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Add new ANA stats to the old GetStats() to count the number of actions taken by each controller.

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This CL only wires up the new stats but does not set the values yet. This will be added in a follow-up CL.

BUG=webrtc:8127

Review-Url: https://codereview.webrtc.org/3011623002
Cr-Commit-Position: refs/heads/master@{#19751}
This commit is contained in:
ivoc 2017-09-08 08:13:19 -07:00 committed by Commit Bot
parent 01410dcc99
commit e1198e068b
27 changed files with 217 additions and 1 deletions
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8
webrtc/api/audio_codecs/audio_encoder.cc
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@ -15,6 +15,10 @@
namespace webrtc {
ANAStats::ANAStats() = default;
ANAStats::~ANAStats() = default;
ANAStats::ANAStats(const ANAStats&) = default;
AudioEncoder::EncodedInfo::EncodedInfo() = default;
AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default;
AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default;
@ -95,4 +99,8 @@ void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}
void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
int max_frame_length_ms) {}
ANAStats AudioEncoder::GetANAStats() const {
return ANAStats();
}
} // namespace webrtc

30
webrtc/api/audio_codecs/audio_encoder.h
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@ -27,6 +27,33 @@ namespace webrtc {
class Clock;
class RtcEventLog;
// Statistics related to Audio Network Adaptation.
struct ANAStats {
ANAStats();
ANAStats(const ANAStats&);
~ANAStats();
// Number of actions taken by the ANA bitrate controller since the start of
// the call. If this value is not set, it indicates that the bitrate
// controller is disabled.
rtc::Optional<uint32_t> bitrate_action_counter;
// Number of actions taken by the ANA channel controller since the start of
// the call. If this value is not set, it indicates that the channel
// controller is disabled.
rtc::Optional<uint32_t> channel_action_counter;
// Number of actions taken by the ANA DTX controller since the start of the
// call. If this value is not set, it indicates that the DTX controller is
// disabled.
rtc::Optional<uint32_t> dtx_action_counter;
// Number of actions taken by the ANA FEC controller since the start of the
// call. If this value is not set, it indicates that the FEC controller is
// disabled.
rtc::Optional<uint32_t> fec_action_counter;
// Number of actions taken by the ANA frame length controller since the start
// of the call. If this value is not set, it indicates that the frame length
// controller is disabled.
rtc::Optional<uint32_t> frame_length_action_counter;
};
// This is the interface class for encoders in AudioCoding module. Each codec
// type must have an implementation of this class.
class AudioEncoder {
@ -203,6 +230,9 @@ class AudioEncoder {
virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
int max_frame_length_ms);
// Get statistics related to audio network adaptation.
virtual ANAStats GetANAStats() const;
protected:
// Subclasses implement this to perform the actual encoding. Called by
// Encode().

10
webrtc/api/statstypes.cc
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@ -586,6 +586,16 @@ const char* StatsReport::Value::display_name() const {
return "googResidualEchoLikelihood";
case kStatsValueNameResidualEchoLikelihoodRecentMax:
return "googResidualEchoLikelihoodRecentMax";
case kStatsValueNameAnaBitrateActionCounter:
return "googAnaBitrateActionCounter";
case kStatsValueNameAnaChannelActionCounter:
return "googAnaChannelActionCounter";
case kStatsValueNameAnaDtxActionCounter:
return "googAnaDtxActionCounter";
case kStatsValueNameAnaFecActionCounter:
return "googAnaFecActionCounter";
case kStatsValueNameAnaFrameLengthActionCounter:
return "googAnaFrameLengthActionCounter";
case kStatsValueNameRetransmitBitrate:
return "googRetransmitBitrate";
case kStatsValueNameRtt:

5
webrtc/api/statstypes.h
View File

@ -208,6 +208,11 @@ class StatsReport {
kStatsValueNameRenderDelayMs,
kStatsValueNameResidualEchoLikelihood,
kStatsValueNameResidualEchoLikelihoodRecentMax,
kStatsValueNameAnaBitrateActionCounter,
kStatsValueNameAnaChannelActionCounter,
kStatsValueNameAnaDtxActionCounter,
kStatsValueNameAnaFecActionCounter,
kStatsValueNameAnaFrameLengthActionCounter,
kStatsValueNameRetransmitBitrate,
kStatsValueNameRtt,
kStatsValueNameSecondaryDecodedRate,

1
webrtc/audio/audio_send_stream.cc
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@ -333,6 +333,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
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();
return stats;
}

2
webrtc/audio/audio_send_stream_unittest.cc
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@ -301,6 +301,8 @@ struct ConfigHelper {
.WillRepeatedly(Return(kCallStats));
EXPECT_CALL(*channel_proxy_, GetRemoteRTCPReportBlocks())
.WillRepeatedly(Return(report_blocks));
EXPECT_CALL(*channel_proxy_, GetANAStatistics())
.WillRepeatedly(Return(ANAStats()));
EXPECT_CALL(voice_engine_, transmit_mixer())
.WillRepeatedly(Return(&transmit_mixer_));

2
webrtc/call/audio_send_stream.h
View File

@ -15,6 +15,7 @@
#include <string>
#include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_encoder_factory.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/call/transport.h"
@ -61,6 +62,7 @@ class AudioSendStream {
float residual_echo_likelihood = -1.0f;
float residual_echo_likelihood_recent_max = -1.0f;
bool typing_noise_detected = false;
ANAStats ana_statistics;
};
struct Config {

2
webrtc/media/base/mediachannel.h
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@ -15,6 +15,7 @@
#include <string>
#include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h"
#include "webrtc/api/optional.h"
#include "webrtc/api/rtpparameters.h"
#include "webrtc/api/rtpreceiverinterface.h"
@ -641,6 +642,7 @@ struct VoiceSenderInfo : public MediaSenderInfo {
float residual_echo_likelihood;
float residual_echo_likelihood_recent_max;
bool typing_noise_detected;
webrtc::ANAStats ana_statistics;
};
struct VoiceReceiverInfo : public MediaReceiverInfo {

1
webrtc/media/engine/webrtcvoiceengine.cc
View File

@ -2260,6 +2260,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
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;
info->senders.push_back(sinfo);
}

16
webrtc/media/engine/webrtcvoiceengine_unittest.cc
View File

@ -546,6 +546,12 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
stats.echo_return_loss_enhancement = 1234;
stats.residual_echo_likelihood = 0.432f;
stats.residual_echo_likelihood_recent_max = 0.6f;
stats.ana_statistics.bitrate_action_counter = rtc::Optional<uint32_t>(321);
stats.ana_statistics.channel_action_counter = rtc::Optional<uint32_t>(432);
stats.ana_statistics.dtx_action_counter = rtc::Optional<uint32_t>(543);
stats.ana_statistics.fec_action_counter = rtc::Optional<uint32_t>(654);
stats.ana_statistics.frame_length_action_counter =
rtc::Optional<uint32_t>(765);
stats.typing_noise_detected = true;
return stats;
}
@ -577,6 +583,16 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
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.ana_statistics.bitrate_action_counter,
stats.ana_statistics.bitrate_action_counter);
EXPECT_EQ(info.ana_statistics.channel_action_counter,
stats.ana_statistics.channel_action_counter);
EXPECT_EQ(info.ana_statistics.dtx_action_counter,
stats.ana_statistics.dtx_action_counter);
EXPECT_EQ(info.ana_statistics.fec_action_counter,
stats.ana_statistics.fec_action_counter);
EXPECT_EQ(info.ana_statistics.frame_length_action_counter,
stats.ana_statistics.frame_length_action_counter);
EXPECT_EQ(info.typing_noise_detected,
stats.typing_noise_detected && is_sending);
}

1
webrtc/modules/audio_coding/BUILD.gn
View File

@ -913,6 +913,7 @@ rtc_static_library("audio_network_adaptor") {
deps = [
"../..:webrtc_common",
"../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
"../../logging:rtc_event_log_api",
"../../rtc_base:protobuf_utils",

10
webrtc/modules/audio_coding/acm2/audio_coding_module.cc
View File

@ -204,6 +204,8 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const override;
ANAStats GetANAStats() const override;
private:
struct InputData {
uint32_t input_timestamp;
@ -1273,6 +1275,14 @@ void AudioCodingModuleImpl::GetDecodingCallStatistics(
receiver_.GetDecodingCallStatistics(call_stats);
}
ANAStats AudioCodingModuleImpl::GetANAStats() const {
rtc::CritScope lock(&acm_crit_sect_);
if (encoder_stack_)
return encoder_stack_->GetANAStats();
// If no encoder is set, return default stats.
return ANAStats();
}
} // namespace
AudioCodingModule::Config::Config()

6
webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
View File

@ -135,6 +135,12 @@ void AudioNetworkAdaptorImpl::StopDebugDump() {
debug_dump_writer_.reset(nullptr);
}
ANAStats AudioNetworkAdaptorImpl::GetStats() const {
// TODO(ivoc): Actually implement the stat.
// Tracking bug: https://crbug.com/webrtc/8127
return ANAStats();
}
void AudioNetworkAdaptorImpl::DumpNetworkMetrics() {
if (debug_dump_writer_)
debug_dump_writer_->DumpNetworkMetrics(last_metrics_, rtc::TimeMillis());

2
webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
View File

@ -58,6 +58,8 @@ class AudioNetworkAdaptorImpl final : public AudioNetworkAdaptor {
void StopDebugDump() override;
ANAStats GetStats() const override;
private:
void DumpNetworkMetrics();

32
webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
View File

@ -270,4 +270,36 @@ TEST(AudioNetworkAdaptorImplTest, LogRuntimeConfigOnGetEncoderRuntimeConfig) {
states.audio_network_adaptor->GetEncoderRuntimeConfig();
}
TEST(AudioNetworkAdaptorImplTest, TestANAStats) {
auto states = CreateAudioNetworkAdaptor();
// Simulate some adaptation, otherwise the stats will not show anything.
AudioEncoderRuntimeConfig config1, config2;
config1.bitrate_bps = rtc::Optional<int>(32000);
config1.enable_fec = rtc::Optional<bool>(true);
config2.bitrate_bps = rtc::Optional<int>(16000);
config2.enable_fec = rtc::Optional<bool>(false);
EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
.WillOnce(SetArgPointee<0>(config1));
states.audio_network_adaptor->GetEncoderRuntimeConfig();
EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
.WillOnce(SetArgPointee<0>(config2));
states.audio_network_adaptor->GetEncoderRuntimeConfig();
auto ana_stats = states.audio_network_adaptor->GetStats();
// Check that the default stats are returned, as these have not been
// implemented yet). Tracking bug: https://crbug.com/8127
auto default_stats = ANAStats();
EXPECT_EQ(ana_stats.bitrate_action_counter,
default_stats.bitrate_action_counter);
EXPECT_EQ(ana_stats.channel_action_counter,
default_stats.channel_action_counter);
EXPECT_EQ(ana_stats.dtx_action_counter, default_stats.dtx_action_counter);
EXPECT_EQ(ana_stats.fec_action_counter, default_stats.fec_action_counter);
EXPECT_EQ(ana_stats.frame_length_action_counter,
default_stats.frame_length_action_counter);
}
} // namespace webrtc

4
webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
View File

@ -11,6 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
#include "webrtc/api/audio_codecs/audio_encoder.h"
#include "webrtc/api/optional.h"
namespace webrtc {
@ -38,7 +39,6 @@ struct AudioEncoderRuntimeConfig {
// encoder based on network metrics.
class AudioNetworkAdaptor {
public:
virtual ~AudioNetworkAdaptor() = default;
virtual void SetUplinkBandwidth(int uplink_bandwidth_bps) = 0;
@ -60,6 +60,8 @@ class AudioNetworkAdaptor {
virtual void StartDebugDump(FILE* file_handle) = 0;
virtual void StopDebugDump() = 0;
virtual ANAStats GetStats() const = 0;
};
} // namespace webrtc

