Fix code style and gyp files in cast to build cast_unittest

media/cast/audio_receiver/audio_decoder.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 "base/logging.h"
 #include "media/cast/audio_receiver/audio_decoder.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 {
 
 AudioDecoder::AudioDecoder(scoped_refptr<CastThread> cast_thread,
                            const AudioReceiverConfig& audio_config)
-    : cast_thread_(cast_thread),
-      have_received_packets_(false) {
-  audio_decoder_ = webrtc::AudioCodingModule::Create(0);
+    : audio_decoder_(webrtc::AudioCodingModule::Create(0)),
+      have_received_packets_(false),
+      cast_thread_(cast_thread) {
   audio_decoder_->InitializeReceiver();
 
   webrtc::CodecInst receive_codec;
   switch (audio_config.codec) {
     case kPcm16:
       receive_codec.pltype = audio_config.rtp_payload_type;
       strncpy(receive_codec.plname, "L16", 4);
       receive_codec.plfreq = audio_config.frequency;
       receive_codec.pacsize = -1;
       receive_codec.channels = audio_config.channels;
@@ skipping 13 matching lines @@
   }
   if (audio_decoder_->RegisterReceiveCodec(receive_codec) != 0) {
     DCHECK(false) << "Failed to register receive codec";
   }
 
   audio_decoder_->SetMaximumPlayoutDelay(audio_config.rtp_max_delay_ms);
   audio_decoder_->SetPlayoutMode(webrtc::streaming);
 }
 
 AudioDecoder::~AudioDecoder() {
-  webrtc::AudioCodingModule::Destroy(audio_decoder_);
 }
 
 bool AudioDecoder::GetRawAudioFrame(int number_of_10ms_blocks,
                                     int desired_frequency,
                                     PcmAudioFrame* audio_frame,
                                     uint32* rtp_timestamp) {
   DCHECK(cast_thread_->CurrentlyOn(CastThread::AUDIO_DECODER));
   if (!have_received_packets_) return false;
 
   for (int i = 0; i < number_of_10ms_blocks; ++i) {
@@ skipping 27 matching lines @@
 }
 
 void AudioDecoder::IncomingParsedRtpPacket(const uint8* payload_data,
                                            int payload_size,
                                            const RtpCastHeader& rtp_header) {
   have_received_packets_ = true;
   audio_decoder_->IncomingPacket(payload_data, payload_size, rtp_header.webrtc);
 }
 
 }  // namespace cast
-}  // namespace media
+}  // namespace media