// Copyright (c) 2012 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#ifndef CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_
#define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_

#include <vector>

#include "base/callback.h"
#include "base/memory/ref_counted.h"
#include "base/synchronization/lock.h"
#include "base/threading/thread_checker.h"
#include "content/common/content_export.h"
#include "content/renderer/media/media_stream_audio_renderer.h"
#include "content/renderer/media/webrtc_audio_device_impl.h"
#include "content/renderer/media/webrtc_local_audio_track.h"

namespace media {
class AudioBus;
class AudioFifo;
class AudioOutputDevice;
class AudioParameters;
}

namespace content {

class WebRtcAudioCapturer;

// WebRtcLocalAudioRenderer is a MediaStreamAudioRenderer designed for rendering
// local audio media stream tracks,
// http://dev.w3.org/2011/webrtc/editor/getusermedia.html#mediastreamtrack
// It also implements media::AudioRendererSink::RenderCallback to render audio
// data provided from a WebRtcLocalAudioTrack source.
// When the audio layer in the browser process asks for data to render, this
// class provides the data by implementing the WebRtcAudioCapturerSink
// interface, i.e., we are a sink seen from the WebRtcAudioCapturer perspective.
// TODO(henrika): improve by using similar principles as in RTCVideoRenderer
// which register itself to the video track when the provider is started and
// deregisters itself when it is stopped.
// Tracking this at http://crbug.com/164813.
class CONTENT_EXPORT WebRtcLocalAudioRenderer
    : NON_EXPORTED_BASE(public MediaStreamAudioRenderer),
      NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
      NON_EXPORTED_BASE(public WebRtcAudioCapturerSink) {
 public:
  // Creates a local renderer and registers a capturing |source| object.
  // The |source| is owned by the WebRtcAudioDeviceImpl.
  // Called on the main thread.
  WebRtcLocalAudioRenderer(WebRtcLocalAudioTrack* audio_track,
                           int source_render_view_id,
                           int session_id,
                           int sample_rate,
                           int frames_per_buffer);

  // MediaStreamAudioRenderer implementation.
  // Called on the main thread.
  virtual void Start() OVERRIDE;
  virtual void Stop() OVERRIDE;
  virtual void Play() OVERRIDE;
  virtual void Pause() OVERRIDE;
  virtual void SetVolume(float volume) OVERRIDE;
  virtual base::TimeDelta GetCurrentRenderTime() const OVERRIDE;
  virtual bool IsLocalRenderer() const OVERRIDE;

  const base::TimeDelta& total_render_time() const {
    return total_render_time_;
  }

 protected:
  virtual ~WebRtcLocalAudioRenderer();

 private:
  // WebRtcAudioCapturerSink implementation.

  // Called on the AudioInputDevice worker thread.
  virtual int CaptureData(const std::vector<int>& channels,
                          const int16* audio_data,
                          int sample_rate,
                          int number_of_channels,
                          int number_of_frames,
                          int audio_delay_milliseconds,
                          int current_volume,
                          bool need_audio_processing,
                          bool key_pressed) OVERRIDE;

  // Can be called on different user thread.
  virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE;

  // media::AudioRendererSink::RenderCallback implementation.
  // Render() is called on the AudioOutputDevice thread and OnRenderError()
  // on the IO thread.
  virtual int Render(media::AudioBus* audio_bus,
                     int audio_delay_milliseconds) OVERRIDE;
  virtual void OnRenderError() OVERRIDE;

  void StartSink();

  // The audio track which provides data to render. Given that this class
  // implements local loopback, the audio track is getting data from a capture
  // instance like a selected microphone and forwards the recorded data to its
  // sinks. The recorded data is stored in a FIFO and consumed
  // by this class when the sink asks for new data.
  // The WebRtcAudioCapturer is today created by WebRtcAudioDeviceImpl.
  scoped_refptr<WebRtcLocalAudioTrack> audio_track_;

  // The render view in which the audio is rendered into |sink_|.
  const int source_render_view_id_;
  const int session_id_;

  // The sink (destination) for rendered audio.
  scoped_refptr<media::AudioOutputDevice> sink_;

  // Used to DCHECK that we are called on the correct thread.
  base::ThreadChecker thread_checker_;

  // Contains copies of captured audio frames.
  scoped_ptr<media::AudioFifo> loopback_fifo_;

  // Stores last time a render callback was received. The time difference
  // between a new time stamp and this value can be used to derive the
  // total render time.
  base::Time last_render_time_;

  // Keeps track of total time audio has been rendered.
  base::TimeDelta total_render_time_;

  // The audio parameters used by the renderer.
  media::AudioParameters audio_params_;

  // Set when playing, cleared when paused.
  bool playing_;

  // Protects |loopback_fifo_|, |playing_| and |sink_|.
  mutable base::Lock thread_lock_;

  // The preferred sample rate and buffer sizes provided via the ctor.
  const int sample_rate_;
  const int frames_per_buffer_;

  // The preferred device id of the output device or empty for the default
  // output device.
  const std::string output_device_id_;

  // Cache value for the volume.
  float volume_;

  // Flag to start the sink only once. Used to log correctly in UMA.
  bool sink_started_;

  DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer);
};

}  // namespace content

#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_