Remove use of VoECodec in video/call tests.

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/config.h"
 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/test/call_test.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()),
@@ skipping 167 matching lines @@
     for (size_t i = 0; i < num_video_streams; ++i)
       video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
     video_send_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId));
   }
 
   if (num_audio_streams > 0) {
     audio_send_config_ = AudioSendStream::Config(send_transport);
     audio_send_config_.voe_channel_id = voe_send_.channel_id;
     audio_send_config_.rtp.ssrc = kAudioSendSsrc;
+    audio_send_config_.send_codec_spec.codec_inst =
+        CodecInst{kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000};

[sprang_webrtc, 2016/10/26 09:11:10]

How is this picked up in the receive config? I don't see an explicit reference
to it..

[the sun, 2016/10/26 09:22:48]

Supported decoders are configured for AudioReceiveStream using the decoder
factory - see line 33.

[sprang_webrtc, 2016/10/26 09:30:34]

Ah, gotcha.

   }
 }
 
 void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
   RTC_DCHECK(video_receive_configs_.empty());
   RTC_DCHECK(allocated_decoders_.empty());
   if (num_video_streams_ > 0) {
     RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
     VideoReceiveStream::Config video_config(rtcp_send_transport);
     video_config.rtp.remb = true;
     video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
     for (const RtpExtension& extension : video_send_config_.rtp.extensions)
       video_config.rtp.extensions.push_back(extension);
     video_config.renderer = &fake_renderer_;
     for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) {
       VideoReceiveStream::Decoder decoder =
           test::CreateMatchingDecoder(video_send_config_.encoder_settings);
       allocated_decoders_.push_back(
           std::unique_ptr<VideoDecoder>(decoder.decoder));
       video_config.decoders.clear();
       video_config.decoders.push_back(decoder);
       video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
       video_receive_configs_.push_back(video_config.Copy());
     }
   }
 
   RTC_DCHECK(num_audio_streams_ <= 1);

[sprang_webrtc, 2016/10/26 09:11:11]

drive-by nit: RTC_DCHECK_LE

[the sun, 2016/10/26 09:22:48]

Done.

   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);
   }
@@ skipping 43 matching lines @@
 void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) {
   frame_generator_capturer_->SetFakeRotation(rotation);
 }
 
 void CallTest::CreateAudioStreams() {
   audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
   for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
     audio_receive_streams_.push_back(
         receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i]));
   }
-  CodecInst isac = {kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000};
-  EXPECT_EQ(0, voe_send_.codec->SetSendCodec(voe_send_.channel_id, isac));
 }
 
 void CallTest::DestroyStreams() {
   if (video_send_stream_)
     sender_call_->DestroyVideoSendStream(video_send_stream_);
   video_send_stream_ = nullptr;
   for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
     receiver_call_->DestroyVideoReceiveStream(video_recv_stream);
 
   if (audio_send_stream_)
     sender_call_->DestroyAudioSendStream(audio_send_stream_);
   audio_send_stream_ = nullptr;
   for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
     receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
   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,
                                     decoder_factory_));
   VoEBase::ChannelConfig config;
   config.enable_voice_pacing = true;
   voe_send_.channel_id = voe_send_.base->CreateChannel(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,
                                     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();
-  voe_recv_.codec = nullptr;
 
   voe_send_.base->DeleteChannel(voe_send_.channel_id);
   voe_send_.channel_id = -1;
   voe_send_.base->Release();
   voe_send_.base = nullptr;
-  voe_send_.codec->Release();
-  voe_send_.codec = nullptr;
 
   VoiceEngine::Delete(voe_send_.voice_engine);
   voe_send_.voice_engine = nullptr;
   VoiceEngine::Delete(voe_recv_.voice_engine);
   voe_recv_.voice_engine = nullptr;
 }
 
 const int CallTest::kDefaultWidth;
 const int CallTest::kDefaultHeight;
 const int CallTest::kDefaultFramerate;
@@ skipping 84 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc