Revert of Adding audio to video_quality_test. (patchset #10 id:230001 of https://codereview.webrtc.org/2136573002/ )

Reason for revert:
This CL breaks https://build.chromium.org/p/client.webrtc/waterfall?builder=Win64%20Debug%20(Clang)

Need to align values to struct Params {} in a proper way. Relanding will follow.

Original issue's description:
> Adding audio to video_quality_test.
>
> This CL adds an audio loopback to video_quality_test (only RunWithVideoRenderer)
>
> BUG=
>
> Committed: https://crrev.com/65a6578e339f52eb5bc400c5715e60498e4af2c1
> Cr-Commit-Position: refs/heads/master@{#13784}

TBR=stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review-Url: https://codereview.webrtc.org/2249163002
Cr-Commit-Position: refs/heads/master@{#13785}

webrtc/video/screenshare_loopback.cc

@@ -235,11 +235,7 @@ void Loopback() {
       {"screenshare", 0.0, 0.0, flags::DurationSecs(), flags::OutputFilename(),
        flags::GraphTitle()},
       pipe_config,
-      flags::FLAGS_logs,
+      flags::FLAGS_logs};
-      {},  // ss.
-      false,  // audio.
-      false,  // audio_video_sync.
-      };
 
   std::vector<std::string> stream_descriptors;
   stream_descriptors.push_back(flags::Stream0());
@@ -255,7 +251,7 @@ void Loopback() {
   if (flags::DurationSecs()) {
     test.RunWithAnalyzer(params);
   } else {
-    test.RunWithRenderers(params);
+    test.RunWithVideoRenderer(params);
   }
 }
 }  // namespace webrtc

webrtc/video/video_loopback.cc

@@ -191,11 +191,6 @@ DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur");
 
 DEFINE_bool(use_fec, false, "Use forward error correction.");
 
-DEFINE_bool(audio, false, "Add audio stream");
-
-DEFINE_bool(audio_video_sync, false, "Sync audio and video stream (no effect if"
-    " audio is false)");
-
 DEFINE_string(
     force_fieldtrials,
     "",
@@ -241,10 +236,7 @@ void Loopback() {
       {"video", 0.0, 0.0, flags::DurationSecs(), flags::OutputFilename(),
        flags::GraphTitle()},
       pipe_config,
-      flags::FLAGS_logs,
+      flags::FLAGS_logs};
-      {},  // ss.
-      flags::FLAGS_audio,
-      flags::FLAGS_audio_video_sync};
 
   std::vector<std::string> stream_descriptors;
   stream_descriptors.push_back(flags::Stream0());
@@ -260,7 +252,7 @@ void Loopback() {
   if (flags::DurationSecs()) {
     test.RunWithAnalyzer(params);
   } else {
-    test.RunWithRenderers(params);
+    test.RunWithVideoRenderer(params);
   }
 }
 }  // namespace webrtc

webrtc/video/video_quality_test.cc

@@ -34,68 +34,15 @@
 #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,
                       public rtc::VideoSinkInterface<VideoFrame>,
@@ -1055,7 +1002,6 @@ void VideoQualityTest::CreateCapturer(VideoCaptureInput* input) {
 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 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
     fclose(graph_data_output_file);
 }
 
-void VideoQualityTest::RunWithRenderers(const Params& params) {
+void VideoQualityTest::RunWithVideoRenderer(const Params& params) {
   params_ = params;
   CheckParams();
 
@@ -1177,15 +1123,6 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
   // 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 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
 
   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 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
 
   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;
+  receive_stream->Start();
-  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();
   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();
+  receive_stream->Stop();
 
-  call->DestroyVideoReceiveStream(video_receive_stream);
+  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

webrtc/video/video_quality_test.h

@@ -70,8 +70,6 @@ class VideoQualityTest : public test::CallTest {
       // If empty, bitrates are generated in VP9Impl automatically.
       std::vector<SpatialLayer> spatial_layers;
     } ss;
-    bool audio;
-    bool audio_video_sync;
   };
   // (*) Set to -1.1 if generating graph data for simulcast or SVC and the
   // selected stream/layer doesn't have the same resolution as the largest
@@ -79,7 +77,7 @@ class VideoQualityTest : public test::CallTest {
 
   VideoQualityTest();
   void RunWithAnalyzer(const Params& params);
-  void RunWithRenderers(const Params& params);
+  void RunWithVideoRenderer(const Params& params);
 
   static void FillScalabilitySettings(
       Params* params,