chromium/third_party/webrtc/audio/audio_state.cc

/*
 *  Copyright (c) 2015 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 "audio/audio_state.h"

#include <algorithm>
#include <memory>
#include <utility>
#include <vector>

#include "api/audio/audio_device.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/time_delta.h"
#include "audio/audio_receive_stream.h"
#include "audio/audio_send_stream.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"

namespace webrtc {
namespace internal {

AudioState::AudioState(const AudioState::Config& config)
    : … { … }

AudioState::~AudioState() { … }

AudioProcessing* AudioState::audio_processing() { … }

AudioTransport* AudioState::audio_transport() { … }

void AudioState::AddReceivingStream(
    webrtc::AudioReceiveStreamInterface* stream) { … }

void AudioState::RemoveReceivingStream(
    webrtc::AudioReceiveStreamInterface* stream) { … }

void AudioState::AddSendingStream(webrtc::AudioSendStream* stream,
                                  int sample_rate_hz,
                                  size_t num_channels) { … }

void AudioState::RemoveSendingStream(webrtc::AudioSendStream* stream) { … }

void AudioState::SetPlayout(bool enabled) { … }

void AudioState::SetRecording(bool enabled) { … }

void AudioState::SetStereoChannelSwapping(bool enable) { … }

void AudioState::UpdateAudioTransportWithSendingStreams() { … }

void AudioState::UpdateNullAudioPollerState() { … }
}  // namespace internal

rtc::scoped_refptr<AudioState> AudioState::Create(
    const AudioState::Config& config) { … }
}  // namespace webrtc