blob: 5c4a0638b321fbd9f804a311134dfdaa352b8071 [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/base/criticalsection.h"
#include "webrtc/base/task_queue.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class CriticalSectionWrapper;
// 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 {
virtual ~AudioDeviceBuffer();
void SetId(uint32_t id) {};
int32_t RegisterAudioCallback(AudioTransport* audio_callback);
int32_t InitPlayout();
int32_t InitRecording();
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 num_samples);
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 num_samples);
virtual int32_t GetPlayoutData(void* audio_buffer);
// TODO(henrika): these methods should not be used and does not contain any
// valid implementation. Investigate the possibility to either remove them
// or add a proper implementation if needed.
int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]);
int32_t StopInputFileRecording();
int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]);
int32_t StopOutputFileRecording();
int32_t SetTypingStatus(bool typing_status);
// Playout and recording parameters can change on the fly. e.g. at device
// switch. These methods ensures that the callback methods always use the
// latest parameters.
void UpdatePlayoutParameters();
void UpdateRecordingParameters();
// Posts the first delayed task in the task queue and starts the periodic
// timer.
void StartTimer();
// 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 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
// each task is serialized by the task queue.
void UpdateRecStats(const void* audio_buffer, size_t num_samples);
void UpdatePlayStats(const void* audio_buffer, size_t num_samples);
// Ensures that methods are called on the same thread as the thread that
// creates this object.
rtc::ThreadChecker thread_checker_;
// Raw pointer to AudioTransport instance. Supplied to RegisterAudioCallback()
// and it must outlive this object.
AudioTransport* audio_transport_cb_;
// TODO(henrika): given usage of thread checker, it should be possible to
// remove all locks in this class.
rtc::CriticalSection _critSect;
rtc::CriticalSection _critSectCb;
// 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_;
// Ensures that the timer is only started once.
bool timer_has_started_;
// 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_;
// selected recording channel (left/right/both)
AudioDeviceModule::ChannelType rec_channel_;
// Number of bytes per audio sample (2 or 4).
size_t rec_bytes_per_sample_;
size_t play_bytes_per_sample_;
// Number of audio samples/bytes per 10ms.
size_t rec_samples_per_10ms_;
size_t rec_bytes_per_10ms_;
size_t play_samples_per_10ms_;
size_t play_bytes_per_10ms_;
// Buffer used for recorded audio samples. Size is currently fixed
// but it should be changed to be dynamic and correspond to
// |play_bytes_per_10ms_|. TODO(henrika): avoid using fixed (max) size.
std::unique_ptr<int8_t[]> rec_buffer_;
// Buffer used for audio samples to be played out. Size is currently fixed
// but it should be changed to be dynamic and correspond to
// |play_bytes_per_10ms_|. TODO(henrika): avoid using fixed (max) size.
std::unique_ptr<int8_t[]> play_buffer_;
// AGC parameters.
uint32_t current_mic_level_;
uint32_t new_mic_level_;
// 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_;
// Contains a clock-drift measurement.
int clock_drift_;
// Counts number of times LogStats() has been called.
size_t num_stat_reports_;
// Total number of recording callbacks where the source provides 10ms audio
// data each time.
uint64_t rec_callbacks_;
// Total number of recording callbacks stored at the last timer task.
uint64_t last_rec_callbacks_;
// Total number of playback callbacks where the sink asks for 10ms audio
// data each time.
uint64_t play_callbacks_;
// Total number of playout callbacks stored at the last timer task.
uint64_t last_play_callbacks_;
// Total number of recorded audio samples.
uint64_t rec_samples_;
// Total number of recorded samples stored at the previous timer task.
uint64_t last_rec_samples_;
// Total number of played audio samples.
uint64_t play_samples_;
// 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 playout callback.
uint64_t last_playout_time_;
// An array where the position corresponds to time differences (in
// milliseconds) between two successive playout callbacks, and the stored
// value is the number of times a given time difference was found.
// Writing to the array is done without a lock since it is only read once at
// destruction when no audio is running.
uint32_t playout_diff_times_[kMaxDeltaTimeInMs + 1] = {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(). Only modified on the task
// queue thread.
int16_t max_rec_level_;
// 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_;
// 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_;
} // namespace webrtc