xref: /aosp_15_r20/external/webrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h (revision d9f758449e529ab9291ac668be2861e7a55c2422)
1 /*
2  *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #ifndef TEST_PC_E2E_ANALYZER_AUDIO_DEFAULT_AUDIO_QUALITY_ANALYZER_H_
12 #define TEST_PC_E2E_ANALYZER_AUDIO_DEFAULT_AUDIO_QUALITY_ANALYZER_H_
13 
14 #include <map>
15 #include <string>
16 
17 #include "absl/strings/string_view.h"
18 #include "api/numerics/samples_stats_counter.h"
19 #include "api/test/audio_quality_analyzer_interface.h"
20 #include "api/test/metrics/metrics_logger.h"
21 #include "api/test/track_id_stream_info_map.h"
22 #include "api/units/time_delta.h"
23 #include "rtc_base/synchronization/mutex.h"
24 
25 namespace webrtc {
26 namespace webrtc_pc_e2e {
27 
28 struct AudioStreamStats {
29   SamplesStatsCounter expand_rate;
30   SamplesStatsCounter accelerate_rate;
31   SamplesStatsCounter preemptive_rate;
32   SamplesStatsCounter speech_expand_rate;
33   SamplesStatsCounter average_jitter_buffer_delay_ms;
34   SamplesStatsCounter preferred_buffer_size_ms;
35 };
36 
37 class DefaultAudioQualityAnalyzer : public AudioQualityAnalyzerInterface {
38  public:
39   explicit DefaultAudioQualityAnalyzer(
40       test::MetricsLogger* const metrics_logger);
41 
42   void Start(std::string test_case_name,
43              TrackIdStreamInfoMap* analyzer_helper) override;
44   void OnStatsReports(
45       absl::string_view pc_label,
46       const rtc::scoped_refptr<const RTCStatsReport>& report) override;
47   void Stop() override;
48 
49   // Returns audio quality stats per stream label.
50   std::map<std::string, AudioStreamStats> GetAudioStreamsStats() const;
51 
52  private:
53   struct StatsSample {
54     uint64_t total_samples_received = 0;
55     uint64_t concealed_samples = 0;
56     uint64_t removed_samples_for_acceleration = 0;
57     uint64_t inserted_samples_for_deceleration = 0;
58     uint64_t silent_concealed_samples = 0;
59     TimeDelta jitter_buffer_delay = TimeDelta::Zero();
60     TimeDelta jitter_buffer_target_delay = TimeDelta::Zero();
61     uint64_t jitter_buffer_emitted_count = 0;
62   };
63 
64   std::string GetTestCaseName(const std::string& stream_label) const;
65 
66   test::MetricsLogger* const metrics_logger_;
67 
68   std::string test_case_name_;
69   TrackIdStreamInfoMap* analyzer_helper_;
70 
71   mutable Mutex lock_;
72   std::map<std::string, AudioStreamStats> streams_stats_ RTC_GUARDED_BY(lock_);
73   std::map<std::string, TrackIdStreamInfoMap::StreamInfo> stream_info_
74       RTC_GUARDED_BY(lock_);
75   std::map<std::string, StatsSample> last_stats_sample_ RTC_GUARDED_BY(lock_);
76 };
77 
78 }  // namespace webrtc_pc_e2e
79 }  // namespace webrtc
80 
81 #endif  // TEST_PC_E2E_ANALYZER_AUDIO_DEFAULT_AUDIO_QUALITY_ANALYZER_H_
82