Remove use of VoECodec in video/call tests.

webrtc/call/call_perf_tests.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
@@ skipping 22 matching lines @@
 #include "webrtc/test/fake_audio_device.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/frame_generator.h"
 #include "webrtc/test/frame_generator_capturer.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/rtp_rtcp_observer.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/perf_test.h"
 #include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/voice_engine/include/voe_codec.h"
-#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
-#include "webrtc/voice_engine/include/voe_video_sync.h"
 
 using webrtc::test::DriftingClock;
 using webrtc::test::FakeAudioDevice;
 
 namespace webrtc {
 
 class CallPerfTest : public test::CallTest {
  protected:
   enum class FecMode {
     kOn, kOff
@@ skipping 89 matching lines @@
                                       float video_ntp_speed,
                                       float video_rtp_speed,
                                       float audio_rtp_speed) {
   const char* kSyncGroup = "av_sync";
   const uint32_t kAudioSendSsrc = 1234;
   const uint32_t kAudioRecvSsrc = 5678;
 
   metrics::Reset();
   VoiceEngine* voice_engine = VoiceEngine::Create();
   VoEBase* voe_base = VoEBase::GetInterface(voice_engine);
-  VoECodec* voe_codec = VoECodec::GetInterface(voice_engine);
   const std::string audio_filename =
       test::ResourcePath("voice_engine/audio_long16", "pcm");
   ASSERT_STRNE("", audio_filename.c_str());
   FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), audio_filename,
                                     audio_rtp_speed);
   EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr, decoder_factory_));
   VoEBase::ChannelConfig config;
   config.enable_voice_pacing = true;
   int send_channel_id = voe_base->CreateChannel(config);
   int recv_channel_id = voe_base->CreateChannel();
@@ skipping 53 matching lines @@
   receive_transport.SetReceiver(sender_call_->Receiver());
 
   test::FakeDecoder fake_decoder;
 
   CreateSendConfig(1, 0, &video_send_transport);
   CreateMatchingReceiveConfigs(&receive_transport);
 
   AudioSendStream::Config audio_send_config(&audio_send_transport);
   audio_send_config.voe_channel_id = send_channel_id;
   audio_send_config.rtp.ssrc = kAudioSendSsrc;
+  audio_send_config.send_codec_spec.codec_inst =
+      CodecInst{103, "ISAC", 16000, 480, 1, 32000};
   AudioSendStream* audio_send_stream =
       sender_call_->CreateAudioSendStream(audio_send_config);
 
-  CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
-  EXPECT_EQ(0, voe_codec->SetSendCodec(send_channel_id, isac));
-
   video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
   if (fec == FecMode::kOn) {
     video_send_config_.rtp.ulpfec.red_payload_type = kRedPayloadType;
     video_send_config_.rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
     video_receive_configs_[0].rtp.ulpfec.red_payload_type = kRedPayloadType;
     video_receive_configs_[0].rtp.ulpfec.ulpfec_payload_type =
         kUlpfecPayloadType;
   }
   video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
   video_receive_configs_[0].renderer = &observer;
@@ skipping 45 matching lines @@
   receive_transport.StopSending();
 
   DestroyStreams();
 
   sender_call_->DestroyAudioSendStream(audio_send_stream);
   receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
 
   voe_base->DeleteChannel(send_channel_id);
   voe_base->DeleteChannel(recv_channel_id);
   voe_base->Release();
-  voe_codec->Release();
 
   DestroyCalls();
 
   VoiceEngine::Delete(voice_engine);
 
   observer.PrintResults();
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.AVSyncOffsetInMs"));
 }
 
 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoNtpDrift) {
@@ skipping 418 matching lines @@
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     test::FrameGeneratorCapturer* frame_generator_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc