| Index: webrtc/modules/audio_device/audio_device_buffer.h
|
| diff --git a/webrtc/modules/audio_device/audio_device_buffer.h b/webrtc/modules/audio_device/audio_device_buffer.h
|
| index 6967ebd757147762b5291abd25d8c1a7940e0b0e..7e9f3e3eec30105a5971727002fe9b8bb39d87eb 100644
|
| --- a/webrtc/modules/audio_device/audio_device_buffer.h
|
| +++ b/webrtc/modules/audio_device/audio_device_buffer.h
|
| @@ -30,14 +30,22 @@ class AudioDeviceObserver;
|
|
|
| class AudioDeviceBuffer {
|
| public:
|
| + enum LogState {
|
| + LOG_START = 0,
|
| + LOG_STOP,
|
| + LOG_ACTIVE,
|
| + };
|
| +
|
| AudioDeviceBuffer();
|
| virtual ~AudioDeviceBuffer();
|
|
|
| void SetId(uint32_t id) {};
|
| int32_t RegisterAudioCallback(AudioTransport* audio_callback);
|
|
|
| - int32_t InitPlayout();
|
| - int32_t InitRecording();
|
| + void StartPlayout();
|
| + void StartRecording();
|
| + void StopPlayout();
|
| + void StopRecording();
|
|
|
| int32_t SetRecordingSampleRate(uint32_t fsHz);
|
| int32_t SetPlayoutSampleRate(uint32_t fsHz);
|
| @@ -72,16 +80,17 @@ class AudioDeviceBuffer {
|
| int32_t SetTypingStatus(bool typing_status);
|
|
|
| private:
|
| - // Posts the first delayed task in the task queue and starts the periodic
|
| - // timer.
|
| - void StartTimer();
|
| + // Starts/stops periodic logging of audio stats.
|
| + void StartPeriodicLogging();
|
| + void StopPeriodicLogging();
|
|
|
| // Called periodically on the internal thread created by the TaskQueue.
|
| - void LogStats();
|
| -
|
| - // Clears all members tracking stats for recording and playout.
|
| - void ResetRecStats();
|
| - void ResetPlayStats();
|
| + // Updates some stats but dooes it on the task queue to ensure that access of
|
| + // members is serialized hence avoiding usage of locks.
|
| + // state = LOG_START => members are initialized and the timer starts.
|
| + // state = LOG_STOP => no logs are printed and the timer stops.
|
| + // state = LOG_ACTIVE => logs are printed and the timer is kept alive.
|
| + void LogStats(LogState state);
|
|
|
| // Updates counters in each play/record callback but does it on the task
|
| // queue to ensure that they can be read by LogStats() without any locks since
|
| @@ -89,6 +98,11 @@ class AudioDeviceBuffer {
|
| void UpdateRecStats(int16_t max_abs, size_t num_samples);
|
| void UpdatePlayStats(int16_t max_abs, size_t num_samples);
|
|
|
| + // Clears all members tracking stats for recording and playout.
|
| + // These methods both run on the task queue.
|
| + void ResetRecStats();
|
| + void ResetPlayStats();
|
| +
|
| // Ensures that methods are called on the same thread as the thread that
|
| // creates this object.
|
| rtc::ThreadChecker thread_checker_;
|
| @@ -107,8 +121,11 @@ class AudioDeviceBuffer {
|
| // each task.
|
| rtc::TaskQueue task_queue_;
|
|
|
| - // Ensures that the timer is only started once.
|
| - bool timer_has_started_;
|
| + // Keeps track of if playout/recording are active or not. A combination
|
| + // of these states are used to determine when to start and stop the timer.
|
| + // Only used on the creating thread and not used to control any media flow.
|
| + bool playing_;
|
| + bool recording_;
|
|
|
| // Sample rate in Hertz.
|
| uint32_t rec_sample_rate_;
|
| @@ -173,8 +190,8 @@ class AudioDeviceBuffer {
|
| // Total number of played samples stored at the previous timer task.
|
| uint64_t last_play_samples_;
|
|
|
| - // Time stamp of last stat report.
|
| - uint64_t last_log_stat_time_;
|
| + // Time stamp of last timer task (drives logging).
|
| + uint64_t last_timer_task_time_;
|
|
|
| // Time stamp of last playout callback.
|
| uint64_t last_playout_time_;
|
| @@ -196,18 +213,19 @@ class AudioDeviceBuffer {
|
| // where a new measurement is done twice per second.
|
| int16_t max_play_level_;
|
|
|
| - // Counts number of times we detect "no audio" corresponding to a case where
|
| - // all level measurements since the last log has been exactly zero.
|
| - // In other words: this counter is incremented only if 20 measurements
|
| - // (two per second) in a row equals zero. The member is only incremented on
|
| - // the task queue and max once every 10th second.
|
| - size_t num_rec_level_is_zero_;
|
| -
|
| // Counts number of audio callbacks modulo 50 to create a signal when
|
| // a new storage of audio stats shall be done.
|
| // Only updated on the OS-specific audio thread that drives audio.
|
| int16_t rec_stat_count_;
|
| int16_t play_stat_count_;
|
| +
|
| + // Time stamps of when playout and recording starts.
|
| + uint64_t play_start_time_;
|
| + uint64_t rec_start_time_;
|
| +
|
| + // Set to true at construction and modified to false as soon as one audio-
|
| + // level estimate larger than zero is detected.
|
| + bool only_silence_recorded_;
|
| };
|
|
|
| } // namespace webrtc
|
|
|