecParameters(), rtp_parameters.codecs[0]);
ASSERT_EQ(2u, cfg.decoders.size());
EXPECT_EQ(101, cfg.decoders[0].payload_type);
EXPECT_EQ("H264", cfg.decoders[0].video_format.name);
const auto it0 =
cfg.decoders[0].video_format.parameters.find(kH264FmtpSpropParameterSets);
ASSERT_TRUE(it0 != cfg.decoders[0].video_format.parameters.end());
EXPECT_EQ("uvw", it0->second);
EXPECT_EQ(102, cfg.decoders[1].payload_type);
EXPECT_EQ("H264", cfg.decoders[1].video_format.name);
const auto it1 =
cfg.decoders[1].video_format.parameters.find(kH264FmtpSpropParameterSets);
ASSERT_TRUE(it1 != cfg.decoders[1].video_format.parameters.end());
EXPECT_EQ("xyz", it1->second);
}
// Test that RtpParameters for receive stream has one encoding and it has
// the correct SSRC.
TEST_F(WebRtcVideoChannelTest, GetRtpReceiveParametersSsrc) {
AddRecvStream();
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpReceiveParameters(last_ssrc_);
ASSERT_EQ(1u, rtp_parameters.encodings.size());
EXPECT_EQ(last_ssrc_, rtp_parameters.encodings[0].ssrc);
}
// Test that if we set/get parameters multiple times, we get the same results.
TEST_F(WebRtcVideoChannelTest, SetAndGetRtpReceiveParameters) {
AddRecvStream();
cricket::VideoRecvParameters parameters;
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) Reason for revert: The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change: function removeVideoCodec(offerSdp) { - offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n', - 'a=rtpmap:100 XVP8/90000\r\n'); + offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/, + 'a=rtpmap:$1 XVP8/90000\r\n'); return offerSdp; } Original issue's description: > Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) > > Reason for revert: > Breaks chromium.fyi test: > WebRtcBrowserTest.NegotiateUnsupportedVideoCodec > > Original issue's description: > > Stop using hardcoded payload types for video codecs > > > > This CL stops using hardcoded payload types for different video codecs > > and will dynamically assign them payload types incrementally from 96 to > > 127 instead. > > > > This CL: > > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in > > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for > > internally supported software codecs instead. The purpose is to > > streamline the payload type assignment in webrtcvideoengine2.cc which > > will now have two encoder factories of the same > > WebRtcVideoEncoderFactory type; one internal and one external. > > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec > > instead. > > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and > > moves the create function to the internal encoder factory instead. > > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an > > interface without any static functions. > > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies > > the internal and external codecs and assigns them payload types > > incrementally from 96 to 127. > > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about > > what payload types will be used. > > > > BUG=webrtc:6677,webrtc:6705 > > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org > > > > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b > > Cr-Commit-Position: refs/heads/master@{#15135} > > TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6677,webrtc:6705 > > Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591 > Cr-Commit-Position: refs/heads/master@{#15140} TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6677,webrtc:6705 Review-Url: https://codereview.webrtc.org/2511933002 Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 01:34:11 -08:00
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
webrtc::RtpParameters initial_params =
channel_->GetRtpReceiveParameters(last_ssrc_);
// ... And this shouldn't change the params returned by
// GetRtpReceiveParameters.
EXPECT_EQ(initial_params, channel_->GetRtpReceiveParameters(last_ssrc_));
}
// Test that GetRtpReceiveParameters returns parameters correctly when SSRCs
// aren't signaled. It should always return an empty "RtpEncodingParameters",
// even after a packet is received and the unsignaled SSRC is known.
TEST_F(WebRtcVideoChannelTest, GetRtpReceiveParametersWithUnsignaledSsrc) {
// Call necessary methods to configure receiving a default stream as
// soon as it arrives.
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
// Call GetRtpReceiveParameters before configured to receive an unsignaled
// stream. Should return nothing.
