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/*
* Copyright (c) 2012 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 MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
#define MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
#include <stddef.h>
#include <stdint.h>
#include <atomic>
#include <memory>
#include "api/audio/audio_device_defines.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
#include "api/task_queue/task_queue_factory.h"
#include "rtc_base/buffer.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/timestamp_aligner.h"
namespace webrtc {
// Delta times between two successive playout callbacks are limited to this
// value before added to an internal array.
const size_t kMaxDeltaTimeInMs = 500;
// TODO(henrika): remove when no longer used by external client.
const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
class AudioDeviceBuffer {
public:
enum LogState {
LOG_START = 0,
LOG_STOP,
LOG_ACTIVE,
};
struct Stats {
void ResetRecStats() {
rec_callbacks = 0;
rec_samples = 0;
max_rec_level = 0;
}
void ResetPlayStats() {
play_callbacks = 0;
play_samples = 0;
max_play_level = 0;
}
// Total number of recording callbacks where the source provides 10ms audio
// data each time.
uint64_t rec_callbacks = 0;
// Total number of playback callbacks where the sink asks for 10ms audio
// data each time.
uint64_t play_callbacks = 0;
// Total number of recorded audio samples.
uint64_t rec_samples = 0;
// Total number of played audio samples.
uint64_t play_samples = 0;
// Contains max level (max(abs(x))) of recorded audio packets over the last
// 10 seconds where a new measurement is done twice per second. The level
// is reset to zero at each call to LogStats().
int16_t max_rec_level = 0;
// Contains max level of recorded audio packets over the last 10 seconds
// where a new measurement is done twice per second.
int16_t max_play_level = 0;
};
// If `create_detached` is true, the created buffer can be used on another
// thread compared to the one on which it was created. It's useful for
// testing.
explicit AudioDeviceBuffer(TaskQueueFactory* task_queue_factory,
bool create_detached = false);
virtual ~AudioDeviceBuffer();
int32_t RegisterAudioCallback(AudioTransport* audio_callback);
void StartPlayout();
void StartRecording();
void StopPlayout();
void StopRecording();
int32_t SetRecordingSampleRate(uint32_t fsHz);
int32_t SetPlayoutSampleRate(uint32_t fsHz);
uint32_t RecordingSampleRate() const;
uint32_t PlayoutSampleRate() const;
int32_t SetRecordingChannels(size_t channels);
int32_t SetPlayoutChannels(size_t channels);
size_t RecordingChannels() const;
size_t PlayoutChannels() const;
// TODO(bugs.webrtc.org/13621) Deprecate this function
virtual int32_t SetRecordedBuffer(const void* audio_buffer,
size_t samples_per_channel);
virtual int32_t SetRecordedBuffer(
const void* audio_buffer,
size_t samples_per_channel,
absl::optional<int64_t> capture_timestamp_ns);
virtual void SetVQEData(int play_delay_ms, int rec_delay_ms);
virtual int32_t DeliverRecordedData();
uint32_t NewMicLevel() const;
virtual int32_t RequestPlayoutData(size_t samples_per_channel);
virtual int32_t GetPlayoutData(void* audio_buffer);
int32_t SetTypingStatus(bool typing_status);
private:
// Starts/stops periodic logging of audio stats.
void StartPeriodicLogging();
void StopPeriodicLogging();
// Called periodically on the internal thread created by the TaskQueue.
// 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. These counters are later
// (periodically) read by LogStats() using a lock.
void UpdateRecStats(int16_t max_abs, size_t samples_per_channel);
void UpdatePlayStats(int16_t max_abs, size_t samples_per_channel);
// Clears all members tracking stats for recording and playout.
// These methods both run on the task queue.
void ResetRecStats();
void ResetPlayStats();
// This object lives on the main (creating) thread and most methods are
// called on that same thread. When audio has started some methods will be
// called on either a native audio thread for playout or a native thread for
// recording. Some members are not annotated since they are "protected by
// design" and adding e.g. a race checker can cause failures for very few
// edge cases and it is IMHO not worth the risk to use them in this class.
// TODO(henrika): see if it is possible to refactor and annotate all members.
// Main thread on which this object is created.
SequenceChecker main_thread_checker_;
Mutex lock_;
// Task queue used to invoke LogStats() periodically. Tasks are executed on a
// worker thread but it does not necessarily have to be the same thread for
// each task.
std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue_;
// Raw pointer to AudioTransport instance. Supplied to RegisterAudioCallback()
// and it must outlive this object. It is not possible to change this member
// while any media is active. It is possible to start media without calling
// RegisterAudioCallback() but that will lead to ignored audio callbacks in
// both directions where native audio will be active but no audio samples will
// be transported.
AudioTransport* audio_transport_cb_;
// Sample rate in Hertz. Accessed atomically.
std::atomic<uint32_t> rec_sample_rate_;
std::atomic<uint32_t> play_sample_rate_;
// Number of audio channels. Accessed atomically.
std::atomic<size_t> rec_channels_;
std::atomic<size_t> play_channels_;
// 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_ RTC_GUARDED_BY(main_thread_checker_);
bool recording_ RTC_GUARDED_BY(main_thread_checker_);
// Buffer used for audio samples to be played out. Size can be changed
// dynamically. The 16-bit samples are interleaved, hence the size is
// proportional to the number of channels.
rtc::BufferT<int16_t> play_buffer_;
// Byte buffer used for recorded audio samples. Size can be changed
// dynamically.
rtc::BufferT<int16_t> rec_buffer_;
// Contains true of a key-press has been detected.
bool typing_status_;
// Delay values used by the AEC.
int play_delay_ms_;
int rec_delay_ms_;
// Capture timestamp.
absl::optional<int64_t> capture_timestamp_ns_;
// The last time the Timestamp Aligner was used to estimate clock offset
// between system clock and capture time from audio.
// This is used to prevent estimating the clock offset too often.
absl::optional<int64_t> align_offsync_estimation_time_;
// Counts number of times LogStats() has been called.
size_t num_stat_reports_ RTC_GUARDED_BY(task_queue_);
// Time stamp of last timer task (drives logging).
int64_t last_timer_task_time_ RTC_GUARDED_BY(task_queue_);
// Counts number of audio callbacks modulo 50 to create a signal when
// a new storage of audio stats shall be done.
int16_t rec_stat_count_;
int16_t play_stat_count_;
// Time stamps of when playout and recording starts.
int64_t play_start_time_ RTC_GUARDED_BY(main_thread_checker_);
int64_t rec_start_time_ RTC_GUARDED_BY(main_thread_checker_);
// Contains counters for playout and recording statistics.
Stats stats_ RTC_GUARDED_BY(lock_);
// Stores current stats at each timer task. Used to calculate differences
// between two successive timer events.
Stats last_stats_ RTC_GUARDED_BY(task_queue_);
// 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_;
// Set to true when logging of audio stats is enabled for the first time in
// StartPeriodicLogging() and set to false by StopPeriodicLogging().
// Setting this member to false prevents (possiby invalid) log messages from
// being printed in the LogStats() task.
bool log_stats_ RTC_GUARDED_BY(task_queue_);
// Used for converting capture timestaps (received from AudioRecordThread
// via AudioRecordJni::DataIsRecorded) to RTC clock.
rtc::TimestampAligner timestamp_aligner_;
// Should *never* be defined in production builds. Only used for testing.
// When defined, the output signal will be replaced by a sinus tone at 440Hz.
#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
double phase_;
#endif
};
} // namespace webrtc
#endif // MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_