admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Use the new AudioProcessing statistics everywhere.

Browse Source 
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}
This commit is contained in:
Ivo Creusen 2017-11-24 17:29:59 +01:00 committed by Commit Bot
parent 606c882972
commit 56d460902e
27 changed files with 322 additions and 267 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

3
api/BUILD.gn
View File

@ -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") {

5
api/DEPS
View File

@ -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",
],
}

25
api/mediastreaminterface.cc
View File

@ -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<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;
}

10
api/mediastreaminterface.h
View File

@ -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<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.

1
audio/BUILD.gn
View File

@ -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",

25
audio/audio_send_stream.cc
View File

@ -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<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;
}

2
audio/audio_send_stream.h
View File

@ -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);

55
audio/audio_send_stream_unittest.cc
View File

@ -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<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 @@ 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<int64_t>(kCallStats.bytesSent), stats.bytes_sent);
EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
@ -446,12 +450,17 @@ TEST(AudioSendStreamTest, GetStats) {
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);
}

13
audio/test/audio_stats_test.cc
View File

@ -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();

1
call/BUILD.gn
View File

@ -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",
]

11
call/audio_send_stream.h
View File

@ -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

4
media/base/mediachannel.h
View File

@ -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 {

5
media/engine/fakewebrtccall.cc
View File

@ -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) {

2
media/engine/fakewebrtccall.h
View File

@ -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_;

16
media/engine/webrtcvoiceengine.cc
View File

@ -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);
}

34
media/engine/webrtcvoiceengine_unittest.cc
View File

@ -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,

11
modules/audio_processing/BUILD.gn
View File

@ -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",

11
modules/audio_processing/audio_processing_impl.cc
View File

@ -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) {

4
modules/audio_processing/audio_processing_unittest.cc
View File

@ -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);

33
modules/audio_processing/include/audio_processing.h
View File

@ -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<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;

22
modules/audio_processing/include/audio_processing_statistics.cc Normal file
View File

@ -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

51
modules/audio_processing/include/audio_processing_statistics.h Normal file
View File

@ -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_

2
modules/audio_processing/include/mock_audio_processing.h
View File

@ -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();
}

12
pc/rtcstatscollector.cc
View File

@ -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<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;
}

6
pc/rtcstatscollector_unittest.cc
View File

@ -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<MediaStreamTrackInterface> remote_audio_track =

74
pc/statscollector.cc
View File

@ -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<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::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<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 @@ 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);
}
}

151
pc/statscollector_unittest.cc
View File

@ -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<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 @@ 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<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 @@ 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);