Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.

webrtc/test/call_test.cc

 /*
  *  Copyright (c) 2014 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/base/checks.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 
 namespace webrtc {
 namespace test {
 
 namespace {
 const int kVideoRotationRtpExtensionId = 4;
 }
 
 CallTest::CallTest()
     : clock_(Clock::GetRealTimeClock()),
       video_send_config_(nullptr),
       video_send_stream_(nullptr),
       audio_send_config_(nullptr),
       audio_send_stream_(nullptr),
       fake_encoder_(clock_),
       num_video_streams_(1),
       num_audio_streams_(0),
+      decoder_factory_(CreateBuiltinAudioDecoderFactory()),
       fake_send_audio_device_(nullptr),
       fake_recv_audio_device_(nullptr) {}
 
 CallTest::~CallTest() {
 }
 
 void CallTest::RunBaseTest(BaseTest* test) {
   num_video_streams_ = test->GetNumVideoStreams();
   num_audio_streams_ = test->GetNumAudioStreams();
   RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
@@ skipping 177 matching lines @@
   }
 
   RTC_DCHECK(num_audio_streams_ <= 1);
   if (num_audio_streams_ == 1) {
     RTC_DCHECK(voe_send_.channel_id >= 0);
     AudioReceiveStream::Config audio_config;
     audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
     audio_config.rtcp_send_transport = rtcp_send_transport;
     audio_config.voe_channel_id = voe_recv_.channel_id;
     audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
+    audio_config.decoder_factory = decoder_factory_;
     audio_receive_configs_.push_back(audio_config);
   }
 }
 
 void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
                                                      float speed) {
   VideoStream stream = video_encoder_config_.streams.back();
   frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
       video_send_stream_->Input(), stream.width, stream.height,
       stream.max_framerate * speed, clock));
@@ skipping 58 matching lines @@
   video_receive_streams_.clear();
 
   allocated_decoders_.clear();
 }
 
 void CallTest::CreateVoiceEngines() {
   CreateFakeAudioDevices();
   voe_send_.voice_engine = VoiceEngine::Create();
   voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
   voe_send_.codec = VoECodec::GetInterface(voe_send_.voice_engine);
-  EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr));
+  EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr,
+                                    decoder_factory_));
   Config voe_config;
   voe_config.Set<VoicePacing>(new VoicePacing(true));
   voe_send_.channel_id = voe_send_.base->CreateChannel(voe_config);
   EXPECT_GE(voe_send_.channel_id, 0);
 
   voe_recv_.voice_engine = VoiceEngine::Create();
   voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine);
   voe_recv_.codec = VoECodec::GetInterface(voe_recv_.voice_engine);
-  EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), nullptr));
+  EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), nullptr,
+                                    decoder_factory_));
   voe_recv_.channel_id = voe_recv_.base->CreateChannel();
   EXPECT_GE(voe_recv_.channel_id, 0);
 }
 
 void CallTest::DestroyVoiceEngines() {
   voe_recv_.base->DeleteChannel(voe_recv_.channel_id);
   voe_recv_.channel_id = -1;
   voe_recv_.base->Release();
   voe_recv_.base = nullptr;
   voe_recv_.codec->Release();
@@ skipping 95 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc