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Fix standard GetStats to not modify NetEq state.

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Add a get_and_clear_legacy_stats flag to AudioReceiveStream::GetStats,
to distinguish calls from standard GetStats and legacy GetStats.

Add const method NetEq::CurrentNetworkStatistics to get current
values of stateless NetEq stats. Standard GetStats will then call this
method instead of NetEq::NetworkStatistics.

Bug: webrtc:11622
Change-Id: I3833a246a9e39b18c99657a738da22c6e2bd5f5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183600
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32092}
This commit is contained in:
Niels Möller 2020-09-14 10:47:50 +02:00 committed by Commit Bot
parent 71d7c8e3cd
commit 6b4d962947
29 changed files with 143 additions and 81 deletions
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3
api/neteq/neteq.h
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@ -274,6 +274,9 @@ class NetEq {
// after the call.
virtual int NetworkStatistics(NetEqNetworkStatistics* stats) = 0;
// Current values only, not resetting any state.
virtual NetEqNetworkStatistics CurrentNetworkStatistics() const = 0;
// Returns a copy of this class's lifetime statistics. These statistics are
// never reset.
virtual NetEqLifetimeStatistics GetLifetimeStatistics() const = 0;

5
audio/audio_receive_stream.cc
View File

@ -173,7 +173,8 @@ void AudioReceiveStream::Stop() {
audio_state()->RemoveReceivingStream(this);
}
webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats(
bool get_and_clear_legacy_stats) const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
webrtc::AudioReceiveStream::Stats stats;
stats.remote_ssrc = config_.rtp.remote_ssrc;
@ -210,7 +211,7 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
rtc::TimeMillis());
// Get jitter buffer and total delay (alg + jitter + playout) stats.
auto ns = channel_receive_->GetNetworkStatistics();
auto ns = channel_receive_->GetNetworkStatistics(get_and_clear_legacy_stats);
stats.fec_packets_received = ns.fecPacketsReceived;
stats.fec_packets_discarded = ns.fecPacketsDiscarded;
stats.jitter_buffer_ms = ns.currentBufferSize;

3
audio/audio_receive_stream.h
View File

@ -67,7 +67,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
void Reconfigure(const webrtc::AudioReceiveStream::Config& config) override;
void Start() override;
void Stop() override;
webrtc::AudioReceiveStream::Stats GetStats() const override;
webrtc::AudioReceiveStream::Stats GetStats(
bool get_and_clear_legacy_stats) const override;
void SetSink(AudioSinkInterface* sink) override;
void SetGain(float gain) override;
bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;

5
audio/audio_receive_stream_unittest.cc
View File

@ -146,7 +146,7 @@ struct ConfigHelper {
.WillOnce(Return(kTotalOutputEnergy));
EXPECT_CALL(*channel_receive_, GetTotalOutputDuration())
.WillOnce(Return(kTotalOutputDuration));
EXPECT_CALL(*channel_receive_, GetNetworkStatistics())
EXPECT_CALL(*channel_receive_, GetNetworkStatistics(_))
.WillOnce(Return(kNetworkStats));
EXPECT_CALL(*channel_receive_, GetDecodingCallStatistics())
.WillOnce(Return(kAudioDecodeStats));
@ -219,7 +219,8 @@ TEST(AudioReceiveStreamTest, GetStats) {
ConfigHelper helper(use_null_audio_processing);
auto recv_stream = helper.CreateAudioReceiveStream();
helper.SetupMockForGetStats();
AudioReceiveStream::Stats stats = recv_stream->GetStats();
AudioReceiveStream::Stats stats =
recv_stream->GetStats(/*get_and_clear_legacy_stats=*/true);
EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
EXPECT_EQ(kCallStats.payload_bytes_rcvd, stats.payload_bytes_rcvd);
EXPECT_EQ(kCallStats.header_and_padding_bytes_rcvd,

