232 lines
8.3 KiB
C++
232 lines
8.3 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
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#define MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <atomic>
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#include "api/task_queue/task_queue_factory.h"
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#include "modules/audio_device/include/audio_device_defines.h"
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#include "rtc_base/buffer.h"
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#include "rtc_base/synchronization/mutex.h"
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#include "rtc_base/task_queue.h"
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#include "rtc_base/thread_annotations.h"
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#include "rtc_base/thread_checker.h"
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namespace webrtc {
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// Delta times between two successive playout callbacks are limited to this
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// value before added to an internal array.
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const size_t kMaxDeltaTimeInMs = 500;
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// TODO(henrika): remove when no longer used by external client.
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const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
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class AudioDeviceBuffer {
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public:
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enum LogState {
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LOG_START = 0,
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LOG_STOP,
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LOG_ACTIVE,
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};
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struct Stats {
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void ResetRecStats() {
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rec_callbacks = 0;
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rec_samples = 0;
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max_rec_level = 0;
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}
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void ResetPlayStats() {
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play_callbacks = 0;
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play_samples = 0;
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max_play_level = 0;
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}
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// Total number of recording callbacks where the source provides 10ms audio
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// data each time.
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uint64_t rec_callbacks = 0;
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// Total number of playback callbacks where the sink asks for 10ms audio
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// data each time.
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uint64_t play_callbacks = 0;
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// Total number of recorded audio samples.
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uint64_t rec_samples = 0;
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// Total number of played audio samples.
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uint64_t play_samples = 0;
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// Contains max level (max(abs(x))) of recorded audio packets over the last
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// 10 seconds where a new measurement is done twice per second. The level
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// is reset to zero at each call to LogStats().
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int16_t max_rec_level = 0;
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// Contains max level of recorded audio packets over the last 10 seconds
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// where a new measurement is done twice per second.
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int16_t max_play_level = 0;
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};
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explicit AudioDeviceBuffer(TaskQueueFactory* task_queue_factory);
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virtual ~AudioDeviceBuffer();
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int32_t RegisterAudioCallback(AudioTransport* audio_callback);
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void StartPlayout();
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void StartRecording();
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void StopPlayout();
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void StopRecording();
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int32_t SetRecordingSampleRate(uint32_t fsHz);
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int32_t SetPlayoutSampleRate(uint32_t fsHz);
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uint32_t RecordingSampleRate() const;
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uint32_t PlayoutSampleRate() const;
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int32_t SetRecordingChannels(size_t channels);
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int32_t SetPlayoutChannels(size_t channels);
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size_t RecordingChannels() const;
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size_t PlayoutChannels() const;
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virtual int32_t SetRecordedBuffer(const void* audio_buffer,
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size_t samples_per_channel);
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virtual void SetVQEData(int play_delay_ms, int rec_delay_ms);
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virtual int32_t DeliverRecordedData();
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uint32_t NewMicLevel() const;
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virtual int32_t RequestPlayoutData(size_t samples_per_channel);
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virtual int32_t GetPlayoutData(void* audio_buffer);
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int32_t SetTypingStatus(bool typing_status);
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private:
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// Starts/stops periodic logging of audio stats.
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void StartPeriodicLogging();
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void StopPeriodicLogging();
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// Called periodically on the internal thread created by the TaskQueue.
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// Updates some stats but dooes it on the task queue to ensure that access of
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// members is serialized hence avoiding usage of locks.
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// state = LOG_START => members are initialized and the timer starts.
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// state = LOG_STOP => no logs are printed and the timer stops.
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// state = LOG_ACTIVE => logs are printed and the timer is kept alive.
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void LogStats(LogState state);
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// Updates counters in each play/record callback. These counters are later
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// (periodically) read by LogStats() using a lock.
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void UpdateRecStats(int16_t max_abs, size_t samples_per_channel);
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void UpdatePlayStats(int16_t max_abs, size_t samples_per_channel);
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// Clears all members tracking stats for recording and playout.
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// These methods both run on the task queue.
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void ResetRecStats();
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void ResetPlayStats();
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// This object lives on the main (creating) thread and most methods are
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// called on that same thread. When audio has started some methods will be
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// called on either a native audio thread for playout or a native thread for
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// recording. Some members are not annotated since they are "protected by
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// design" and adding e.g. a race checker can cause failures for very few
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// edge cases and it is IMHO not worth the risk to use them in this class.
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// TODO(henrika): see if it is possible to refactor and annotate all members.
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// Main thread on which this object is created.
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rtc::ThreadChecker main_thread_checker_;
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Mutex lock_;
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// Task queue used to invoke LogStats() periodically. Tasks are executed on a
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// worker thread but it does not necessarily have to be the same thread for
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// each task.
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rtc::TaskQueue task_queue_;
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// Raw pointer to AudioTransport instance. Supplied to RegisterAudioCallback()
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// and it must outlive this object. It is not possible to change this member
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// while any media is active. It is possible to start media without calling
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// RegisterAudioCallback() but that will lead to ignored audio callbacks in
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// both directions where native audio will be active but no audio samples will
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// be transported.
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AudioTransport* audio_transport_cb_;
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// Sample rate in Hertz. Accessed atomically.
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std::atomic<uint32_t> rec_sample_rate_;
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std::atomic<uint32_t> play_sample_rate_;
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// Number of audio channels. Accessed atomically.
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std::atomic<size_t> rec_channels_;
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std::atomic<size_t> play_channels_;
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// Keeps track of if playout/recording are active or not. A combination
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// of these states are used to determine when to start and stop the timer.
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// Only used on the creating thread and not used to control any media flow.
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bool playing_ RTC_GUARDED_BY(main_thread_checker_);
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bool recording_ RTC_GUARDED_BY(main_thread_checker_);
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// Buffer used for audio samples to be played out. Size can be changed
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// dynamically. The 16-bit samples are interleaved, hence the size is
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// proportional to the number of channels.
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rtc::BufferT<int16_t> play_buffer_;
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// Byte buffer used for recorded audio samples. Size can be changed
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// dynamically.
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rtc::BufferT<int16_t> rec_buffer_;
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// Contains true of a key-press has been detected.
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bool typing_status_;
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// Delay values used by the AEC.
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int play_delay_ms_;
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int rec_delay_ms_;
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// Counts number of times LogStats() has been called.
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size_t num_stat_reports_ RTC_GUARDED_BY(task_queue_);
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// Time stamp of last timer task (drives logging).
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int64_t last_timer_task_time_ RTC_GUARDED_BY(task_queue_);
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// Counts number of audio callbacks modulo 50 to create a signal when
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// a new storage of audio stats shall be done.
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int16_t rec_stat_count_;
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int16_t play_stat_count_;
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// Time stamps of when playout and recording starts.
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int64_t play_start_time_ RTC_GUARDED_BY(main_thread_checker_);
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int64_t rec_start_time_ RTC_GUARDED_BY(main_thread_checker_);
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// Contains counters for playout and recording statistics.
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Stats stats_ RTC_GUARDED_BY(lock_);
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// Stores current stats at each timer task. Used to calculate differences
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// between two successive timer events.
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Stats last_stats_ RTC_GUARDED_BY(task_queue_);
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// Set to true at construction and modified to false as soon as one audio-
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// level estimate larger than zero is detected.
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bool only_silence_recorded_;
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// Set to true when logging of audio stats is enabled for the first time in
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// StartPeriodicLogging() and set to false by StopPeriodicLogging().
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// Setting this member to false prevents (possiby invalid) log messages from
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// being printed in the LogStats() task.
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bool log_stats_ RTC_GUARDED_BY(task_queue_);
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// Should *never* be defined in production builds. Only used for testing.
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// When defined, the output signal will be replaced by a sinus tone at 440Hz.
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#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
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double phase_;
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#endif
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
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