Adding audio to video_quality_test.

webrtc/video/video_quality_test.cc

Index: webrtc/video/video_quality_test.cc
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index 66210d08643e62fc93bccff52f0eea2ba70f8b04..ed0aae51928058f04068bcd6a21f21f6b3246750 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -34,6 +34,8 @@
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/video_renderer.h"
 #include "webrtc/video/video_quality_test.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
 
 namespace webrtc {
 
@@ -1110,6 +1112,12 @@ void VideoQualityTest::RunWithVideoRenderer(const Params& params) {
   // match the full stack tests.
   Call::Config call_config;
   call_config.bitrate_config = params_.common.call_bitrate_config;
+
+  CreateVoiceEngine();

[stefan-webrtc, 2016/07/11 15:01:10]

I think we should add an option to configure with or without audio in
loopback.cc which uses this class.

[minyue-webrtc, 2016/07/12 10:09:36]

Done.

+  AudioState::Config audio_state_config;
+  audio_state_config.voice_engine = voe_.voice_engine;
+  call_config.audio_state = AudioState::Create(audio_state_config);
+
   std::unique_ptr<Call> call(Call::Create(call_config));
 
   test::LayerFilteringTransport transport(
@@ -1124,6 +1132,7 @@ void VideoQualityTest::RunWithVideoRenderer(const Params& params) {
 
   video_send_config_.local_renderer = local_preview.get();
   video_receive_configs_[stream_id].renderer = loopback_video.get();
+  // video_receive_configs_[stream_id].sync_group = kSyncGroup;

[stefan-webrtc, 2016/07/11 15:01:10]

Would be good to be able to test sync. Maybe have an option to enable sync and
use the mic?

[minyue-webrtc, 2016/07/12 10:09:36]

Done. So we can use the actually device instead of fake_audio_device_

 
   video_send_config_.suspend_below_min_bitrate =
       params_.common.suspend_below_min_bitrate;
@@ -1142,24 +1151,98 @@ void VideoQualityTest::RunWithVideoRenderer(const Params& params) {
 
   video_send_stream_ =
       call->CreateVideoSendStream(video_send_config_, video_encoder_config_);
-  VideoReceiveStream* receive_stream =
+  VideoReceiveStream* video_receive_stream =
       call->CreateVideoReceiveStream(video_receive_configs_[stream_id].Copy());
   CreateCapturer(video_send_stream_->Input());
 
-  receive_stream->Start();
+  audio_send_config_ = AudioSendStream::Config(&transport);
+  audio_send_config_.voe_channel_id = voe_.send_channel_id;
+  audio_send_config_.rtp.ssrc = kAudioSendSsrc;
+  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.decoder_factory = decoder_factory_;
+  audio_receive_configs_.push_back(audio_config);
+  AudioReceiveStream* audio_receive_stream =
+      call->CreateAudioReceiveStream(audio_receive_configs_[0]);
+
+  const CodecInst kOpusInst = {120, "OPUS", 48000, 960, 2, 64000};
+  EXPECT_EQ(0, voe_.codec->SetSendCodec(voe_.send_channel_id, kOpusInst));
+
+  // Start sending and receiving video.
+  video_receive_stream->Start();
   video_send_stream_->Start();
   capturer_->Start();
 
+  // Start receiving audio.
+  audio_receive_stream->Start();
+  EXPECT_EQ(0, voe_.base->StartPlayout(voe_.receive_channel_id));
+  EXPECT_EQ(0, voe_.base->StartReceive(voe_.receive_channel_id));
+
+  // Start sending audio.
+  audio_send_stream_->Start();
+  EXPECT_EQ(0, voe_.base->StartSend(voe_.send_channel_id));
+  fake_audio_device_->Start();
+
   test::PressEnterToContinue();
 
+  // Stop sending audio.
+  fake_audio_device_->Stop();
+  EXPECT_EQ(0, voe_.base->StopSend(voe_.send_channel_id));
+  audio_send_stream_->Stop();
+
+  // Stop receiving audio.
+  EXPECT_EQ(0, voe_.base->StopReceive(voe_.receive_channel_id));
+  EXPECT_EQ(0, voe_.base->StopPlayout(voe_.receive_channel_id));
+  audio_receive_stream->Stop();
+
+  // Stop receiving and sending video.
   capturer_->Stop();
   video_send_stream_->Stop();
-  receive_stream->Stop();
+  video_receive_stream->Stop();
 
-  call->DestroyVideoReceiveStream(receive_stream);
+  call->DestroyVideoReceiveStream(video_receive_stream);
   call->DestroyVideoSendStream(video_send_stream_);
+  call->DestroyAudioSendStream(audio_send_stream_);
+  call->DestroyAudioReceiveStream(audio_receive_stream);
 
   transport.StopSending();
+  DestroyVoiceEngine();
+}
+
+void VideoQualityTest::CreateVoiceEngine() {
+  fake_audio_device_.reset(new test::FakeAudioDevice(
+      clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"),
+      test::DriftingClock::kNoDrift));
+  voe_.voice_engine = VoiceEngine::Create();
+  voe_.base = VoEBase::GetInterface(voe_.voice_engine);
+  voe_.codec = VoECodec::GetInterface(voe_.voice_engine);
+  EXPECT_EQ(0, voe_.base->Init(fake_audio_device_.get(), nullptr,
+                               decoder_factory_));
+  Config voe_config;
+  voe_config.Set<VoicePacing>(new VoicePacing(true));
+  voe_.send_channel_id = voe_.base->CreateChannel(voe_config);
+  EXPECT_GE(voe_.send_channel_id, 0);
+  voe_.receive_channel_id = voe_.base->CreateChannel();
+  EXPECT_GE(voe_.receive_channel_id, 0);
+}
+
+void VideoQualityTest::DestroyVoiceEngine() {
+  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;
+
+  VoiceEngine::Delete(voe_.voice_engine);
+  voe_.voice_engine = nullptr;
 }
 
 }  // namespace webrtc