Reland of Add full stack tests for MediaCodec.

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.
  */
 
@@ skipping 24 matching lines @@
       event_log_(RtcEventLog::CreateNull()),
       sender_call_transport_controller_(nullptr),
       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),
       num_flexfec_streams_(0),
-      decoder_factory_(CreateBuiltinAudioDecoderFactory()),
-      encoder_factory_(CreateBuiltinAudioEncoderFactory()),
+      audio_decoder_factory_(CreateBuiltinAudioDecoderFactory()),
+      audio_encoder_factory_(CreateBuiltinAudioEncoderFactory()),
       task_queue_("CallTestTaskQueue"),
       fake_send_audio_device_(nullptr),
       fake_recv_audio_device_(nullptr) {}
 
 CallTest::~CallTest() {
   task_queue_.SendTask([this]() {
     fake_send_audio_device_.reset();
     fake_recv_audio_device_.reset();
     frame_generator_capturer_.reset();
   });
@@ skipping 159 matching lines @@
         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 =
         rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
             {kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}}});
-    audio_send_config_.encoder_factory = encoder_factory_;
+    audio_send_config_.encoder_factory = audio_encoder_factory_;
   }
 
   // TODO(brandtr): Update this when we support multistream protection.
   if (num_flexfec_streams > 0) {
     video_send_config_.rtp.flexfec.payload_type = kFlexfecPayloadType;
     video_send_config_.rtp.flexfec.ssrc = kFlexfecSendSsrc;
     video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]};
   }
 }
 
@@ skipping 22 matching lines @@
   }
 
   RTC_DCHECK_GE(1, num_audio_streams_);
   if (num_audio_streams_ == 1) {
     RTC_DCHECK_LE(0, voe_send_.channel_id);
     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_config.decoder_factory = audio_decoder_factory_;
     audio_config.decoder_map = {{kAudioSendPayloadType, {"opus", 48000, 2}}};
     audio_receive_configs_.push_back(audio_config);
   }
 
   // TODO(brandtr): Update this when we support multistream protection.
   RTC_DCHECK(num_flexfec_streams_ <= 1);
   if (num_flexfec_streams_ == 1) {
     FlexfecReceiveStream::Config config(rtcp_send_transport);
     config.payload_type = kFlexfecPayloadType;
     config.remote_ssrc = kFlexfecSendSsrc;
@@ skipping 139 matching lines @@
 }
 
 void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) {
   frame_generator_capturer_->SetFakeRotation(rotation);
 }
 
 void CallTest::CreateVoiceEngines() {
   voe_send_.voice_engine = VoiceEngine::Create();
   voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
   EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(),
-                                    apm_send_.get(), decoder_factory_));
+                                    apm_send_.get(), audio_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);
   EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(),
-                                    apm_recv_.get(), decoder_factory_));
+                                    apm_recv_.get(), audio_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;
 
@@ skipping 154 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc