Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.

webrtc/call/call_unittest.cc

 /*
  *  Copyright (c) 2015 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 <list>
 #include <memory>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/audio_state.h"
 #include "webrtc/call.h"
+#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace {
 
 struct CallHelper {
-  CallHelper() {
+  explicit CallHelper(
+      rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr)
+      : voice_engine_(decoder_factory) {
     webrtc::AudioState::Config audio_state_config;
     audio_state_config.voice_engine = &voice_engine_;
     webrtc::Call::Config config;
     config.audio_state = webrtc::AudioState::Create(audio_state_config);
     call_.reset(webrtc::Call::Create(config));
   }
 
   webrtc::Call* operator->() { return call_.get(); }
 
  private:
@@ skipping 12 matching lines @@
   CallHelper call;
   AudioSendStream::Config config(nullptr);
   config.rtp.ssrc = 42;
   config.voe_channel_id = 123;
   AudioSendStream* stream = call->CreateAudioSendStream(config);
   EXPECT_NE(stream, nullptr);
   call->DestroyAudioSendStream(stream);
 }
 
 TEST(CallTest, CreateDestroy_AudioReceiveStream) {
-  CallHelper call;
+  rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory(
+      new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>);
+  CallHelper call(decoder_factory);
   AudioReceiveStream::Config config;
   config.rtp.remote_ssrc = 42;
   config.voe_channel_id = 123;
+  config.decoder_factory = decoder_factory;
   AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
   EXPECT_NE(stream, nullptr);
   call->DestroyAudioReceiveStream(stream);
 }
 
 TEST(CallTest, CreateDestroy_AudioSendStreams) {
   CallHelper call;
   AudioSendStream::Config config(nullptr);
   config.voe_channel_id = 123;
   std::list<AudioSendStream*> streams;
   for (int i = 0; i < 2; ++i) {
     for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
       config.rtp.ssrc = ssrc;
       AudioSendStream* stream = call->CreateAudioSendStream(config);
       EXPECT_NE(stream, nullptr);
       if (ssrc & 1) {
         streams.push_back(stream);
       } else {
         streams.push_front(stream);
       }
     }
     for (auto s : streams) {
       call->DestroyAudioSendStream(s);
     }
     streams.clear();
   }
 }
 
 TEST(CallTest, CreateDestroy_AudioReceiveStreams) {
-  CallHelper call;
+  rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory(
+      new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>);
+  CallHelper call(decoder_factory);
   AudioReceiveStream::Config config;
   config.voe_channel_id = 123;
+  config.decoder_factory = decoder_factory;
   std::list<AudioReceiveStream*> streams;
   for (int i = 0; i < 2; ++i) {
     for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
       config.rtp.remote_ssrc = ssrc;
       AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
       EXPECT_NE(stream, nullptr);
       if (ssrc & 1) {
         streams.push_back(stream);
       } else {
         streams.push_front(stream);
       }
     }
     for (auto s : streams) {
       call->DestroyAudioReceiveStream(s);
     }
     streams.clear();
   }
 }
 }  // namespace webrtc