Use the new AudioProcessing statistics everywhere.
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 <ivoc@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20877}diff --git a/api/BUILD.gn b/api/BUILD.gn
index aa5db93..8a76f25 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -124,6 +124,9 @@
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 1e8e347..a537633 100644
--- a/api/DEPS
+++ b/api/DEPS
@@ -38,4 +38,9 @@
".*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 92bca16..4821da8 100644
--- a/api/mediastreaminterface.cc
+++ b/api/mediastreaminterface.cc
@@ -21,19 +21,18 @@
AudioProcessorStats stats;
GetStats(&stats);
AudioProcessorStatistics new_stats;
- new_stats.aec_divergent_filter_fraction =
- rtc::Optional<double>(stats.aec_divergent_filter_fraction);
- new_stats.aec_quality_min = rtc::Optional<double>(stats.aec_quality_min);
- new_stats.echo_delay_median_ms =
- rtc::Optional<int32_t>(stats.echo_delay_median_ms);
- new_stats.echo_delay_std_ms = rtc::Optional<int32_t>(stats.echo_delay_std_ms);
- new_stats.echo_return_loss = rtc::Optional<double>(stats.echo_return_loss);
- new_stats.echo_return_loss_enhancement =
- rtc::Optional<double>(stats.echo_return_loss_enhancement);
- new_stats.residual_echo_likelihood =
- rtc::Optional<double>(stats.residual_echo_likelihood);
- new_stats.residual_echo_likelihood_recent_max =
- rtc::Optional<double>(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 2cc5923..80595e0 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 @@
// regular stats struct when all users have been updated.
struct AudioProcessorStatistics {
bool typing_noise_detected = false;
- rtc::Optional<double> echo_return_loss;
- rtc::Optional<double> echo_return_loss_enhancement;
- rtc::Optional<int32_t> echo_delay_median_ms;
- rtc::Optional<int32_t> echo_delay_std_ms;
- rtc::Optional<double> aec_quality_min;
- rtc::Optional<double> residual_echo_likelihood;
- rtc::Optional<double> residual_echo_likelihood_recent_max;
- rtc::Optional<double> aec_divergent_filter_fraction;
+ AudioProcessingStats apm_statistics;
};
// Get audio processor statistics.
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 038338c..2a29ceb 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -102,6 +102,7 @@
"../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 8583ed0..2a5628c 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -277,6 +277,11 @@
}
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 @@
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 @@
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<internal::AudioState*>(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 ef89269..08bdddb 100644
--- a/audio/audio_send_stream.h
+++ b/audio/audio_send_stream.h
@@ -58,6 +58,8 @@
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 c67eb9b..145a8e2 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 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 @@
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 @@
testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
rtc::scoped_refptr<MockAudioProcessing> audio_processing_;
MockTransmitMixer transmit_mixer_;
- AudioProcessing::AudioProcessingStatistics audio_processing_stats_;
+ AudioProcessingStats audio_processing_stats_;
SimulatedClock simulated_clock_;
PacketRouter packet_router_;
testing::NiceMock<MockPacedSender> pacer_;
@@ -429,7 +433,7 @@
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<int64_t>(kCallStats.bytesSent), stats.bytes_sent);
EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
@@ -446,12 +450,17 @@
EXPECT_EQ(static_cast<int32_t>(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 a1fecb8..ee225c0 100644
--- a/audio/test/audio_stats_test.cc
+++ b/audio/test/audio_stats_test.cc
@@ -57,13 +57,12 @@
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 3c834f1..7404504 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -29,6 +29,7 @@
"../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 9ed0d1e..4912182 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 @@
// 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 @@
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 0b84ff7..7306ebf 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 @@
// 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 @@
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 9c266bc..3d95e51 100644
--- a/media/engine/fakewebrtccall.cc
+++ b/media/engine/fakewebrtccall.cc
@@ -64,6 +64,11 @@
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 3d0825d..e598e90 100644
--- a/media/engine/fakewebrtccall.h
+++ b/media/engine/fakewebrtccall.h
@@ -65,6 +65,8 @@
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 808c5d2..0ae975d 100644
--- a/media/engine/webrtcvoiceengine.cc
+++ b/media/engine/webrtcvoiceengine.cc
@@ -923,10 +923,10 @@
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 @@
// 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 @@
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 b03c8df..0a165f5 100644
--- a/media/engine/webrtcvoiceengine_unittest.cc
+++ b/media/engine/webrtcvoiceengine_unittest.cc
@@ -557,13 +557,12 @@
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 @@
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 fa3f3bd..fdc4783 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -233,6 +233,7 @@
defines = []
deps = [
":aec_dump_interface",
+ ":audio_processing_statistics",
"..:module_api",
"../..:webrtc_common",
"../../api:array_view",
@@ -311,6 +312,16 @@
]
}
+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 4a39bb9..b5ef42e 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -1557,13 +1557,6 @@
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 @@
}
// 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 @@
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 c3402dd..4da0621 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -3026,7 +3026,7 @@
}
// 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 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 a4245c6..6c8f4db 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 @@
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<double> echo_return_loss;
- // ERLE = 10log_10(P_echo / P_out)
- rtc::Optional<double> echo_return_loss_enhancement;
- // Fraction of time that the AEC linear filter is divergent, in a 1-second
- // non-overlapped aggregation window.
- rtc::Optional<double> 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<int32_t> delay_median_ms;
- rtc::Optional<int32_t> delay_standard_deviation_ms;
-
- // Residual echo detector likelihood.
- rtc::Optional<double> residual_echo_likelihood;
- // Maximum residual echo likelihood from the last time period.
