From 56d460902e1609f7e34d530a463435a71adb2577 Mon Sep 17 00:00:00 2001 From: Ivo Creusen Date: Fri, 24 Nov 2017 17:29:59 +0100 Subject: [PATCH] Use the new AudioProcessing statistics everywhere. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM. Bug: webrtc:8563, b/67926135 Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24 Reviewed-on: https://webrtc-review.googlesource.com/25621 Commit-Queue: Ivo Creusen Reviewed-by: Henrik Boström Reviewed-by: Karl Wiberg Cr-Commit-Position: refs/heads/master@{#20877} --- api/BUILD.gn | 3 + api/DEPS | 5 + api/mediastreaminterface.cc | 25 ++- api/mediastreaminterface.h | 10 +- audio/BUILD.gn | 1 + audio/audio_send_stream.cc | 25 +-- audio/audio_send_stream.h | 2 + audio/audio_send_stream_unittest.cc | 55 ++++--- audio/test/audio_stats_test.cc | 13 +- call/BUILD.gn | 1 + call/audio_send_stream.h | 11 +- media/base/mediachannel.h | 4 + media/engine/fakewebrtccall.cc | 5 + media/engine/fakewebrtccall.h | 2 + media/engine/webrtcvoiceengine.cc | 16 +- media/engine/webrtcvoiceengine_unittest.cc | 34 ++-- modules/audio_processing/BUILD.gn | 11 ++ .../audio_processing/audio_processing_impl.cc | 11 +- .../audio_processing_unittest.cc | 4 +- .../include/audio_processing.h | 33 +--- .../include/audio_processing_statistics.cc | 22 +++ .../include/audio_processing_statistics.h | 51 ++++++ .../include/mock_audio_processing.h | 2 + pc/rtcstatscollector.cc | 12 +- pc/rtcstatscollector_unittest.cc | 6 +- pc/statscollector.cc | 74 +++------ pc/statscollector_unittest.cc | 151 +++++++++++------- 27 files changed, 322 insertions(+), 267 deletions(-) create mode 100644 modules/audio_processing/include/audio_processing_statistics.cc create mode 100644 modules/audio_processing/include/audio_processing_statistics.h diff --git a/api/BUILD.gn b/api/BUILD.gn index aa5db9373c..8a76f2548b 100644 --- a/api/BUILD.gn +++ b/api/BUILD.gn @@ -124,6 +124,9 @@ rtc_source_set("mediastream_interface_and_implicit_video_frame_api") { sources = [ "mediastreaminterface.h", ] + deps = [ + "../modules/audio_processing:audio_processing_statistics", + ] } rtc_source_set("libjingle_api_deprecated_headers") { diff --git a/api/DEPS b/api/DEPS index 1e8e347924..a537633981 100644 --- a/api/DEPS +++ b/api/DEPS @@ -38,4 +38,9 @@ specific_include_rules = { ".*i420_buffer\.h": [ "+system_wrappers/include/aligned_malloc.h", ], + + # Needed to use the APM statistics. + "mediastreaminterface.h": [ + "+modules/audio_processing/include/audio_processing_statistics.h", + ], } diff --git a/api/mediastreaminterface.cc b/api/mediastreaminterface.cc index 92bca162af..4821da83b3 100644 --- a/api/mediastreaminterface.cc +++ b/api/mediastreaminterface.cc @@ -21,19 +21,18 @@ AudioProcessorInterface::GetStats(bool /*has_remote_tracks*/) { AudioProcessorStats stats; GetStats(&stats); AudioProcessorStatistics new_stats; - new_stats.aec_divergent_filter_fraction = - rtc::Optional(stats.aec_divergent_filter_fraction); - new_stats.aec_quality_min = rtc::Optional(stats.aec_quality_min); - new_stats.echo_delay_median_ms = - rtc::Optional(stats.echo_delay_median_ms); - new_stats.echo_delay_std_ms = rtc::Optional(stats.echo_delay_std_ms); - new_stats.echo_return_loss = rtc::Optional(stats.echo_return_loss); - new_stats.echo_return_loss_enhancement = - rtc::Optional(stats.echo_return_loss_enhancement); - new_stats.residual_echo_likelihood = - rtc::Optional(stats.residual_echo_likelihood); - new_stats.residual_echo_likelihood_recent_max = - rtc::Optional(stats.residual_echo_likelihood_recent_max); + new_stats.apm_statistics.divergent_filter_fraction = + stats.aec_divergent_filter_fraction; + new_stats.apm_statistics.delay_median_ms = stats.echo_delay_median_ms; + new_stats.apm_statistics.delay_standard_deviation_ms = + stats.echo_delay_std_ms; + new_stats.apm_statistics.echo_return_loss = stats.echo_return_loss; + new_stats.apm_statistics.echo_return_loss_enhancement = + stats.echo_return_loss_enhancement; + new_stats.apm_statistics.residual_echo_likelihood = + stats.residual_echo_likelihood; + new_stats.apm_statistics.residual_echo_likelihood_recent_max = + stats.residual_echo_likelihood_recent_max; return new_stats; } diff --git a/api/mediastreaminterface.h b/api/mediastreaminterface.h index 2cc5923d7e..80595e09ab 100644 --- a/api/mediastreaminterface.h +++ b/api/mediastreaminterface.h @@ -29,6 +29,7 @@ // mediachannel.h, which is no longer a dependency of this file. #include "media/base/videosinkinterface.h" #include "media/base/videosourceinterface.h" +#include "modules/audio_processing/include/audio_processing_statistics.h" #include "rtc_base/ratetracker.h" #include "rtc_base/refcount.h" #include "rtc_base/scoped_ref_ptr.h" @@ -255,14 +256,7 @@ class AudioProcessorInterface : public rtc::RefCountInterface { // regular stats struct when all users have been updated. struct AudioProcessorStatistics { bool typing_noise_detected = false; - rtc::Optional echo_return_loss; - rtc::Optional echo_return_loss_enhancement; - rtc::Optional echo_delay_median_ms; - rtc::Optional echo_delay_std_ms; - rtc::Optional aec_quality_min; - rtc::Optional residual_echo_likelihood; - rtc::Optional residual_echo_likelihood_recent_max; - rtc::Optional aec_divergent_filter_fraction; + AudioProcessingStats apm_statistics; }; // Get audio processor statistics. diff --git a/audio/BUILD.gn b/audio/BUILD.gn index 038338c03a..2a29ceb22b 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -102,6 +102,7 @@ if (rtc_include_tests) { "../call:rtp_receiver", "../modules/audio_device:mock_audio_device", "../modules/audio_mixer:audio_mixer_impl", + "../modules/audio_processing:audio_processing_statistics", "../modules/congestion_controller:congestion_controller", "../modules/congestion_controller:mock_congestion_controller", "../modules/pacing:mock_paced_sender", diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc index 8583ed0e5f..2a5628cfa9 100644 --- a/audio/audio_send_stream.cc +++ b/audio/audio_send_stream.cc @@ -277,6 +277,11 @@ void AudioSendStream::SetMuted(bool muted) { } webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { + return GetStats(true); +} + +webrtc::AudioSendStream::Stats AudioSendStream::GetStats( + bool has_remote_tracks) const { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); webrtc::AudioSendStream::Stats stats; stats.local_ssrc = config_.rtp.ssrc; @@ -289,10 +294,6 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { if (call_stats.rttMs > 0) { stats.rtt_ms = call_stats.rttMs; } - // TODO(solenberg): [was ajm]: Re-enable this metric once we have a reliable - // implementation. - stats.aec_quality_min = -1; - if (config_.send_codec_spec) { const auto& spec = *config_.send_codec_spec; stats.codec_name = spec.format.name; @@ -323,23 +324,13 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { stats.total_input_energy = base->transmit_mixer()->GetTotalInputEnergy(); stats.total_input_duration = base->transmit_mixer()->GetTotalInputDuration(); - RTC_DCHECK(audio_state_->audio_processing()); - auto audio_processing_stats = - audio_state_->audio_processing()->GetStatistics(); - stats.echo_delay_median_ms = audio_processing_stats.delay_median; - stats.echo_delay_std_ms = audio_processing_stats.delay_standard_deviation; - stats.echo_return_loss = audio_processing_stats.echo_return_loss.instant(); - stats.echo_return_loss_enhancement = - audio_processing_stats.echo_return_loss_enhancement.instant(); - stats.residual_echo_likelihood = - audio_processing_stats.residual_echo_likelihood; - stats.residual_echo_likelihood_recent_max = - audio_processing_stats.residual_echo_likelihood_recent_max; - internal::AudioState* audio_state = static_cast(audio_state_.get()); stats.typing_noise_detected = audio_state->typing_noise_detected(); stats.ana_statistics = channel_proxy_->GetANAStatistics(); + RTC_DCHECK(audio_state_->audio_processing()); + stats.apm_statistics = + audio_state_->audio_processing()->GetStatistics(has_remote_tracks); return stats; } diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h index ef892691b9..08bdddb203 100644 --- a/audio/audio_send_stream.h +++ b/audio/audio_send_stream.h @@ -58,6 +58,8 @@ class AudioSendStream final : public webrtc::AudioSendStream, int duration_ms) override; void SetMuted(bool muted) override; webrtc::AudioSendStream::Stats GetStats() const override; + webrtc::AudioSendStream::Stats GetStats( + bool has_remote_tracks) const override; void SignalNetworkState(NetworkState state); bool DeliverRtcp(const uint8_t* packet, size_t length); diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc index c67eb9ba00..145a8e2419 100644 --- a/audio/audio_send_stream_unittest.cc +++ b/audio/audio_send_stream_unittest.cc @@ -19,6 +19,7 @@ #include "call/rtp_transport_controller_send_interface.h" #include "logging/rtc_event_log/mock/mock_rtc_event_log.h" #include "modules/audio_mixer/audio_mixer_impl.h" +#include "modules/audio_processing/include/audio_processing_statistics.h" #include "modules/audio_processing/include/mock_audio_processing.h" #include "modules/congestion_controller/include/mock/mock_congestion_observer.h" #include "modules/congestion_controller/include/send_side_congestion_controller.h" @@ -50,11 +51,13 @@ const uint32_t kSsrc = 1234; const char* kCName = "foo_name"; const int kAudioLevelId = 2; const int kTransportSequenceNumberId = 4; -const int kEchoDelayMedian = 254; -const int kEchoDelayStdDev = -3; -const int kEchoReturnLoss = -65; -const int kEchoReturnLossEnhancement = 101; -const float kResidualEchoLikelihood = -1.0f; +const int32_t kEchoDelayMedian = 254; +const int32_t kEchoDelayStdDev = -3; +const double kDivergentFilterFraction = 0.2f; +const double kEchoReturnLoss = -65; +const double kEchoReturnLossEnhancement = 101; +const double kResidualEchoLikelihood = -1.0f; +const double kResidualEchoLikelihoodMax = 23.0f; const int32_t kSpeechInputLevel = 96; const double kTotalInputEnergy = 0.25; const double kTotalInputDuration = 0.5; @@ -310,17 +313,18 @@ struct ConfigHelper { EXPECT_CALL(transmit_mixer_, typing_noise_detected()) .WillRepeatedly(Return(true)); - // We have to set the instantaneous value, the average, min and max. We only - // care about the instantaneous value, so we set all to the same value. - audio_processing_stats_.echo_return_loss.Set( - kEchoReturnLoss, kEchoReturnLoss, kEchoReturnLoss, kEchoReturnLoss); - audio_processing_stats_.echo_return_loss_enhancement.Set( - kEchoReturnLossEnhancement, kEchoReturnLossEnhancement, - kEchoReturnLossEnhancement, kEchoReturnLossEnhancement); - audio_processing_stats_.delay_median = kEchoDelayMedian; - audio_processing_stats_.delay_standard_deviation = kEchoDelayStdDev; + audio_processing_stats_.echo_return_loss = kEchoReturnLoss; + audio_processing_stats_.echo_return_loss_enhancement = + kEchoReturnLossEnhancement; + audio_processing_stats_.delay_median_ms = kEchoDelayMedian; + audio_processing_stats_.delay_standard_deviation_ms = kEchoDelayStdDev; + audio_processing_stats_.divergent_filter_fraction = + kDivergentFilterFraction; + audio_processing_stats_.residual_echo_likelihood = kResidualEchoLikelihood; + audio_processing_stats_.residual_echo_likelihood_recent_max = + kResidualEchoLikelihoodMax; - EXPECT_CALL(*audio_processing_, GetStatistics()) + EXPECT_CALL(*audio_processing_, GetStatistics(true)) .WillRepeatedly(Return(audio_processing_stats_)); } @@ -331,7 +335,7 @@ struct ConfigHelper { testing::StrictMock* channel_proxy_ = nullptr; rtc::scoped_refptr audio_processing_; MockTransmitMixer transmit_mixer_; - AudioProcessing::AudioProcessingStatistics audio_processing_stats_; + AudioProcessingStats audio_processing_stats_; SimulatedClock simulated_clock_; PacketRouter packet_router_; testing::NiceMock pacer_; @@ -429,7 +433,7 @@ TEST(AudioSendStreamTest, GetStats) { helper.transport(), helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats(), rtc::nullopt); helper.SetupMockForGetStats(); - AudioSendStream::Stats stats = send_stream.GetStats(); + AudioSendStream::Stats stats = send_stream.GetStats(true); EXPECT_EQ(kSsrc, stats.local_ssrc); EXPECT_EQ(static_cast(kCallStats.bytesSent), stats.bytes_sent); EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent); @@ -446,12 +450,17 @@ TEST(AudioSendStreamTest, GetStats) { EXPECT_EQ(static_cast(kSpeechInputLevel), stats.audio_level); EXPECT_EQ(kTotalInputEnergy, stats.total_input_energy); EXPECT_EQ(kTotalInputDuration, stats.total_input_duration); - EXPECT_EQ(-1, stats.aec_quality_min); - EXPECT_EQ(kEchoDelayMedian, stats.echo_delay_median_ms); - EXPECT_EQ(kEchoDelayStdDev, stats.echo_delay_std_ms); - EXPECT_EQ(kEchoReturnLoss, stats.echo_return_loss); - EXPECT_EQ(kEchoReturnLossEnhancement, stats.echo_return_loss_enhancement); - EXPECT_EQ(kResidualEchoLikelihood, stats.residual_echo_likelihood); + EXPECT_EQ(kEchoDelayMedian, stats.apm_statistics.delay_median_ms); + EXPECT_EQ(kEchoDelayStdDev, stats.apm_statistics.delay_standard_deviation_ms); + EXPECT_EQ(kEchoReturnLoss, stats.apm_statistics.echo_return_loss); + EXPECT_EQ(kEchoReturnLossEnhancement, + stats.apm_statistics.echo_return_loss_enhancement); + EXPECT_EQ(kDivergentFilterFraction, + stats.apm_statistics.divergent_filter_fraction); + EXPECT_EQ(kResidualEchoLikelihood, + stats.apm_statistics.residual_echo_likelihood); + EXPECT_EQ(kResidualEchoLikelihoodMax, + stats.apm_statistics.residual_echo_likelihood_recent_max); EXPECT_TRUE(stats.typing_noise_detected); } diff --git a/audio/test/audio_stats_test.cc b/audio/test/audio_stats_test.cc index a1fecb8298..ee225c0b50 100644 --- a/audio/test/audio_stats_test.cc +++ b/audio/test/audio_stats_test.cc @@ -57,13 +57,12 @@ class NoLossTest : public AudioEndToEndTest { EXPECT_EQ(0, send_stats.audio_level); // send_stats.total_input_energy // send_stats.total_input_duration - EXPECT_EQ(-1.0f, send_stats.aec_quality_min); - EXPECT_EQ(-1, send_stats.echo_delay_median_ms); - EXPECT_EQ(-1, send_stats.echo_delay_std_ms); - EXPECT_EQ(-100, send_stats.echo_return_loss); - EXPECT_EQ(-100, send_stats.echo_return_loss_enhancement); - EXPECT_EQ(0.0f, send_stats.residual_echo_likelihood); - EXPECT_EQ(0.0f, send_stats.residual_echo_likelihood_recent_max); + EXPECT_FALSE(send_stats.apm_statistics.delay_median_ms); + EXPECT_FALSE(send_stats.apm_statistics.delay_standard_deviation_ms); + EXPECT_FALSE(send_stats.apm_statistics.echo_return_loss); + EXPECT_FALSE(send_stats.apm_statistics.echo_return_loss_enhancement); + EXPECT_FALSE(send_stats.apm_statistics.residual_echo_likelihood); + EXPECT_FALSE(send_stats.apm_statistics.residual_echo_likelihood_recent_max); EXPECT_EQ(false, send_stats.typing_noise_detected); AudioReceiveStream::Stats recv_stats = receive_stream()->GetStats(); diff --git a/call/BUILD.gn b/call/BUILD.gn index 3c834f105b..740450466a 100644 --- a/call/BUILD.gn +++ b/call/BUILD.gn @@ -29,6 +29,7 @@ rtc_source_set("call_interfaces") { "../api:optional", "../api:transport_api", "../api/audio_codecs:audio_codecs_api", + "../modules/audio_processing:audio_processing_statistics", "../rtc_base:rtc_base", "../rtc_base:rtc_base_approved", ] diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h index 9ed0d1e63f..4912182c12 100644 --- a/call/audio_send_stream.h +++ b/call/audio_send_stream.h @@ -22,6 +22,7 @@ #include "api/optional.h" #include "api/rtpparameters.h" #include "call/rtp_config.h" +#include "modules/audio_processing/include/audio_processing_statistics.h" #include "rtc_base/scoped_ref_ptr.h" #include "typedefs.h" // NOLINT(build/include) @@ -54,15 +55,10 @@ class AudioSendStream { // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy double total_input_energy = 0.0; double total_input_duration = 0.0; - float aec_quality_min = -1.0f; - int32_t echo_delay_median_ms = -1; - int32_t echo_delay_std_ms = -1; - int32_t echo_return_loss = -100; - int32_t echo_return_loss_enhancement = -100; - float residual_echo_likelihood = -1.0f; - float residual_echo_likelihood_recent_max = -1.0f; bool typing_noise_detected = false; + ANAStats ana_statistics; + AudioProcessingStats apm_statistics; }; struct Config { @@ -157,6 +153,7 @@ class AudioSendStream { virtual void SetMuted(bool muted) = 0; virtual Stats GetStats() const = 0; + virtual Stats GetStats(bool has_remote_tracks) const = 0; }; } // namespace webrtc diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h index 0b84ff79cc..7306ebf430 100644 --- a/media/base/mediachannel.h +++ b/media/base/mediachannel.h @@ -27,6 +27,7 @@ #include "media/base/streamparams.h" #include "media/base/videosinkinterface.h" #include "media/base/videosourceinterface.h" +#include "modules/audio_processing/include/audio_processing_statistics.h" #include "rtc_base/basictypes.h" #include "rtc_base/buffer.h" #include "rtc_base/copyonwritebuffer.h" @@ -624,6 +625,8 @@ struct VoiceSenderInfo : public MediaSenderInfo { // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy double total_input_energy; double total_input_duration; + // TODO(bugs.webrtc.org/8572): Remove APM stats from this struct, since they + // are no longer needed now that we have apm_statistics. float aec_quality_min; int echo_delay_median_ms; int echo_delay_std_ms; @@ -633,6 +636,7 @@ struct VoiceSenderInfo : public MediaSenderInfo { float residual_echo_likelihood_recent_max; bool typing_noise_detected; webrtc::ANAStats ana_statistics; + webrtc::AudioProcessingStats apm_statistics; }; struct VoiceReceiverInfo : public MediaReceiverInfo { diff --git a/media/engine/fakewebrtccall.cc b/media/engine/fakewebrtccall.cc index 9c266bc572..3d95e511e6 100644 --- a/media/engine/fakewebrtccall.cc +++ b/media/engine/fakewebrtccall.cc @@ -64,6 +64,11 @@ webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const { return stats_; } +webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats( + bool /*has_remote_tracks*/) const { + return stats_; +} + FakeAudioReceiveStream::FakeAudioReceiveStream( int id, const webrtc::AudioReceiveStream::Config& config) : id_(id), config_(config) { diff --git a/media/engine/fakewebrtccall.h b/media/engine/fakewebrtccall.h index 3d0825d237..e598e9014a 100644 --- a/media/engine/fakewebrtccall.h +++ b/media/engine/fakewebrtccall.h @@ -65,6 +65,8 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream { int duration_ms) override; void SetMuted(bool muted) override; webrtc::AudioSendStream::Stats GetStats() const override; + webrtc::AudioSendStream::Stats GetStats( + bool has_remote_tracks) const override; int id_ = -1; TelephoneEvent latest_telephone_event_; diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc index 808c5d2d1d..0ae975dce0 100644 --- a/media/engine/webrtcvoiceengine.cc +++ b/media/engine/webrtcvoiceengine.cc @@ -923,10 +923,10 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream return muted_; } - webrtc::AudioSendStream::Stats GetStats() const { + webrtc::AudioSendStream::Stats GetStats(bool has_remote_tracks) const { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); RTC_DCHECK(stream_); - return stream_->GetStats(); + return stream_->GetStats(has_remote_tracks); } // Starts the sending by setting ourselves as a sink to the AudioSource to @@ -2200,7 +2200,8 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) { // Get SSRC and stats for each sender. RTC_DCHECK_EQ(info->senders.size(), 0U); for (const auto& stream : send_streams_) { - webrtc::AudioSendStream::Stats stats = stream.second->GetStats(); + webrtc::AudioSendStream::Stats stats = + stream.second->GetStats(recv_streams_.size() > 0); VoiceSenderInfo sinfo; sinfo.add_ssrc(stats.local_ssrc); sinfo.bytes_sent = stats.bytes_sent; @@ -2215,16 +2216,9 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) { sinfo.audio_level = stats.audio_level; sinfo.total_input_energy = stats.total_input_energy; sinfo.total_input_duration = stats.total_input_duration; - sinfo.aec_quality_min = stats.aec_quality_min; - sinfo.echo_delay_median_ms = stats.echo_delay_median_ms; - sinfo.echo_delay_std_ms = stats.echo_delay_std_ms; - sinfo.echo_return_loss = stats.echo_return_loss; - sinfo.echo_return_loss_enhancement = stats.echo_return_loss_enhancement; - sinfo.residual_echo_likelihood = stats.residual_echo_likelihood; - sinfo.residual_echo_likelihood_recent_max = - stats.residual_echo_likelihood_recent_max; sinfo.typing_noise_detected = (send_ ? stats.typing_noise_detected : false); sinfo.ana_statistics = stats.ana_statistics; + sinfo.apm_statistics = stats.apm_statistics; info->senders.push_back(sinfo); } diff --git a/media/engine/webrtcvoiceengine_unittest.cc b/media/engine/webrtcvoiceengine_unittest.cc index b03c8df5b8..0a165f5748 100644 --- a/media/engine/webrtcvoiceengine_unittest.cc +++ b/media/engine/webrtcvoiceengine_unittest.cc @@ -557,13 +557,12 @@ class WebRtcVoiceEngineTestFake : public testing::Test { stats.jitter_ms = 12; stats.rtt_ms = 345; stats.audio_level = 678; - stats.aec_quality_min = 9.01f; - stats.echo_delay_median_ms = 234; - stats.echo_delay_std_ms = 567; - stats.echo_return_loss = 890; - stats.echo_return_loss_enhancement = 1234; - stats.residual_echo_likelihood = 0.432f; - stats.residual_echo_likelihood_recent_max = 0.6f; + stats.apm_statistics.delay_median_ms = 234; + stats.apm_statistics.delay_standard_deviation_ms = 567; + stats.apm_statistics.echo_return_loss = 890; + stats.apm_statistics.echo_return_loss_enhancement = 1234; + stats.apm_statistics.residual_echo_likelihood = 0.432f; + stats.apm_statistics.residual_echo_likelihood_recent_max = 0.6f; stats.ana_statistics.bitrate_action_counter = 321; stats.ana_statistics.channel_action_counter = 432; stats.ana_statistics.dtx_action_counter = 543; @@ -593,15 +592,18 @@ class WebRtcVoiceEngineTestFake : public testing::Test { EXPECT_EQ(info.jitter_ms, stats.jitter_ms); EXPECT_EQ(info.rtt_ms, stats.rtt_ms); EXPECT_EQ(info.audio_level, stats.audio_level); - EXPECT_EQ(info.aec_quality_min, stats.aec_quality_min); - EXPECT_EQ(info.echo_delay_median_ms, stats.echo_delay_median_ms); - EXPECT_EQ(info.echo_delay_std_ms, stats.echo_delay_std_ms); - EXPECT_EQ(info.echo_return_loss, stats.echo_return_loss); - EXPECT_EQ(info.echo_return_loss_enhancement, - stats.echo_return_loss_enhancement); - EXPECT_EQ(info.residual_echo_likelihood, stats.residual_echo_likelihood); - EXPECT_EQ(info.residual_echo_likelihood_recent_max, - stats.residual_echo_likelihood_recent_max); + EXPECT_EQ(info.apm_statistics.delay_median_ms, + stats.apm_statistics.delay_median_ms); + EXPECT_EQ(info.apm_statistics.delay_standard_deviation_ms, + stats.apm_statistics.delay_standard_deviation_ms); + EXPECT_EQ(info.apm_statistics.echo_return_loss, + stats.apm_statistics.echo_return_loss); + EXPECT_EQ(info.apm_statistics.echo_return_loss_enhancement, + stats.apm_statistics.echo_return_loss_enhancement); + EXPECT_EQ(info.apm_statistics.residual_echo_likelihood, + stats.apm_statistics.residual_echo_likelihood); + EXPECT_EQ(info.apm_statistics.residual_echo_likelihood_recent_max, + stats.apm_statistics.residual_echo_likelihood_recent_max); EXPECT_EQ(info.ana_statistics.bitrate_action_counter, stats.ana_statistics.bitrate_action_counter); EXPECT_EQ(info.ana_statistics.channel_action_counter, diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn index fa3f3bdee4..fdc4783edf 100644 --- a/modules/audio_processing/BUILD.gn +++ b/modules/audio_processing/BUILD.gn @@ -233,6 +233,7 @@ rtc_static_library("audio_processing") { defines = [] deps = [ ":aec_dump_interface", + ":audio_processing_statistics", "..:module_api", "../..:webrtc_common", "../../api:array_view", @@ -311,6 +312,16 @@ rtc_static_library("audio_processing") { ] } +rtc_source_set("audio_processing_statistics") { + sources = [ + "include/audio_processing_statistics.cc", + "include/audio_processing_statistics.h", + ] + deps = [ + "../../api:optional", + ] +} + rtc_source_set("aec_dump_interface") { sources = [ "include/aec_dump.cc", diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc index 4a39bb9deb..b5ef42e5dc 100644 --- a/modules/audio_processing/audio_processing_impl.cc +++ b/modules/audio_processing/audio_processing_impl.cc @@ -1557,13 +1557,6 @@ AudioProcessing::AudioProcessingStatistics::AudioProcessingStatistics( AudioProcessing::AudioProcessingStatistics::~AudioProcessingStatistics() = default; -AudioProcessing::AudioProcessingStats::AudioProcessingStats() = default; - -AudioProcessing::AudioProcessingStats::AudioProcessingStats( - const AudioProcessingStats& other) = default; - -AudioProcessing::AudioProcessingStats::~AudioProcessingStats() = default; - // TODO(ivoc): Remove this when GetStatistics() becomes pure virtual. AudioProcessing::AudioProcessingStatistics AudioProcessing::GetStatistics() const { @@ -1571,7 +1564,7 @@ AudioProcessing::AudioProcessingStatistics AudioProcessing::GetStatistics() } // TODO(ivoc): Remove this when GetStatistics() becomes pure virtual. -AudioProcessing::AudioProcessingStats AudioProcessing::GetStatistics( +AudioProcessingStats AudioProcessing::GetStatistics( bool has_remote_tracks) const { return AudioProcessingStats(); } @@ -1611,7 +1604,7 @@ AudioProcessing::AudioProcessingStatistics AudioProcessingImpl::GetStatistics() return stats; } -AudioProcessing::AudioProcessingStats AudioProcessingImpl::GetStatistics( +AudioProcessingStats AudioProcessingImpl::GetStatistics( bool has_remote_tracks) const { AudioProcessingStats stats; if (has_remote_tracks) { diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc index c3402dd500..4da0621abf 100644 --- a/modules/audio_processing/audio_processing_unittest.cc +++ b/modules/audio_processing/audio_processing_unittest.cc @@ -3026,7 +3026,7 @@ TEST(MAYBE_ApmStatistics, AEC2EnabledTest) { } // Test statistics interface. - AudioProcessing::AudioProcessingStats stats = apm->GetStatistics(true); + AudioProcessingStats stats = apm->GetStatistics(true); // We expect all statistics to be set and have a sensible value. ASSERT_TRUE(stats.residual_echo_likelihood); EXPECT_GE(*stats.residual_echo_likelihood, 0.0); @@ -3085,7 +3085,7 @@ TEST(MAYBE_ApmStatistics, AECMEnabledTest) { } // Test statistics interface. - AudioProcessing::AudioProcessingStats stats = apm->GetStatistics(true); + AudioProcessingStats stats = apm->GetStatistics(true); // We expect only the residual echo detector statistics to be set and have a // sensible value. EXPECT_TRUE(stats.residual_echo_likelihood); diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h index a4245c6b34..6c8f4dbc02 100644 --- a/modules/audio_processing/include/audio_processing.h +++ b/modules/audio_processing/include/audio_processing.h @@ -22,6 +22,7 @@ #include "api/optional.h" #include "modules/audio_processing/beamformer/array_util.h" +#include "modules/audio_processing/include/audio_processing_statistics.h" #include "modules/audio_processing/include/config.h" #include "rtc_base/arraysize.h" #include "rtc_base/deprecation.h" @@ -564,38 +565,6 @@ class AudioProcessing : public rtc::RefCountInterface { float residual_echo_likelihood_recent_max = -1.0f; }; - // This version of the stats uses Optionals, it will replace the regular - // AudioProcessingStatistics struct. - struct AudioProcessingStats { - AudioProcessingStats(); - AudioProcessingStats(const AudioProcessingStats& other); - ~AudioProcessingStats(); - - // AEC Statistics. - // ERL = 10log_10(P_far / P_echo) - rtc::Optional echo_return_loss; - // ERLE = 10log_10(P_echo / P_out) - rtc::Optional echo_return_loss_enhancement; - // Fraction of time that the AEC linear filter is divergent, in a 1-second - // non-overlapped aggregation window. - rtc::Optional divergent_filter_fraction; - - // The delay metrics consists of the delay median and standard deviation. It - // also consists of the fraction of delay estimates that can make the echo - // cancellation perform poorly. The values are aggregated until the first - // call to |GetStatistics()| and afterwards aggregated and updated every - // second. Note that if there are several clients pulling metrics from - // |GetStatistics()| during a session the first call from any of them will - // change to one second aggregation window for all. - rtc::Optional delay_median_ms; - rtc::Optional delay_standard_deviation_ms; - - // Residual echo detector likelihood. - rtc::Optional residual_echo_likelihood; - // Maximum residual echo likelihood from the last time period. - rtc::Optional residual_echo_likelihood_recent_max; - }; - // TODO(ivoc): Make this pure virtual when all subclasses have been updated. virtual AudioProcessingStatistics GetStatistics() const; diff --git a/modules/audio_processing/include/audio_processing_statistics.cc b/modules/audio_processing/include/audio_processing_statistics.cc new file mode 100644 index 0000000000..7139ee502e --- /dev/null +++ b/modules/audio_processing/include/audio_processing_statistics.cc @@ -0,0 +1,22 @@ +/* + * Copyright 2017 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 "modules/audio_processing/include/audio_processing_statistics.h" + +namespace webrtc { + +AudioProcessingStats::AudioProcessingStats() = default; + +AudioProcessingStats::AudioProcessingStats(const AudioProcessingStats& other) = + default; + +AudioProcessingStats::~AudioProcessingStats() = default; + +} // namespace webrtc diff --git a/modules/audio_processing/include/audio_processing_statistics.h b/modules/audio_processing/include/audio_processing_statistics.h new file mode 100644 index 0000000000..7dbc90735a --- /dev/null +++ b/modules/audio_processing/include/audio_processing_statistics.h @@ -0,0 +1,51 @@ +/* + * Copyright 2017 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. + */ + +#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_ +#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_ + +#include "api/optional.h" + +namespace webrtc { +// This version of the stats uses Optionals, it will replace the regular +// AudioProcessingStatistics struct. +struct AudioProcessingStats { + AudioProcessingStats(); + AudioProcessingStats(const AudioProcessingStats& other); + ~AudioProcessingStats(); + + // AEC Statistics. + // ERL = 10log_10(P_far / P_echo) + rtc::Optional echo_return_loss; + // ERLE = 10log_10(P_echo / P_out) + rtc::Optional echo_return_loss_enhancement; + // Fraction of time that the AEC linear filter is divergent, in a 1-second + // non-overlapped aggregation window. + rtc::Optional divergent_filter_fraction; + + // The delay metrics consists of the delay median and standard deviation. It + // also consists of the fraction of delay estimates that can make the echo + // cancellation perform poorly. The values are aggregated until the first + // call to |GetStatistics()| and afterwards aggregated and updated every + // second. Note that if there are several clients pulling metrics from + // |GetStatistics()| during a session the first call from any of them will + // change to one second aggregation window for all. + rtc::Optional delay_median_ms; + rtc::Optional delay_standard_deviation_ms; + + // Residual echo detector likelihood. + rtc::Optional residual_echo_likelihood; + // Maximum residual echo likelihood from the last time period. + rtc::Optional residual_echo_likelihood_recent_max; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_ diff --git a/modules/audio_processing/include/mock_audio_processing.h b/modules/audio_processing/include/mock_audio_processing.h index 7cbd166d01..f2bdc2f241 100644 --- a/modules/audio_processing/include/mock_audio_processing.h +++ b/modules/audio_processing/include/mock_audio_processing.h @@ -15,6 +15,7 @@ #include "modules/audio_processing/include/aec_dump.h" #include "modules/audio_processing/include/audio_processing.h" +#include "modules/audio_processing/include/audio_processing_statistics.h" #include "test/gmock.h" namespace webrtc { @@ -199,6 +200,7 @@ class MockAudioProcessing : public AudioProcessing { MOCK_METHOD0(UpdateHistogramsOnCallEnd, void()); MOCK_CONST_METHOD0(GetStatistics, AudioProcessingStatistics()); + MOCK_CONST_METHOD1(GetStatistics, AudioProcessingStats(bool)); virtual MockEchoCancellation* echo_cancellation() const { return echo_cancellation_.get(); } diff --git a/pc/rtcstatscollector.cc b/pc/rtcstatscollector.cc index a23ab30185..aa901796bf 100644 --- a/pc/rtcstatscollector.cc +++ b/pc/rtcstatscollector.cc @@ -404,13 +404,13 @@ ProduceMediaStreamTrackStatsFromVoiceSenderInfo( audio_track_stats->total_audio_energy = voice_sender_info.total_input_energy; audio_track_stats->total_samples_duration = voice_sender_info.total_input_duration; - if (voice_sender_info.echo_return_loss != -100) { - audio_track_stats->echo_return_loss = static_cast( - voice_sender_info.echo_return_loss); + 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.echo_return_loss_enhancement != -100) { - audio_track_stats->echo_return_loss_enhancement = static_cast( - voice_sender_info.echo_return_loss_enhancement); + 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; } diff --git a/pc/rtcstatscollector_unittest.cc b/pc/rtcstatscollector_unittest.cc index d6c21cebfa..413b13b616 100644 --- a/pc/rtcstatscollector_unittest.cc +++ b/pc/rtcstatscollector_unittest.cc @@ -1516,8 +1516,8 @@ TEST_F(RTCStatsCollectorTest, voice_sender_info_ssrc1.audio_level = 32767; voice_sender_info_ssrc1.total_input_energy = 0.25; voice_sender_info_ssrc1.total_input_duration = 0.5; - voice_sender_info_ssrc1.echo_return_loss = 42; - voice_sender_info_ssrc1.echo_return_loss_enhancement = 52; + voice_sender_info_ssrc1.apm_statistics.echo_return_loss = 42.0; + voice_sender_info_ssrc1.apm_statistics.echo_return_loss_enhancement = 52.0; // Uses default values, the corresponding stats object should contain // undefined members. @@ -1527,8 +1527,6 @@ TEST_F(RTCStatsCollectorTest, voice_sender_info_ssrc2.audio_level = 0; voice_sender_info_ssrc2.total_input_energy = 0.0; voice_sender_info_ssrc2.total_input_duration = 0.0; - voice_sender_info_ssrc2.echo_return_loss = -100; - voice_sender_info_ssrc2.echo_return_loss_enhancement = -100; // Remote audio track rtc::scoped_refptr remote_audio_track = diff --git a/pc/statscollector.cc b/pc/statscollector.cc index 6b27dfd290..0344c988fa 100644 --- a/pc/statscollector.cc +++ b/pc/statscollector.cc @@ -99,50 +99,39 @@ void ExtractCommonReceiveProperties(const cricket::MediaReceiverInfo& info, report->AddString(StatsReport::kStatsValueNameCodecName, info.codec_name); } -void SetAudioProcessingStats( - StatsReport* report, - bool typing_noise_detected, - rtc::Optional echo_return_loss, - rtc::Optional echo_return_loss_enhancement, - rtc::Optional echo_delay_median_ms, - rtc::Optional aec_quality_min, - rtc::Optional echo_delay_std_ms, - rtc::Optional residual_echo_likelihood, - rtc::Optional residual_echo_likelihood_recent_max) { +void SetAudioProcessingStats(StatsReport* report, + bool typing_noise_detected, + const AudioProcessingStats& apm_stats) { report->AddBoolean(StatsReport::kStatsValueNameTypingNoiseState, typing_noise_detected); - // TODO(ivoc): Remove the checks for default values once the whole stat - // chain uses optionals. - if (aec_quality_min && *aec_quality_min >= 0.0) { - report->AddFloat(StatsReport::kStatsValueNameEchoCancellationQualityMin, - *aec_quality_min); - } - if (echo_delay_median_ms && *echo_delay_median_ms >= 0) { + if (apm_stats.delay_median_ms) { report->AddInt(StatsReport::kStatsValueNameEchoDelayMedian, - *echo_delay_median_ms); + *apm_stats.delay_median_ms); } - if (echo_delay_std_ms && *echo_delay_std_ms >= 0) { + if (apm_stats.delay_standard_deviation_ms) { report->AddInt(StatsReport::kStatsValueNameEchoDelayStdDev, - *echo_delay_std_ms); + *apm_stats.delay_standard_deviation_ms); } - // These can take on valid negative values. - if (echo_return_loss) { + if (apm_stats.echo_return_loss) { report->AddInt(StatsReport::kStatsValueNameEchoReturnLoss, - static_cast(*echo_return_loss)); + *apm_stats.echo_return_loss); } - if (echo_return_loss_enhancement) { + if (apm_stats.echo_return_loss_enhancement) { report->AddInt(StatsReport::kStatsValueNameEchoReturnLossEnhancement, - static_cast(*echo_return_loss_enhancement)); + *apm_stats.echo_return_loss_enhancement); } - if (residual_echo_likelihood && *residual_echo_likelihood >= 0.0) { + if (apm_stats.residual_echo_likelihood) { report->AddFloat(StatsReport::kStatsValueNameResidualEchoLikelihood, - *residual_echo_likelihood); + static_cast(*apm_stats.residual_echo_likelihood)); } - if (residual_echo_likelihood_recent_max && - *residual_echo_likelihood_recent_max >= 0.0) { + if (apm_stats.residual_echo_likelihood_recent_max) { report->AddFloat( StatsReport::kStatsValueNameResidualEchoLikelihoodRecentMax, - *residual_echo_likelihood_recent_max); + static_cast(*apm_stats.residual_echo_likelihood_recent_max)); + } + if (apm_stats.divergent_filter_fraction) { + report->AddFloat(StatsReport::kStatsValueNameAecDivergentFilterFraction, + static_cast(*apm_stats.divergent_filter_fraction)); } } @@ -204,16 +193,8 @@ void ExtractStats(const cricket::VoiceReceiverInfo& info, StatsReport* report) { void ExtractStats(const cricket::VoiceSenderInfo& info, StatsReport* report) { ExtractCommonSendProperties(info, report); - // TODO(ivoc): Update VoiceSenderInfo to pass Optionals all the way from APM. - SetAudioProcessingStats( - report, info.typing_noise_detected, - rtc::Optional(info.echo_return_loss), - rtc::Optional(info.echo_return_loss_enhancement), - rtc::Optional(info.echo_delay_median_ms), - rtc::Optional(info.aec_quality_min), - rtc::Optional(info.echo_delay_std_ms), - rtc::Optional(info.residual_echo_likelihood), - rtc::Optional(info.residual_echo_likelihood_recent_max)); + SetAudioProcessingStats(report, info.typing_noise_detected, + info.apm_statistics); const FloatForAdd floats[] = { { StatsReport::kStatsValueNameTotalAudioEnergy, info.total_input_energy }, @@ -1033,17 +1014,8 @@ void StatsCollector::UpdateReportFromAudioTrack(AudioTrackInterface* track, AudioProcessorInterface::AudioProcessorStatistics stats = audio_processor->GetStats(has_remote_tracks); - SetAudioProcessingStats( - report, stats.typing_noise_detected, stats.echo_return_loss, - stats.echo_return_loss_enhancement, stats.echo_delay_median_ms, - stats.aec_quality_min, stats.echo_delay_std_ms, - stats.residual_echo_likelihood, - stats.residual_echo_likelihood_recent_max); - - if (stats.aec_divergent_filter_fraction) { - report->AddFloat(StatsReport::kStatsValueNameAecDivergentFilterFraction, - *stats.aec_divergent_filter_fraction); - } + SetAudioProcessingStats(report, stats.typing_noise_detected, + stats.apm_statistics); } } diff --git a/pc/statscollector_unittest.cc b/pc/statscollector_unittest.cc index d65e3a61c0..ffae504e88 100644 --- a/pc/statscollector_unittest.cc +++ b/pc/statscollector_unittest.cc @@ -84,19 +84,19 @@ class FakeAudioProcessor : public webrtc::AudioProcessorInterface { stats->echo_return_loss = 2; stats->echo_return_loss_enhancement = 3; stats->echo_delay_median_ms = 4; - stats->aec_quality_min = 5.1f; stats->echo_delay_std_ms = 6; } AudioProcessorInterface::AudioProcessorStatistics GetStats( - bool /*has_recv_streams*/) override { + bool has_recv_streams) override { AudioProcessorStatistics stats; stats.typing_noise_detected = true; - stats.echo_return_loss = rtc::Optional(2.0); - stats.echo_return_loss_enhancement = rtc::Optional(3.0); - stats.echo_delay_median_ms = rtc::Optional(4); - stats.aec_quality_min = rtc::Optional(5.1); - stats.echo_delay_std_ms = rtc::Optional(6); + if (has_recv_streams) { + stats.apm_statistics.echo_return_loss = 2.0; + stats.apm_statistics.echo_return_loss_enhancement = 3.0; + stats.apm_statistics.delay_median_ms = 4; + stats.apm_statistics.delay_standard_deviation_ms = 5; + } return stats; } }; @@ -137,7 +137,6 @@ class FakeAudioProcessorWithInitValue : public webrtc::AudioProcessorInterface { stats->echo_return_loss = -100; stats->echo_return_loss_enhancement = -100; stats->echo_delay_median_ms = -1; - stats->aec_quality_min = -1.0f; stats->echo_delay_std_ms = -1; } @@ -409,37 +408,70 @@ void VerifyVoiceSenderInfoReport(const StatsReport* report, EXPECT_TRUE(GetValue( report, StatsReport::kStatsValueNameJitterReceived, &value_in_report)); EXPECT_EQ(rtc::ToString(sinfo.jitter_ms), value_in_report); - EXPECT_TRUE(GetValue( - report, StatsReport::kStatsValueNameEchoCancellationQualityMin, - &value_in_report)); - EXPECT_EQ(rtc::ToString(sinfo.aec_quality_min), value_in_report); - EXPECT_TRUE(GetValue( - report, StatsReport::kStatsValueNameEchoDelayMedian, &value_in_report)); - EXPECT_EQ(rtc::ToString(sinfo.echo_delay_median_ms), - value_in_report); - EXPECT_TRUE(GetValue( - report, StatsReport::kStatsValueNameEchoDelayStdDev, &value_in_report)); - EXPECT_EQ(rtc::ToString(sinfo.echo_delay_std_ms), - value_in_report); - EXPECT_TRUE(GetValue( - report, StatsReport::kStatsValueNameEchoReturnLoss, &value_in_report)); - EXPECT_EQ(rtc::ToString(sinfo.echo_return_loss), - value_in_report); - EXPECT_TRUE(GetValue( - report, StatsReport::kStatsValueNameEchoReturnLossEnhancement, - &value_in_report)); - EXPECT_EQ(rtc::ToString(sinfo.echo_return_loss_enhancement), - value_in_report); - EXPECT_TRUE(GetValue(report, - StatsReport::kStatsValueNameResidualEchoLikelihood, - &value_in_report)); - EXPECT_EQ(rtc::ToString(sinfo.residual_echo_likelihood), - value_in_report); - EXPECT_TRUE(GetValue( - report, StatsReport::kStatsValueNameResidualEchoLikelihoodRecentMax, - &value_in_report)); - EXPECT_EQ(rtc::ToString(sinfo.residual_echo_likelihood_recent_max), - value_in_report); + if (sinfo.apm_statistics.delay_median_ms) { + EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameEchoDelayMedian, + &value_in_report)); + EXPECT_EQ(rtc::ToString(*sinfo.apm_statistics.delay_median_ms), + value_in_report); + } else { + EXPECT_FALSE(GetValue(report, StatsReport::kStatsValueNameEchoDelayMedian, + &value_in_report)); + } + if (sinfo.apm_statistics.delay_standard_deviation_ms) { + EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameEchoDelayStdDev, + &value_in_report)); + EXPECT_EQ( + rtc::ToString(*sinfo.apm_statistics.delay_standard_deviation_ms), + value_in_report); + } else { + EXPECT_FALSE(GetValue(report, StatsReport::kStatsValueNameEchoDelayStdDev, + &value_in_report)); + } + if (sinfo.apm_statistics.echo_return_loss) { + EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameEchoReturnLoss, + &value_in_report)); + EXPECT_EQ(rtc::ToString(*sinfo.apm_statistics.echo_return_loss), + value_in_report); + } else { + EXPECT_FALSE(GetValue(report, StatsReport::kStatsValueNameEchoReturnLoss, + &value_in_report)); + } + if (sinfo.apm_statistics.echo_return_loss_enhancement) { + EXPECT_TRUE(GetValue(report, + StatsReport::kStatsValueNameEchoReturnLossEnhancement, + &value_in_report)); + EXPECT_EQ( + rtc::ToString(*sinfo.apm_statistics.echo_return_loss_enhancement), + value_in_report); + } else { + EXPECT_FALSE(GetValue(report, + StatsReport::kStatsValueNameEchoReturnLossEnhancement, + &value_in_report)); + } + if (sinfo.apm_statistics.residual_echo_likelihood) { + EXPECT_TRUE(GetValue(report, + StatsReport::kStatsValueNameResidualEchoLikelihood, + &value_in_report)); + EXPECT_EQ( + rtc::ToString(*sinfo.apm_statistics.residual_echo_likelihood), + value_in_report); + } else { + EXPECT_FALSE(GetValue(report, + StatsReport::kStatsValueNameResidualEchoLikelihood, + &value_in_report)); + } + if (sinfo.apm_statistics.residual_echo_likelihood_recent_max) { + EXPECT_TRUE(GetValue( + report, StatsReport::kStatsValueNameResidualEchoLikelihoodRecentMax, + &value_in_report)); + EXPECT_EQ(rtc::ToString( + *sinfo.apm_statistics.residual_echo_likelihood_recent_max), + value_in_report); + } else { + EXPECT_FALSE(GetValue( + report, StatsReport::kStatsValueNameResidualEchoLikelihoodRecentMax, + &value_in_report)); + } EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameAudioInputLevel, &value_in_report)); EXPECT_EQ(rtc::ToString(sinfo.audio_level), value_in_report); @@ -510,33 +542,27 @@ void InitVoiceSenderInfo(cricket::VoiceSenderInfo* voice_sender_info) { voice_sender_info->echo_return_loss_enhancement = 109; voice_sender_info->echo_delay_median_ms = 110; voice_sender_info->echo_delay_std_ms = 111; - voice_sender_info->aec_quality_min = 112.0f; voice_sender_info->typing_noise_detected = false; - voice_sender_info->ana_statistics.bitrate_action_counter = 113; - voice_sender_info->ana_statistics.channel_action_counter = 114; - voice_sender_info->ana_statistics.dtx_action_counter = 115; - voice_sender_info->ana_statistics.fec_action_counter = 116; - voice_sender_info->ana_statistics.frame_length_increase_counter = 117; - voice_sender_info->ana_statistics.frame_length_decrease_counter = 118; - voice_sender_info->ana_statistics.uplink_packet_loss_fraction = 119.0; + voice_sender_info->ana_statistics.bitrate_action_counter = 112; + voice_sender_info->ana_statistics.channel_action_counter = 113; + voice_sender_info->ana_statistics.dtx_action_counter = 114; + voice_sender_info->ana_statistics.fec_action_counter = 115; + voice_sender_info->ana_statistics.frame_length_increase_counter = 116; + voice_sender_info->ana_statistics.frame_length_decrease_counter = 117; + voice_sender_info->ana_statistics.uplink_packet_loss_fraction = 118.0; } void UpdateVoiceSenderInfoFromAudioTrack( AudioTrackInterface* audio_track, - cricket::VoiceSenderInfo* voice_sender_info) { + cricket::VoiceSenderInfo* voice_sender_info, + bool has_remote_tracks) { audio_track->GetSignalLevel(&voice_sender_info->audio_level); - webrtc::AudioProcessorInterface::AudioProcessorStats audio_processor_stats; - audio_track->GetAudioProcessor()->GetStats(&audio_processor_stats); + webrtc::AudioProcessorInterface::AudioProcessorStatistics + audio_processor_stats = + audio_track->GetAudioProcessor()->GetStats(has_remote_tracks); voice_sender_info->typing_noise_detected = audio_processor_stats.typing_noise_detected; - voice_sender_info->echo_return_loss = audio_processor_stats.echo_return_loss; - voice_sender_info->echo_return_loss_enhancement = - audio_processor_stats.echo_return_loss_enhancement; - voice_sender_info->echo_delay_median_ms = - audio_processor_stats.echo_delay_median_ms; - voice_sender_info->aec_quality_min = audio_processor_stats.aec_quality_min; - voice_sender_info->echo_delay_std_ms = - audio_processor_stats.echo_delay_std_ms; + voice_sender_info->apm_statistics = audio_processor_stats.apm_statistics; } void InitVoiceReceiverInfo(cricket::VoiceReceiverInfo* voice_receiver_info) { @@ -730,7 +756,8 @@ class StatsCollectorTest : public testing::Test { // Verifies the values in the track report. if (voice_sender_info) { - UpdateVoiceSenderInfoFromAudioTrack(audio_track, voice_sender_info); + UpdateVoiceSenderInfoFromAudioTrack(audio_track, voice_sender_info, + stats_read->receivers.size() > 0); VerifyVoiceSenderInfoReport(report, *voice_sender_info); } if (voice_receiver_info) { @@ -1746,7 +1773,8 @@ TEST_F(StatsCollectorTest, FilterOutNegativeInitialValues) { // Some of the contents in |voice_sender_info| needs to be updated from the // |audio_track_|. - UpdateVoiceSenderInfoFromAudioTrack(local_track.get(), &voice_sender_info); + UpdateVoiceSenderInfoFromAudioTrack(local_track.get(), &voice_sender_info, + true); cricket::VoiceReceiverInfo voice_receiver_info; voice_receiver_info.add_ssrc(kSsrcOfTrack); @@ -1974,7 +2002,8 @@ TEST_F(StatsCollectorTest, LocalAndRemoteTracksWithSameSsrc) { // Some of the contents in |voice_sender_info| needs to be updated from the // |audio_track_|. - UpdateVoiceSenderInfoFromAudioTrack(audio_track_.get(), &voice_sender_info); + UpdateVoiceSenderInfoFromAudioTrack(audio_track_.get(), &voice_sender_info, + true); cricket::VoiceReceiverInfo voice_receiver_info; InitVoiceReceiverInfo(&voice_receiver_info);