Cast: Added support for AES-CTR crypto.

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 "base/bind.h"
 #include "base/logging.h"
 #include "base/message_loop/message_loop.h"
 #include "media/cast/cast_defines.h"
 #include "media/cast/cast_environment.h"
 #include "third_party/webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
 #include "third_party/webrtc/modules/interface/module_common_types.h"
 
 namespace media {
 namespace cast {
 
 // 48KHz, 2 channels and 100 ms.
 static const int kMaxNumberOfSamples = 48 * 2 * 100;
 
 // This class is only called from the cast audio encoder thread.
 class WebrtEncodedDataCallback : public webrtc::AudioPacketizationCallback {
  public:
   WebrtEncodedDataCallback(scoped_refptr<CastEnvironment> cast_environment,
                            AudioCodec codec,
                            int frequency)
       : codec_(codec),
         frequency_(frequency),
         cast_environment_(cast_environment),
-        last_timestamp_(0) {}
+        last_timestamp_(0),
+        frame_id_(0) {}
 
   virtual int32 SendData(
       webrtc::FrameType /*frame_type*/,
       uint8 /*payload_type*/,
       uint32 timestamp,
       const uint8* payload_data,
       uint16 payload_size,
       const webrtc::RTPFragmentationHeader* /*fragmentation*/) OVERRIDE {
     scoped_ptr<EncodedAudioFrame> audio_frame(new EncodedAudioFrame());
     audio_frame->codec = codec_;
     audio_frame->samples = timestamp - last_timestamp_;
+    audio_frame->frame_id = frame_id_++;
     DCHECK(audio_frame->samples <= kMaxNumberOfSamples);
     last_timestamp_ = timestamp;
     audio_frame->data.insert(audio_frame->data.begin(),
                              payload_data,
                              payload_data + payload_size);
 
     cast_environment_->PostTask(CastEnvironment::MAIN, FROM_HERE,
         base::Bind(*frame_encoded_callback_, base::Passed(&audio_frame),
                    recorded_time_));
     return 0;
   }
 
   void SetEncodedCallbackInfo(
       const base::TimeTicks& recorded_time,
       const AudioEncoder::FrameEncodedCallback* frame_encoded_callback) {
     recorded_time_ = recorded_time;
     frame_encoded_callback_ = frame_encoded_callback;
   }
 
  private:
   const AudioCodec codec_;
   const int frequency_;
   scoped_refptr<CastEnvironment> cast_environment_;
   uint32 last_timestamp_;
+  uint32 frame_id_;
   base::TimeTicks recorded_time_;
   const AudioEncoder::FrameEncodedCallback* frame_encoded_callback_;
 };
 
 AudioEncoder::AudioEncoder(scoped_refptr<CastEnvironment> cast_environment,
                            const AudioSenderConfig& audio_config)
     : cast_environment_(cast_environment),
       audio_encoder_(webrtc::AudioCodingModule::Create(0)),
       webrtc_encoder_callback_(
           new WebrtEncodedDataCallback(cast_environment, audio_config.codec,
@@ skipping 90 matching lines @@
   // Note: Not all insert of 10 ms will generate a callback with encoded data.
   webrtc_encoder_callback_->SetEncodedCallbackInfo(recorded_time,
                                                    &frame_encoded_callback);
   for (size_t i = 0; i < number_of_10ms_blocks; ++i) {
     audio_encoder_->Process();
   }
 }
 
 }  // namespace cast
 }  // namespace media