chromium/third_party/blink/renderer/modules/webrtc/webrtc_audio_device_impl.cc

// Copyright 2013 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#include "third_party/blink/renderer/modules/webrtc/webrtc_audio_device_impl.h"

#include "base/containers/contains.h"
#include "base/logging.h"
#include "base/metrics/histogram_macros.h"
#include "base/strings/stringprintf.h"
#include "base/trace_event/trace_event.h"
#include "media/base/audio_bus.h"
#include "media/base/audio_parameters.h"
#include "media/base/audio_timestamp_helper.h"
#include "media/base/sample_rates.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
#include "third_party/blink/renderer/modules/mediastream/processed_local_audio_source.h"
#include "third_party/blink/renderer/modules/webrtc/webrtc_audio_renderer.h"

AudioParameters;
ChannelLayout;

namespace blink {

namespace {

void SendLogMessage(const std::string& message) { … }

}  // namespace

WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()
    : … { … }

WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl() { … }

void WebRtcAudioDeviceImpl::RenderData(
    media::AudioBus* audio_bus,
    int sample_rate,
    base::TimeDelta audio_delay,
    base::TimeDelta* current_time,
    const media::AudioGlitchInfo& glitch_info) { … }

void WebRtcAudioDeviceImpl::RemoveAudioRenderer(
    blink::WebRtcAudioRenderer* renderer) { … }

void WebRtcAudioDeviceImpl::AudioRendererThreadStopped() { … }

void WebRtcAudioDeviceImpl::SetOutputDeviceForAec(
    const String& output_device_id) { … }

int32_t WebRtcAudioDeviceImpl::RegisterAudioCallback(
    webrtc::AudioTransport* audio_callback) { … }

int32_t WebRtcAudioDeviceImpl::Init() { … }

int32_t WebRtcAudioDeviceImpl::Terminate() { … }

bool WebRtcAudioDeviceImpl::Initialized() const { … }

int32_t WebRtcAudioDeviceImpl::PlayoutIsAvailable(bool* available) { … }

bool WebRtcAudioDeviceImpl::PlayoutIsInitialized() const { … }

int32_t WebRtcAudioDeviceImpl::RecordingIsAvailable(bool* available) { … }

bool WebRtcAudioDeviceImpl::RecordingIsInitialized() const { … }

int32_t WebRtcAudioDeviceImpl::StartPlayout() { … }

int32_t WebRtcAudioDeviceImpl::StopPlayout() { … }

bool WebRtcAudioDeviceImpl::Playing() const { … }

int32_t WebRtcAudioDeviceImpl::StartRecording() { … }

int32_t WebRtcAudioDeviceImpl::StopRecording() { … }

bool WebRtcAudioDeviceImpl::Recording() const { … }

int32_t WebRtcAudioDeviceImpl::PlayoutDelay(uint16_t* delay_ms) const { … }

bool WebRtcAudioDeviceImpl::SetAudioRenderer(
    blink::WebRtcAudioRenderer* renderer) { … }

void WebRtcAudioDeviceImpl::AddAudioCapturer(
    ProcessedLocalAudioSource* capturer) { … }

void WebRtcAudioDeviceImpl::RemoveAudioCapturer(
    ProcessedLocalAudioSource* capturer) { … }

void WebRtcAudioDeviceImpl::AddPlayoutSink(
    blink::WebRtcPlayoutDataSource::Sink* sink) { … }

void WebRtcAudioDeviceImpl::RemovePlayoutSink(
    blink::WebRtcPlayoutDataSource::Sink* sink) { … }

std::optional<webrtc::AudioDeviceModule::Stats>
WebRtcAudioDeviceImpl::GetStats() const { … }

base::UnguessableToken
WebRtcAudioDeviceImpl::GetAuthorizedDeviceSessionIdForAudioRenderer() { … }

}  // namespace blink