Added A/V sync tests with drifting clocks.

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 218 matching lines @@
     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::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
+                                                     float drift) {
+  VideoStream stream = video_encoder_config_.streams.back();
+  frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
+      video_send_stream_->Input(), stream.width, stream.height,
+      stream.max_framerate / (1. + drift), clock));
+}
+
 void CallTest::CreateFrameGeneratorCapturer() {
   VideoStream stream = video_encoder_config_.streams.back();
   frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
       video_send_stream_->Input(), stream.width, stream.height,
       stream.max_framerate, clock_));
 }
 
+void CallTest::CreateFakeAudioDevicesWithDrift(float drift) {
+  fake_send_audio_device_.reset(new FakeAudioDevice(
+      clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"), drift));
+  fake_recv_audio_device_.reset(new FakeAudioDevice(
+      clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"), 0));

[stefan-webrtc, 2016/02/09 14:02:20]

This always overwrites the drifting device

[danilchap, 2016/02/09 14:56:49]

oops, didn't notice this function is not used.

+}
+
 void CallTest::CreateFakeAudioDevices() {
   fake_send_audio_device_.reset(new FakeAudioDevice(
-      clock_, test::ResourcePath("voice_engine/audio_long16", "pcm")));
+      clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"), 0));
   fake_recv_audio_device_.reset(new FakeAudioDevice(
-      clock_, test::ResourcePath("voice_engine/audio_long16", "pcm")));
+      clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"), 0));
 }
 
 void CallTest::CreateVideoStreams() {
   RTC_DCHECK(video_send_stream_ == nullptr);
   RTC_DCHECK(video_receive_streams_.empty());
   RTC_DCHECK(audio_send_stream_ == nullptr);
   RTC_DCHECK(audio_receive_streams_.empty());
 
   video_send_stream_ = sender_call_->CreateVideoSendStream(
       video_send_config_, video_encoder_config_);
@@ skipping 174 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc