Stop using old AudioCodingModule::RegisterReceiveCodec overloads

webrtc/modules/audio_coding/test/opus_test.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/audio_coding/test/opus_test.h"
 
 #include <assert.h>
 
 #include <string>
 
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/test/TestStereo.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine_configurations.h"
 
 namespace webrtc {
@@ skipping 59 matching lines @@
   WebRtcOpus_DecoderInit(opus_stereo_decoder_);
 
   ASSERT_TRUE(acm_receiver_.get() != NULL);
   EXPECT_EQ(0, acm_receiver_->InitializeReceiver());
 
   // Register Opus stereo as receiving codec.
   CodecInst opus_codec_param;
   int codec_id = acm_receiver_->Codec("opus", 48000, 2);
   EXPECT_EQ(0, acm_receiver_->Codec(codec_id, &opus_codec_param));
   payload_type_ = opus_codec_param.pltype;
-  EXPECT_EQ(0, acm_receiver_->RegisterReceiveCodec(opus_codec_param));
+  EXPECT_EQ(true,
+            acm_receiver_->RegisterReceiveCodec(
+                opus_codec_param.pltype, CodecInstToSdp(opus_codec_param)));
 
   // Create and connect the channel.
   channel_a2b_ = new TestPackStereo;
   channel_a2b_->RegisterReceiverACM(acm_receiver_.get());
 
   //
   // Test Stereo.
   //
 
   channel_a2b_->set_codec_mode(kStereo);
@@ skipping 44 matching lines @@
   //
   // Test Mono.
   //
   channel_a2b_->set_codec_mode(kMono);
   audio_channels = 1;
   test_cntr++;
   OpenOutFile(test_cntr);
 
   // Register Opus mono as receiving codec.
   opus_codec_param.channels = 1;
-  EXPECT_EQ(0, acm_receiver_->RegisterReceiveCodec(opus_codec_param));
+  EXPECT_EQ(true,
+            acm_receiver_->RegisterReceiveCodec(
+                opus_codec_param.pltype, CodecInstToSdp(opus_codec_param)));
 
   // Run Opus with 2.5 ms frame size.
   Run(channel_a2b_, audio_channels, 32000, 120);
 
   // Run Opus with 5 ms frame size.
   Run(channel_a2b_, audio_channels, 32000, 240);
 
   // Run Opus with 10 ms frame size.
   Run(channel_a2b_, audio_channels, 32000, 480);
 
@@ skipping 204 matching lines @@
   out_file_.Open(file_name, 48000, "wb");
   file_stream.str("");
   file_name = file_stream.str();
   file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
       << test_number << ".pcm";
   file_name = file_stream.str();
   out_file_standalone_.Open(file_name, 48000, "wb");
 }
 
 }  // namespace webrtc