8
audio/channel_receive.cc
View File

@ -128,7 +128,8 @@ class ChannelReceive : public ChannelReceiveInterface {
double GetTotalOutputDuration() const override;
// Stats.
NetworkStatistics GetNetworkStatistics() const override;
NetworkStatistics GetNetworkStatistics(
bool get_and_clear_legacy_stats) const override;
AudioDecodingCallStats GetDecodingCallStatistics() const override;
// Audio+Video Sync.
@ -801,10 +802,11 @@ void ChannelReceive::SetDepacketizerToDecoderFrameTransformer(
InitFrameTransformerDelegate(std::move(frame_transformer));
}
NetworkStatistics ChannelReceive::GetNetworkStatistics() const {
NetworkStatistics ChannelReceive::GetNetworkStatistics(
bool get_and_clear_legacy_stats) const {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
NetworkStatistics stats;
acm_receiver_.GetNetworkStatistics(&stats);
acm_receiver_.GetNetworkStatistics(&stats, get_and_clear_legacy_stats);
return stats;
}

3
audio/channel_receive.h
View File

@ -99,7 +99,8 @@ class ChannelReceiveInterface : public RtpPacketSinkInterface {
virtual double GetTotalOutputDuration() const = 0;
// Stats.
virtual NetworkStatistics GetNetworkStatistics() const = 0;
virtual NetworkStatistics GetNetworkStatistics(
bool get_and_clear_legacy_stats) const = 0;
virtual AudioDecodingCallStats GetDecodingCallStatistics() const = 0;
// Audio+Video Sync.

5
audio/mock_voe_channel_proxy.h
View File

@ -35,7 +35,10 @@ class MockChannelReceive : public voe::ChannelReceiveInterface {
(override));
MOCK_METHOD(void, ResetReceiverCongestionControlObjects, (), (override));
MOCK_METHOD(CallReceiveStatistics, GetRTCPStatistics, (), (const, override));
MOCK_METHOD(NetworkStatistics, GetNetworkStatistics, (), (const, override));
MOCK_METHOD(NetworkStatistics,
GetNetworkStatistics,
(bool),
(const, override));
MOCK_METHOD(AudioDecodingCallStats,
GetDecodingCallStatistics,
(),

3
audio/test/audio_stats_test.cc
View File

@ -65,7 +65,8 @@ class NoLossTest : public AudioEndToEndTest {
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();
AudioReceiveStream::Stats recv_stats =
receive_stream()->GetStats(/*get_and_clear_legacy_stats=*/true);
EXPECT_PRED2(IsNear, kBytesSent, recv_stats.payload_bytes_rcvd);
EXPECT_PRED2(IsNear, kPacketsSent, recv_stats.packets_rcvd);
EXPECT_EQ(0u, recv_stats.packets_lost);

3
call/audio_receive_stream.h
View File

@ -167,7 +167,8 @@ class AudioReceiveStream {
// When a stream is stopped, it can't receive, process or deliver packets.
virtual void Stop() = 0;
virtual Stats GetStats() const = 0;
virtual Stats GetStats(bool get_and_clear_legacy_stats) const = 0;
Stats GetStats() { return GetStats(/*get_and_clear_legacy_stats=*/true); }
// Sets an audio sink that receives unmixed audio from the receive stream.
// Ownership of the sink is managed by the caller.

3
media/base/fake_media_engine.cc
View File

@ -183,7 +183,8 @@ absl::optional<int> FakeVoiceMediaChannel::GetBaseMinimumPlayoutDelayMs(
}
return absl::nullopt;
}
bool FakeVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
bool FakeVoiceMediaChannel::GetStats(VoiceMediaInfo* info,
bool get_and_clear_legacy_stats) {
return false;
}
void FakeVoiceMediaChannel::SetRawAudioSink(

2
media/base/fake_media_engine.h
View File

@ -349,7 +349,7 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
absl::optional<int> GetBaseMinimumPlayoutDelayMs(
uint32_t ssrc) const override;
bool GetStats(VoiceMediaInfo* info) override;
bool GetStats(VoiceMediaInfo* info, bool get_and_clear_legacy_stats) override;
void SetRawAudioSink(
uint32_t ssrc,

3
media/base/media_channel.h
View File

@ -834,7 +834,8 @@ class VoiceMediaChannel : public MediaChannel, public Delayable {
// DTMF event 0-9, *, #, A-D.
virtual bool InsertDtmf(uint32_t ssrc, int event, int duration) = 0;
// Gets quality stats for the channel.
virtual bool GetStats(VoiceMediaInfo* info) = 0;
virtual bool GetStats(VoiceMediaInfo* info,
bool get_and_clear_legacy_stats) = 0;
virtual void SetRawAudioSink(
uint32_t ssrc,

3
media/engine/fake_webrtc_call.cc
View File

@ -100,7 +100,8 @@ void FakeAudioReceiveStream::Reconfigure(
config_ = config;
}
webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats(
bool get_and_clear_legacy_stats) const {
return stats_;
}

3
media/engine/fake_webrtc_call.h
View File

@ -104,7 +104,8 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
void Start() override { started_ = true; }
void Stop() override { started_ = false; }
webrtc::AudioReceiveStream::Stats GetStats() const override;
webrtc::AudioReceiveStream::Stats GetStats(
bool get_and_clear_legacy_stats) const override;
void SetSink(webrtc::AudioSinkInterface* sink) override;
void SetGain(float gain) override;
bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override {

11
media/engine/webrtc_voice_engine.cc
View File

@ -1249,10 +1249,11 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
}
}
webrtc::AudioReceiveStream::Stats GetStats() const {
webrtc::AudioReceiveStream::Stats GetStats(
bool get_and_clear_legacy_stats) const {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
RTC_DCHECK(stream_);
return stream_->GetStats();
return stream_->GetStats(get_and_clear_legacy_stats);
}
void SetRawAudioSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) {
@ -2300,7 +2301,8 @@ void WebRtcVoiceMediaChannel::OnReadyToSend(bool ready) {
ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
}
bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info,
bool get_and_clear_legacy_stats) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::GetStats");
RTC_DCHECK(worker_thread_checker_.IsCurrent());
RTC_DCHECK(info);
@ -2353,7 +2355,8 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
continue;
}
}
webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats();
webrtc::AudioReceiveStream::Stats stats =
stream.second->GetStats(get_and_clear_legacy_stats);
VoiceReceiverInfo rinfo;
rinfo.add_ssrc(stats.remote_ssrc);
rinfo.payload_bytes_rcvd = stats.payload_bytes_rcvd;

2
media/engine/webrtc_voice_engine.h
View File

@ -195,7 +195,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
void OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) override;
void OnReadyToSend(bool ready) override;
bool GetStats(VoiceMediaInfo* info) override;
bool GetStats(VoiceMediaInfo* info, bool get_and_clear_legacy_stats) override;
// Set the audio sink for an existing stream.
void SetRawAudioSink(

21
media/engine/webrtc_voice_engine_unittest.cc
View File

@ -2346,7 +2346,8 @@ TEST_P(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
{
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
cricket::VoiceMediaInfo info;
EXPECT_EQ(true, channel_->GetStats(&info));
EXPECT_EQ(true,
channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true));
// We have added 4 send streams. We should see empty stats for all.
EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size());
@ -2365,7 +2366,8 @@ TEST_P(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
cricket::VoiceMediaInfo info;
EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY));
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
EXPECT_EQ(true, channel_->GetStats(&info));
EXPECT_EQ(true,
channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true));
EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size());
EXPECT_EQ(0u, info.receivers.size());
}
@ -2377,7 +2379,8 @@ TEST_P(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
SetAudioReceiveStreamStats();
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
EXPECT_EQ(true, channel_->GetStats(&info));
EXPECT_EQ(true,
channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true));
EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size());
EXPECT_EQ(1u, info.receivers.size());
VerifyVoiceReceiverInfo(info.receivers[0]);
@ -2550,7 +2553,8 @@ TEST_P(WebRtcVoiceEngineTestFake, GetStats) {
{
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
cricket::VoiceMediaInfo info;
EXPECT_EQ(true, channel_->GetStats(&info));
EXPECT_EQ(true,
channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true));
// We have added one send stream. We should see the stats we've set.
EXPECT_EQ(1u, info.senders.size());
@ -2565,7 +2569,8 @@ TEST_P(WebRtcVoiceEngineTestFake, GetStats) {
cricket::VoiceMediaInfo info;
SetSend(true);
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
EXPECT_EQ(true, channel_->GetStats(&info));
EXPECT_EQ(true,
channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true));
VerifyVoiceSenderInfo(info.senders[0], true);
VerifyVoiceSendRecvCodecs(info);
}
@ -2575,7 +2580,8 @@ TEST_P(WebRtcVoiceEngineTestFake, GetStats) {
cricket::VoiceMediaInfo info;
EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY));
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
EXPECT_EQ(true, channel_->GetStats(&info));
EXPECT_EQ(true,
channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true));
EXPECT_EQ(1u, info.senders.size());
EXPECT_EQ(0u, info.receivers.size());
}
@ -2587,7 +2593,8 @@ TEST_P(WebRtcVoiceEngineTestFake, GetStats) {
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
SetAudioReceiveStreamStats();
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
EXPECT_EQ(true, channel_->GetStats(&info));
EXPECT_EQ(true,
channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true));
EXPECT_EQ(1u, info.senders.size());
EXPECT_EQ(1u, info.receivers.size());
VerifyVoiceReceiverInfo(info.receivers[0]);

49
modules/audio_coding/acm2/acm_receiver.cc
View File

@ -244,26 +244,45 @@ absl::optional<std::pair<int, SdpAudioFormat>> AcmReceiver::LastDecoder()
return std::make_pair(last_decoder_->payload_type, last_decoder_->sdp_format);
}
void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) const {
void AcmReceiver::GetNetworkStatistics(
NetworkStatistics* acm_stat,
bool get_and_clear_legacy_stats /* = true */) const {
NetEqNetworkStatistics neteq_stat;
// NetEq function always returns zero, so we don't check the return value.
neteq_->NetworkStatistics(&neteq_stat);
if (get_and_clear_legacy_stats) {
// NetEq function always returns zero, so we don't check the return value.
neteq_->NetworkStatistics(&neteq_stat);
acm_stat->currentPacketLossRate = neteq_stat.packet_loss_rate;
acm_stat->currentExpandRate = neteq_stat.expand_rate;
acm_stat->currentSpeechExpandRate = neteq_stat.speech_expand_rate;
acm_stat->currentPreemptiveRate = neteq_stat.preemptive_rate;
acm_stat->currentAccelerateRate = neteq_stat.accelerate_rate;
acm_stat->currentSecondaryDecodedRate = neteq_stat.secondary_decoded_rate;
acm_stat->currentSecondaryDiscardedRate =
neteq_stat.secondary_discarded_rate;
acm_stat->addedSamples = neteq_stat.added_zero_samples;
acm_stat->meanWaitingTimeMs = neteq_stat.mean_waiting_time_ms;
acm_stat->medianWaitingTimeMs = neteq_stat.median_waiting_time_ms;
acm_stat->minWaitingTimeMs = neteq_stat.min_waiting_time_ms;
acm_stat->maxWaitingTimeMs = neteq_stat.max_waiting_time_ms;
} else {
neteq_stat = neteq_->CurrentNetworkStatistics();
acm_stat->currentPacketLossRate = 0;
acm_stat->currentExpandRate = 0;
acm_stat->currentSpeechExpandRate = 0;
acm_stat->currentPreemptiveRate = 0;
acm_stat->currentAccelerateRate = 0;
acm_stat->currentSecondaryDecodedRate = 0;
acm_stat->currentSecondaryDiscardedRate = 0;
acm_stat->addedSamples = 0;
acm_stat->meanWaitingTimeMs = -1;
acm_stat->medianWaitingTimeMs = -1;
acm_stat->minWaitingTimeMs = -1;
acm_stat->maxWaitingTimeMs = 1;
}
acm_stat->currentBufferSize = neteq_stat.current_buffer_size_ms;
acm_stat->preferredBufferSize = neteq_stat.preferred_buffer_size_ms;
acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found ? true : false;
acm_stat->currentPacketLossRate = neteq_stat.packet_loss_rate;
acm_stat->currentExpandRate = neteq_stat.expand_rate;
acm_stat->currentSpeechExpandRate = neteq_stat.speech_expand_rate;
acm_stat->currentPreemptiveRate = neteq_stat.preemptive_rate;
acm_stat->currentAccelerateRate = neteq_stat.accelerate_rate;
acm_stat->currentSecondaryDecodedRate = neteq_stat.secondary_decoded_rate;
acm_stat->currentSecondaryDiscardedRate = neteq_stat.secondary_discarded_rate;
acm_stat->addedSamples = neteq_stat.added_zero_samples;
acm_stat->meanWaitingTimeMs = neteq_stat.mean_waiting_time_ms;
acm_stat->medianWaitingTimeMs = neteq_stat.median_waiting_time_ms;
acm_stat->minWaitingTimeMs = neteq_stat.min_waiting_time_ms;
acm_stat->maxWaitingTimeMs = neteq_stat.max_waiting_time_ms;
NetEqLifetimeStatistics neteq_lifetime_stat = neteq_->GetLifetimeStatistics();
acm_stat->totalSamplesReceived = neteq_lifetime_stat.total_samples_received;

3
modules/audio_coding/acm2/acm_receiver.h
View File

@ -138,7 +138,8 @@ class AcmReceiver {
// Output:
// - statistics : The current network statistics.
//
void GetNetworkStatistics(NetworkStatistics* statistics) const;
void GetNetworkStatistics(NetworkStatistics* statistics,
bool get_and_clear_legacy_stats = true) const;
//
// Flushes the NetEq packet and speech buffers.

37
modules/audio_coding/neteq/neteq_impl.cc
View File

@ -387,17 +387,9 @@ int NetEqImpl::FilteredCurrentDelayMs() const {
int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
MutexLock lock(&mutex_);
assert(decoder_database_.get());
const size_t total_samples_in_buffers =
packet_buffer_->NumSamplesInBuffer(decoder_frame_length_) +
sync_buffer_->FutureLength();
assert(controller_.get());
stats->preferred_buffer_size_ms = controller_->TargetLevelMs();
stats->jitter_peaks_found = controller_->PeakFound();
stats_->GetNetworkStatistics(fs_hz_, total_samples_in_buffers,
decoder_frame_length_, stats);
*stats = CurrentNetworkStatisticsInternal();
stats_->GetNetworkStatistics(decoder_frame_length_, stats);
// Compensate for output delay chain.
stats->current_buffer_size_ms += output_delay_chain_ms_;
stats->preferred_buffer_size_ms += output_delay_chain_ms_;
stats->mean_waiting_time_ms += output_delay_chain_ms_;
stats->median_waiting_time_ms += output_delay_chain_ms_;
stats->min_waiting_time_ms += output_delay_chain_ms_;
@ -405,6 +397,31 @@ int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
return 0;
}
NetEqNetworkStatistics NetEqImpl::CurrentNetworkStatistics() const {
MutexLock lock(&mutex_);
return CurrentNetworkStatisticsInternal();
}
NetEqNetworkStatistics NetEqImpl::CurrentNetworkStatisticsInternal() const {
assert(decoder_database_.get());
NetEqNetworkStatistics stats;
const size_t total_samples_in_buffers =
packet_buffer_->NumSamplesInBuffer(decoder_frame_length_) +
sync_buffer_->FutureLength();
assert(controller_.get());
stats.preferred_buffer_size_ms = controller_->TargetLevelMs();
stats.jitter_peaks_found = controller_->PeakFound();
RTC_DCHECK_GT(fs_hz_, 0);
stats.current_buffer_size_ms =
static_cast<uint16_t>(total_samples_in_buffers * 1000 / fs_hz_);
// Compensate for output delay chain.
stats.current_buffer_size_ms += output_delay_chain_ms_;
stats.preferred_buffer_size_ms += output_delay_chain_ms_;
return stats;
}
NetEqLifetimeStatistics NetEqImpl::GetLifetimeStatistics() const {
MutexLock lock(&mutex_);
return stats_->GetLifetimeStatistics();

