blob: ed8b1f77228de3110dee5d74c9368a18b1778e13 [file] [log] [blame]
* 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.
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/rtc_base/buffer.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/rtc_base/task_queue.h"
#include "webrtc/rtc_base/thread_annotations.h"
#include "webrtc/rtc_base/thread_checker.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/typedefs.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 AudioDeviceObserver;
class AudioDeviceBuffer {
enum LogState {
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;
virtual ~AudioDeviceBuffer();
void SetId(uint32_t id) {};
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);
int32_t RecordingSampleRate() const;
int32_t PlayoutSampleRate() const;
int32_t SetRecordingChannels(size_t channels);
int32_t SetPlayoutChannels(size_t channels);
size_t RecordingChannels() const;
size_t PlayoutChannels() const;
int32_t SetRecordingChannel(const AudioDeviceModule::ChannelType channel);
int32_t RecordingChannel(AudioDeviceModule::ChannelType& channel) const;
virtual int32_t SetRecordedBuffer(const void* audio_buffer,
size_t samples_per_channel);
int32_t SetCurrentMicLevel(uint32_t level);
virtual void SetVQEData(int play_delay_ms, int rec_delay_ms, int clock_drift);
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);
// Called on iOS where the native audio layer can be interrupted by other
// audio applications. This method can then be used to reset internal
// states and detach thread checkers to allow for a new audio session where
// native callbacks may come from a new set of I/O threads.
void NativeAudioInterrupted();
// 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 failuries 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.
rtc::ThreadChecker main_thread_checker_;
// Native (platform specific) audio thread driving the playout side.
rtc::ThreadChecker playout_thread_checker_;
// Native (platform specific) audio thread driving the recording side.
rtc::ThreadChecker recording_thread_checker_;
rtc::CriticalSection 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.
rtc::TaskQueue 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 acive but no audio samples will
// be transported.
AudioTransport* audio_transport_cb_;
// The members below that are not annotated are protected by design. They are
// all set on the main thread (verified by |main_thread_checker_|) and then
// read on either the playout or recording audio thread. But, media will never
// be active when the member is set; hence no conflict exists. It is too
// complex to ensure and verify that this is actually the case.
// Sample rate in Hertz.
uint32_t rec_sample_rate_;
uint32_t play_sample_rate_;
// Number of audio channels.
size_t rec_channels_;
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_ACCESS_ON(main_thread_checker_);
bool recording_ RTC_ACCESS_ON(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_ RTC_ACCESS_ON(playout_thread_checker_);
// Byte buffer used for recorded audio samples. Size can be changed
// dynamically.
rtc::BufferT<int16_t> rec_buffer_ RTC_ACCESS_ON(recording_thread_checker_);
// AGC parameters.
#if !defined(WEBRTC_WIN)
uint32_t current_mic_level_ RTC_ACCESS_ON(recording_thread_checker_);
// Windows uses a dedicated thread for volume APIs.
uint32_t current_mic_level_;
uint32_t new_mic_level_ RTC_ACCESS_ON(recording_thread_checker_);
// Contains true of a key-press has been detected.
bool typing_status_ RTC_ACCESS_ON(recording_thread_checker_);
// Delay values used by the AEC.
int play_delay_ms_ RTC_ACCESS_ON(recording_thread_checker_);
int rec_delay_ms_ RTC_ACCESS_ON(recording_thread_checker_);
// Contains a clock-drift measurement.
int clock_drift_ RTC_ACCESS_ON(recording_thread_checker_);
// Counts number of times LogStats() has been called.
size_t num_stat_reports_ RTC_ACCESS_ON(task_queue_);
// Time stamp of last timer task (drives logging).
int64_t last_timer_task_time_ RTC_ACCESS_ON(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_ RTC_ACCESS_ON(recording_thread_checker_);
int16_t play_stat_count_ RTC_ACCESS_ON(playout_thread_checker_);
// Time stamps of when playout and recording starts.
int64_t play_start_time_ RTC_ACCESS_ON(main_thread_checker_);
int64_t rec_start_time_ RTC_ACCESS_ON(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_ACCESS_ON(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_ACCESS_ON(task_queue_);
// 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.
double phase_ RTC_ACCESS_ON(playout_thread_checker_);
} // namespace webrtc