EXPECT_EQ(webrtc::RtpParameters(), channel_->GetRtpReceiveParameters(0));
// Set a sink for an unsignaled stream.
cricket::FakeVideoRenderer renderer;
// Value of "0" means "unsignaled stream".
EXPECT_TRUE(channel_->SetSink(0, &renderer));
// Call GetRtpReceiveParameters before the SSRC is known. Value of "0"
// in this method means "unsignaled stream".
webrtc::RtpParameters rtp_parameters = channel_->GetRtpReceiveParameters(0);
ASSERT_EQ(1u, rtp_parameters.encodings.size());
EXPECT_FALSE(rtp_parameters.encodings[0].ssrc);
// Receive VP8 packet.
uint8_t data[kMinRtpPacketLen];
cricket::RtpHeader rtpHeader;
rtpHeader.payload_type = GetEngineCodec("VP8").id;
rtpHeader.seq_num = rtpHeader.timestamp = 0;
rtpHeader.ssrc = kIncomingUnsignalledSsrc;
cricket::SetRtpHeader(data, sizeof(data), rtpHeader);
rtc::CopyOnWriteBuffer packet(data, sizeof(data));
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
// The |ssrc| member should still be unset.
rtp_parameters = channel_->GetRtpReceiveParameters(0);
ASSERT_EQ(1u, rtp_parameters.encodings.size());
EXPECT_FALSE(rtp_parameters.encodings[0].ssrc);
}
void WebRtcVideoChannelTest::TestReceiverLocalSsrcConfiguration(
bool receiver_first) {
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
const uint32_t kSenderSsrc = 0xC0FFEE;
const uint32_t kSecondSenderSsrc = 0xBADCAFE;
const uint32_t kReceiverSsrc = 0x4711;
const uint32_t kExpectedDefaultReceiverSsrc = 1;
if (receiver_first) {
AddRecvStream(StreamParams::CreateLegacy(kReceiverSsrc));
std::vector<FakeVideoReceiveStream*> receive_streams =
fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(1u, receive_streams.size());
// Default local SSRC when we have no sender.
EXPECT_EQ(kExpectedDefaultReceiverSsrc,
receive_streams[0]->GetConfig().rtp.local_ssrc);
}
AddSendStream(StreamParams::CreateLegacy(kSenderSsrc));
if (!receiver_first)
AddRecvStream(StreamParams::CreateLegacy(kReceiverSsrc));
std::vector<FakeVideoReceiveStream*> receive_streams =
fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(1u, receive_streams.size());
EXPECT_EQ(kSenderSsrc, receive_streams[0]->GetConfig().rtp.local_ssrc);
// Removing first sender should fall back to another (in this case the second)
// local send stream's SSRC.
AddSendStream(StreamParams::CreateLegacy(kSecondSenderSsrc));
ASSERT_TRUE(channel_->RemoveSendStream(kSenderSsrc));
receive_streams = fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(1u, receive_streams.size());
EXPECT_EQ(kSecondSenderSsrc, receive_streams[0]->GetConfig().rtp.local_ssrc);
// Removing the last sender should fall back to default local SSRC.
ASSERT_TRUE(channel_->RemoveSendStream(kSecondSenderSsrc));
receive_streams = fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(1u, receive_streams.size());
EXPECT_EQ(kExpectedDefaultReceiverSsrc,
receive_streams[0]->GetConfig().rtp.local_ssrc);
}
TEST_F(WebRtcVideoChannelTest, ConfiguresLocalSsrc) {
TestReceiverLocalSsrcConfiguration(false);
}
TEST_F(WebRtcVideoChannelTest, ConfiguresLocalSsrcOnExistingReceivers) {
TestReceiverLocalSsrcConfiguration(true);
}
class WebRtcVideoChannelSimulcastTest : public ::testing::Test {
public:
WebRtcVideoChannelSimulcastTest()
: fake_call_(),
encoder_factory_(new cricket::FakeWebRtcVideoEncoderFactory),
decoder_factory_(new cricket::FakeWebRtcVideoDecoderFactory),
mock_rate_allocator_factory_(
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
std::make_unique<webrtc::MockVideoBitrateAllocatorFactory>()),
engine_(std::unique_ptr<cricket::FakeWebRtcVideoEncoderFactory>(
encoder_factory_),
std::unique_ptr<cricket::FakeWebRtcVideoDecoderFactory>(
decoder_factory_)),
last_ssrc_(0) {}
void SetUp() override {
encoder_factory_->AddSupportedVideoCodecType("VP8");
channel_.reset(engine_.CreateMediaChannel(
&fake_call_, GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
mock_rate_allocator_factory_.get()));
channel_->OnReadyToSend(true);
last_ssrc_ = 123;
}
protected:
void VerifySimulcastSettings(const VideoCodec& codec,
int capture_width,
int capture_height,
size_t num_configured_streams,
size_t expected_num_streams,
bool screenshare,
bool conference_mode) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(codec);
parameters.conference_mode = conference_mode;
ASSERT_TRUE(channel_->SetSendParameters(parameters));
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
RTC_DCHECK(num_configured_streams <= ssrcs.size());
ssrcs.resize(num_configured_streams);
AddSendStream(CreateSimStreamParams("cname", ssrcs));
// Send a full-size frame to trigger a stream reconfiguration to use all
// expected simulcast layers.
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(capture_width, capture_height,
rtc::kNumMicrosecsPerSec / 30);
VideoOptions options;
if (screenshare)
options.is_screencast = screenshare;
EXPECT_TRUE(
channel_->SetVideoSend(ssrcs.front(), &options, &frame_forwarder));
// Fetch the latest stream since SetVideoSend() may recreate it if the
// screen content setting is changed.
FakeVideoSendStream* stream = fake_call_.GetVideoSendStreams().front();
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
auto rtp_parameters = channel_->GetRtpSendParameters(kSsrcs3[0]);
EXPECT_EQ(num_configured_streams, rtp_parameters.encodings.size());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(expected_num_streams, video_streams.size());
EXPECT_LE(expected_num_streams, stream->GetConfig().rtp.ssrcs.size());
std::vector<webrtc::VideoStream> expected_streams;
if (num_configured_streams > 1 || conference_mode) {
expected_streams = GetSimulcastConfig(
num_configured_streams, capture_width, capture_height,
webrtc::kDefaultBitratePriority, kDefaultQpMax,
screenshare && conference_mode, true);
if (screenshare && conference_mode) {
for (const webrtc::VideoStream& stream : expected_streams) {
// Never scale screen content.
EXPECT_EQ(stream.width, rtc::checked_cast<size_t>(capture_width));
EXPECT_EQ(stream.height, rtc::checked_cast<size_t>(capture_height));
}
}
} else {
webrtc::VideoStream stream;
stream.width = capture_width;
stream.height = capture_height;
stream.max_framerate = kDefaultVideoMaxFramerate;
stream.min_bitrate_bps = webrtc::kDefaultMinVideoBitrateBps;
stream.target_bitrate_bps = stream.max_bitrate_bps =
GetMaxDefaultBitrateBps(capture_width, capture_height);
stream.max_qp = kDefaultQpMax;
expected_streams.push_back(stream);
}
ASSERT_EQ(expected_streams.size(), video_streams.size());
size_t num_streams = video_streams.size();
int total_max_bitrate_bps = 0;
for (size_t i = 0; i < num_streams; ++i) {
EXPECT_EQ(expected_streams[i].width, video_streams[i].width);
EXPECT_EQ(expected_streams[i].height, video_streams[i].height);
EXPECT_GT(video_streams[i].max_framerate, 0);
EXPECT_EQ(expected_streams[i].max_framerate,
video_streams[i].max_framerate);
EXPECT_GT(video_streams[i].min_bitrate_bps, 0);
EXPECT_EQ(expected_streams[i].min_bitrate_bps,
video_streams[i].min_bitrate_bps);
EXPECT_GT(video_streams[i].target_bitrate_bps, 0);
EXPECT_EQ(expected_streams[i].target_bitrate_bps,
video_streams[i].target_bitrate_bps);
EXPECT_GT(video_streams[i].max_bitrate_bps, 0);
EXPECT_EQ(expected_streams[i].max_bitrate_bps,
video_streams[i].max_bitrate_bps);
EXPECT_GT(video_streams[i].max_qp, 0);
EXPECT_EQ(expected_streams[i].max_qp, video_streams[i].max_qp);
EXPECT_EQ(num_configured_streams > 1 || conference_mode,
Reland "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers." This reverts commit e27e0aca9411b6990fcdf56d8a3475569ee5fd2f. Reason for revert: <INSERT REASONING HERE> Original change's description: > Revert "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers." > > This reverts commit d2ed0a4c9e7f04060d8e3358eb0006c31579bb86. > > Reason for revert: Breaks downstream projects. > > Original change's description: > > Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers. > > > > temporal_layer_thresholds_bps served only one purpose: its size was used > > to infer number of temporal layers. I replaced it with num_temporal_layers, > > which does what is says. > > > > The practical reason for this change is the need to have possibility to > > distinguish between cases when VP9 SVC temporal layering was/not set > > through field trial. That was not possible with > > temporal_layer_thresholds_bps[] because empty vector means 1 temporal > > layer. > > > > Bug: webrtc:8518 > > Change-Id: I275ec3a8c74e8ba409eb049878199f132a20ec51 > > Reviewed-on: https://webrtc-review.googlesource.com/58084 > > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22230} > > TBR=sprang@webrtc.org,stefan@webrtc.org,ssilkin@webrtc.org > > Change-Id: Ic2940f7f78a74312170940d51ad8967cde8ad42f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8518 > Reviewed-on: https://webrtc-review.googlesource.com/58902 > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22234} TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,ssilkin@webrtc.org Change-Id: I1900c6b845b9baa9430fb72c3f4e7f2a44b3a8b1 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8518 Reviewed-on: https://webrtc-review.googlesource.com/59160 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22256}
2018-03-01 15:11:29 +01:00
expected_streams[i].num_temporal_layers.has_value());
if (conference_mode) {
EXPECT_EQ(expected_streams[i].num_temporal_layers,
video_streams[i].num_temporal_layers);
}
if (i == num_streams - 1) {
total_max_bitrate_bps += video_streams[i].max_bitrate_bps;
} else {
total_max_bitrate_bps += video_streams[i].target_bitrate_bps;
}
}
EXPECT_TRUE(channel_->SetVideoSend(ssrcs.front(), nullptr, nullptr));
}
FakeVideoSendStream* AddSendStream() {
return AddSendStream(StreamParams::CreateLegacy(last_ssrc_++));
}
FakeVideoSendStream* AddSendStream(const StreamParams& sp) {
size_t num_streams = fake_call_.GetVideoSendStreams().size();
EXPECT_TRUE(channel_->AddSendStream(sp));
std::vector<FakeVideoSendStream*> streams =
fake_call_.GetVideoSendStreams();
EXPECT_EQ(num_streams + 1, streams.size());
return streams[streams.size() - 1];
}
std::vector<FakeVideoSendStream*> GetFakeSendStreams() {
return fake_call_.GetVideoSendStreams();
}
FakeVideoReceiveStream* AddRecvStream() {
return AddRecvStream(StreamParams::CreateLegacy(last_ssrc_++));
}
FakeVideoReceiveStream* AddRecvStream(const StreamParams& sp) {
size_t num_streams = fake_call_.GetVideoReceiveStreams().size();
EXPECT_TRUE(channel_->AddRecvStream(sp));
std::vector<FakeVideoReceiveStream*> streams =
fake_call_.GetVideoReceiveStreams();
EXPECT_EQ(num_streams + 1, streams.size());
return streams[streams.size() - 1];
}
webrtc::RtcEventLogNull event_log_;
FakeCall fake_call_;
cricket::FakeWebRtcVideoEncoderFactory* encoder_factory_;
cricket::FakeWebRtcVideoDecoderFactory* decoder_factory_;
std::unique_ptr<webrtc::MockVideoBitrateAllocatorFactory>
mock_rate_allocator_factory_;
WebRtcVideoEngine engine_;
std::unique_ptr<VideoMediaChannel> channel_;
uint32_t last_ssrc_;
};
TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsWith2SimulcastStreams) {
VerifySimulcastSettings(cricket::VideoCodec("VP8"), 640, 360, 2, 2, false,
true);
}
TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsWith3SimulcastStreams) {
VerifySimulcastSettings(cricket::VideoCodec("VP8"), 1280, 720, 3, 3, false,
true);
}
// Test that we normalize send codec format size in simulcast.
TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsWithOddSizeInSimulcast) {
VerifySimulcastSettings(cricket::VideoCodec("VP8"), 541, 271, 2, 2, false,
true);
}
TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsForScreenshare) {
VerifySimulcastSettings(cricket::VideoCodec("VP8"), 1280, 720, 3, 3, true,
false);
}
TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsForSimulcastScreenshare) {
VerifySimulcastSettings(cricket::VideoCodec("VP8"), 1280, 720, 3, 2, true,
true);
}
TEST_F(WebRtcVideoChannelSimulcastTest, SimulcastScreenshareWithoutConference) {
VerifySimulcastSettings(cricket::VideoCodec("VP8"), 1280, 720, 3, 3, true,
false);
}
TEST_F(WebRtcVideoChannelBaseTest, GetSources) {
EXPECT_THAT(channel_->GetSources(kSsrc), IsEmpty());
EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
EXPECT_TRUE(SetDefaultCodec());
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(renderer_.num_rendered_frames(), 0);
// Send and receive one frame.
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_THAT(channel_->GetSources(kSsrc - 1), IsEmpty());
EXPECT_THAT(channel_->GetSources(kSsrc), SizeIs(1));
EXPECT_THAT(channel_->GetSources(kSsrc + 1), IsEmpty());
webrtc::RtpSource source = channel_->GetSources(kSsrc)[0];
EXPECT_EQ(source.source_id(), kSsrc);
EXPECT_EQ(source.source_type(), webrtc::RtpSourceType::SSRC);
int64_t rtp_timestamp_1 = source.rtp_timestamp();
int64_t timestamp_ms_1 = source.timestamp_ms();
// Send and receive another frame.
SendFrame();
EXPECT_FRAME_WAIT(2, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_THAT(channel_->GetSources(kSsrc - 1), IsEmpty());
EXPECT_THAT(channel_->GetSources(kSsrc), SizeIs(1));
EXPECT_THAT(channel_->GetSources(kSsrc + 1), IsEmpty());
source = channel_->GetSources(kSsrc)[0];
EXPECT_EQ(source.source_id(), kSsrc);
EXPECT_EQ(source.source_type(), webrtc::RtpSourceType::SSRC);
int64_t rtp_timestamp_2 = source.rtp_timestamp();
int64_t timestamp_ms_2 = source.timestamp_ms();
EXPECT_GT(rtp_timestamp_2, rtp_timestamp_1);
EXPECT_GT(timestamp_ms_2, timestamp_ms_1);
}
TEST_F(WebRtcVideoChannelTest, SetsRidsOnSendStream) {
StreamParams sp = CreateSimStreamParams("cname", {123, 456, 789});
std::vector<std::string> rids = {"f", "h", "q"};
std::vector<cricket::RidDescription> rid_descriptions;
for (const auto& rid : rids) {
rid_descriptions.emplace_back(rid, cricket::RidDirection::kSend);
}
sp.set_rids(rid_descriptions);
ASSERT_TRUE(channel_->AddSendStream(sp));
const auto& streams = fake_call_->GetVideoSendStreams();
ASSERT_EQ(1u, streams.size());
auto stream = streams[0];
ASSERT_NE(stream, nullptr);
const auto& config = stream->GetConfig();
EXPECT_THAT(config.rtp.rids, ::testing::ElementsAreArray(rids));
}
} // namespace cricket