webrtc_m130/video/video_receive_stream_unittest.cc

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/*
* Copyright 2017 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 "video/video_receive_stream.h"
#include <algorithm>
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
#include <memory>
#include <utility>
#include <vector>
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/video/function_video_decoder_factory.h"
#include "api/video_codecs/video_decoder.h"
#include "call/rtp_stream_receiver_controller.h"
#include "common_video/test/utilities.h"
#include "media/base/fake_video_renderer.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "modules/utility/include/process_thread.h"
#include "modules/video_coding/encoded_frame.h"
#include "rtc_base/event.h"
#include "system_wrappers/include/clock.h"
#include "test/fake_decoder.h"
#include "test/field_trial.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/time_controller/simulated_time_controller.h"
#include "test/video_decoder_proxy_factory.h"
#include "video/call_stats.h"
namespace webrtc {
namespace {
using ::testing::_;
using ::testing::ElementsAreArray;
using ::testing::Invoke;
using ::testing::IsEmpty;
using ::testing::SizeIs;
constexpr int kDefaultTimeOutMs = 50;
class MockTransport : public Transport {
public:
MOCK_METHOD(bool,
SendRtp,
(const uint8_t*, size_t length, const PacketOptions& options),
(override));
MOCK_METHOD(bool, SendRtcp, (const uint8_t*, size_t length), (override));
};
class MockVideoDecoder : public VideoDecoder {
public:
MOCK_METHOD(int32_t,
InitDecode,
(const VideoCodec*, int32_t number_of_cores),
(override));
MOCK_METHOD(int32_t,
Decode,
(const EncodedImage& input,
bool missing_frames,
int64_t render_time_ms),
(override));
MOCK_METHOD(int32_t,
RegisterDecodeCompleteCallback,
(DecodedImageCallback*),
(override));
MOCK_METHOD(int32_t, Release, (), (override));
MOCK_METHOD(VideoDecoder::DecoderInfo, GetDetcoderInfo, (), (const override));
const char* ImplementationName() const { return "MockVideoDecoder"; }
};
class FrameObjectFake : public EncodedFrame {
public:
void SetPayloadType(uint8_t payload_type) { _payloadType = payload_type; }
void SetRotation(const VideoRotation& rotation) { rotation_ = rotation; }
void SetNtpTime(int64_t ntp_time_ms) { ntp_time_ms_ = ntp_time_ms; }
int64_t ReceivedTime() const override { return 0; }
int64_t RenderTime() const override { return _renderTimeMs; }
};
} // namespace
class VideoReceiveStreamTest : public ::testing::Test {
public:
VideoReceiveStreamTest()
: process_thread_(ProcessThread::Create("TestThread")),
task_queue_factory_(CreateDefaultTaskQueueFactory()),
h264_decoder_factory_(&mock_h264_video_decoder_),
config_(&mock_transport_, &h264_decoder_factory_),
call_stats_(Clock::GetRealTimeClock(), process_thread_.get()) {}
void SetUp() {
constexpr int kDefaultNumCpuCores = 2;
config_.rtp.remote_ssrc = 1111;
config_.rtp.local_ssrc = 2222;
config_.renderer = &fake_renderer_;
VideoReceiveStream::Decoder h264_decoder;
h264_decoder.payload_type = 99;
h264_decoder.video_format = SdpVideoFormat("H264");