- rtc::Optional<double> 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 0000000..7139ee5
--- /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 0000000..7dbc907
--- /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<double> echo_return_loss;
+ // ERLE = 10log_10(P_echo / P_out)
+ rtc::Optional<double> echo_return_loss_enhancement;
+ // Fraction of time that the AEC linear filter is divergent, in a 1-second
+ // non-overlapped aggregation window.
+ rtc::Optional<double> 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<int32_t> delay_median_ms;
+ rtc::Optional<int32_t> delay_standard_deviation_ms;
+
+ // Residual echo detector likelihood.
+ rtc::Optional<double> residual_echo_likelihood;
+ // Maximum residual echo likelihood from the last time period.
+ rtc::Optional<double> 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 7cbd166..f2bdc2f 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 @@
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 a23ab30..aa90179 100644
--- a/pc/rtcstatscollector.cc
+++ b/pc/rtcstatscollector.cc
@@ -404,13 +404,13 @@
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<double>(
- 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<double>(
- 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 d6c21ce..413b13b 100644
--- a/pc/rtcstatscollector_unittest.cc
+++ b/pc/rtcstatscollector_unittest.cc
@@ -1516,8 +1516,8 @@
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 @@
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<MediaStreamTrackInterface> remote_audio_track =
diff --git a/pc/statscollector.cc b/pc/statscollector.cc
index 6b27dfd..0344c98 100644
--- a/pc/statscollector.cc
+++ b/pc/statscollector.cc
@@ -99,50 +99,39 @@
report->AddString(StatsReport::kStatsValueNameCodecName, info.codec_name);
}
-void SetAudioProcessingStats(
- StatsReport* report,
- bool typing_noise_detected,
- rtc::Optional<double> echo_return_loss,
- rtc::Optional<double> echo_return_loss_enhancement,
- rtc::Optional<int32_t> echo_delay_median_ms,
- rtc::Optional<double> aec_quality_min,
- rtc::Optional<int32_t> echo_delay_std_ms,
- rtc::Optional<double> residual_echo_likelihood,
- rtc::Optional<double> 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<int32_t>(*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<int32_t>(*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<float>(*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<float>(*apm_stats.residual_echo_likelihood_recent_max));
+ }
+ if (apm_stats.divergent_filter_fraction) {
+ report->AddFloat(StatsReport::kStatsValueNameAecDivergentFilterFraction,
+ static_cast<float>(*apm_stats.divergent_filter_fraction));
}
}
@@ -204,16 +193,8 @@
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<double>(info.echo_return_loss),
- rtc::Optional<double>(info.echo_return_loss_enhancement),
- rtc::Optional<int32_t>(info.echo_delay_median_ms),
- rtc::Optional<double>(info.aec_quality_min),
- rtc::Optional<int32_t>(info.echo_delay_std_ms),
- rtc::Optional<double>(info.residual_echo_likelihood),
- rtc::Optional<double>(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 @@
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 d65e3a6..ffae504 100644
--- a/pc/statscollector_unittest.cc
+++ b/pc/statscollector_unittest.cc
@@ -84,19 +84,19 @@
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<double>(2.0);
- stats.echo_return_loss_enhancement = rtc::Optional<double>(3.0);
- stats.echo_delay_median_ms = rtc::Optional<int32_t>(4);
- stats.aec_quality_min = rtc::Optional<double>(5.1);
- stats.echo_delay_std_ms = rtc::Optional<int32_t>(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 @@
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 @@
EXPECT_TRUE(GetValue(
report, StatsReport::kStatsValueNameJitterReceived, &value_in_report));
EXPECT_EQ(rtc::ToString<int>(sinfo.jitter_ms), value_in_report);
- EXPECT_TRUE(GetValue(
- report, StatsReport::kStatsValueNameEchoCancellationQualityMin,
- &value_in_report));
- EXPECT_EQ(rtc::ToString<float>(sinfo.aec_quality_min), value_in_report);
- EXPECT_TRUE(GetValue(
- report, StatsReport::kStatsValueNameEchoDelayMedian, &value_in_report));
- EXPECT_EQ(rtc::ToString<int>(sinfo.echo_delay_median_ms),
- value_in_report);
- EXPECT_TRUE(GetValue(
- report, StatsReport::kStatsValueNameEchoDelayStdDev, &value_in_report));
- EXPECT_EQ(rtc::ToString<int>(sinfo.echo_delay_std_ms),
- value_in_report);
- EXPECT_TRUE(GetValue(
- report, StatsReport::kStatsValueNameEchoReturnLoss, &value_in_report));
- EXPECT_EQ(rtc::ToString<int>(sinfo.echo_return_loss),
- value_in_report);
- EXPECT_TRUE(GetValue(
- report, StatsReport::kStatsValueNameEchoReturnLossEnhancement,
- &value_in_report));
- EXPECT_EQ(rtc::ToString<int>(sinfo.echo_return_loss_enhancement),
- value_in_report);
- EXPECT_TRUE(GetValue(report,
- StatsReport::kStatsValueNameResidualEchoLikelihood,
- &value_in_report));
- EXPECT_EQ(rtc::ToString<float>(sinfo.residual_echo_likelihood),
- value_in_report);
- EXPECT_TRUE(GetValue(
- report, StatsReport::kStatsValueNameResidualEchoLikelihoodRecentMax,
- &value_in_report));
- EXPECT_EQ(rtc::ToString<float>(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<int>(*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<int>(*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<int>(*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<int>(*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<float>(*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<float>(
+ *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<int>(sinfo.audio_level), value_in_report);
@@ -510,33 +542,27 @@
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 @@
// 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 @@
// 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 @@
// 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);
