webrtc_m130/test/encoder_settings.cc

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/*
* Copyright (c) 2014 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 "test/encoder_settings.h"
#include <algorithm>
#include "api/scoped_refptr.h"
#include "api/video_codecs/sdp_video_format.h"
#include "call/rtp_config.h"
#include "rtc_base/checks.h"
#include "rtc_base/ref_counted_object.h"
namespace webrtc {
namespace test {
const size_t DefaultVideoStreamFactory::kMaxNumberOfStreams;
const int DefaultVideoStreamFactory::kMaxBitratePerStream[] = {150000, 450000,
1500000};
const int DefaultVideoStreamFactory::kDefaultMinBitratePerStream[] = {
Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ ) Reason for revert: Downstream fixed Original issue's description: > Revert of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #8 id:120001 of https://codereview.webrtc.org/2705603002/ ) > > Reason for revert: > Breaks downstream > > Original issue's description: > > Fixes a bug where a video stream can get stuck in the suspended state. > > > > This happens if a lot of FEC is allocated when the stream becomes suspended. The required bitrate to unsuspend can then be too high so that the padding bitrate we are allowed to generate is not enough. > > > > This CL also switches the tests from using ISAC to OPUS as RampUpTest.UpDownUpAudioVideoTransportSequenceNumberRtx relies on audio BWE to work (which is only compatible with OPUS). I don't know why it didn't fail before. > > > > BUG=webrtc:7178 > > > > Review-Url: https://codereview.webrtc.org/2705603002 > > Cr-Commit-Position: refs/heads/master@{#16739} > > Committed: https://chromium.googlesource.com/external/webrtc/+/a518a39963d34616d8f0e94991c7f5fbb5affb38 > > TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7178 > > Review-Url: https://codereview.webrtc.org/2703393002 > Cr-Commit-Position: refs/heads/master@{#16751} > Committed: https://chromium.googlesource.com/external/webrtc/+/b80bdcafed6c529be140da7d9f3e95a00b94219e TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7178 Review-Url: https://codereview.webrtc.org/2704323003 Cr-Commit-Position: refs/heads/master@{#16753}
2017-02-21 07:28:31 -08:00
30000, 200000, 700000};
// static
std::vector<VideoStream> CreateVideoStreams(
int width,
int height,
const webrtc::VideoEncoderConfig& encoder_config) {
RTC_DCHECK(encoder_config.number_of_streams <=
DefaultVideoStreamFactory::kMaxNumberOfStreams);
std::vector<VideoStream> stream_settings(encoder_config.number_of_streams);
int bitrate_left_bps = 0;
if (encoder_config.max_bitrate_bps > 0) {
bitrate_left_bps = encoder_config.max_bitrate_bps;
} else {
for (size_t stream_num = 0; stream_num < encoder_config.number_of_streams;
++stream_num) {
bitrate_left_bps +=
DefaultVideoStreamFactory::kMaxBitratePerStream[stream_num];
}
}
for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
stream_settings[i].width =
(i + 1) * width / encoder_config.number_of_streams;
stream_settings[i].height =
(i + 1) * height / encoder_config.number_of_streams;
stream_settings[i].max_framerate = 30;
stream_settings[i].max_qp = 56;
stream_settings[i].min_bitrate_bps =
DefaultVideoStreamFactory::kDefaultMinBitratePerStream[i];
// Use configured values instead of default values if set.
const VideoStream stream = (i < encoder_config.simulcast_layers.size())
? encoder_config.simulcast_layers[i]
: VideoStream();
int max_bitrate_bps =
stream.max_bitrate_bps > 0
? stream.max_bitrate_bps
: DefaultVideoStreamFactory::kMaxBitratePerStream[i];
max_bitrate_bps = std::min(bitrate_left_bps, max_bitrate_bps);
int target_bitrate_bps = stream.target_bitrate_bps > 0
? stream.target_bitrate_bps
: max_bitrate_bps;
target_bitrate_bps = std::min(max_bitrate_bps, target_bitrate_bps);
if (stream.min_bitrate_bps > 0) {
RTC_DCHECK_LE(stream.min_bitrate_bps, target_bitrate_bps);
stream_settings[i].min_bitrate_bps = stream.min_bitrate_bps;
}
if (stream.max_framerate > 0) {
stream_settings[i].max_framerate = stream.max_framerate;
}
if (stream.num_temporal_layers) {
RTC_DCHECK_GE(*stream.num_temporal_layers, 1);
stream_settings[i].num_temporal_layers = stream.num_temporal_layers;
}
if (stream.scale_resolution_down_by >= 1.0) {
stream_settings[i].width = width / stream.scale_resolution_down_by;
stream_settings[i].height = height / stream.scale_resolution_down_by;
}
stream_settings[i].scalability_mode = stream.scalability_mode;
stream_settings[i].target_bitrate_bps = target_bitrate_bps;
stream_settings[i].max_bitrate_bps = max_bitrate_bps;
stream_settings[i].active =
encoder_config.number_of_streams == 1 || stream.active;
bitrate_left_bps -= stream_settings[i].target_bitrate_bps;
}
stream_settings[encoder_config.number_of_streams - 1].max_bitrate_bps +=
bitrate_left_bps;
stream_settings[0].bitrate_priority = encoder_config.bitrate_priority;
return stream_settings;
}
DefaultVideoStreamFactory::DefaultVideoStreamFactory() {}
std::vector<VideoStream> DefaultVideoStreamFactory::CreateEncoderStreams(
int width,
int height,
const webrtc::VideoEncoderConfig& encoder_config) {
return CreateVideoStreams(width, height, encoder_config);
}
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
void FillEncoderConfiguration(VideoCodecType codec_type,
size_t num_streams,
VideoEncoderConfig* configuration) {
RTC_DCHECK_LE(num_streams, DefaultVideoStreamFactory::kMaxNumberOfStreams);
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
configuration->codec_type = codec_type;
configuration->number_of_streams = num_streams;
configuration->video_stream_factory =
rtc::make_ref_counted<DefaultVideoStreamFactory>();
configuration->max_bitrate_bps = 0;
configuration->simulcast_layers = std::vector<VideoStream>(num_streams);
for (size_t i = 0; i < num_streams; ++i) {
configuration->max_bitrate_bps +=
DefaultVideoStreamFactory::kMaxBitratePerStream[i];
}
}
VideoReceiveStream::Decoder CreateMatchingDecoder(
int payload_type,
const std::string& payload_name) {
VideoReceiveStream::Decoder decoder;
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
decoder.payload_type = payload_type;
decoder.video_format = SdpVideoFormat(payload_name);
return decoder;
}
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
VideoReceiveStream::Decoder CreateMatchingDecoder(
const VideoSendStream::Config& config) {
return CreateMatchingDecoder(config.rtp.payload_type,
config.rtp.payload_name);
}
} // namespace test
} // namespace webrtc