h264_decoder.video_format.parameters.insert(
{"sprop-parameter-sets", "Z0IACpZTBYmI,aMljiA=="});
config_.decoders.push_back(h264_decoder);
clock_ = Clock::GetRealTimeClock();
timing_ = new VCMTiming(clock_);
video_receive_stream_ =
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::internal::VideoReceiveStream>(
task_queue_factory_.get(), &rtp_stream_receiver_controller_,
kDefaultNumCpuCores, &packet_router_, config_.Copy(),
process_thread_.get(), &call_stats_, clock_, timing_);
}
protected:
std::unique_ptr<ProcessThread> process_thread_;
const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
test::VideoDecoderProxyFactory h264_decoder_factory_;
VideoReceiveStream::Config config_;
CallStats call_stats_;
MockVideoDecoder mock_h264_video_decoder_;
cricket::FakeVideoRenderer fake_renderer_;
MockTransport mock_transport_;
PacketRouter packet_router_;
RtpStreamReceiverController rtp_stream_receiver_controller_;
std::unique_ptr<webrtc::internal::VideoReceiveStream> video_receive_stream_;
Clock* clock_;
VCMTiming* timing_;
};
TEST_F(VideoReceiveStreamTest, CreateFrameFromH264FmtpSpropAndIdr) {
constexpr uint8_t idr_nalu[] = {0x05, 0xFF, 0xFF, 0xFF};
RtpPacketToSend rtppacket(nullptr);
uint8_t* payload = rtppacket.AllocatePayload(sizeof(idr_nalu));
memcpy(payload, idr_nalu, sizeof(idr_nalu));
rtppacket.SetMarker(true);
rtppacket.SetSsrc(1111);
rtppacket.SetPayloadType(99);
rtppacket.SetSequenceNumber(1);
rtppacket.SetTimestamp(0);
rtc::Event init_decode_event_;
EXPECT_CALL(mock_h264_video_decoder_, InitDecode(_, _))
.WillOnce(Invoke([&init_decode_event_](const VideoCodec* config,
int32_t number_of_cores) {
init_decode_event_.Set();
return 0;
}));
EXPECT_CALL(mock_h264_video_decoder_, RegisterDecodeCompleteCallback(_));
video_receive_stream_->Start();
EXPECT_CALL(mock_h264_video_decoder_, Decode(_, false, _));
RtpPacketReceived parsed_packet;
ASSERT_TRUE(parsed_packet.Parse(rtppacket.data(), rtppacket.size()));
rtp_stream_receiver_controller_.OnRtpPacket(parsed_packet);
EXPECT_CALL(mock_h264_video_decoder_, Release());
// Make sure the decoder thread had a chance to run.
init_decode_event_.Wait(kDefaultTimeOutMs);
}
TEST_F(VideoReceiveStreamTest, PlayoutDelay) {
const VideoPlayoutDelay kPlayoutDelayMs = {123, 321};
std::unique_ptr<FrameObjectFake> test_frame(new FrameObjectFake());
test_frame->SetId(0);
test_frame->SetPlayoutDelay(kPlayoutDelayMs);
video_receive_stream_->OnCompleteFrame(std::move(test_frame));
EXPECT_EQ(kPlayoutDelayMs.min_ms, timing_->min_playout_delay());
EXPECT_EQ(kPlayoutDelayMs.max_ms, timing_->max_playout_delay());
// Check that the biggest minimum delay is chosen.
video_receive_stream_->SetMinimumPlayoutDelay(400);
EXPECT_EQ(400, timing_->min_playout_delay());
// Check base minimum delay validation.
EXPECT_FALSE(video_receive_stream_->SetBaseMinimumPlayoutDelayMs(12345));
EXPECT_FALSE(video_receive_stream_->SetBaseMinimumPlayoutDelayMs(-1));
EXPECT_TRUE(video_receive_stream_->SetBaseMinimumPlayoutDelayMs(500));
EXPECT_EQ(500, timing_->min_playout_delay());
// Check that intermidiate values are remembered and the biggest remembered
// is chosen.
video_receive_stream_->SetBaseMinimumPlayoutDelayMs(0);
EXPECT_EQ(400, timing_->min_playout_delay());
video_receive_stream_->SetMinimumPlayoutDelay(0);
EXPECT_EQ(123, timing_->min_playout_delay());
}
Revert "Reland "Preserve min and max playout delay from RTP header extension"" This reverts commit 87bed4793ff8f463202f442381339626d0b27f0d. Reason for revert: <INSERT REASONING HERE> Original change's description: > Reland "Preserve min and max playout delay from RTP header extension" > > This reverts commit f31cc08ba01ed403e89255b5f3f38d5dbdde855e. > > Reason for revert: Reland with fixes > > Original change's description: > > Revert "Preserve min and max playout delay from RTP header extension" > > > > This reverts commit 85ba9972c42544c4771e394c9aa1d20bf5d09a91. > > > > Reason for revert: Audio might be more out of sync than needed due to jitter in received video stream. > > > > Original change's description: > > > Preserve min and max playout delay from RTP header extension > > > > > > Audio and video synchronization can sometimes override the minimum > > > and maximum playout delay that is set through the RTP header > > > extension. This CL makes sure that the playout delay always is > > > within the limits set by the RTP header extension. > > > > > > Bug: webrtc:10886 > > > Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645 > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#28980} > > > > TBR=stefan@webrtc.org,kron@webrtc.org > > > > Change-Id: I417e440d8a7e04ab3e19faa4454b704d2b971cd7 > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Bug: webrtc:10886 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150652 > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#28984} > > TBR=stefan@webrtc.org,kron@webrtc.org > > Change-Id: I5a3908a8c45f7faedab6f009b22df81d674e13a0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10886 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150653 > Reviewed-by: Johannes Kron <kron@webrtc.org> > Commit-Queue: Johannes Kron <kron@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28985} TBR=stefan@webrtc.org,kron@webrtc.org Change-Id: Id2e5d1ff804881e956a07fa4ae0f8301895dcc95 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10886 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150654 Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28986}
2019-08-28 12:41:11 +00:00
TEST_F(VideoReceiveStreamTest, PlayoutDelayPreservesDefaultMaxValue) {
const int default_max_playout_latency = timing_->max_playout_delay();
const VideoPlayoutDelay kPlayoutDelayMs = {123, -1};
std::unique_ptr<FrameObjectFake> test_frame(new FrameObjectFake());
test_frame->SetId(0);
test_frame->SetPlayoutDelay(kPlayoutDelayMs);
video_receive_stream_->OnCompleteFrame(std::move(test_frame));
// Ensure that -1 preserves default maximum value from `timing_`.
Revert "Reland "Preserve min and max playout delay from RTP header extension"" This reverts commit 87bed4793ff8f463202f442381339626d0b27f0d. Reason for revert: <INSERT REASONING HERE> Original change's description: > Reland "Preserve min and max playout delay from RTP header extension" > > This reverts commit f31cc08ba01ed403e89255b5f3f38d5dbdde855e. > > Reason for revert: Reland with fixes > > Original change's description: > > Revert "Preserve min and max playout delay from RTP header extension" > > > > This reverts commit 85ba9972c42544c4771e394c9aa1d20bf5d09a91. > > > > Reason for revert: Audio might be more out of sync than needed due to jitter in received video stream. > > > > Original change's description: > > > Preserve min and max playout delay from RTP header extension > > > > > > Audio and video synchronization can sometimes override the minimum > > > and maximum playout delay that is set through the RTP header > > > extension. This CL makes sure that the playout delay always is > > > within the limits set by the RTP header extension. > > > > > > Bug: webrtc:10886 > > > Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645 > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#28980} > > > > TBR=stefan@webrtc.org,kron@webrtc.org > > > > Change-Id: I417e440d8a7e04ab3e19faa4454b704d2b971cd7 > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Bug: webrtc:10886 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150652 > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#28984} > > TBR=stefan@webrtc.org,kron@webrtc.org > > Change-Id: I5a3908a8c45f7faedab6f009b22df81d674e13a0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10886 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150653 > Reviewed-by: Johannes Kron <kron@webrtc.org> > Commit-Queue: Johannes Kron <kron@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28985} TBR=stefan@webrtc.org,kron@webrtc.org Change-Id: Id2e5d1ff804881e956a07fa4ae0f8301895dcc95 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10886 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150654 Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28986}
2019-08-28 12:41:11 +00:00
EXPECT_EQ(kPlayoutDelayMs.min_ms, timing_->min_playout_delay());
EXPECT_NE(kPlayoutDelayMs.max_ms, timing_->max_playout_delay());
EXPECT_EQ(default_max_playout_latency, timing_->max_playout_delay());
}
Revert "Reland "Preserve min and max playout delay from RTP header extension"" This reverts commit 87bed4793ff8f463202f442381339626d0b27f0d. Reason for revert: <INSERT REASONING HERE> Original change's description: > Reland "Preserve min and max playout delay from RTP header extension" > > This reverts commit f31cc08ba01ed403e89255b5f3f38d5dbdde855e. > > Reason for revert: Reland with fixes > > Original change's description: > > Revert "Preserve min and max playout delay from RTP header extension" > > > > This reverts commit 85ba9972c42544c4771e394c9aa1d20bf5d09a91. > > > > Reason for revert: Audio might be more out of sync than needed due to jitter in received video stream. > > > > Original change's description: > > > Preserve min and max playout delay from RTP header extension > > > > > > Audio and video synchronization can sometimes override the minimum > > > and maximum playout delay that is set through the RTP header > > > extension. This CL makes sure that the playout delay always is > > > within the limits set by the RTP header extension. > > > > > > Bug: webrtc:10886 > > > Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645 > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#28980} > > > > TBR=stefan@webrtc.org,kron@webrtc.org > > > > Change-Id: I417e440d8a7e04ab3e19faa4454b704d2b971cd7 > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Bug: webrtc:10886 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150652 > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#28984} > > TBR=stefan@webrtc.org,kron@webrtc.org > > Change-Id: I5a3908a8c45f7faedab6f009b22df81d674e13a0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10886 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150653 > Reviewed-by: Johannes Kron <kron@webrtc.org> > Commit-Queue: Johannes Kron <kron@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28985} TBR=stefan@webrtc.org,kron@webrtc.org Change-Id: Id2e5d1ff804881e956a07fa4ae0f8301895dcc95 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10886 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150654 Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28986}
2019-08-28 12:41:11 +00:00
TEST_F(VideoReceiveStreamTest, PlayoutDelayPreservesDefaultMinValue) {
const int default_min_playout_latency = timing_->min_playout_delay();
const VideoPlayoutDelay kPlayoutDelayMs = {-1, 321};
std::unique_ptr<FrameObjectFake> test_frame(new FrameObjectFake());
test_frame->SetId(0);
test_frame->SetPlayoutDelay(kPlayoutDelayMs);
video_receive_stream_->OnCompleteFrame(std::move(test_frame));
Reland "Preserve min and max playout delay from RTP header extension" This reverts commit f31cc08ba01ed403e89255b5f3f38d5dbdde855e. Reason for revert: Reland with fixes Original change's description: > Revert "Preserve min and max playout delay from RTP header extension" > > This reverts commit 85ba9972c42544c4771e394c9aa1d20bf5d09a91. > > Reason for revert: Audio might be more out of sync than needed due to jitter in received video stream. > > Original change's description: > > Preserve min and max playout delay from RTP header extension > > > > Audio and video synchronization can sometimes override the minimum > > and maximum playout delay that is set through the RTP header > > extension. This CL makes sure that the playout delay always is > > within the limits set by the RTP header extension. > > > > Bug: webrtc:10886 > > Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645 > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#28980} > > TBR=stefan@webrtc.org,kron@webrtc.org > > Change-Id: I417e440d8a7e04ab3e19faa4454b704d2b971cd7 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10886 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150652 > Reviewed-by: Johannes Kron <kron@webrtc.org> > Commit-Queue: Johannes Kron <kron@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28984} TBR=stefan@webrtc.org,kron@webrtc.org Change-Id: I5a3908a8c45f7faedab6f009b22df81d674e13a0 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10886 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150653 Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28985}
2019-08-28 12:40:45 +00:00
// Ensure that -1 preserves default minimum value from `timing_`.
Revert "Reland "Preserve min and max playout delay from RTP header extension"" This reverts commit 87bed4793ff8f463202f442381339626d0b27f0d. Reason for revert: <INSERT REASONING HERE> Original change's description: > Reland "Preserve min and max playout delay from RTP header extension" > > This reverts commit f31cc08ba01ed403e89255b5f3f38d5dbdde855e. > > Reason for revert: Reland with fixes > > Original change's description: > > Revert "Preserve min and max playout delay from RTP header extension" > > > > This reverts commit 85ba9972c42544c4771e394c9aa1d20bf5d09a91. > > > > Reason for revert: Audio might be more out of sync than needed due to jitter in received video stream. > > > > Original change's description: > > > Preserve min and max playout delay from RTP header extension > > > > > > Audio and video synchronization can sometimes override the minimum > > > and maximum playout delay that is set through the RTP header > > > extension. This CL makes sure that the playout delay always is > > > within the limits set by the RTP header extension. > > > > > > Bug: webrtc:10886 > > > Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645 > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#28980} > > > > TBR=stefan@webrtc.org,kron@webrtc.org > > > > Change-Id: I417e440d8a7e04ab3e19faa4454b704d2b971cd7 > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Bug: webrtc:10886 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150652 > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#28984} > > TBR=stefan@webrtc.org,kron@webrtc.org > > Change-Id: I5a3908a8c45f7faedab6f009b22df81d674e13a0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10886 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150653 > Reviewed-by: Johannes Kron <kron@webrtc.org> > Commit-Queue: Johannes Kron <kron@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28985} TBR=stefan@webrtc.org,kron@webrtc.org Change-Id: Id2e5d1ff804881e956a07fa4ae0f8301895dcc95 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10886 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150654 Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28986}
2019-08-28 12:41:11 +00:00
EXPECT_NE(kPlayoutDelayMs.min_ms, timing_->min_playout_delay());
EXPECT_EQ(kPlayoutDelayMs.max_ms, timing_->max_playout_delay());
EXPECT_EQ(default_min_playout_latency, timing_->min_playout_delay());
}
class VideoReceiveStreamTestWithFakeDecoder : public ::testing::Test {
public:
VideoReceiveStreamTestWithFakeDecoder()
: fake_decoder_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
[]() { return std::make_unique<test::FakeDecoder>(); }),
process_thread_(ProcessThread::Create("TestThread")),
task_queue_factory_(CreateDefaultTaskQueueFactory()),
config_(&mock_transport_, &fake_decoder_factory_),
call_stats_(Clock::GetRealTimeClock(), process_thread_.get()) {}
void SetUp() {
config_.rtp.remote_ssrc = 1111;
config_.rtp.local_ssrc = 2222;
config_.renderer = &fake_renderer_;
VideoReceiveStream::Decoder fake_decoder;
fake_decoder.payload_type = 99;
fake_decoder.video_format = SdpVideoFormat("VP8");
config_.decoders.push_back(fake_decoder);
clock_ = Clock::GetRealTimeClock();
ReCreateReceiveStream(VideoReceiveStream::RecordingState());
}
void ReCreateReceiveStream(VideoReceiveStream::RecordingState state) {
constexpr int kDefaultNumCpuCores = 2;
video_receive_stream_ = nullptr;
timing_ = new VCMTiming(clock_);
video_receive_stream_.reset(new webrtc::internal::VideoReceiveStream(
task_queue_factory_.get(), &rtp_stream_receiver_controller_,
kDefaultNumCpuCores, &packet_router_, config_.Copy(),
process_thread_.get(), &call_stats_, clock_, timing_));
video_receive_stream_->SetAndGetRecordingState(std::move(state), false);
}
protected:
test::FunctionVideoDecoderFactory fake_decoder_factory_;
std::unique_ptr<ProcessThread> process_thread_;
const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
VideoReceiveStream::Config config_;
CallStats call_stats_;
cricket::FakeVideoRenderer fake_renderer_;
MockTransport mock_transport_;
PacketRouter packet_router_;
RtpStreamReceiverController rtp_stream_receiver_controller_;
std::unique_ptr<webrtc::internal::VideoReceiveStream> video_receive_stream_;
Clock* clock_;
VCMTiming* timing_;
};
TEST_F(VideoReceiveStreamTestWithFakeDecoder, PassesNtpTime) {
const int64_t kNtpTimestamp = 12345;
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
auto test_frame = std::make_unique<FrameObjectFake>();
test_frame->SetPayloadType(99);
test_frame->SetId(0);
test_frame->SetNtpTime(kNtpTimestamp);
video_receive_stream_->Start();
video_receive_stream_->OnCompleteFrame(std::move(test_frame));
EXPECT_TRUE(fake_renderer_.WaitForRenderedFrame(kDefaultTimeOutMs));
EXPECT_EQ(kNtpTimestamp, fake_renderer_.ntp_time_ms());
}
TEST_F(VideoReceiveStreamTestWithFakeDecoder, PassesRotation) {
const webrtc::VideoRotation kRotation = webrtc::kVideoRotation_180;
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
auto test_frame = std::make_unique<FrameObjectFake>();
test_frame->SetPayloadType(99);
test_frame->SetId(0);
test_frame->SetRotation(kRotation);
video_receive_stream_->Start();
video_receive_stream_->OnCompleteFrame(std::move(test_frame));
EXPECT_TRUE(fake_renderer_.WaitForRenderedFrame(kDefaultTimeOutMs));
EXPECT_EQ(kRotation, fake_renderer_.rotation());
}
TEST_F(VideoReceiveStreamTestWithFakeDecoder, PassesPacketInfos) {
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
auto test_frame = std::make_unique<FrameObjectFake>();
test_frame->SetPayloadType(99);
test_frame->SetId(0);
RtpPacketInfos packet_infos = CreatePacketInfos(3);
test_frame->SetPacketInfos(packet_infos);
video_receive_stream_->Start();
video_receive_stream_->OnCompleteFrame(std::move(test_frame));
EXPECT_TRUE(fake_renderer_.WaitForRenderedFrame(kDefaultTimeOutMs));
EXPECT_THAT(fake_renderer_.packet_infos(), ElementsAreArray(packet_infos));
}
TEST_F(VideoReceiveStreamTestWithFakeDecoder, RenderedFrameUpdatesGetSources) {
constexpr uint32_t kSsrc = 1111;
constexpr uint32_t kCsrc = 9001;
constexpr uint32_t kRtpTimestamp = 12345;
// Prepare one video frame with per-packet information.
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
auto test_frame = std::make_unique<FrameObjectFake>();
test_frame->SetPayloadType(99);
test_frame->SetId(0);
RtpPacketInfos packet_infos;
{
RtpPacketInfos::vector_type infos;
RtpPacketInfo info;
info.set_ssrc(kSsrc);
info.set_csrcs({kCsrc});
info.set_rtp_timestamp(kRtpTimestamp);
info.set_receive_time(clock_->CurrentTime() - TimeDelta::Millis(5000));
infos.push_back(info);
info.set_receive_time(clock_->CurrentTime() - TimeDelta::Millis(3000));
infos.push_back(info);
info.set_receive_time(clock_->CurrentTime() - TimeDelta::Millis(2000));
infos.push_back(info);
info.set_receive_time(clock_->CurrentTime() - TimeDelta::Millis(4000));
infos.push_back(info);
packet_infos = RtpPacketInfos(std::move(infos));
}
test_frame->SetPacketInfos(packet_infos);
// Start receive stream.
video_receive_stream_->Start();
EXPECT_THAT(video_receive_stream_->GetSources(), IsEmpty());
// Render one video frame.
int64_t timestamp_ms_min = clock_->TimeInMilliseconds();
video_receive_stream_->OnCompleteFrame(std::move(test_frame));
EXPECT_TRUE(fake_renderer_.WaitForRenderedFrame(kDefaultTimeOutMs));
int64_t timestamp_ms_max = clock_->TimeInMilliseconds();
// Verify that the per-packet information is passed to the renderer.
EXPECT_THAT(fake_renderer_.packet_infos(), ElementsAreArray(packet_infos));
// Verify that the per-packet information also updates `GetSources()`.
std::vector<RtpSource> sources = video_receive_stream_->GetSources();
ASSERT_THAT(sources, SizeIs(2));
{
auto it = std::find_if(sources.begin(), sources.end(),
[](const RtpSource& source) {
return source.source_type() == RtpSourceType::SSRC;
});
ASSERT_NE(it, sources.end());
EXPECT_EQ(it->source_id(), kSsrc);
EXPECT_EQ(it->source_type(), RtpSourceType::SSRC);
EXPECT_EQ(it->rtp_timestamp(), kRtpTimestamp);
EXPECT_GE(it->timestamp_ms(), timestamp_ms_min);
EXPECT_LE(it->timestamp_ms(), timestamp_ms_max);
}
{
auto it = std::find_if(sources.begin(), sources.end(),
[](const RtpSource& source) {
return source.source_type() == RtpSourceType::CSRC;
});
ASSERT_NE(it, sources.end());
EXPECT_EQ(it->source_id(), kCsrc);
EXPECT_EQ(it->source_type(), RtpSourceType::CSRC);
EXPECT_EQ(it->rtp_timestamp(), kRtpTimestamp);
EXPECT_GE(it->timestamp_ms(), timestamp_ms_min);
EXPECT_LE(it->timestamp_ms(), timestamp_ms_max);
}
}
std::unique_ptr<FrameObjectFake> MakeFrame(VideoFrameType frame_type,
int picture_id) {
auto frame = std::make_unique<FrameObjectFake>();
frame->SetPayloadType(99);
frame->SetId(picture_id);
frame->SetFrameType(frame_type);
return frame;
}
TEST_F(VideoReceiveStreamTestWithFakeDecoder,
PassesFrameWhenEncodedFramesCallbackSet) {
testing::MockFunction<void(const RecordableEncodedFrame&)> callback;
video_receive_stream_->Start();
// Expect a keyframe request to be generated
EXPECT_CALL(mock_transport_, SendRtcp);
EXPECT_CALL(callback, Call);
video_receive_stream_->SetAndGetRecordingState(
VideoReceiveStream::RecordingState(callback.AsStdFunction()), true);
video_receive_stream_->OnCompleteFrame(
MakeFrame(VideoFrameType::kVideoFrameKey, 0));
EXPECT_TRUE(fake_renderer_.WaitForRenderedFrame(kDefaultTimeOutMs));
video_receive_stream_->Stop();
}
TEST_F(VideoReceiveStreamTestWithFakeDecoder,
MovesEncodedFrameDispatchStateWhenReCreating) {
testing::MockFunction<void(const RecordableEncodedFrame&)> callback;
video_receive_stream_->Start();
// Expect a key frame request over RTCP.
EXPECT_CALL(mock_transport_, SendRtcp).Times(1);
video_receive_stream_->SetAndGetRecordingState(
VideoReceiveStream::RecordingState(callback.AsStdFunction()), true);
video_receive_stream_->Stop();
VideoReceiveStream::RecordingState old_state =
video_receive_stream_->SetAndGetRecordingState(
VideoReceiveStream::RecordingState(), false);
ReCreateReceiveStream(std::move(old_state));
video_receive_stream_->Stop();
}
class VideoReceiveStreamTestWithSimulatedClock : public ::testing::Test {
public:
class FakeDecoder2 : public test::FakeDecoder {
public:
explicit FakeDecoder2(std::function<void()> decode_callback)
: callback_(decode_callback) {}
int32_t Decode(const EncodedImage& input,
bool missing_frames,
int64_t render_time_ms) override {
int32_t result =
FakeDecoder::Decode(input, missing_frames, render_time_ms);
callback_();
return result;
}
private:
std::function<void()> callback_;
};
static VideoReceiveStream::Config GetConfig(
Transport* transport,
VideoDecoderFactory* decoder_factory,
rtc::VideoSinkInterface<webrtc::VideoFrame>* renderer) {
VideoReceiveStream::Config config(transport);
config.rtp.remote_ssrc = 1111;
config.rtp.local_ssrc = 2222;
config.renderer = renderer;
Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory." This reverts commit a4f23ad0ce4382e3a11bc6a8c1f9f6183e722fd8. Reason for revert: Downstream fix landed. TBR=mflodman@webrtc.org Original change's description: > Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory." > > This reverts commit acb9d8365a5f9eb1e2a9e9902690d62dab1e5759. > > Reason for revert: Break downstream stuff. > > Original change's description: > > Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory. > > > > Bug: webrtc:9106 > > Change-Id: I85712f3ab6a734d3fad7819491d3b8e3388b47e7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180342 > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Magnus Flodman <mflodman@webrtc.org> > > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#31834} > > TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org > > Change-Id: I6cfdb85a154a78135839f84edf5f69673d5ab715 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9106 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180807 > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31835} TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org # Not skipping CQ checks because this is a reland. Bug: webrtc:9106 Change-Id: I03b3e68532107bec37bcc6e47a5489c84fe91ef9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180808 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31866}
2020-08-03 15:55:10 +00:00
config.decoder_factory = decoder_factory;
VideoReceiveStream::Decoder fake_decoder;
fake_decoder.payload_type = 99;
fake_decoder.video_format = SdpVideoFormat("VP8");
config.decoders.push_back(fake_decoder);
return config;
}
VideoReceiveStreamTestWithSimulatedClock()
Use newer version of TimeDelta and TimeStamp factories in webrtc find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 11:16:00 +01:00
: time_controller_(Timestamp::Millis(4711)),
fake_decoder_factory_([this] {
return std::make_unique<FakeDecoder2>([this] { OnFrameDecoded(); });
}),
process_thread_(time_controller_.CreateProcessThread("ProcessThread")),
config_(GetConfig(&mock_transport_,
&fake_decoder_factory_,
&fake_renderer_)),
call_stats_(time_controller_.GetClock(), process_thread_.get()),
video_receive_stream_(time_controller_.GetTaskQueueFactory(),
&rtp_stream_receiver_controller_,
/*num_cores=*/2,
&packet_router_,
config_.Copy(),
process_thread_.get(),
&call_stats_,
time_controller_.GetClock(),
new VCMTiming(time_controller_.GetClock())) {
video_receive_stream_.Start();
}
void OnFrameDecoded() { event_->Set(); }
void PassEncodedFrameAndWait(std::unique_ptr<EncodedFrame> frame) {
event_ = std::make_unique<rtc::Event>();
// This call will eventually end up in the Decoded method where the
// event is set.
video_receive_stream_.OnCompleteFrame(std::move(frame));
event_->Wait(rtc::Event::kForever);
}
protected:
GlobalSimulatedTimeController time_controller_;
test::FunctionVideoDecoderFactory fake_decoder_factory_;
std::unique_ptr<ProcessThread> process_thread_;
MockTransport mock_transport_;
cricket::FakeVideoRenderer fake_renderer_;
VideoReceiveStream::Config config_;
CallStats call_stats_;
PacketRouter packet_router_;
RtpStreamReceiverController rtp_stream_receiver_controller_;
webrtc::internal::VideoReceiveStream video_receive_stream_;
std::unique_ptr<rtc::Event> event_;
};
TEST_F(VideoReceiveStreamTestWithSimulatedClock,
RequestsKeyFramesUntilKeyFrameReceived) {
Use newer version of TimeDelta and TimeStamp factories in webrtc find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 11:16:00 +01:00
auto tick = TimeDelta::Millis(
internal::VideoReceiveStream::kMaxWaitForKeyFrameMs / 2);
EXPECT_CALL(mock_transport_, SendRtcp).Times(1);
video_receive_stream_.GenerateKeyFrame();
PassEncodedFrameAndWait(MakeFrame(VideoFrameType::kVideoFrameDelta, 0));
time_controller_.AdvanceTime(tick);
PassEncodedFrameAndWait(MakeFrame(VideoFrameType::kVideoFrameDelta, 1));
testing::Mock::VerifyAndClearExpectations(&mock_transport_);
// T+200ms: still no key frame received, expect key frame request sent again.
EXPECT_CALL(mock_transport_, SendRtcp).Times(1);
time_controller_.AdvanceTime(tick);
PassEncodedFrameAndWait(MakeFrame(VideoFrameType::kVideoFrameDelta, 2));
testing::Mock::VerifyAndClearExpectations(&mock_transport_);
// T+200ms: now send a key frame - we should not observe new key frame
// requests after this.
EXPECT_CALL(mock_transport_, SendRtcp).Times(0);
PassEncodedFrameAndWait(MakeFrame(VideoFrameType::kVideoFrameKey, 3));
time_controller_.AdvanceTime(2 * tick);
PassEncodedFrameAndWait(MakeFrame(VideoFrameType::kVideoFrameDelta, 4));
}
} // namespace webrtc