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Fix echo return loss stats and add to RTCAudioSourceStats.

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This solves two problems:
* Echo return loss stats weren't being gathered in Chrome, because they
  need to be taken from the audio processor attached to the track
  rather than the audio send stream.
* The standardized location is in RTCAudioSourceStats, not
  RTCMediaStreamTrackStats. For now, will populate the stats in both
  locations.

Bug: webrtc:12770
Change-Id: I47eaf7f2b50b914a1be84156aa831e27497d07e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223182
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34344}
This commit is contained in:
Taylor Brandstetter
2021-06-20 14:59:48 -07:00
committed by WebRTC LUCI CQ
parent 2e3edc1da9
commit a27cfbffdf
5 changed files with 148 additions and 15 deletions
Download Patch File Download Diff File Expand all files Collapse all files

96
pc/rtc_stats_collector_unittest.cc
View File

@ -200,14 +200,34 @@ std::unique_ptr<cricket::Candidate> CreateFakeCandidate(
return candidate;
}
class FakeAudioProcessor : public AudioProcessorInterface {
public:
FakeAudioProcessor() {}
~FakeAudioProcessor() {}
private:
AudioProcessorInterface::AudioProcessorStatistics GetStats(
bool has_recv_streams) override {
AudioProcessorStatistics stats;
stats.apm_statistics.echo_return_loss = 2.0;
stats.apm_statistics.echo_return_loss_enhancement = 3.0;
return stats;
}
};
class FakeAudioTrackForStats : public MediaStreamTrack<AudioTrackInterface> {
public:
static rtc::scoped_refptr<FakeAudioTrackForStats> Create(
const std::string& id,
MediaStreamTrackInterface::TrackState state) {
MediaStreamTrackInterface::TrackState state,
bool create_fake_audio_processor) {
rtc::scoped_refptr<FakeAudioTrackForStats> audio_track_stats(
new rtc::RefCountedObject<FakeAudioTrackForStats>(id));
audio_track_stats->set_state(state);
if (create_fake_audio_processor) {
audio_track_stats->processor_ =
rtc::make_ref_counted<FakeAudioProcessor>();
}
return audio_track_stats;
}
@ -222,8 +242,11 @@ class FakeAudioTrackForStats : public MediaStreamTrack<AudioTrackInterface> {
void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override {}
bool GetSignalLevel(int* level) override { return false; }
rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor() override {
return nullptr;
return processor_;
}
private:
rtc::scoped_refptr<FakeAudioProcessor> processor_;
};
class FakeVideoTrackSourceForStats : public VideoTrackSourceInterface {
@ -308,9 +331,11 @@ class FakeVideoTrackForStats : public MediaStreamTrack<VideoTrackInterface> {
rtc::scoped_refptr<MediaStreamTrackInterface> CreateFakeTrack(
cricket::MediaType media_type,
const std::string& track_id,
MediaStreamTrackInterface::TrackState track_state) {
MediaStreamTrackInterface::TrackState track_state,
bool create_fake_audio_processor = false) {
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
return FakeAudioTrackForStats::Create(track_id, track_state);
return FakeAudioTrackForStats::Create(track_id, track_state,
create_fake_audio_processor);
} else {
RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
return FakeVideoTrackForStats::Create(track_id, track_state, nullptr);
@ -2580,6 +2605,9 @@ TEST_F(RTCStatsCollectorTest, RTCAudioSourceStatsCollectedForSenderWithTrack) {
voice_media_info.senders[0].audio_level = 32767; // [0,32767]
voice_media_info.senders[0].total_input_energy = 2.0;
voice_media_info.senders[0].total_input_duration = 3.0;
voice_media_info.senders[0].apm_statistics.echo_return_loss = 42.0;
voice_media_info.senders[0].apm_statistics.echo_return_loss_enhancement =
52.0;
auto* voice_media_channel = pc_->AddVoiceChannel("AudioMid", "TransportName");
voice_media_channel->SetStats(voice_media_info);
stats_->SetupLocalTrackAndSender(cricket::MEDIA_TYPE_AUDIO,
@ -2595,6 +2623,8 @@ TEST_F(RTCStatsCollectorTest, RTCAudioSourceStatsCollectedForSenderWithTrack) {
expected_audio.audio_level = 1.0; // [0,1]
expected_audio.total_audio_energy = 2.0;
expected_audio.total_samples_duration = 3.0;
expected_audio.echo_return_loss = 42.0;
expected_audio.echo_return_loss_enhancement = 52.0;
ASSERT_TRUE(report->Get(expected_audio.id()));
EXPECT_EQ(report->Get(expected_audio.id())->cast_to<RTCAudioSourceStats>(),
@ -3056,6 +3086,64 @@ TEST_F(RTCStatsCollectorTest,
EXPECT_FALSE(report->Get("RTCVideoSource_42"));
}
// Test collecting echo return loss stats from the audio processor attached to
// the track, rather than the voice sender info.
TEST_F(RTCStatsCollectorTest, CollectEchoReturnLossFromTrackAudioProcessor) {
rtc::scoped_refptr<MediaStream> local_stream =
MediaStream::Create("LocalStreamId");
pc_->mutable_local_streams()->AddStream(local_stream);
// Local audio track
rtc::scoped_refptr<MediaStreamTrackInterface> local_audio_track =
CreateFakeTrack(cricket::MEDIA_TYPE_AUDIO, "LocalAudioTrackID",
MediaStreamTrackInterface::kEnded,
/*create_fake_audio_processor=*/true);
local_stream->AddTrack(
static_cast<AudioTrackInterface*>(local_audio_track.get()));
cricket::VoiceSenderInfo voice_sender_info_ssrc1;
voice_sender_info_ssrc1.local_stats.push_back(cricket::SsrcSenderInfo());
voice_sender_info_ssrc1.local_stats[0].ssrc = 1;
stats_->CreateMockRtpSendersReceiversAndChannels(
{std::make_pair(local_audio_track.get(), voice_sender_info_ssrc1)}, {},
{}, {}, {local_stream->id()}, {});
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
RTCMediaStreamTrackStats expected_local_audio_track_ssrc1(
IdForType<RTCMediaStreamTrackStats>(report), report->timestamp_us(),
RTCMediaStreamTrackKind::kAudio);
expected_local_audio_track_ssrc1.track_identifier = local_audio_track->id();
expected_local_audio_track_ssrc1.media_source_id =
"RTCAudioSource_11"; // Attachment ID = SSRC + 10
expected_local_audio_track_ssrc1.remote_source = false;
expected_local_audio_track_ssrc1.ended = true;
expected_local_audio_track_ssrc1.detached = false;
expected_local_audio_track_ssrc1.echo_return_loss = 2.0;
expected_local_audio_track_ssrc1.echo_return_loss_enhancement = 3.0;
ASSERT_TRUE(report->Get(expected_local_audio_track_ssrc1.id()))
<< "Did not find " << expected_local_audio_track_ssrc1.id() << " in "
<< report->ToJson();
EXPECT_EQ(expected_local_audio_track_ssrc1,
report->Get(expected_local_audio_track_ssrc1.id())
->cast_to<RTCMediaStreamTrackStats>());
RTCAudioSourceStats expected_audio("RTCAudioSource_11",
report->timestamp_us());
expected_audio.track_identifier = "LocalAudioTrackID";
expected_audio.kind = "audio";
expected_audio.audio_level = 0;
expected_audio.total_audio_energy = 0;
expected_audio.total_samples_duration = 0;
expected_audio.echo_return_loss = 2.0;
expected_audio.echo_return_loss_enhancement = 3.0;
ASSERT_TRUE(report->Get(expected_audio.id()));
EXPECT_EQ(report->Get(expected_audio.id())->cast_to<RTCAudioSourceStats>(),
expected_audio);
}
TEST_F(RTCStatsCollectorTest, GetStatsWithSenderSelector) {
ExampleStatsGraph graph = SetupExampleStatsGraphForSelectorTests();
// Expected stats graph when filtered by sender: