chromium/third_party/webrtc/pc/remote_audio_source.cc

/*
 *  Copyright 2014 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 "pc/remote_audio_source.h"

#include <stddef.h>

#include <memory>
#include <string>
#include <utility>

#include "absl/algorithm/container.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_format.h"
#include "rtc_base/trace_event.h"

namespace webrtc {

// This proxy is passed to the underlying media engine to receive audio data as
// they come in. The data will then be passed back up to the RemoteAudioSource
// which will fan it out to all the sinks that have been added to it.
class RemoteAudioSource::AudioDataProxy : public AudioSinkInterface { … };

RemoteAudioSource::RemoteAudioSource(
    TaskQueueBase* worker_thread,
    OnAudioChannelGoneAction on_audio_channel_gone_action)
    : … { … }

RemoteAudioSource::~RemoteAudioSource() { … }

void RemoteAudioSource::Start(
    cricket::VoiceMediaReceiveChannelInterface* media_channel,
    absl::optional<uint32_t> ssrc) { … }

void RemoteAudioSource::Stop(
    cricket::VoiceMediaReceiveChannelInterface* media_channel,
    absl::optional<uint32_t> ssrc) { … }

void RemoteAudioSource::SetState(SourceState new_state) { … }

MediaSourceInterface::SourceState RemoteAudioSource::state() const { … }

bool RemoteAudioSource::remote() const { … }

void RemoteAudioSource::SetVolume(double volume) { … }

void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) { … }

void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) { … }

void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) { … }

void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) { … }

void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) { … }

void RemoteAudioSource::OnAudioChannelGone() { … }

}  // namespace webrtc