audio_device_buffer.cc | Explore in Territory

/*
 *  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 "modules/audio_device/audio_device_buffer.h"

#include <string.h>

#include <cmath>
#include <cstddef>
#include <cstdint>

#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/metrics.h"

namespace webrtc {

static const char kTimerQueueName[] = …;

// Time between two sucessive calls to LogStats().
static const size_t kTimerIntervalInSeconds = …;
static const size_t kTimerIntervalInMilliseconds = …;
// Min time required to qualify an audio session as a "call". If playout or
// recording has been active for less than this time we will not store any
// logs or UMA stats but instead consider the call as too short.
static const size_t kMinValidCallTimeTimeInSeconds = …;
static const size_t kMinValidCallTimeTimeInMilliseconds = …;
#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
static const double k2Pi = 6.28318530717959;
#endif

AudioDeviceBuffer::AudioDeviceBuffer(TaskQueueFactory* task_queue_factory,
                                     bool create_detached)
    : … { … }

AudioDeviceBuffer::~AudioDeviceBuffer() { … }

int32_t AudioDeviceBuffer::RegisterAudioCallback(
    AudioTransport* audio_callback) { … }

void AudioDeviceBuffer::StartPlayout() { … }

void AudioDeviceBuffer::StartRecording() { … }

void AudioDeviceBuffer::StopPlayout() { … }

void AudioDeviceBuffer::StopRecording() { … }

int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz) { … }

int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz) { … }

uint32_t AudioDeviceBuffer::RecordingSampleRate() const { … }

uint32_t AudioDeviceBuffer::PlayoutSampleRate() const { … }

int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels) { … }

int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels) { … }

size_t AudioDeviceBuffer::RecordingChannels() const { … }

size_t AudioDeviceBuffer::PlayoutChannels() const { … }

int32_t AudioDeviceBuffer::SetTypingStatus(bool typing_status) { … }

void AudioDeviceBuffer::SetVQEData(int play_delay_ms, int rec_delay_ms) { … }

int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
                                             size_t samples_per_channel) { … }

int32_t AudioDeviceBuffer::SetRecordedBuffer(
    const void* audio_buffer,
    size_t samples_per_channel,
    absl::optional<int64_t> capture_timestamp_ns) { … }

int32_t AudioDeviceBuffer::DeliverRecordedData() { … }

int32_t AudioDeviceBuffer::RequestPlayoutData(size_t samples_per_channel) { … }

int32_t AudioDeviceBuffer::GetPlayoutData(void* audio_buffer) { … }

void AudioDeviceBuffer::StartPeriodicLogging() { … }

void AudioDeviceBuffer::StopPeriodicLogging() { … }

void AudioDeviceBuffer::LogStats(LogState state) { … }

void AudioDeviceBuffer::ResetRecStats() { … }

void AudioDeviceBuffer::ResetPlayStats() { … }

void AudioDeviceBuffer::UpdateRecStats(int16_t max_abs,
                                       size_t samples_per_channel) { … }

void AudioDeviceBuffer::UpdatePlayStats(int16_t max_abs,
                                        size_t samples_per_channel) { … }

}  // namespace webrtc
chromium/third_party/webrtc/modules/audio_device/audio_device_buffer.cc