2
webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
View File

@ -40,6 +40,8 @@ class MockAudioNetworkAdaptor : public AudioNetworkAdaptor {
MOCK_METHOD1(StartDebugDump, void(FILE* file_handle));
MOCK_METHOD0(StopDebugDump, void());
MOCK_CONST_METHOD0(GetStats, ANAStats());
};
} // namespace webrtc

7
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
View File

@ -773,4 +773,11 @@ void AudioEncoderOpus::MaybeUpdateUplinkBandwidth() {
}
}
ANAStats AudioEncoderOpus::GetANAStats() const {
if (audio_network_adaptor_) {
return audio_network_adaptor_->GetStats();
}
return ANAStats();
}
} // namespace webrtc

1
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
View File

@ -116,6 +116,7 @@ class AudioEncoderOpus final : public AudioEncoder {
void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
void SetReceiverFrameLengthRange(int min_frame_length_ms,
int max_frame_length_ms) override;
ANAStats GetANAStats() const override;
rtc::ArrayView<const int> supported_frame_lengths_ms() const {
return config_.supported_frame_lengths_ms;
}

2
webrtc/modules/audio_coding/include/audio_coding_module.h
View File

@ -823,6 +823,8 @@ class AudioCodingModule {
virtual void GetDecodingCallStatistics(
AudioDecodingCallStats* call_stats) const = 0;
virtual ANAStats GetANAStats() const = 0;
};
} // namespace webrtc

20
webrtc/pc/statscollector.cc
View File

@ -224,6 +224,26 @@ void ExtractStats(const cricket::VoiceSenderInfo& info, StatsReport* report) {
}
}
report->AddString(StatsReport::kStatsValueNameMediaType, "audio");
if (info.ana_statistics.bitrate_action_counter) {
report->AddInt(StatsReport::kStatsValueNameAnaBitrateActionCounter,
*info.ana_statistics.bitrate_action_counter);
}
if (info.ana_statistics.channel_action_counter) {
report->AddInt(StatsReport::kStatsValueNameAnaChannelActionCounter,
*info.ana_statistics.channel_action_counter);
}
if (info.ana_statistics.dtx_action_counter) {
report->AddInt(StatsReport::kStatsValueNameAnaDtxActionCounter,
*info.ana_statistics.dtx_action_counter);
}
if (info.ana_statistics.fec_action_counter) {
report->AddInt(StatsReport::kStatsValueNameAnaFecActionCounter,
*info.ana_statistics.fec_action_counter);
}
if (info.ana_statistics.frame_length_action_counter) {
report->AddInt(StatsReport::kStatsValueNameAnaFrameLengthActionCounter,
*info.ana_statistics.frame_length_action_counter);
}
}
void ExtractStats(const cricket::VideoReceiverInfo& info, StatsReport* report) {

