Reland of Adding audio to video_quality_test.
The original commit was https://codereview.webrtc.org/2136573002/. BUG= Review-Url: https://codereview.webrtc.org/2259783002 Cr-Commit-Position: refs/heads/master@{#13818}
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@ -223,19 +223,21 @@ void Loopback() {
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call_bitrate_config.start_bitrate_bps = flags::StartBitrateKbps() * 1000;
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call_bitrate_config.max_bitrate_bps = flags::MaxBitrateKbps() * 1000;
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VideoQualityTest::Params params{
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{flags::Width(), flags::Height(), flags::Fps(),
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flags::MinBitrateKbps() * 1000, flags::TargetBitrateKbps() * 1000,
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flags::MaxBitrateKbps() * 1000, false, flags::Codec(),
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flags::NumTemporalLayers(), flags::SelectedTL(),
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flags::MinTransmitBitrateKbps() * 1000, flags::FLAGS_send_side_bwe,
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false, call_bitrate_config},
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{}, // Video specific.
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{true, flags::SlideChangeInterval(), flags::ScrollDuration()},
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{"screenshare", 0.0, 0.0, flags::DurationSecs(), flags::OutputFilename(),
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flags::GraphTitle()},
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pipe_config,
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flags::FLAGS_logs};
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VideoQualityTest::Params params;
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params.common = {flags::Width(), flags::Height(), flags::Fps(),
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flags::MinBitrateKbps() * 1000, flags::TargetBitrateKbps() * 1000,
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flags::MaxBitrateKbps() * 1000, false, flags::Codec(),
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flags::NumTemporalLayers(), flags::SelectedTL(),
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flags::MinTransmitBitrateKbps() * 1000, flags::FLAGS_send_side_bwe,
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false, call_bitrate_config};
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params.screenshare = {true, flags::SlideChangeInterval(),
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flags::ScrollDuration()};
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params.analyzer = {"screenshare", 0.0, 0.0, flags::DurationSecs(),
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flags::OutputFilename(), flags::GraphTitle()};
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params.pipe = pipe_config;
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params.logs = flags::FLAGS_logs;
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params.audio = false;
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params.audio_video_sync = false;
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std::vector<std::string> stream_descriptors;
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stream_descriptors.push_back(flags::Stream0());
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@ -251,7 +253,7 @@ void Loopback() {
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if (flags::DurationSecs()) {
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test.RunWithAnalyzer(params);
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} else {
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test.RunWithVideoRenderer(params);
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test.RunWithRenderers(params);
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}
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}
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} // namespace webrtc
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@ -191,6 +191,11 @@ DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur");
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DEFINE_bool(use_fec, false, "Use forward error correction.");
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DEFINE_bool(audio, false, "Add audio stream");
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DEFINE_bool(audio_video_sync, false, "Sync audio and video stream (no effect if"
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" audio is false)");
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DEFINE_string(
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force_fieldtrials,
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"",
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@ -224,19 +229,20 @@ void Loopback() {
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call_bitrate_config.start_bitrate_bps = flags::StartBitrateKbps() * 1000;
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call_bitrate_config.max_bitrate_bps = flags::MaxBitrateKbps() * 1000;
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VideoQualityTest::Params params{
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{flags::Width(), flags::Height(), flags::Fps(),
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flags::MinBitrateKbps() * 1000, flags::TargetBitrateKbps() * 1000,
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flags::MaxBitrateKbps() * 1000, flags::FLAGS_suspend_below_min_bitrate,
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flags::Codec(), flags::NumTemporalLayers(), flags::SelectedTL(),
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0, // No min transmit bitrate.
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flags::FLAGS_send_side_bwe, flags::FLAGS_use_fec, call_bitrate_config},
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{flags::Clip()},
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{}, // Screenshare specific.
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{"video", 0.0, 0.0, flags::DurationSecs(), flags::OutputFilename(),
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flags::GraphTitle()},
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pipe_config,
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flags::FLAGS_logs};
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VideoQualityTest::Params params;
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params.common = {flags::Width(), flags::Height(), flags::Fps(),
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flags::MinBitrateKbps() * 1000, flags::TargetBitrateKbps() * 1000,
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flags::MaxBitrateKbps() * 1000, flags::FLAGS_suspend_below_min_bitrate,
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flags::Codec(), flags::NumTemporalLayers(), flags::SelectedTL(),
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0, // No min transmit bitrate.
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flags::FLAGS_send_side_bwe, flags::FLAGS_use_fec, call_bitrate_config};
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params.video = {flags::Clip()};
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params.analyzer = {"video", 0.0, 0.0, flags::DurationSecs(),
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flags::OutputFilename(), flags::GraphTitle()};
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params.pipe = pipe_config;
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params.logs = flags::FLAGS_logs;
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params.audio = flags::FLAGS_audio,
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params.audio_video_sync = flags::FLAGS_audio_video_sync;
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std::vector<std::string> stream_descriptors;
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stream_descriptors.push_back(flags::Stream0());
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@ -252,7 +258,7 @@ void Loopback() {
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if (flags::DurationSecs()) {
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test.RunWithAnalyzer(params);
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} else {
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test.RunWithVideoRenderer(params);
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test.RunWithRenderers(params);
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}
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}
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} // namespace webrtc
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@ -34,15 +34,68 @@
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#include "webrtc/test/testsupport/fileutils.h"
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#include "webrtc/test/video_renderer.h"
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#include "webrtc/video/video_quality_test.h"
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#include "webrtc/voice_engine/include/voe_base.h"
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#include "webrtc/voice_engine/include/voe_codec.h"
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namespace {
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constexpr int kSendStatsPollingIntervalMs = 1000;
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constexpr int kPayloadTypeH264 = 122;
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constexpr int kPayloadTypeVP8 = 123;
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constexpr int kPayloadTypeVP9 = 124;
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constexpr size_t kMaxComparisons = 10;
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constexpr char kSyncGroup[] = "av_sync";
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constexpr int kOpusMinBitrate = 6000;
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constexpr int kOpusBitrateFb = 32000;
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struct VoiceEngineState {
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VoiceEngineState()
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: voice_engine(nullptr),
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base(nullptr),
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codec(nullptr),
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send_channel_id(-1),
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receive_channel_id(-1) {}
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webrtc::VoiceEngine* voice_engine;
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webrtc::VoEBase* base;
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webrtc::VoECodec* codec;
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int send_channel_id;
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int receive_channel_id;
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};
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void CreateVoiceEngine(VoiceEngineState* voe,
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rtc::scoped_refptr<webrtc::AudioDecoderFactory>
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decoder_factory) {
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voe->voice_engine = webrtc::VoiceEngine::Create();
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voe->base = webrtc::VoEBase::GetInterface(voe->voice_engine);
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voe->codec = webrtc::VoECodec::GetInterface(voe->voice_engine);
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EXPECT_EQ(0, voe->base->Init(nullptr, nullptr, decoder_factory));
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webrtc::Config voe_config;
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voe_config.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
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voe->send_channel_id = voe->base->CreateChannel(voe_config);
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EXPECT_GE(voe->send_channel_id, 0);
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voe->receive_channel_id = voe->base->CreateChannel();
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EXPECT_GE(voe->receive_channel_id, 0);
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}
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void DestroyVoiceEngine(VoiceEngineState* voe) {
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voe->base->DeleteChannel(voe->send_channel_id);
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voe->send_channel_id = -1;
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voe->base->DeleteChannel(voe->receive_channel_id);
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voe->receive_channel_id = -1;
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voe->base->Release();
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voe->base = nullptr;
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voe->codec->Release();
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voe->codec = nullptr;
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webrtc::VoiceEngine::Delete(voe->voice_engine);
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voe->voice_engine = nullptr;
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}
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} // namespace
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namespace webrtc {
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static const int kSendStatsPollingIntervalMs = 1000;
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static const int kPayloadTypeH264 = 122;
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static const int kPayloadTypeVP8 = 123;
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static const int kPayloadTypeVP9 = 124;
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static const size_t kMaxComparisons = 10;
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class VideoAnalyzer : public PacketReceiver,
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public Transport,
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public rtc::VideoSinkInterface<VideoFrame>,
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@ -1002,6 +1055,7 @@ void VideoQualityTest::CreateCapturer(VideoCaptureInput* input) {
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void VideoQualityTest::RunWithAnalyzer(const Params& params) {
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params_ = params;
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RTC_CHECK(!params_.audio);
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// TODO(ivica): Merge with RunWithRenderer and use a flag / argument to
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// differentiate between the analyzer and the renderer case.
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CheckParams();
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@ -1099,7 +1153,7 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
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fclose(graph_data_output_file);
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}
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void VideoQualityTest::RunWithVideoRenderer(const Params& params) {
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void VideoQualityTest::RunWithRenderers(const Params& params) {
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params_ = params;
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CheckParams();
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@ -1123,6 +1177,15 @@ void VideoQualityTest::RunWithVideoRenderer(const Params& params) {
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// match the full stack tests.
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Call::Config call_config;
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call_config.bitrate_config = params_.common.call_bitrate_config;
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::VoiceEngineState voe;
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if (params_.audio) {
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CreateVoiceEngine(&voe, decoder_factory_);
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AudioState::Config audio_state_config;
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audio_state_config.voice_engine = voe.voice_engine;
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call_config.audio_state = AudioState::Create(audio_state_config);
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}
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std::unique_ptr<Call> call(Call::Create(call_config));
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test::LayerFilteringTransport transport(
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@ -1137,6 +1200,8 @@ void VideoQualityTest::RunWithVideoRenderer(const Params& params) {
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video_send_config_.local_renderer = local_preview.get();
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video_receive_configs_[stream_id].renderer = loopback_video.get();
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if (params_.audio && params_.audio_video_sync)
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video_receive_configs_[stream_id].sync_group = kSyncGroup;
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video_send_config_.suspend_below_min_bitrate =
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params_.common.suspend_below_min_bitrate;
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@ -1155,24 +1220,91 @@ void VideoQualityTest::RunWithVideoRenderer(const Params& params) {
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video_send_stream_ =
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call->CreateVideoSendStream(video_send_config_, video_encoder_config_);
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VideoReceiveStream* receive_stream =
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VideoReceiveStream* video_receive_stream =
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call->CreateVideoReceiveStream(video_receive_configs_[stream_id].Copy());
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CreateCapturer(video_send_stream_->Input());
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receive_stream->Start();
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AudioReceiveStream* audio_receive_stream = nullptr;
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if (params_.audio) {
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audio_send_config_ = AudioSendStream::Config(&transport);
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audio_send_config_.voe_channel_id = voe.send_channel_id;
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audio_send_config_.rtp.ssrc = kAudioSendSsrc;
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// Add extension to enable audio send side BWE, and allow audio bit rate
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// adaptation.
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audio_send_config_.rtp.extensions.clear();
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if (params_.common.send_side_bwe) {
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audio_send_config_.rtp.extensions.push_back(webrtc::RtpExtension(
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webrtc::RtpExtension::kTransportSequenceNumberUri,
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test::kTransportSequenceNumberExtensionId));
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audio_send_config_.min_bitrate_kbps = kOpusMinBitrate / 1000;
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audio_send_config_.max_bitrate_kbps = kOpusBitrateFb / 1000;
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}
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audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_);
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AudioReceiveStream::Config audio_config;
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audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
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audio_config.rtcp_send_transport = &transport;
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audio_config.voe_channel_id = voe.receive_channel_id;
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audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
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audio_config.rtp.transport_cc = params_.common.send_side_bwe;
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audio_config.rtp.extensions = audio_send_config_.rtp.extensions;
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audio_config.decoder_factory = decoder_factory_;
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if (params_.audio_video_sync)
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audio_config.sync_group = kSyncGroup;
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audio_receive_stream =call->CreateAudioReceiveStream(audio_config);
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const CodecInst kOpusInst = {120, "OPUS", 48000, 960, 2, 64000};
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EXPECT_EQ(0, voe.codec->SetSendCodec(voe.send_channel_id, kOpusInst));
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}
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// Start sending and receiving video.
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video_receive_stream->Start();
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video_send_stream_->Start();
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capturer_->Start();
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if (params_.audio) {
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// Start receiving audio.
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audio_receive_stream->Start();
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EXPECT_EQ(0, voe.base->StartPlayout(voe.receive_channel_id));
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EXPECT_EQ(0, voe.base->StartReceive(voe.receive_channel_id));
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// Start sending audio.
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audio_send_stream_->Start();
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EXPECT_EQ(0, voe.base->StartSend(voe.send_channel_id));
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}
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test::PressEnterToContinue();
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if (params_.audio) {
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// Stop sending audio.
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EXPECT_EQ(0, voe.base->StopSend(voe.send_channel_id));
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audio_send_stream_->Stop();
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// Stop receiving audio.
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EXPECT_EQ(0, voe.base->StopReceive(voe.receive_channel_id));
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EXPECT_EQ(0, voe.base->StopPlayout(voe.receive_channel_id));
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audio_receive_stream->Stop();
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}
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// Stop receiving and sending video.
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capturer_->Stop();
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video_send_stream_->Stop();
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receive_stream->Stop();
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video_receive_stream->Stop();
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call->DestroyVideoReceiveStream(receive_stream);
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call->DestroyVideoReceiveStream(video_receive_stream);
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call->DestroyVideoSendStream(video_send_stream_);
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if (params_.audio) {
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call->DestroyAudioSendStream(audio_send_stream_);
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call->DestroyAudioReceiveStream(audio_receive_stream);
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}
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transport.StopSending();
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if (params_.audio)
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DestroyVoiceEngine(&voe);
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}
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} // namespace webrtc
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@ -62,7 +62,7 @@ class VideoQualityTest : public test::CallTest {
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} analyzer;
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FakeNetworkPipe::Config pipe;
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bool logs;
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struct { // Spatial scalability.
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struct { // Spatial scalability.
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std::vector<VideoStream> streams; // If empty, one stream is assumed.
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size_t selected_stream;
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int num_spatial_layers;
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@ -70,6 +70,8 @@ class VideoQualityTest : public test::CallTest {
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// If empty, bitrates are generated in VP9Impl automatically.
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std::vector<SpatialLayer> spatial_layers;
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} ss;
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bool audio;
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bool audio_video_sync;
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};
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// (*) Set to -1.1 if generating graph data for simulcast or SVC and the
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// selected stream/layer doesn't have the same resolution as the largest
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@ -77,7 +79,7 @@ class VideoQualityTest : public test::CallTest {
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VideoQualityTest();
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void RunWithAnalyzer(const Params& params);
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void RunWithVideoRenderer(const Params& params);
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void RunWithRenderers(const Params& params);
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static void FillScalabilitySettings(
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Params* params,
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