Refactoring full stack and loopback tests

webrtc/video/loopback.cc

Index: webrtc/video/loopback.cc
diff --git a/webrtc/video/loopback.cc b/webrtc/video/loopback.cc
index 9083500c1e6a3a80583812f32a97c99e320595b2..2a80b101126f62b031933c9656f9d03f3acb233e 100644
--- a/webrtc/video/loopback.cc
+++ b/webrtc/video/loopback.cc
@@ -27,8 +27,8 @@
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/layer_filtering_transport.h"
+#include "webrtc/test/frame_generator.h"
 #include "webrtc/test/run_loop.h"
-#include "webrtc/test/testsupport/trace_to_stderr.h"
 #include "webrtc/test/video_capturer.h"
 #include "webrtc/test/video_renderer.h"
 #include "webrtc/typedefs.h"
@@ -36,12 +36,6 @@
 namespace webrtc {
 namespace test {
 
-static const int kAbsSendTimeExtensionId = 7;
-
-static const uint32_t kSendSsrc = 0x654321;
-static const uint32_t kSendRtxSsrc = 0x654322;
-static const uint32_t kReceiverLocalSsrc = 0x123456;
-
 static const uint8_t kRtxVideoPayloadType = 96;
 static const uint8_t kVideoPayloadTypeVP8 = 124;
 static const uint8_t kVideoPayloadTypeVP9 = 125;
@@ -50,28 +44,27 @@ Loopback::Loopback(const Config& config)
     : config_(config), clock_(Clock::GetRealTimeClock()) {
 }
 
-Loopback::~Loopback() {
+void Loopback::TestBody() {
+  // Empty. Use Run() instead.

[sprang_webrtc, 2015/09/11 08:01:34]

Should this ever be reached?

[ivica, 2015/09/14 17:12:55]

Whoops, rebase error. This file should've been removed completely.

 }
 
 void Loopback::Run() {
-  rtc::scoped_ptr<test::TraceToStderr> trace_to_stderr_;
-  if (config_.logs)
-    trace_to_stderr_.reset(new test::TraceToStderr);
-
   rtc::scoped_ptr<test::VideoRenderer> local_preview(
-      test::VideoRenderer::Create("Local Preview", config_.width,
-                                  config_.height));
+      test::VideoRenderer::Create("Local Preview", config_.shared.width,
+                                  config_.shared.height));
   rtc::scoped_ptr<test::VideoRenderer> loopback_video(
-      test::VideoRenderer::Create("Loopback Video", config_.width,
-                                  config_.height));
+      test::VideoRenderer::Create("Loopback Video", config_.shared.width,
+                                  config_.shared.height));
+
+  test::DirectTransport transport(config_.shared.pipe);
+  Call::Config call_config(&transport);
 
-  Call::Config call_config;
   call_config.bitrate_config.min_bitrate_bps =
-      static_cast<int>(config_.min_bitrate_kbps) * 1000;
+      static_cast<int>(config_.shared.min_bitrate_bps);
   call_config.bitrate_config.start_bitrate_bps =
-      static_cast<int>(config_.start_bitrate_kbps) * 1000;
+      static_cast<int>(config_.start_bitrate_bps);
   call_config.bitrate_config.max_bitrate_bps =
-      static_cast<int>(config_.max_bitrate_kbps) * 1000;
+      static_cast<int>(config_.shared.max_bitrate_bps);
   rtc::scoped_ptr<Call> call(Call::Create(call_config));
 
   FakeNetworkPipe::Config pipe_config;
@@ -88,54 +81,36 @@ void Loopback::Run() {
   // Loopback, call sends to itself.
   send_transport.SetReceiver(call->Receiver());
 
-  VideoSendStream::Config send_config(&send_transport);
-  send_config.rtp.ssrcs.push_back(kSendSsrc);
-  send_config.rtp.rtx.ssrcs.push_back(kSendRtxSsrc);
-  send_config.rtp.rtx.payload_type = kRtxVideoPayloadType;
-  send_config.rtp.nack.rtp_history_ms = 1000;
-  send_config.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
-
-  send_config.local_renderer = local_preview.get();
   rtc::scoped_ptr<VideoEncoder> encoder;
-  if (config_.codec == "VP8") {
-    encoder.reset(VideoEncoder::Create(VideoEncoder::kVp8));
-  } else if (config_.codec == "VP9") {
-    encoder.reset(VideoEncoder::Create(VideoEncoder::kVp9));
-  } else {
-    // Codec not supported.
-    RTC_NOTREACHED() << "Codec not supported!";
-    return;
-  }
-  const int payload_type =
-      config_.codec == "VP8" ? kVideoPayloadTypeVP8 : kVideoPayloadTypeVP9;
-  send_config.encoder_settings.encoder = encoder.get();
-  send_config.encoder_settings.payload_name = config_.codec;
-  send_config.encoder_settings.payload_type = payload_type;
-
-  VideoEncoderConfig encoder_config(CreateEncoderConfig());
+  SendConfigDefaultValues(&config_.shared);
+  SetupFromSharedConfig(config_.shared, &encoder);
+  send_config_.local_renderer = local_preview.get();
+  send_config_.encoder_settings.payload_type = kVideoPayloadType;
+  // VideoSendStream::Config send_config;
+  // send_config.rtp.ssrcs.push_back(kSendSsrc);              // (!)
+  // send_config.rtp.rtx.ssrcs.push_back(kSendRtxSsrc);       // (!)
+  send_config_.rtp.rtx.payload_type = kRtxVideoPayloadType; // (!)

[sprang_webrtc, 2015/09/11 08:01:34]

What's with the comments here?

[ivica, 2015/09/14 17:12:55]

Same as above, file deleted.

+
+  rtc::scoped_ptr<test::FrameGenerator> frame_generator;
+  SetupScreenshare(config_.shared, &codec_settings_, &frame_generator);

[sprang_webrtc, 2015/09/11 08:01:34]

Is this only used with screenshare?

[ivica, 2015/09/14 17:12:55]

Same as above, file deleted.

 
   VideoSendStream* send_stream =
-      call->CreateVideoSendStream(send_config, encoder_config);
-
-  rtc::scoped_ptr<test::VideoCapturer> capturer(CreateCapturer(send_stream));
-
-  VideoReceiveStream::Config receive_config(&send_transport);
-  receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0];
-  receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
-  receive_config.rtp.nack.rtp_history_ms = 1000;
-  receive_config.rtp.remb = true;
-  receive_config.rtp.rtx[payload_type].ssrc = kSendRtxSsrc;
-  receive_config.rtp.rtx[payload_type].payload_type = kRtxVideoPayloadType;
-  receive_config.rtp.extensions.push_back(
+      call->CreateVideoSendStream(send_config_, encoder_config_);
+
+  test::FrameGeneratorCapturer* capturer = new test::FrameGeneratorCapturer(
+      clock_, send_stream->Input(), frame_generator.release(),
+      config_.shared.fps);
+  EXPECT_TRUE(capturer->Init());
+
+  receive_configs_[0].rtp.rtx[kVideoPayloadType].ssrc = kSendRtxSsrcs[0];
+  receive_configs_[0].rtp.rtx[kVideoPayloadType].payload_type =
+      kRtxVideoPayloadType;
+  receive_configs_[0].rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
-  receive_config.renderer = loopback_video.get();
-  VideoReceiveStream::Decoder decoder =
-      test::CreateMatchingDecoder(send_config.encoder_settings);
-  receive_config.decoders.push_back(decoder);
+  receive_configs_[0].renderer = loopback_video.get();
 
   VideoReceiveStream* receive_stream =
-      call->CreateVideoReceiveStream(receive_config);
+      call->CreateVideoReceiveStream(receive_configs_[0]);
 
   receive_stream->Start();
   send_stream->Start();
@@ -150,34 +125,10 @@ void Loopback::Run() {
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(send_stream);
 
-  delete decoder.decoder;
+  // delete decoder.decoder;

[sprang_webrtc, 2015/09/11 08:01:34]

Remove

[ivica, 2015/09/14 17:12:55]

Same as above, file deleted.

 
   send_transport.StopSending();
 }
 
-VideoEncoderConfig Loopback::CreateEncoderConfig() {
-  VideoEncoderConfig encoder_config;
-  encoder_config.streams = test::CreateVideoStreams(1);
-  VideoStream* stream = &encoder_config.streams[0];
-  stream->width = config_.width;
-  stream->height = config_.height;
-  stream->min_bitrate_bps = static_cast<int>(config_.min_bitrate_kbps) * 1000;
-  stream->max_bitrate_bps = static_cast<int>(config_.max_bitrate_kbps) * 1000;
-  stream->target_bitrate_bps =
-      static_cast<int>(config_.max_bitrate_kbps) * 1000;
-  stream->max_framerate = config_.fps;
-  stream->max_qp = 56;
-  if (config_.num_temporal_layers != 0) {
-    stream->temporal_layer_thresholds_bps.resize(config_.num_temporal_layers -
-                                                 1);
-  }
-  return encoder_config;
-}
-
-test::VideoCapturer* Loopback::CreateCapturer(VideoSendStream* send_stream) {
-  return test::VideoCapturer::Create(send_stream->Input(), config_.width,
-                                     config_.height, config_.fps, clock_);
-}
-
 }  // namespace test
 }  // namespace webrtc