WIP: Move the creation of AudioCodecFactory into PeerConnectionFactory.

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

 /*
  *  Copyright (c) 2011 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/neteq/include/neteq.h"
 
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>  // memset
 
 #include <algorithm>
 #include <memory>
 #include <set>
 #include <string>
 #include <vector>
 
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/sha1digest.h"
 #include "webrtc/base/stringencode.h"
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
@@ skipping 229 matching lines @@
 NetEqDecodingTest::NetEqDecodingTest()
     : neteq_(NULL),
       config_(),
       sim_clock_(0),
       output_sample_rate_(kInitSampleRateHz),
       algorithmic_delay_ms_(0) {
   config_.sample_rate_hz = kInitSampleRateHz;
 }
 
 void NetEqDecodingTest::SetUp() {
-  neteq_ = NetEq::Create(config_);
+  neteq_ = NetEq::Create(config_, CreateBuiltinAudioDecoderFactory());
   NetEqNetworkStatistics stat;
   ASSERT_EQ(0, neteq_->NetworkStatistics(&stat));
   algorithmic_delay_ms_ = stat.current_buffer_size_ms;
   ASSERT_TRUE(neteq_);
   LoadDecoders();
 }
 
 void NetEqDecodingTest::TearDown() {
   delete neteq_;
 }
@@ skipping 1223 matching lines @@
     timestamp += kSamples;
 
     // Pull audio once.
     ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
   // Verify speech output.
   EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
 }
 }  // namespace webrtc