[Cast] Remove AudioDecoder's dependency on WebRTC, and refactor/clean-up AudioReceiver.

media/cast/audio_receiver/audio_decoder.h

Index: media/cast/audio_receiver/audio_decoder.h
diff --git a/media/cast/audio_receiver/audio_decoder.h b/media/cast/audio_receiver/audio_decoder.h
index 6bafdd1465b2aad4633b26d4ad8260f9bb4badae..6d8251017b9335664c16e93b9f78610f4278e869 100644
--- a/media/cast/audio_receiver/audio_decoder.h
+++ b/media/cast/audio_receiver/audio_decoder.h
@@ -6,61 +6,39 @@
 #define MEDIA_CAST_AUDIO_RECEIVER_AUDIO_DECODER_H_
 
 #include "base/callback.h"
-#include "base/synchronization/lock.h"
+#include "base/memory/ref_counted.h"
+#include "media/base/audio_bus.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/framer/cast_message_builder.h"
-#include "media/cast/framer/frame_id_map.h"
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
-
-namespace webrtc {
-class AudioCodingModule;
-}
+#include "media/cast/transport/cast_transport_config.h"
 
 namespace media {
 namespace cast {
 
-typedef std::map<uint32, uint32> FrameIdRtpTimestampMap;
-
-// Thread safe class.
 class AudioDecoder {
  public:
-  AudioDecoder(scoped_refptr<CastEnvironment> cast_environment,
-               const AudioReceiverConfig& audio_config,
-               RtpPayloadFeedback* incoming_payload_feedback);
+  AudioDecoder(const scoped_refptr<CastEnvironment>& cast_environment,
+               const AudioReceiverConfig& audio_config);
   virtual ~AudioDecoder();
 
-  // Extract a raw audio frame from the decoder.
-  // Set the number of desired 10ms blocks and frequency.
-  // Should be called from the cast audio decoder thread; however that is not
-  // required.
-  bool GetRawAudioFrame(int number_of_10ms_blocks,
-                        int desired_frequency,
-                        PcmAudioFrame* audio_frame,
-                        uint32* rtp_timestamp);
-
-  // Insert an RTP packet to the decoder.
-  // Should be called from the main cast thread; however that is not required.
-  void IncomingParsedRtpPacket(const uint8* payload_data,
-                               size_t payload_size,
-                               const RtpCastHeader& rtp_header);
+  CastInitializationStatus InitializationResult() const;

[hubbe, 2014/03/28 19:50:59]

Add a comment for this function.

[miu, 2014/03/28 23:51:58]

Done.

 
-  bool TimeToSendNextCastMessage(base::TimeTicks* time_to_send);
-  void SendCastMessage();
+  // In the normal case, |encoded_frame->frame_id| will be
+  // monotonically-increasing by 1 for each successive call to this method.
+  // When it is not, the decoder will assume one or more packets have been
+  // dropped, and will perform recovery actions.
+  typedef base::Callback<void(scoped_ptr<AudioBus>,

[hubbe, 2014/03/28 19:50:59]

I think you should move the typedef before the comment and possible add a
comment for the typedef as well.

[miu, 2014/03/28 23:51:58]

Done.  IIRC, Chromium style wants type decls before method definitions too.  I
meant to fix this, but forgot.

+                              bool is_continuous)> DecodeFrameCallback;
+  void DecodeFrame(scoped_ptr<transport::EncodedAudioFrame> encoded_frame,

[hubbe, 2014/03/28 19:50:59]

Why does this function take a callback at all?
Couldn't it just decode the audio in encoded_frame and return it?
If not, perhaps explain why in the comment?

[miu, 2014/03/28 23:51:58]

Done.

+                   const DecodeFrameCallback& callback);
 
  private:
-  scoped_refptr<CastEnvironment> cast_environment_;
-
-  // The webrtc AudioCodingModule is thread safe.
-  scoped_ptr<webrtc::AudioCodingModule> audio_decoder_;
-
-  FrameIdMap frame_id_map_;
-  CastMessageBuilder cast_message_builder_;
+  class ImplBase;
+  class OpusImpl;
+  class Pcm16Impl;
 
-  base::Lock lock_;
-  bool have_received_packets_;
-  FrameIdRtpTimestampMap frame_id_rtp_timestamp_map_;
-  uint32 last_played_out_timestamp_;
+  const scoped_refptr<CastEnvironment> cast_environment_;
+  scoped_refptr<ImplBase> impl_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
 };