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/*
* Copyright (c) 2019 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 "test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h"
#include "api/stats/rtc_stats.h"
#include "api/stats/rtcstats_objects.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace webrtc_pc_e2e {
void DefaultAudioQualityAnalyzer::Start(std::string test_case_name,
TrackIdStreamInfoMap* analyzer_helper) {
test_case_name_ = std::move(test_case_name);
analyzer_helper_ = analyzer_helper;
}
void DefaultAudioQualityAnalyzer::OnStatsReports(
absl::string_view pc_label,
const rtc::scoped_refptr<const RTCStatsReport>& report) {
// TODO(https://crbug.com/webrtc/11789): use "inbound-rtp" instead of "track"
// stats when required audio metrics moved there
auto stats = report->GetStatsOfType<RTCMediaStreamTrackStats>();
for (auto& stat : stats) {
if (!stat->kind.is_defined() ||
!(*stat->kind == RTCMediaStreamTrackKind::kAudio) ||
!*stat->remote_source) {
continue;
}
StatsSample sample;
sample.total_samples_received =
stat->total_samples_received.ValueOrDefault(0ul);
sample.concealed_samples = stat->concealed_samples.ValueOrDefault(0ul);
sample.removed_samples_for_acceleration =
stat->removed_samples_for_acceleration.ValueOrDefault(0ul);
sample.inserted_samples_for_deceleration =
stat->inserted_samples_for_deceleration.ValueOrDefault(0ul);
sample.silent_concealed_samples =
stat->silent_concealed_samples.ValueOrDefault(0ul);
sample.jitter_buffer_delay =
TimeDelta::Seconds(stat->jitter_buffer_delay.ValueOrDefault(0.));
sample.jitter_buffer_target_delay =
TimeDelta::Seconds(stat->jitter_buffer_target_delay.ValueOrDefault(0.));
sample.jitter_buffer_emitted_count =
stat->jitter_buffer_emitted_count.ValueOrDefault(0ul);
const std::string stream_label = std::string(
analyzer_helper_->GetStreamLabelFromTrackId(*stat->track_identifier));
MutexLock lock(&lock_);
StatsSample prev_sample = last_stats_sample_[stream_label];
RTC_CHECK_GE(sample.total_samples_received,
prev_sample.total_samples_received);
double total_samples_diff = static_cast<double>(
sample.total_samples_received - prev_sample.total_samples_received);
if (total_samples_diff == 0) {
return;
}
AudioStreamStats& audio_stream_stats = streams_stats_[stream_label];
audio_stream_stats.expand_rate.AddSample(
(sample.concealed_samples - prev_sample.concealed_samples) /
total_samples_diff);
audio_stream_stats.accelerate_rate.AddSample(
(sample.removed_samples_for_acceleration -
prev_sample.removed_samples_for_acceleration) /
total_samples_diff);
audio_stream_stats.preemptive_rate.AddSample(
(sample.inserted_samples_for_deceleration -
prev_sample.inserted_samples_for_deceleration) /
total_samples_diff);
int64_t speech_concealed_samples =
sample.concealed_samples - sample.silent_concealed_samples;
int64_t prev_speech_concealed_samples =
prev_sample.concealed_samples - prev_sample.silent_concealed_samples;
audio_stream_stats.speech_expand_rate.AddSample(
(speech_concealed_samples - prev_speech_concealed_samples) /
total_samples_diff);
int64_t jitter_buffer_emitted_count_diff =
sample.jitter_buffer_emitted_count -
prev_sample.jitter_buffer_emitted_count;
if (jitter_buffer_emitted_count_diff > 0) {
TimeDelta jitter_buffer_delay_diff =
sample.jitter_buffer_delay - prev_sample.jitter_buffer_delay;
TimeDelta jitter_buffer_target_delay_diff =
sample.jitter_buffer_target_delay -
prev_sample.jitter_buffer_target_delay;
audio_stream_stats.average_jitter_buffer_delay_ms.AddSample(
jitter_buffer_delay_diff.ms<double>() /
jitter_buffer_emitted_count_diff);
audio_stream_stats.preferred_buffer_size_ms.AddSample(
jitter_buffer_target_delay_diff.ms<double>() /
jitter_buffer_emitted_count_diff);
}
last_stats_sample_[stream_label] = sample;
}
}
std::string DefaultAudioQualityAnalyzer::GetTestCaseName(
const std::string& stream_label) const {
return test_case_name_ + "/" + stream_label;
}
void DefaultAudioQualityAnalyzer::Stop() {
using ::webrtc::test::ImproveDirection;
MutexLock lock(&lock_);
for (auto& item : streams_stats_) {
ReportResult("expand_rate", item.first, item.second.expand_rate, "unitless",
ImproveDirection::kSmallerIsBetter);
ReportResult("accelerate_rate", item.first, item.second.accelerate_rate,
"unitless", ImproveDirection::kSmallerIsBetter);
ReportResult("preemptive_rate", item.first, item.second.preemptive_rate,
"unitless", ImproveDirection::kSmallerIsBetter);
ReportResult("speech_expand_rate", item.first,
item.second.speech_expand_rate, "unitless",
ImproveDirection::kSmallerIsBetter);
ReportResult("average_jitter_buffer_delay_ms", item.first,
item.second.average_jitter_buffer_delay_ms, "ms",
ImproveDirection::kNone);
ReportResult("preferred_buffer_size_ms", item.first,
item.second.preferred_buffer_size_ms, "ms",
ImproveDirection::kNone);
}
}
std::map<std::string, AudioStreamStats>
DefaultAudioQualityAnalyzer::GetAudioStreamsStats() const {
MutexLock lock(&lock_);
return streams_stats_;
}
void DefaultAudioQualityAnalyzer::ReportResult(
const std::string& metric_name,
const std::string& stream_label,
const SamplesStatsCounter& counter,
const std::string& unit,
webrtc::test::ImproveDirection improve_direction) const {
test::PrintResultMeanAndError(
metric_name, /*modifier=*/"", GetTestCaseName(stream_label),
counter.IsEmpty() ? 0 : counter.GetAverage(),
counter.IsEmpty() ? 0 : counter.GetStandardDeviation(), unit,
/*important=*/false, improve_direction);
}
} // namespace webrtc_pc_e2e
} // namespace webrtc
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