5
modules/audio_coding/neteq/neteq_impl.h
View File

@ -162,6 +162,8 @@ class NetEqImpl : public webrtc::NetEq {
// after the call.
int NetworkStatistics(NetEqNetworkStatistics* stats) override;
NetEqNetworkStatistics CurrentNetworkStatistics() const override;
NetEqLifetimeStatistics GetLifetimeStatistics() const override;
NetEqOperationsAndState GetOperationsAndState() const override;
@ -330,6 +332,9 @@ class NetEqImpl : public webrtc::NetEq {
virtual void UpdatePlcComponents(int fs_hz, size_t channels)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
NetEqNetworkStatistics CurrentNetworkStatisticsInternal() const
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
Clock* const clock_;
mutable Mutex mutex_;

7
modules/audio_coding/neteq/statistics_calculator.cc
View File

@ -312,16 +312,11 @@ void StatisticsCalculator::StoreWaitingTime(int waiting_time_ms) {
operations_and_state_.last_waiting_time_ms = waiting_time_ms;
}
void StatisticsCalculator::GetNetworkStatistics(int fs_hz,
size_t num_samples_in_buffers,
size_t samples_per_packet,
void StatisticsCalculator::GetNetworkStatistics(size_t samples_per_packet,
NetEqNetworkStatistics* stats) {
RTC_DCHECK_GT(fs_hz, 0);
RTC_DCHECK(stats);
stats->added_zero_samples = 0;
stats->current_buffer_size_ms =
static_cast<uint16_t>(num_samples_in_buffers * 1000 / fs_hz);
stats->packet_loss_rate =
CalculateQ14Ratio(lost_timestamps_, timestamps_since_last_report_);

10
modules/audio_coding/neteq/statistics_calculator.h
View File

@ -104,15 +104,11 @@ class StatisticsCalculator {
// period caused not by an actual packet loss, but by a delayed packet.
virtual void LogDelayedPacketOutageEvent(int num_samples, int fs_hz);
// Returns the current network statistics in |stats|. The current sample rate
// is |fs_hz|, the total number of samples in packet buffer and sync buffer
// yet to play out is |num_samples_in_buffers|, and the number of samples per
// packet is |samples_per_packet|. The method does not populate
// Returns the current network statistics in |stats|. The number of samples
// per packet is |samples_per_packet|. The method does not populate
// |preferred_buffer_size_ms|, |jitter_peaks_found| or |clockdrift_ppm|; use
// the PopulateDelayManagerStats method for those.
void GetNetworkStatistics(int fs_hz,
size_t num_samples_in_buffers,
size_t samples_per_packet,
void GetNetworkStatistics(size_t samples_per_packet,
NetEqNetworkStatistics* stats);
// Returns a copy of this class's lifetime statistics. These statistics are

11
modules/audio_coding/neteq/statistics_calculator_unittest.cc
View File

@ -70,14 +70,11 @@ TEST(StatisticsCalculator, ExpandedSamplesCorrection) {
constexpr int k10MsSamples = kSampleRateHz / 100;
constexpr int kPacketSizeMs = 20;
constexpr size_t kSamplesPerPacket = kPacketSizeMs * kSampleRateHz / 1000;
// Assume 2 packets in the buffer.
constexpr size_t kNumSamplesInBuffer = 2 * kSamplesPerPacket;
// Advance time by 10 ms.
stats.IncreaseCounter(k10MsSamples, kSampleRateHz);
stats.GetNetworkStatistics(kSampleRateHz, kNumSamplesInBuffer,
kSamplesPerPacket, &stats_output);
stats.GetNetworkStatistics(kSamplesPerPacket, &stats_output);
EXPECT_EQ(0u, stats_output.expand_rate);
EXPECT_EQ(0u, stats_output.speech_expand_rate);
@ -86,8 +83,7 @@ TEST(StatisticsCalculator, ExpandedSamplesCorrection) {
stats.ExpandedVoiceSamplesCorrection(-100);
stats.ExpandedNoiseSamplesCorrection(-100);
stats.IncreaseCounter(k10MsSamples, kSampleRateHz);
stats.GetNetworkStatistics(kSampleRateHz, kNumSamplesInBuffer,
kSamplesPerPacket, &stats_output);
stats.GetNetworkStatistics(kSamplesPerPacket, &stats_output);
// Expect no change, since negative values are disallowed.
EXPECT_EQ(0u, stats_output.expand_rate);
EXPECT_EQ(0u, stats_output.speech_expand_rate);
@ -96,8 +92,7 @@ TEST(StatisticsCalculator, ExpandedSamplesCorrection) {
stats.ExpandedVoiceSamplesCorrection(50);
stats.ExpandedNoiseSamplesCorrection(200);
stats.IncreaseCounter(k10MsSamples, kSampleRateHz);
stats.GetNetworkStatistics(kSampleRateHz, kNumSamplesInBuffer,
kSamplesPerPacket, &stats_output);
stats.GetNetworkStatistics(kSamplesPerPacket, &stats_output);
// Calculate expected rates in Q14. Expand rate is noise + voice, while
// speech expand rate is only voice.
EXPECT_EQ(((50u + 200u) << 14) / k10MsSamples, stats_output.expand_rate);

3
pc/rtc_stats_collector.cc
View File

@ -1955,7 +1955,8 @@ RTCStatsCollector::PrepareTransceiverStatsInfos_s_w() const {
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
for (const auto& entry : voice_stats) {
if (!entry.first->GetStats(entry.second.get())) {
if (!entry.first->GetStats(entry.second.get(),
/*get_and_clear_legacy_stats=*/false)) {
RTC_LOG(LS_WARNING) << "Failed to get voice stats.";
}
}

3
pc/stats_collector.cc
View File

@ -991,7 +991,8 @@ class VoiceMediaChannelStatsGatherer final : public MediaChannelStatsGatherer {
}
bool GetStatsOnWorkerThread() override {
return voice_media_channel_->GetStats(&voice_media_info);
return voice_media_channel_->GetStats(&voice_media_info,
/*get_and_clear_legacy_stats=*/true);
}
void ExtractStats(StatsCollector* collector) const override {

3
pc/test/fake_peer_connection_for_stats.h
View File

@ -36,7 +36,8 @@ class FakeVoiceMediaChannelForStats : public cricket::FakeVoiceMediaChannel {
}
// VoiceMediaChannel overrides.
bool GetStats(cricket::VoiceMediaInfo* info) override {
bool GetStats(cricket::VoiceMediaInfo* info,
bool get_and_clear_legacy_stats) override {
if (stats_) {
*info = *stats_;
return true;

4
test/scenario/audio_stream.cc
View File

@ -212,7 +212,9 @@ void ReceiveAudioStream::Stop() {
AudioReceiveStream::Stats ReceiveAudioStream::GetStats() const {
AudioReceiveStream::Stats result;
receiver_->SendTask([&] { result = receive_stream_->GetStats(); });
receiver_->SendTask([&] {
result = receive_stream_->GetStats(/*get_and_clear_legacy_stats=*/true);
});
return result;
}

3
video/video_analyzer.cc
View File

@ -522,7 +522,8 @@ void VideoAnalyzer::PollStats() {
}
if (audio_receive_stream_ != nullptr) {
AudioReceiveStream::Stats receive_stats = audio_receive_stream_->GetStats();
AudioReceiveStream::Stats receive_stats =
audio_receive_stream_->GetStats(/*get_and_clear_legacy_stats=*/true);
audio_expand_rate_.AddSample(receive_stats.expand_rate);
audio_accelerate_rate_.AddSample(receive_stats.accelerate_rate);
audio_jitter_buffer_ms_.AddSample(receive_stats.jitter_buffer_ms);