/*
 *  Copyright 2011 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 "webrtc/api/audiotrack.h"

#include "webrtc/base/checks.h"

using rtc::scoped_refptr;

namespace webrtc {

const char MediaStreamTrackInterface::kAudioKind[] = "audio";

// static
scoped_refptr<AudioTrack> AudioTrack::Create(
    const std::string& id,
    const scoped_refptr<AudioSourceInterface>& source) {
  return new rtc::RefCountedObject<AudioTrack>(id, source);
}

AudioTrack::AudioTrack(const std::string& label,
                       const scoped_refptr<AudioSourceInterface>& source)
    : MediaStreamTrack<AudioTrackInterface>(label), audio_source_(source) {
  if (audio_source_) {
    audio_source_->RegisterObserver(this);
    OnChanged();
  }
}

AudioTrack::~AudioTrack() {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  set_state(MediaStreamTrackInterface::kEnded);
  if (audio_source_)
    audio_source_->UnregisterObserver(this);
}

std::string AudioTrack::kind() const {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  return kAudioKind;
}

AudioSourceInterface* AudioTrack::GetSource() const {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  return audio_source_.get();
}

void AudioTrack::AddSink(AudioTrackSinkInterface* sink) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  if (audio_source_)
    audio_source_->AddSink(sink);
}

void AudioTrack::RemoveSink(AudioTrackSinkInterface* sink) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  if (audio_source_)
    audio_source_->RemoveSink(sink);
}

void AudioTrack::OnChanged() {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  if (state() == kFailed)
    return;  // We can't recover from this state (do we ever set it?).

  TrackState new_state = kInitializing;

  // |audio_source_| must be non-null if we ever get here.
  switch (audio_source_->state()) {
    case MediaSourceInterface::kLive:
    case MediaSourceInterface::kMuted:
      new_state = kLive;
      break;
    case MediaSourceInterface::kEnded:
      new_state = kEnded;
      break;
    case MediaSourceInterface::kInitializing:
    default:
      // use kInitializing.
      break;
  }

  set_state(new_state);
}

}  // namespace webrtc