webrtc_audio_source_adapter.cc | Explore in Territory

// Copyright 2016 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#ifdef UNSAFE_BUFFERS_BUILD
// TODO(crbug.com/40285824): Remove this and convert code to safer constructs.
#pragma allow_unsafe_buffers
#endif

#include "remoting/protocol/webrtc_audio_source_adapter.h"

#include <utility>

#include "base/check_op.h"
#include "base/functional/bind.h"
#include "base/observer_list.h"
#include "base/synchronization/lock.h"
#include "base/task/single_thread_task_runner.h"
#include "base/threading/thread_checker.h"
#include "base/time/time.h"
#include "remoting/proto/audio.pb.h"
#include "remoting/protocol/audio_source.h"

namespace remoting::protocol {

static const int kChannels = …;
static const int kBytesPerSample = …;

// Frame size expected by webrtc::AudioTrackSinkInterface.
static constexpr base::TimeDelta kAudioFrameDuration = …;

class WebrtcAudioSourceAdapter::Core { … };

WebrtcAudioSourceAdapter::Core::Core() { … }

WebrtcAudioSourceAdapter::Core::~Core() { … }

void WebrtcAudioSourceAdapter::Core::Start(
    std::unique_ptr<AudioSource> audio_source) { … }

void WebrtcAudioSourceAdapter::Core::Pause(bool pause) { … }

void WebrtcAudioSourceAdapter::Core::AddSink(
    webrtc::AudioTrackSinkInterface* sink) { … }

void WebrtcAudioSourceAdapter::Core::RemoveSink(
    webrtc::AudioTrackSinkInterface* sink) { … }

void WebrtcAudioSourceAdapter::Core::OnAudioPacket(
    std::unique_ptr<AudioPacket> packet) { … }

WebrtcAudioSourceAdapter::WebrtcAudioSourceAdapter(
    scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner)
    : … { … }

WebrtcAudioSourceAdapter::~WebrtcAudioSourceAdapter() { … }

void WebrtcAudioSourceAdapter::Start(
    std::unique_ptr<AudioSource> audio_source) { … }

void WebrtcAudioSourceAdapter::Pause(bool pause) { … }

WebrtcAudioSourceAdapter::SourceState WebrtcAudioSourceAdapter::state() const { … }

bool WebrtcAudioSourceAdapter::remote() const { … }

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

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

void WebrtcAudioSourceAdapter::AddSink(webrtc::AudioTrackSinkInterface* sink) { … }
void WebrtcAudioSourceAdapter::RemoveSink(
    webrtc::AudioTrackSinkInterface* sink) { … }

void WebrtcAudioSourceAdapter::RegisterObserver(
    webrtc::ObserverInterface* observer) { … }
void WebrtcAudioSourceAdapter::UnregisterObserver(
    webrtc::ObserverInterface* observer) { … }

}  // namespace remoting::protocol
chromium/remoting/protocol/webrtc_audio_source_adapter.cc
