[Cast] EncodedAudioFrame+EncodedVideoFrame+reference_time --> EncodedFrame

media/cast/audio_sender/audio_encoder.cc

 // 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.
 
 #include "media/cast/audio_sender/audio_encoder.h"
 
 #include <algorithm>
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
@@ skipping 42 matching lines @@
       rtp_timestamp, frame_id,
       static_cast<int>(frame_size), /* key_frame - unused */ false,
       /*target_bitrate - unused*/ 0);
 }
 
 }  // namespace
 
 
 // Base class that handles the common problem of feeding one or more AudioBus'
 // data into a buffer and then, once the buffer is full, encoding the signal and
-// emitting an EncodedAudioFrame via the FrameEncodedCallback.
+// emitting an EncodedFrame via the FrameEncodedCallback.
 //
 // Subclasses complete the implementation by handling the actual encoding
 // details.
 class AudioEncoder::ImplBase
     : public base::RefCountedThreadSafe<AudioEncoder::ImplBase> {
  public:
   ImplBase(const scoped_refptr<CastEnvironment>& cast_environment,
            transport::AudioCodec codec,
            int num_channels,
            int sampling_rate,
            const FrameEncodedCallback& callback)
       : cast_environment_(cast_environment),
         codec_(codec),
         num_channels_(num_channels),
         samples_per_frame_(sampling_rate / kFramesPerSecond),
         callback_(callback),
         cast_initialization_status_(STATUS_AUDIO_UNINITIALIZED),
         buffer_fill_end_(0),
         frame_id_(0),
         frame_rtp_timestamp_(0) {
+    // Support for max sampling rate of 48KHz, 2 channels, 100 ms duration.
+    const int kMaxSamplesTimesChannelsPerFrame = 48 * 2 * 100;
     if (num_channels_ <= 0 || samples_per_frame_ <= 0 ||
         sampling_rate % kFramesPerSecond != 0 ||
-        samples_per_frame_ * num_channels_ >
-            transport::EncodedAudioFrame::kMaxNumberOfSamples) {
+        samples_per_frame_ * num_channels_ > kMaxSamplesTimesChannelsPerFrame) {
       cast_initialization_status_ = STATUS_INVALID_AUDIO_CONFIGURATION;
     }
   }
 
   CastInitializationStatus InitializationResult() const {
     return cast_initialization_status_;
   }
 
   void EncodeAudio(scoped_ptr<AudioBus> audio_bus,
                    const base::TimeTicks& recorded_time) {
@@ skipping 35 matching lines @@
           samples_per_frame_ - buffer_fill_end_, audio_bus->frames() - src_pos);
       DCHECK_EQ(audio_bus->channels(), num_channels_);
       TransferSamplesIntoBuffer(
           audio_bus.get(), src_pos, buffer_fill_end_, num_samples_to_xfer);
       src_pos += num_samples_to_xfer;
       buffer_fill_end_ += num_samples_to_xfer;
 
       if (buffer_fill_end_ < samples_per_frame_)
         break;
 
-      scoped_ptr<transport::EncodedAudioFrame> audio_frame(
-          new transport::EncodedAudioFrame());
-      audio_frame->codec = codec_;
+      scoped_ptr<transport::EncodedFrame> audio_frame(
+          new transport::EncodedFrame());
+      audio_frame->dependency = transport::EncodedFrame::KEY;
       audio_frame->frame_id = frame_id_;
+      audio_frame->referenced_frame_id = frame_id_;
       audio_frame->rtp_timestamp = frame_rtp_timestamp_;
+      audio_frame->reference_time = frame_capture_time_;
 
       if (EncodeFromFilledBuffer(&audio_frame->data)) {
         LogAudioFrameEncodedEvent(cast_environment_,
                                   cast_environment_->Clock()->NowTicks(),
                                   audio_frame->rtp_timestamp,
                                   audio_frame->frame_id,
                                   audio_frame->data.size());
         cast_environment_->PostTask(
             CastEnvironment::MAIN,
             FROM_HERE,
-            base::Bind(callback_,
-                       base::Passed(&audio_frame),
-                       frame_capture_time_));
+            base::Bind(callback_, base::Passed(&audio_frame)));
       }
 
       // Reset the internal buffer, frame ID, and timestamps for the next frame.
       buffer_fill_end_ = 0;
       ++frame_id_;
       frame_rtp_timestamp_ += samples_per_frame_;
       frame_capture_time_ += frame_duration;
     }
   }
 
@@ skipping 15 matching lines @@
 
   // Subclass' ctor is expected to set this to STATUS_AUDIO_INITIALIZED.
   CastInitializationStatus cast_initialization_status_;
 
  private:
   // In the case where a call to EncodeAudio() cannot completely fill the
   // buffer, this points to the position at which to populate data in a later
   // call.
   int buffer_fill_end_;
 
-  // A counter used to label EncodedAudioFrames.
+  // A counter used to label EncodedFrames.
   uint32 frame_id_;
 
   // The RTP timestamp for the next frame of encoded audio.  This is defined as
   // the number of audio samples encoded so far, plus the estimated number of
   // samples that were missed due to data underruns.  A receiver uses this value
   // to detect gaps in the audio signal data being provided.  Per the spec, RTP
   // timestamp values are allowed to overflow and roll around past zero.
   uint32 frame_rtp_timestamp_;
 
   // The local system time associated with the start of the next frame of
@@ skipping 194 matching lines @@
   cast_environment_->PostTask(CastEnvironment::AUDIO,
                               FROM_HERE,
                               base::Bind(&AudioEncoder::ImplBase::EncodeAudio,
                                          impl_,
                                          base::Passed(&audio_bus),
                                          recorded_time));
 }
 
 }  // namespace cast
 }  // namespace media