5ef2bc1914
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:a518a39963> > 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:b80bdcafedTBR=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}
102 lines
3.7 KiB
C++
102 lines
3.7 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/test/encoder_settings.h"
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#include <algorithm>
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#include <string>
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#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
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#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
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#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
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#include "webrtc/test/fake_decoder.h"
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namespace webrtc {
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namespace test {
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const size_t DefaultVideoStreamFactory::kMaxNumberOfStreams;
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const int DefaultVideoStreamFactory::kMaxBitratePerStream[] = {150000, 450000,
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1500000};
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const int DefaultVideoStreamFactory::kDefaultMinBitratePerStream[] = {
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30000, 200000, 700000};
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// static
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std::vector<VideoStream> CreateVideoStreams(
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int width,
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int height,
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const webrtc::VideoEncoderConfig& encoder_config) {
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RTC_DCHECK(encoder_config.number_of_streams <=
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DefaultVideoStreamFactory::kMaxNumberOfStreams);
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std::vector<VideoStream> stream_settings(encoder_config.number_of_streams);
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int bitrate_left_bps = encoder_config.max_bitrate_bps;
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for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
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stream_settings[i].width =
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(i + 1) * width / encoder_config.number_of_streams;
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stream_settings[i].height =
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(i + 1) * height / encoder_config.number_of_streams;
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stream_settings[i].max_framerate = 30;
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stream_settings[i].min_bitrate_bps =
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DefaultVideoStreamFactory::kDefaultMinBitratePerStream[i];
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stream_settings[i].target_bitrate_bps = stream_settings[i].max_bitrate_bps =
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std::min(bitrate_left_bps,
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DefaultVideoStreamFactory::kMaxBitratePerStream[i]);
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stream_settings[i].max_qp = 56;
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bitrate_left_bps -= stream_settings[i].target_bitrate_bps;
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}
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stream_settings[encoder_config.number_of_streams - 1].max_bitrate_bps +=
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bitrate_left_bps;
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return stream_settings;
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}
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DefaultVideoStreamFactory::DefaultVideoStreamFactory() {}
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std::vector<VideoStream> DefaultVideoStreamFactory::CreateEncoderStreams(
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int width,
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int height,
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const webrtc::VideoEncoderConfig& encoder_config) {
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return CreateVideoStreams(width, height, encoder_config);
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}
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void FillEncoderConfiguration(size_t num_streams,
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VideoEncoderConfig* configuration) {
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RTC_DCHECK_LE(num_streams, DefaultVideoStreamFactory::kMaxNumberOfStreams);
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configuration->number_of_streams = num_streams;
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configuration->video_stream_factory =
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new rtc::RefCountedObject<DefaultVideoStreamFactory>();
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configuration->max_bitrate_bps = 0;
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for (size_t i = 0; i < num_streams; ++i) {
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configuration->max_bitrate_bps +=
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DefaultVideoStreamFactory::kMaxBitratePerStream[i];
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}
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}
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VideoReceiveStream::Decoder CreateMatchingDecoder(
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const VideoSendStream::Config::EncoderSettings& encoder_settings) {
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VideoReceiveStream::Decoder decoder;
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decoder.payload_type = encoder_settings.payload_type;
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decoder.payload_name = encoder_settings.payload_name;
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if (encoder_settings.payload_name == "H264") {
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decoder.decoder = H264Decoder::Create();
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} else if (encoder_settings.payload_name == "VP8") {
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decoder.decoder = VP8Decoder::Create();
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} else if (encoder_settings.payload_name == "VP9") {
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decoder.decoder = VP9Decoder::Create();
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} else {
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decoder.decoder = new FakeDecoder();
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}
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return decoder;
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}
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} // namespace test
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} // namespace webrtc
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