Cast:Adding cast_transport_config and cleaning up

chrome/renderer/media/cast_rtp_stream.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 "chrome/renderer/media/cast_rtp_stream.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/memory/weak_ptr.h"
 #include "chrome/renderer/media/cast_session.h"
 #include "chrome/renderer/media/cast_udp_transport.h"
 #include "content/public/renderer/media_stream_audio_sink.h"
 #include "content/public/renderer/media_stream_video_sink.h"
 #include "media/base/audio_bus.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_defines.h"
 #include "media/cast/cast_sender.h"
+#include "media/cast/transport/cast_transport_config.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
 
 using media::cast::AudioSenderConfig;
 using media::cast::VideoSenderConfig;
 
 namespace {
 const char kCodecNameOpus[] = "OPUS";
 const char kCodecNameVp8[] = "VP8";
 
 CastRtpPayloadParams DefaultOpusPayload() {
@@ skipping 44 matching lines @@
   const CastRtpPayloadParams& payload_params = params.payloads[0];
   config->sender_ssrc = payload_params.ssrc;
   config->incoming_feedback_ssrc = payload_params.feedback_ssrc;
   config->rtp_payload_type = payload_params.payload_type;
   config->use_external_encoder = false;
   config->frequency = payload_params.clock_rate;
   config->channels = payload_params.channels;
   config->bitrate = payload_params.max_bitrate;
   config->aes_key = payload_params.aes_key;
   config->aes_iv_mask = payload_params.aes_iv_mask;
-  config->codec = media::cast::kPcm16;
+  config->codec = media::cast::transport::kPcm16;
   if (payload_params.codec_name == kCodecNameOpus)
-    config->codec = media::cast::kOpus;
+    config->codec = media::cast::transport::kOpus;
   else
     return false;
   return true;
 }
 
 bool ToVideoSenderConfig(const CastRtpParams& params,
                          VideoSenderConfig* config) {
   if (params.payloads.empty())
     return false;
   const CastRtpPayloadParams& payload_params = params.payloads[0];
   config->sender_ssrc = payload_params.ssrc;
   config->incoming_feedback_ssrc = payload_params.feedback_ssrc;
   config->rtp_payload_type = payload_params.payload_type;
   config->use_external_encoder = false;
   config->width = payload_params.width;
   config->height = payload_params.height;
   config->min_bitrate = config->start_bitrate = payload_params.min_bitrate;
   config->max_bitrate = payload_params.max_bitrate;
   config->aes_key = payload_params.aes_key;
   config->aes_iv_mask = payload_params.aes_iv_mask;
   if (payload_params.codec_name == kCodecNameVp8)
-    config->codec = media::cast::kVp8;
+    config->codec = media::cast::transport::kVp8;
   else
     return false;
   return true;
 }
 
 void DeleteAudioBus(scoped_ptr<media::AudioBus> audio_bus) {
   // Do nothing as |audio_bus| will be deleted.
 }
 
 }  // namespace
@@ skipping 172 matching lines @@
 }
 
 void CastRtpStream::Stop() {
   audio_sink_.reset();
   video_sink_.reset();
 }
 
 bool CastRtpStream::IsAudio() const {
   return track_.source().type() == blink::WebMediaStreamSource::TypeAudio;
 }