blob: 019ab2e4da08a78ee4cf750eef6bb2b89a5c3ab8 [file] [log] [blame]
/*
* 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.
*/
#ifndef TEST_PC_E2E_ANALYZER_AUDIO_DEFAULT_AUDIO_QUALITY_ANALYZER_H_
#define TEST_PC_E2E_ANALYZER_AUDIO_DEFAULT_AUDIO_QUALITY_ANALYZER_H_
#include <map>
#include <string>
#include "absl/strings/string_view.h"
#include "api/stats_types.h"
#include "api/test/audio_quality_analyzer_interface.h"
#include "api/test/track_id_stream_label_map.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/numerics/samples_stats_counter.h"
namespace webrtc {
namespace webrtc_pc_e2e {
struct AudioStreamStats {
SamplesStatsCounter expand_rate;
SamplesStatsCounter accelerate_rate;
SamplesStatsCounter preemptive_rate;
SamplesStatsCounter speech_expand_rate;
SamplesStatsCounter preferred_buffer_size_ms;
};
// TODO(bugs.webrtc.org/10430): Migrate to the new GetStats as soon as
// bugs.webrtc.org/10428 is fixed.
class DefaultAudioQualityAnalyzer : public AudioQualityAnalyzerInterface {
public:
void Start(std::string test_case_name,
TrackIdStreamLabelMap* analyzer_helper) override;
void OnStatsReports(absl::string_view pc_label,
const StatsReports& stats_reports) override;
void Stop() override;
// Returns audio quality stats per stream label.
std::map<std::string, AudioStreamStats> GetAudioStreamsStats() const;
private:
const std::string& GetStreamLabelFromStatsReport(
const StatsReport* stats_report) const;
std::string GetTestCaseName(const std::string& stream_label) const;
void ReportResult(const std::string& metric_name,
const std::string& stream_label,
const SamplesStatsCounter& counter,
const std::string& unit) const;
std::string test_case_name_;
TrackIdStreamLabelMap* analyzer_helper_;
rtc::CriticalSection lock_;
std::map<std::string, AudioStreamStats> streams_stats_ RTC_GUARDED_BY(lock_);
};
} // namespace webrtc_pc_e2e
} // namespace webrtc
#endif // TEST_PC_E2E_ANALYZER_AUDIO_DEFAULT_AUDIO_QUALITY_ANALYZER_H_