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Expose jitterBufferEmittedCount in addition to the existing jitterBufferDelay for getStats().

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NetEq currently only passes `jitterBufferDelay` to `getStats()`. We need its paired `jitterBufferEmittedCount` denominator stat for the calculations to be accurate.

Bug: webrtc:10192
Change-Id: I655aea629026ce9101409c2e0f18c2fa57a1c3ab
Reviewed-on: https://webrtc-review.googlesource.com/c/117320
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#26276}
This commit is contained in:
Chen Xing 2019-01-15 15:46:29 +01:00 committed by Commit Bot
parent 876d1d3fed
commit 0acffb5b36
16 changed files with 32 additions and 3 deletions
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1
api/stats/rtcstats_objects.h
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@ -284,6 +284,7 @@ class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
// implemented for audio only).
// https://crbug.com/webrtc/8318
RTCStatsMember<double> jitter_buffer_delay;
RTCStatsMember<uint64_t> jitter_buffer_emitted_count;
// Video-only members
RTCStatsMember<uint32_t> frame_width;
RTCStatsMember<uint32_t> frame_height;

1
audio/audio_receive_stream.cc
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@ -207,6 +207,7 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
stats.jitter_buffer_delay_seconds =
static_cast<double>(ns.jitterBufferDelayMs) /
static_cast<double>(rtc::kNumMillisecsPerSec);
stats.jitter_buffer_emitted_count = ns.jitterBufferEmittedCount;
stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);

6
audio/audio_receive_stream_unittest.cc
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@ -67,8 +67,8 @@ const CallReceiveStatistics kCallStats = {345, 678, 901, 234,
const std::pair<int, SdpAudioFormat> kReceiveCodec =
{123, {"codec_name_recv", 96000, 0}};
const NetworkStatistics kNetworkStats = {
123, 456, false, 789012, 3456, 123, 456, 0, {}, 789, 12,
345, 678, 901, 0, -1, -1, -1, -1, -1, 0};
123, 456, false, 789012, 3456, 123, 456, 789, 0, {}, 789,
12, 345, 678, 901, 0, -1, -1, -1, -1, -1, 0};
const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
struct ConfigHelper {
@ -288,6 +288,8 @@ TEST(AudioReceiveStreamTest, GetStats) {
EXPECT_EQ(static_cast<double>(kNetworkStats.jitterBufferDelayMs) /
static_cast<double>(rtc::kNumMillisecsPerSec),
stats.jitter_buffer_delay_seconds);
EXPECT_EQ(kNetworkStats.jitterBufferEmittedCount,
stats.jitter_buffer_emitted_count);
EXPECT_EQ(Q14ToFloat(kNetworkStats.currentExpandRate), stats.expand_rate);
EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSpeechExpandRate),
stats.speech_expand_rate);

1
call/audio_receive_stream.h
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@ -56,6 +56,7 @@ class AudioReceiveStream {
uint64_t concealed_samples = 0;
uint64_t concealment_events = 0;
double jitter_buffer_delay_seconds = 0.0;
uint64_t jitter_buffer_emitted_count = 0;
// Stats below DO NOT correspond directly to anything in the WebRTC stats
float expand_rate = 0.0f;
float speech_expand_rate = 0.0f;

3
media/base/media_channel.h
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@ -449,7 +449,8 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
double total_output_duration = 0.0;
uint64_t concealed_samples = 0;
uint64_t concealment_events = 0;
double jitter_buffer_delay_seconds = 0;
double jitter_buffer_delay_seconds = 0.0;
uint64_t jitter_buffer_emitted_count = 0;
// Stats below DO NOT correspond directly to anything in the WebRTC stats
// fraction of synthesized audio inserted through expansion.
float expand_rate = 0.0f;

1
media/engine/webrtc_voice_engine.cc
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@ -2244,6 +2244,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
rinfo.concealed_samples = stats.concealed_samples;
rinfo.concealment_events = stats.concealment_events;
rinfo.jitter_buffer_delay_seconds = stats.jitter_buffer_delay_seconds;
rinfo.jitter_buffer_emitted_count = stats.jitter_buffer_emitted_count;
rinfo.expand_rate = stats.expand_rate;
rinfo.speech_expand_rate = stats.speech_expand_rate;
rinfo.secondary_decoded_rate = stats.secondary_decoded_rate;

3
media/engine/webrtc_voice_engine_unittest.cc
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@ -655,6 +655,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
stats.concealed_samples = 234;
stats.concealment_events = 12;
stats.jitter_buffer_delay_seconds = 34;
stats.jitter_buffer_emitted_count = 77;
stats.expand_rate = 5.67f;
stats.speech_expand_rate = 8.90f;
stats.secondary_decoded_rate = 1.23f;
@ -702,6 +703,8 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
EXPECT_EQ(info.concealment_events, stats.concealment_events);
EXPECT_EQ(info.jitter_buffer_delay_seconds,
stats.jitter_buffer_delay_seconds);
EXPECT_EQ(info.jitter_buffer_emitted_count,
stats.jitter_buffer_emitted_count);
EXPECT_EQ(info.expand_rate, stats.expand_rate);
EXPECT_EQ(info.speech_expand_rate, stats.speech_expand_rate);
EXPECT_EQ(info.secondary_decoded_rate, stats.secondary_decoded_rate);

2
modules/audio_coding/acm2/acm_receiver.cc
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@ -245,6 +245,8 @@ void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
acm_stat->concealedSamples = neteq_lifetime_stat.concealed_samples;
acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
acm_stat->jitterBufferEmittedCount =
neteq_lifetime_stat.jitter_buffer_emitted_count;
acm_stat->delayedPacketOutageSamples =
neteq_lifetime_stat.delayed_packet_outage_samples;

1
modules/audio_coding/include/audio_coding_module_typedefs.h
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@ -80,6 +80,7 @@ struct NetworkStatistics {
uint64_t concealedSamples;
uint64_t concealmentEvents;
uint64_t jitterBufferDelayMs;
uint64_t jitterBufferEmittedCount;
// Stats below DO NOT correspond directly to anything in the WebRTC stats
// Loss rate (network + late); fraction between 0 and 1, scaled to Q14.
uint16_t currentPacketLossRate;

1
modules/audio_coding/neteq/include/neteq.h
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@ -70,6 +70,7 @@ struct NetEqLifetimeStatistics {
uint64_t concealed_samples = 0;
uint64_t concealment_events = 0;
uint64_t jitter_buffer_delay_ms = 0;
uint64_t jitter_buffer_emitted_count = 0;
// Below stat is not part of the spec.
uint64_t voice_concealed_samples = 0;
uint64_t delayed_packet_outage_samples = 0;

3
modules/audio_coding/neteq/neteq_unittest.cc
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@ -1656,6 +1656,7 @@ void NetEqDecodingTestFaxMode::TestJitterBufferDelay(bool apply_packet_loss) {
int packets_sent = 0;
int packets_received = 0;
int expected_delay = 0;
uint64_t expected_emitted_count = 0;
while (packets_received < kNumPackets) {
// Insert packet.
if (packets_sent < kNumPackets) {
@ -1679,6 +1680,7 @@ void NetEqDecodingTestFaxMode::TestJitterBufferDelay(bool apply_packet_loss) {
// number of samples that are sent for play out.
int current_delay_ms = packets_delay * kPacketLenMs;
expected_delay += current_delay_ms * kSamples;
expected_emitted_count += kSamples;
}
}
@ -1690,6 +1692,7 @@ void NetEqDecodingTestFaxMode::TestJitterBufferDelay(bool apply_packet_loss) {
// Check jitter buffer delay.
NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
EXPECT_EQ(expected_delay, static_cast<int>(stats.jitter_buffer_delay_ms));
EXPECT_EQ(expected_emitted_count, stats.jitter_buffer_emitted_count);
}
TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithoutLoss) {

1
modules/audio_coding/neteq/statistics_calculator.cc
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@ -246,6 +246,7 @@ void StatisticsCalculator::IncreaseCounter(size_t num_samples, int fs_hz) {
void StatisticsCalculator::JitterBufferDelay(size_t num_samples,
uint64_t waiting_time_ms) {
lifetime_stats_.jitter_buffer_delay_ms += waiting_time_ms * num_samples;
lifetime_stats_.jitter_buffer_emitted_count += num_samples;
}
void StatisticsCalculator::SecondaryDecodedSamples(int num_samples) {

2
pc/rtc_stats_collector.cc
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@ -445,6 +445,8 @@ ProduceMediaStreamTrackStatsFromVoiceReceiverInfo(
}
audio_track_stats->jitter_buffer_delay =
voice_receiver_info.jitter_buffer_delay_seconds;
audio_track_stats->jitter_buffer_emitted_count =
voice_receiver_info.jitter_buffer_emitted_count;
audio_track_stats->total_audio_energy =
voice_receiver_info.total_output_energy;
audio_track_stats->total_samples_received =

2
pc/rtc_stats_collector_unittest.cc
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@ -1425,6 +1425,7 @@ TEST_F(RTCStatsCollectorTest,
voice_receiver_info.concealed_samples = 123;
voice_receiver_info.concealment_events = 12;
voice_receiver_info.jitter_buffer_delay_seconds = 3456;
voice_receiver_info.jitter_buffer_emitted_count = 13;
voice_receiver_info.jitter_buffer_flushes = 7;
voice_receiver_info.delayed_packet_outage_samples = 15;
@ -1460,6 +1461,7 @@ TEST_F(RTCStatsCollectorTest,
expected_remote_audio_track.concealed_samples = 123;
expected_remote_audio_track.concealment_events = 12;
expected_remote_audio_track.jitter_buffer_delay = 3456;
expected_remote_audio_track.jitter_buffer_emitted_count = 13;
expected_remote_audio_track.jitter_buffer_flushes = 7;
expected_remote_audio_track.delayed_packet_outage_samples = 15;
ASSERT_TRUE(report->Get(expected_remote_audio_track.id()));

4
pc/rtc_stats_integrationtest.cc
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@ -611,6 +611,8 @@ class RTCStatsReportVerifier {
*media_stream_track.remote_source) {
verifier.TestMemberIsNonNegative<double>(
media_stream_track.jitter_buffer_delay);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.jitter_buffer_emitted_count);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.total_samples_received);
verifier.TestMemberIsNonNegative<uint64_t>(
@ -623,6 +625,8 @@ class RTCStatsReportVerifier {
media_stream_track.delayed_packet_outage_samples);
} else {
verifier.TestMemberIsUndefined(media_stream_track.jitter_buffer_delay);
verifier.TestMemberIsUndefined(
media_stream_track.jitter_buffer_emitted_count);
verifier.TestMemberIsUndefined(media_stream_track.total_samples_received);
verifier.TestMemberIsUndefined(media_stream_track.concealed_samples);
verifier.TestMemberIsUndefined(media_stream_track.concealment_events);

3
stats/rtcstats_objects.cc
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@ -360,6 +360,7 @@ WEBRTC_RTCSTATS_IMPL(RTCMediaStreamTrackStats, RTCStats, "track",
&detached,
&kind,
&jitter_buffer_delay,
&jitter_buffer_emitted_count,
&frame_width,
&frame_height,
&frames_per_second,
@ -398,6 +399,7 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(std::string&& id,
detached("detached"),
kind("kind", kind),
jitter_buffer_delay("jitterBufferDelay"),
jitter_buffer_emitted_count("jitterBufferEmittedCount"),
frame_width("frameWidth"),
frame_height("frameHeight"),
frames_per_second("framesPerSecond"),
@ -432,6 +434,7 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
detached(other.detached),
kind(other.kind),
jitter_buffer_delay(other.jitter_buffer_delay),
jitter_buffer_emitted_count(other.jitter_buffer_emitted_count),
frame_width(other.frame_width),
frame_height(other.frame_height),
frames_per_second(other.frames_per_second),