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Revert "Fix echo return loss stats and add to RTCAudioSourceStats."

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This reverts commit a27cfbffdfa0bf359628d2164db5b9d6321f9c9c.

Reason for revert: WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsGetStatsPromise failing.

Original change's description:
> Fix echo return loss stats and add to RTCAudioSourceStats.
>
> 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}

TBR=deadbeef@webrtc.org,hbos@webrtc.org,hbos@chromium.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: I6b2587d762f005adef67c0d5121f1b58c3b76688
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12770
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223068
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#34352}
This commit is contained in:
Evan Shrubsole
2021-06-22 07:58:41 +00:00
committed by WebRTC LUCI CQ
parent 9e2b3155ee
commit fe6580fb87
5 changed files with 15 additions and 148 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
api/stats/rtcstats_objects.h
View File

@ -629,8 +629,6 @@ class RTC_EXPORT RTCAudioSourceStats final : public RTCMediaSourceStats {
RTCStatsMember<double> audio_level;
RTCStatsMember<double> total_audio_energy;
RTCStatsMember<double> total_samples_duration;
RTCStatsMember<double> echo_return_loss;
RTCStatsMember<double> echo_return_loss_enhancement;
};
// https://w3c.github.io/webrtc-stats/#dom-rtcvideosourcestats

47
pc/rtc_stats_collector.cc
View File

@ -743,22 +743,10 @@ const std::string& ProduceIceCandidateStats(int64_t timestamp_us,
return stats->id();
}
template <typename StatsType>
void SetAudioProcessingStats(StatsType* stats,
const AudioProcessingStats& apm_stats) {
if (apm_stats.echo_return_loss) {
stats->echo_return_loss = *apm_stats.echo_return_loss;
}
if (apm_stats.echo_return_loss_enhancement) {
stats->echo_return_loss_enhancement =
*apm_stats.echo_return_loss_enhancement;
}
}
std::unique_ptr<RTCMediaStreamTrackStats>
ProduceMediaStreamTrackStatsFromVoiceSenderInfo(
int64_t timestamp_us,
AudioTrackInterface& audio_track,
const AudioTrackInterface& audio_track,
const cricket::VoiceSenderInfo& voice_sender_info,
int attachment_id) {
std::unique_ptr<RTCMediaStreamTrackStats> audio_track_stats(
@ -773,17 +761,13 @@ ProduceMediaStreamTrackStatsFromVoiceSenderInfo(
attachment_id);
audio_track_stats->remote_source = false;
audio_track_stats->detached = false;
// Audio processor may be attached to either the track or the send
// stream, so look in both places.
SetAudioProcessingStats(audio_track_stats.get(),
voice_sender_info.apm_statistics);
auto audio_processor(audio_track.GetAudioProcessor());
if (audio_processor.get()) {
// The |has_remote_tracks| argument is obsolete; makes no difference if it's
// set to true or false.
AudioProcessorInterface::AudioProcessorStatistics ap_stats =
audio_processor->GetStats(/*has_remote_tracks=*/false);
SetAudioProcessingStats(audio_track_stats.get(), ap_stats.apm_statistics);
if (voice_sender_info.apm_statistics.echo_return_loss) {
audio_track_stats->echo_return_loss =
*voice_sender_info.apm_statistics.echo_return_loss;
}
if (voice_sender_info.apm_statistics.echo_return_loss_enhancement) {
audio_track_stats->echo_return_loss_enhancement =
*voice_sender_info.apm_statistics.echo_return_loss_enhancement;
}
return audio_track_stats;
}
@ -1673,8 +1657,6 @@ void RTCStatsCollector::ProduceMediaSourceStats_s(
// create separate media source stats objects on a per-attachment basis.
std::unique_ptr<RTCMediaSourceStats> media_source_stats;
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
AudioTrackInterface* audio_track =
static_cast<AudioTrackInterface*>(track.get());
auto audio_source_stats = std::make_unique<RTCAudioSourceStats>(
RTCMediaSourceStatsIDFromKindAndAttachment(
cricket::MEDIA_TYPE_AUDIO, sender_internal->AttachmentId()),
@ -1695,21 +1677,8 @@ void RTCStatsCollector::ProduceMediaSourceStats_s(
voice_sender_info->total_input_energy;
audio_source_stats->total_samples_duration =
voice_sender_info->total_input_duration;
SetAudioProcessingStats(audio_source_stats.get(),
voice_sender_info->apm_statistics);
}
}
// Audio processor may be attached to either the track or the send
// stream, so look in both places.
auto audio_processor(audio_track->GetAudioProcessor());
if (audio_processor.get()) {
// The |has_remote_tracks| argument is obsolete; makes no difference
// if it's set to true or false.
AudioProcessorInterface::AudioProcessorStatistics ap_stats =
audio_processor->GetStats(/*has_remote_tracks=*/false);
SetAudioProcessingStats(audio_source_stats.get(),
ap_stats.apm_statistics);
}
media_source_stats = std::move(audio_source_stats);
} else {
RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());

96
pc/rtc_stats_collector_unittest.cc
View File

@ -200,34 +200,14 @@ 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,
bool create_fake_audio_processor) {
MediaStreamTrackInterface::TrackState state) {
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;
}
@ -242,11 +222,8 @@ 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 processor_;
return nullptr;
}
private:
rtc::scoped_refptr<FakeAudioProcessor> processor_;
};
class FakeVideoTrackSourceForStats : public VideoTrackSourceInterface {
@ -331,11 +308,9 @@ class FakeVideoTrackForStats : public MediaStreamTrack<VideoTrackInterface> {
rtc::scoped_refptr<MediaStreamTrackInterface> CreateFakeTrack(
cricket::MediaType media_type,
const std::string& track_id,
MediaStreamTrackInterface::TrackState track_state,
bool create_fake_audio_processor = false) {
MediaStreamTrackInterface::TrackState track_state) {
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
return FakeAudioTrackForStats::Create(track_id, track_state,
create_fake_audio_processor);
return FakeAudioTrackForStats::Create(track_id, track_state);
} else {
RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
return FakeVideoTrackForStats::Create(track_id, track_state, nullptr);
@ -2605,9 +2580,6 @@ 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,
@ -2623,8 +2595,6 @@ 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>(),
@ -3086,64 +3056,6 @@ 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:

6
pc/rtc_stats_integrationtest.cc
View File

@ -1082,12 +1082,6 @@ class RTCStatsReportVerifier {
verifier.TestMemberIsNonNegative<double>(audio_source.audio_level);
verifier.TestMemberIsPositive<double>(audio_source.total_audio_energy);
verifier.TestMemberIsPositive<double>(audio_source.total_samples_duration);
// TODO(hbos): |echo_return_loss| and |echo_return_loss_enhancement| are
// flaky on msan bot (sometimes defined, sometimes undefined). Should the
// test run until available or is there a way to have it always be
// defined? crbug.com/627816
verifier.MarkMemberTested(audio_source.echo_return_loss, true);
verifier.MarkMemberTested(audio_source.echo_return_loss_enhancement, true);
return verifier.ExpectAllMembersSuccessfullyTested();
}

12
stats/rtcstats_objects.cc
View File

@ -987,9 +987,7 @@ RTCMediaSourceStats::~RTCMediaSourceStats() {}
WEBRTC_RTCSTATS_IMPL(RTCAudioSourceStats, RTCMediaSourceStats, "media-source",
&audio_level,
&total_audio_energy,
&total_samples_duration,
&echo_return_loss,
&echo_return_loss_enhancement)
&total_samples_duration)
// clang-format on
RTCAudioSourceStats::RTCAudioSourceStats(const std::string& id,
@ -1000,17 +998,13 @@ RTCAudioSourceStats::RTCAudioSourceStats(std::string&& id, int64_t timestamp_us)
: RTCMediaSourceStats(std::move(id), timestamp_us),
audio_level("audioLevel"),
total_audio_energy("totalAudioEnergy"),
total_samples_duration("totalSamplesDuration"),
echo_return_loss("echoReturnLoss"),
echo_return_loss_enhancement("echoReturnLossEnhancement") {}
total_samples_duration("totalSamplesDuration") {}
RTCAudioSourceStats::RTCAudioSourceStats(const RTCAudioSourceStats& other)
: RTCMediaSourceStats(other),
audio_level(other.audio_level),
total_audio_energy(other.total_audio_energy),
total_samples_duration(other.total_samples_duration),
echo_return_loss(other.echo_return_loss),
echo_return_loss_enhancement(other.echo_return_loss_enhancement) {}
total_samples_duration(other.total_samples_duration) {}
RTCAudioSourceStats::~RTCAudioSourceStats() {}