42
webrtc/pc/statscollector_unittest.cc
View File

@ -430,6 +430,37 @@ void VerifyVoiceSenderInfoReport(const StatsReport* report,
report, StatsReport::kStatsValueNameTypingNoiseState, &value_in_report));
std::string typing_detected = sinfo.typing_noise_detected ? "true" : "false";
EXPECT_EQ(typing_detected, value_in_report);
EXPECT_TRUE(GetValue(report,
StatsReport::kStatsValueNameAnaBitrateActionCounter,
&value_in_report));
ASSERT_TRUE(sinfo.ana_statistics.bitrate_action_counter);
EXPECT_EQ(
rtc::ToString<uint32_t>(*sinfo.ana_statistics.bitrate_action_counter),
value_in_report);
EXPECT_TRUE(GetValue(report,
StatsReport::kStatsValueNameAnaChannelActionCounter,
&value_in_report));
ASSERT_TRUE(sinfo.ana_statistics.channel_action_counter);
EXPECT_EQ(
rtc::ToString<uint32_t>(*sinfo.ana_statistics.channel_action_counter),
value_in_report);
EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameAnaDtxActionCounter,
&value_in_report));
ASSERT_TRUE(sinfo.ana_statistics.dtx_action_counter);
EXPECT_EQ(rtc::ToString<uint32_t>(*sinfo.ana_statistics.dtx_action_counter),
value_in_report);
EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameAnaFecActionCounter,
&value_in_report));
ASSERT_TRUE(sinfo.ana_statistics.fec_action_counter);
EXPECT_EQ(rtc::ToString<uint32_t>(*sinfo.ana_statistics.fec_action_counter),
value_in_report);
EXPECT_TRUE(GetValue(report,
StatsReport::kStatsValueNameAnaFrameLengthActionCounter,
&value_in_report));
ASSERT_TRUE(sinfo.ana_statistics.frame_length_action_counter);
EXPECT_EQ(rtc::ToString<uint32_t>(
*sinfo.ana_statistics.frame_length_action_counter),
value_in_report);
}
// Helper methods to avoid duplication of code.
@ -450,6 +481,16 @@ void InitVoiceSenderInfo(cricket::VoiceSenderInfo* voice_sender_info) {
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 =
rtc::Optional<uint32_t>(113);
voice_sender_info->ana_statistics.channel_action_counter =
rtc::Optional<uint32_t>(114);
voice_sender_info->ana_statistics.dtx_action_counter =
rtc::Optional<uint32_t>(115);
voice_sender_info->ana_statistics.fec_action_counter =
rtc::Optional<uint32_t>(116);
voice_sender_info->ana_statistics.frame_length_action_counter =
rtc::Optional<uint32_t>(117);
}
void UpdateVoiceSenderInfoFromAudioTrack(
@ -1988,6 +2029,7 @@ TEST_F(StatsCollectorTest, TwoLocalTracksWithSameSsrc) {
stats.AddLocalAudioTrack(audio_track_, kSsrcOfTrack);
cricket::VoiceSenderInfo voice_sender_info;
InitVoiceSenderInfo(&voice_sender_info);
voice_sender_info.add_ssrc(kSsrcOfTrack);
cricket::VoiceMediaInfo stats_read;

1
webrtc/test/mock_voe_channel_proxy.h
View File

@ -52,6 +52,7 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>());
MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics());
MOCK_CONST_METHOD0(GetDecodingCallStatistics, AudioDecodingCallStats());
MOCK_CONST_METHOD0(GetANAStatistics, ANAStats());
MOCK_CONST_METHOD0(GetSpeechOutputLevel, int());
MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int());
MOCK_CONST_METHOD0(GetTotalOutputEnergy, double());

4
webrtc/voice_engine/channel.cc
View File

@ -2856,6 +2856,10 @@ void Channel::GetDecodingCallStatistics(AudioDecodingCallStats* stats) const {
audio_coding_->GetDecodingCallStatistics(stats);
}
ANAStats Channel::GetANAStatistics() const {
return audio_coding_->GetANAStats();
}
uint32_t Channel::GetDelayEstimate() const {
rtc::CritScope lock(&video_sync_lock_);
return audio_coding_->FilteredCurrentDelayMs() + playout_delay_ms_;

1
webrtc/voice_engine/channel.h
View File

@ -268,6 +268,7 @@ class Channel
// Stats.
int GetNetworkStatistics(NetworkStatistics& stats);
void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
ANAStats GetANAStatistics() const;
// Audio+Video Sync.
uint32_t GetDelayEstimate() const;

5
webrtc/voice_engine/channel_proxy.cc
View File

@ -144,6 +144,11 @@ AudioDecodingCallStats ChannelProxy::GetDecodingCallStatistics() const {
return stats;
}
ANAStats ChannelProxy::GetANAStatistics() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->GetANAStatistics();
}
int ChannelProxy::GetSpeechOutputLevel() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->GetSpeechOutputLevel();

1
webrtc/voice_engine/channel_proxy.h
View File

@ -81,6 +81,7 @@ class ChannelProxy : public RtpPacketSinkInterface {
virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
virtual NetworkStatistics GetNetworkStatistics() const;
virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
virtual ANAStats GetANAStatistics() const;
virtual int GetSpeechOutputLevel() const;
virtual int GetSpeechOutputLevelFullRange() const;
// See description of "totalAudioEnergy" in the WebRTC stats spec: