src/media/cast/audio_receiver/audio_decoder.h

   1 // Copyright 2013 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 #ifndef MEDIA_CAST_AUDIO_RECEIVER_AUDIO_DECODER_H_
   6 #define MEDIA_CAST_AUDIO_RECEIVER_AUDIO_DECODER_H_
   7
   8 #include "base/callback.h"
   9 #include "base/synchronization/lock.h"
  10 #include "media/cast/cast_config.h"
  11 #include "media/cast/cast_environment.h"
  12 #include "media/cast/framer/cast_message_builder.h"
  13 #include "media/cast/framer/frame_id_map.h"
  14 #include "media/cast/rtp_receiver/rtp_receiver_defines.h"
  15
  16 namespace webrtc {
  17 class AudioCodingModule;
  18 }
  19
  20 namespace media {
  21 namespace cast {
  22
  23 typedef std::map<uint32, uint32> FrameIdRtpTimestampMap;
  24
  25 // Thread safe class.
  26 class AudioDecoder {
  27  public:
  28   AudioDecoder(scoped_refptr<CastEnvironment> cast_environment,
  29                const AudioReceiverConfig& audio_config,
  30                RtpPayloadFeedback* incoming_payload_feedback);
  31   virtual ~AudioDecoder();
  32
  33   // Extract a raw audio frame from the decoder.
  34   // Set the number of desired 10ms blocks and frequency.
  35   // Should be called from the cast audio decoder thread; however that is not
  36   // required.
  37   bool GetRawAudioFrame(int number_of_10ms_blocks,
  38                         int desired_frequency,
  39                         PcmAudioFrame* audio_frame,
  40                         uint32* rtp_timestamp);
  41
  42   // Insert an RTP packet to the decoder.
  43   // Should be called from the main cast thread; however that is not required.
  44   void IncomingParsedRtpPacket(const uint8* payload_data,
  45                                size_t payload_size,
  46                                const RtpCastHeader& rtp_header);
  47
  48   bool TimeToSendNextCastMessage(base::TimeTicks* time_to_send);
  49   void SendCastMessage();
  50
  51  private:
  52   scoped_refptr<CastEnvironment> cast_environment_;
  53
  54   // The webrtc AudioCodingModule is thread safe.
  55   scoped_ptr<webrtc::AudioCodingModule> audio_decoder_;
  56
  57   FrameIdMap frame_id_map_;
  58   CastMessageBuilder cast_message_builder_;
  59
  60   base::Lock lock_;
  61   bool have_received_packets_;
  62   FrameIdRtpTimestampMap frame_id_rtp_timestamp_map_;
  63   uint32 last_played_out_timestamp_;
  64
  65   DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
  66 };
  67
  68 }  // namespace cast
  69 }  // namespace media
  70
  71 #endif  // MEDIA_CAST_AUDIO_RECEIVER_AUDIO_DECODER_H_