Revert of 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 51160c91c6a2e15b23f8b4c90d536e58eba8478f..a401d6dd42aec5a7dc8a143b77de3e656b3672ec 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -34,67 +34,14 @@
 #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 {
-
-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::Config voe_config;
-  voe_config.Set<webrtc::VoicePacing>(new webrtc::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 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;
-}
-
-}  // namespace
 
 namespace webrtc {
+
+static const int kSendStatsPollingIntervalMs = 1000;
+static const int kPayloadTypeH264 = 122;
+static const int kPayloadTypeVP8 = 123;
+static const int kPayloadTypeVP9 = 124;
+static const size_t kMaxComparisons = 10;
 
 class VideoAnalyzer : public PacketReceiver,
                       public Transport,
@@ -1055,7 +1002,6 @@
 void VideoQualityTest::RunWithAnalyzer(const Params& params) {
   params_ = params;
 
-  RTC_CHECK(!params_.audio);
   // TODO(ivica): Merge with RunWithRenderer and use a flag / argument to
   // differentiate between the analyzer and the renderer case.
   CheckParams();
@@ -1153,7 +1099,7 @@
     fclose(graph_data_output_file);
 }
 
-void VideoQualityTest::RunWithRenderers(const Params& params) {
+void VideoQualityTest::RunWithVideoRenderer(const Params& params) {
   params_ = params;
   CheckParams();
 
@@ -1177,15 +1123,6 @@
   // match the full stack tests.
   Call::Config call_config;
   call_config.bitrate_config = params_.common.call_bitrate_config;
-
-  ::VoiceEngineState voe;
-  if (params_.audio) {
-    CreateVoiceEngine(&voe, decoder_factory_);
-    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(
@@ -1200,8 +1137,6 @@
 
   video_send_config_.local_renderer = local_preview.get();
   video_receive_configs_[stream_id].renderer = loopback_video.get();
-  if (params_.audio && params_.audio_video_sync)
-    video_receive_configs_[stream_id].sync_group = kSyncGroup;
 
   video_send_config_.suspend_below_min_bitrate =
       params_.common.suspend_below_min_bitrate;
@@ -1220,91 +1155,24 @@
 
   video_send_stream_ =
       call->CreateVideoSendStream(video_send_config_, video_encoder_config_);
-  VideoReceiveStream* video_receive_stream =
+  VideoReceiveStream* receive_stream =
       call->CreateVideoReceiveStream(video_receive_configs_[stream_id].Copy());
   CreateCapturer(video_send_stream_->Input());
 
-  AudioReceiveStream* audio_receive_stream = nullptr;
-  if (params_.audio) {
-    audio_send_config_ = AudioSendStream::Config(&transport);
-    audio_send_config_.voe_channel_id = voe.send_channel_id;
-    audio_send_config_.rtp.ssrc = kAudioSendSsrc;
-
-    // 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_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));
-  }
-
-  // Start sending and receiving video.
-  video_receive_stream->Start();
+  receive_stream->Start();
   video_send_stream_->Start();
   capturer_->Start();
 
-  if (params_.audio) {
-    // 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));
-  }
-
   test::PressEnterToContinue();
 
-  if (params_.audio) {
-    // Stop sending audio.
-    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();
-  video_receive_stream->Stop();
-
-  call->DestroyVideoReceiveStream(video_receive_stream);
+  receive_stream->Stop();
+
+  call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(video_send_stream_);
 
-  if (params_.audio) {
-     call->DestroyAudioSendStream(audio_send_stream_);
-     call->DestroyAudioReceiveStream(audio_receive_stream);
-  }
-
   transport.StopSending();
-  if (params_.audio)
-    DestroyVoiceEngine(&voe);
 }
 
 }  // namespace webrtc