Cast:Adding cast_transport_config and cleaning up

media/cast/audio_receiver/audio_decoder_unittest.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/test/simple_test_tick_clock.h"
 #include "media/cast/audio_receiver/audio_decoder.h"
 #include "media/cast/cast_environment.h"
 #include "media/cast/test/fake_task_runner.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
@@ skipping 34 matching lines @@
   scoped_refptr<test::FakeTaskRunner> task_runner_;
   scoped_refptr<CastEnvironment> cast_environment_;
   scoped_ptr<AudioDecoder> audio_decoder_;
 };
 
 TEST_F(AudioDecoderTest, Pcm16MonoNoResampleOnePacket) {
   AudioReceiverConfig audio_config;
   audio_config.rtp_payload_type = 127;
   audio_config.frequency = 16000;
   audio_config.channels = 1;
-  audio_config.codec = kPcm16;
+  audio_config.codec = transport::kPcm16;
   audio_config.use_external_decoder = false;
   Configure(audio_config);
 
   RtpCastHeader rtp_header;
   rtp_header.webrtc.header.payloadType = 127;
   rtp_header.webrtc.header.sequenceNumber = 1234;
   rtp_header.webrtc.header.timestamp = 0x87654321;
   rtp_header.webrtc.header.ssrc = 0x12345678;
   rtp_header.webrtc.header.paddingLength = 0;
   rtp_header.webrtc.header.headerLength = 12;
@@ skipping 28 matching lines @@
   for (size_t i = 10; i < audio_frame.samples.size(); ++i) {
     EXPECT_EQ(0x3412, audio_frame.samples[i]);
   }
 }
 
 TEST_F(AudioDecoderTest, Pcm16StereoNoResampleTwoPackets) {
   AudioReceiverConfig audio_config;
   audio_config.rtp_payload_type = 127;
   audio_config.frequency = 16000;
   audio_config.channels = 2;
-  audio_config.codec = kPcm16;
+  audio_config.codec = transport::kPcm16;
   audio_config.use_external_decoder = false;
   Configure(audio_config);
 
   RtpCastHeader rtp_header;
   rtp_header.frame_id = 0;
   rtp_header.webrtc.header.payloadType = 127;
   rtp_header.webrtc.header.sequenceNumber = 1234;
   rtp_header.webrtc.header.timestamp = 0x87654321;
   rtp_header.webrtc.header.ssrc = 0x12345678;
   rtp_header.webrtc.header.paddingLength = 0;
@@ skipping 49 matching lines @@
   audio_decoder_->SendCastMessage();
   testing_clock_.Advance(base::TimeDelta::FromMilliseconds(33));
   audio_decoder_->SendCastMessage();
 }
 
 TEST_F(AudioDecoderTest, Pcm16Resample) {
   AudioReceiverConfig audio_config;
   audio_config.rtp_payload_type = 127;
   audio_config.frequency = 16000;
   audio_config.channels = 2;
-  audio_config.codec = kPcm16;
+  audio_config.codec = transport::kPcm16;
   audio_config.use_external_decoder = false;
   Configure(audio_config);
 
   RtpCastHeader rtp_header;
   rtp_header.webrtc.header.payloadType = 127;
   rtp_header.webrtc.header.sequenceNumber = 1234;
   rtp_header.webrtc.header.timestamp = 0x87654321;
   rtp_header.webrtc.header.ssrc = 0x12345678;
   rtp_header.webrtc.header.paddingLength = 0;
   rtp_header.webrtc.header.headerLength = 12;
@@ skipping 26 matching lines @@
   // Resampling makes the variance worse.
   for (size_t i = 100 * audio_config.channels; i < audio_frame.samples.size();
       ++i) {
     EXPECT_NEAR(0x3412, audio_frame.samples[i], 400);
     if (0x3412 == audio_frame.samples[i])  count++;
   }
 }
 
 }  // namespace cast
 }  // namespace media