Remove use of VoECodec in video/call tests.

webrtc/video/video_quality_test.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 "webrtc/video/video_quality_test.h"
@@ skipping 19 matching lines @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/include/cpu_info.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/layer_filtering_transport.h"
 #include "webrtc/test/run_loop.h"
 #include "webrtc/test/statistics.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/vcm_capturer.h"
 #include "webrtc/test/video_renderer.h"
 #include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/voice_engine/include/voe_codec.h"
 
 namespace {
 
 constexpr int kSendStatsPollingIntervalMs = 1000;
 constexpr int kPayloadTypeH264 = 122;
 constexpr int kPayloadTypeVP8 = 123;
 constexpr int kPayloadTypeVP9 = 124;
 constexpr size_t kMaxComparisons = 10;
 constexpr char kSyncGroup[] = "av_sync";
 constexpr int kOpusMinBitrate = 6000;
 constexpr int kOpusBitrateFb = 32000;
 
 struct VoiceEngineState {
   VoiceEngineState()
       : voice_engine(nullptr),
         base(nullptr),
-        codec(nullptr),
         send_channel_id(-1),
         receive_channel_id(-1) {}
 
   webrtc::VoiceEngine* voice_engine;
   webrtc::VoEBase* base;
-  webrtc::VoECodec* codec;
   int send_channel_id;
   int receive_channel_id;
 };
 
 void CreateVoiceEngine(VoiceEngineState* voe,
                        rtc::scoped_refptr<webrtc::AudioDecoderFactory>
                            decoder_factory) {
   voe->voice_engine = webrtc::VoiceEngine::Create();
   voe->base = webrtc::VoEBase::GetInterface(voe->voice_engine);
-  voe->codec = webrtc::VoECodec::GetInterface(voe->voice_engine);
   EXPECT_EQ(0, voe->base->Init(nullptr, nullptr, decoder_factory));
   webrtc::VoEBase::ChannelConfig config;
   config.enable_voice_pacing = true;
   voe->send_channel_id = voe->base->CreateChannel(config);
   EXPECT_GE(voe->send_channel_id, 0);
   voe->receive_channel_id = voe->base->CreateChannel();
   EXPECT_GE(voe->receive_channel_id, 0);
 }
 
 void DestroyVoiceEngine(VoiceEngineState* voe) {
   voe->base->DeleteChannel(voe->send_channel_id);
   voe->send_channel_id = -1;
   voe->base->DeleteChannel(voe->receive_channel_id);
   voe->receive_channel_id = -1;
   voe->base->Release();
   voe->base = nullptr;
-  voe->codec->Release();
-  voe->codec = nullptr;
 
   webrtc::VoiceEngine::Delete(voe->voice_engine);
   voe->voice_engine = nullptr;
 }
 
 class VideoStreamFactory
     : public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface {
  public:
   explicit VideoStreamFactory(const std::vector<webrtc::VideoStream>& streams)
       : streams_(streams) {}
@@ skipping 1221 matching lines @@
     // Add extension to enable audio send side BWE, and allow audio bit rate
     // adaptation.
     audio_send_config_.rtp.extensions.clear();
     if (params_.common.send_side_bwe) {
       audio_send_config_.rtp.extensions.push_back(webrtc::RtpExtension(
           webrtc::RtpExtension::kTransportSequenceNumberUri,
           test::kTransportSequenceNumberExtensionId));
       audio_send_config_.min_bitrate_kbps = kOpusMinBitrate / 1000;
       audio_send_config_.max_bitrate_kbps = kOpusBitrateFb / 1000;
     }
+    audio_send_config_.send_codec_spec.codec_inst =
+        CodecInst{120, "OPUS", 48000, 960, 2, 64000};
 
     audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_);
 
     AudioReceiveStream::Config audio_config;
     audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
     audio_config.rtcp_send_transport = &transport;
     audio_config.voe_channel_id = voe.receive_channel_id;
     audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
     audio_config.rtp.transport_cc = params_.common.send_side_bwe;
     audio_config.rtp.extensions = audio_send_config_.rtp.extensions;
     audio_config.decoder_factory = decoder_factory_;
     if (params_.audio_video_sync)
       audio_config.sync_group = kSyncGroup;
 
     audio_receive_stream = call->CreateAudioReceiveStream(audio_config);
-
-    const CodecInst kOpusInst = {120, "OPUS", 48000, 960, 2, 64000};
-    EXPECT_EQ(0, voe.codec->SetSendCodec(voe.send_channel_id, kOpusInst));
   }
 
   StartEncodedFrameLogs(video_receive_stream);
   StartEncodedFrameLogs(video_send_stream_);
 
   // Start sending and receiving video.
   video_receive_stream->Start();
   video_send_stream_->Start();
   capturer_->Start();
 
@@ skipping 58 matching lines @@
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.common.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
 
 }  // namespace webrtc