Refactor FakeAudioDevice to have separate methods for starting recording and playout.

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.
  */
 
@@ skipping 122 matching lines @@
   if (num_audio_streams_ > 0)
     DestroyVoiceEngines();
 }
 
 void CallTest::Start() {
   if (video_send_stream_)
     video_send_stream_->Start();
   for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
     video_recv_stream->Start();
   if (audio_send_stream_) {
-    fake_send_audio_device_->Start();
+    fake_send_audio_device_->StartRecordingSine(1000, 256);
     audio_send_stream_->Start();
     EXPECT_EQ(0, voe_send_.base->StartSend(voe_send_.channel_id));
   }
   for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
     audio_recv_stream->Start();
   if (!audio_receive_streams_.empty()) {
-    fake_recv_audio_device_->Start();
+    fake_recv_audio_device_->StartPlayout();
     EXPECT_EQ(0, voe_recv_.base->StartPlayout(voe_recv_.channel_id));
   }
   for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
     flexfec_recv_stream->Start();
   if (frame_generator_capturer_.get() != NULL)
     frame_generator_capturer_->Start();
 }
 
 void CallTest::Stop() {
   if (frame_generator_capturer_.get() != NULL)
     frame_generator_capturer_->Stop();
   for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
     flexfec_recv_stream->Stop();
   if (!audio_receive_streams_.empty()) {
-    fake_recv_audio_device_->Stop();
+    fake_recv_audio_device_->StopPlayout();
     EXPECT_EQ(0, voe_recv_.base->StopPlayout(voe_recv_.channel_id));
   }
   for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
     audio_recv_stream->Stop();
   if (audio_send_stream_) {
-    fake_send_audio_device_->Stop();
+    fake_send_audio_device_->StopRecording();
     EXPECT_EQ(0, voe_send_.base->StopSend(voe_send_.channel_id));
     audio_send_stream_->Stop();
   }
   for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
     video_recv_stream->Stop();
   if (video_send_stream_)
     video_send_stream_->Stop();
 }
 
 void CallTest::CreateCalls(const Call::Config& sender_config,
@@ skipping 120 matching lines @@
                                             int width,
                                             int height) {
   frame_generator_capturer_.reset(
       test::FrameGeneratorCapturer::Create(width, height, framerate, clock_));
   video_send_stream_->SetSource(
       frame_generator_capturer_.get(),
       VideoSendStream::DegradationPreference::kBalanced);
 }
 
 void CallTest::CreateFakeAudioDevices() {
-  fake_send_audio_device_.reset(new FakeAudioDevice(
-      clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"),
-      DriftingClock::kNoDrift));
-  fake_recv_audio_device_.reset(new FakeAudioDevice(
-      clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"),
-      DriftingClock::kNoDrift));
+  fake_send_audio_device_.reset(
+      new FakeAudioDevice(clock_, DriftingClock::kNoDrift));
+  fake_recv_audio_device_.reset(
+      new FakeAudioDevice(clock_, DriftingClock::kNoDrift));
 }
 
 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_.Copy(), video_encoder_config_.Copy());
@@ skipping 181 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc