Cast:Adding signaling and infrastructure for adjustable delay

media/cast/audio_receiver/audio_receiver.h

 // Copyright 2013 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 MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_
 #define MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_
 
 #include "base/basictypes.h"
 #include "base/callback.h"
 #include "base/macros.h"
@@ skipping 35 matching lines @@
                       public base::SupportsWeakPtr<AudioReceiver> {
  public:
   AudioReceiver(scoped_refptr<CastEnvironment> cast_environment,
                 const AudioReceiverConfig& audio_config,
                 transport::PacedPacketSender* const packet_sender);
 
   virtual ~AudioReceiver();
 
   // Extract a raw audio frame from the cast receiver.
   // Actual decoding will be preformed on a designated audio_decoder thread.
-  void GetRawAudioFrame(int number_of_10ms_blocks, int desired_frequency,
+  void GetRawAudioFrame(int number_of_10ms_blocks,
+                        int desired_frequency,
                         const AudioFrameDecodedCallback& callback);
 
   // Extract an encoded audio frame from the cast receiver.
   void GetEncodedAudioFrame(const AudioFrameEncodedCallback& callback);
 
   // Should only be called from the main cast thread.
   void IncomingPacket(scoped_ptr<Packet> packet);
 
+  // Update target audio delay used to compute the playout time. Rtcp
+  // will also be updated (will be included in all outgoing reports).
+  void SetTargetDelay(base::TimeDelta target_delay);
+
  protected:
-  void IncomingParsedRtpPacket(const uint8* payload_data, size_t payload_size,
+  void IncomingParsedRtpPacket(const uint8* payload_data,
+                               size_t payload_size,
                                const RtpCastHeader& rtp_header);
 
  private:
   friend class LocalRtpAudioData;
   friend class LocalRtpAudioFeedback;
 
   void CastFeedback(const RtcpCastMessage& cast_message);
 
   // Time to pull out the audio even though we are missing data.
   void PlayoutTimeout();
 
   bool PostEncodedAudioFrame(
       const AudioFrameEncodedCallback& callback,
       bool next_frame,
       scoped_ptr<transport::EncodedAudioFrame>* encoded_frame);
 
   // Actual decoding implementation - should be called under the audio decoder
   // thread.
-  void DecodeAudioFrameThread(int number_of_10ms_blocks, int desired_frequency,
+  void DecodeAudioFrameThread(int number_of_10ms_blocks,
+                              int desired_frequency,
                               const AudioFrameDecodedCallback callback);
   void ReturnDecodedFrameWithPlayoutDelay(
-      scoped_ptr<PcmAudioFrame> audio_frame, uint32 rtp_timestamp,
+      scoped_ptr<PcmAudioFrame> audio_frame,
+      uint32 rtp_timestamp,
       const AudioFrameDecodedCallback callback);
 
   // Return the playout time based on the current time and rtp timestamp.
   base::TimeTicks GetPlayoutTime(base::TimeTicks now, uint32 rtp_timestamp);
 
   void InitializeTimers();
 
   // Decrypts the data within the |audio_frame| and replaces the data with the
   // decrypted string.
   bool DecryptAudioFrame(scoped_ptr<transport::EncodedAudioFrame>* audio_frame);
@@ skipping 41 matching lines @@
   // it allows the event to be transmitted via RTCP.
   RtpTimestamp frame_id_to_rtp_timestamp_[256];
 
   DISALLOW_COPY_AND_ASSIGN(AudioReceiver);
 };
 
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_