Add tests for verifying transport feedback for audio and video.

webrtc/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.
  */
 #include "webrtc/base/checks.h"
@@ skipping 42 matching lines @@
   CreateSenderCall(send_config);
   if (test->ShouldCreateReceivers()) {
     Call::Config recv_config(test->GetReceiverCallConfig());
     if (num_audio_streams_ > 0) {
       AudioState::Config audio_state_config;
       audio_state_config.voice_engine = voe_recv_.voice_engine;
       recv_config.audio_state = AudioState::Create(audio_state_config);
     }
     CreateReceiverCall(recv_config);
   }
+  test->OnCallsCreated(sender_call_.get(), receiver_call_.get());

[mflodman, 2016/01/14 08:53:16]

Any particular reason for moving this?

[stefan-webrtc, 2016/01/14 08:59:16]

There was a reason. The previous version of the test had a separate transport
class which monitored packets, and it needed access to the receiver call to
create its DirectTransport.

Now I have refactored even more, and this is no longer needed, but I think it
makes sense to call OnCallsCreated as early as possible anyway to give the tests
more freedom.

   send_transport_.reset(test->CreateSendTransport(sender_call_.get()));
   receive_transport_.reset(test->CreateReceiveTransport());
-  test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
 
   if (test->ShouldCreateReceivers()) {
     send_transport_->SetReceiver(receiver_call_->Receiver());
     receive_transport_->SetReceiver(sender_call_->Receiver());
   } else {
     // Sender-only call delivers to itself.
     send_transport_->SetReceiver(sender_call_->Receiver());
     receive_transport_->SetReceiver(nullptr);
   }
 
@@ skipping 14 matching lines @@
 
   if (num_video_streams_ > 0) {
     CreateVideoStreams();
     test->OnVideoStreamsCreated(video_send_stream_, video_receive_streams_);
   }
   if (num_audio_streams_ > 0) {
     CreateAudioStreams();
     test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_);
   }
 
-  CreateFrameGeneratorCapturer();
-  test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
+  if (num_video_streams_ > 0) {
+    CreateFrameGeneratorCapturer();
+    test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
+  }
 
   Start();
   test->PerformTest();
   send_transport_->StopSending();
   receive_transport_->StopSending();
   Stop();
 
   DestroyStreams();
   DestroyCalls();
   if (num_audio_streams_ > 0)
@@ skipping 60 matching lines @@
   sender_call_.reset();
   receiver_call_.reset();
 }
 
 void CallTest::CreateSendConfig(size_t num_video_streams,
                                 size_t num_audio_streams,
                                 Transport* send_transport) {
   RTC_DCHECK(num_video_streams <= kNumSsrcs);
   RTC_DCHECK_LE(num_audio_streams, 1u);
   RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0);
-  video_send_config_ = VideoSendStream::Config(send_transport);
-  video_send_config_.encoder_settings.encoder = &fake_encoder_;
-  video_send_config_.encoder_settings.payload_name = "FAKE";
-  video_send_config_.encoder_settings.payload_type = kFakeVideoSendPayloadType;
-  video_send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
-  video_encoder_config_.streams = test::CreateVideoStreams(num_video_streams);
-  for (size_t i = 0; i < num_video_streams; ++i)
-    video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
-  video_send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId));
+  if (num_video_streams > 0) {
+    video_send_config_ = VideoSendStream::Config(send_transport);
+    video_send_config_.encoder_settings.encoder = &fake_encoder_;
+    video_send_config_.encoder_settings.payload_name = "FAKE";
+    video_send_config_.encoder_settings.payload_type =
+        kFakeVideoSendPayloadType;
+    video_send_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
+    video_encoder_config_.streams = test::CreateVideoStreams(num_video_streams);
+    for (size_t i = 0; i < num_video_streams; ++i)
+      video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
+    video_send_config_.rtp.extensions.push_back(RtpExtension(
+        RtpExtension::kVideoRotation, 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;
   }
 }
 
 void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
-  RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
   RTC_DCHECK(video_receive_configs_.empty());
   RTC_DCHECK(allocated_decoders_.empty());
-  RTC_DCHECK(num_audio_streams_ == 0 || voe_send_.channel_id >= 0);
-  VideoReceiveStream::Config video_config(rtcp_send_transport);
-  video_config.rtp.remb = true;
-  video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
-  for (const RtpExtension& extension : video_send_config_.rtp.extensions)
-    video_config.rtp.extensions.push_back(extension);
-  for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) {
-    VideoReceiveStream::Decoder decoder =
-        test::CreateMatchingDecoder(video_send_config_.encoder_settings);
-    allocated_decoders_.push_back(decoder.decoder);
-    video_config.decoders.clear();
-    video_config.decoders.push_back(decoder);
-    video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
-    video_receive_configs_.push_back(video_config);
+  if (num_video_streams_ > 0) {
+    RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
+    VideoReceiveStream::Config video_config(rtcp_send_transport);
+    video_config.rtp.remb = true;
+    video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
+    for (const RtpExtension& extension : video_send_config_.rtp.extensions)
+      video_config.rtp.extensions.push_back(extension);
+    for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) {
+      VideoReceiveStream::Decoder decoder =
+          test::CreateMatchingDecoder(video_send_config_.encoder_settings);
+      allocated_decoders_.push_back(decoder.decoder);
+      video_config.decoders.clear();
+      video_config.decoders.push_back(decoder);
+      video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
+      video_receive_configs_.push_back(video_config);
+    }
   }
 
   RTC_DCHECK(num_audio_streams_ <= 1);
   if (num_audio_streams_ == 1) {
+    RTC_DCHECK(voe_send_.channel_id >= 0);
     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_receive_configs_.push_back(audio_config);
   }
 }
 
 void CallTest::CreateFrameGeneratorCapturer() {
@@ skipping 195 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc