Unit test:Make sure WebRtcAudioRenderer always plays sink after starting it.

content/renderer/media/webrtc_audio_renderer_unittest.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc_audio_renderer.h"
 
 #include <string>
 #include <utility>
 #include <vector>
 
@@ skipping 98 matching lines @@
             device_id, device_id == kInvalidOutputDeviceId
                            ? media::OUTPUT_DEVICE_STATUS_ERROR_INTERNAL
                            : media::OUTPUT_DEVICE_STATUS_OK,
             media::AudioParameters(
                 media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                 media::CHANNEL_LAYOUT_STEREO, kHardwareSampleRate, 16,
                 kHardwareBufferSize)));
     if (device_id != kInvalidOutputDeviceId) {
       mock_sink_ = fake_sink;
       EXPECT_CALL(*mock_sink_.get(), Start());
+      EXPECT_CALL(*mock_sink_.get(), Play());
     }
 
     MockCreateAudioRendererSink(source_type, render_frame_id, session_id,
                                 device_id, security_origin);
     return fake_sink;
   }
 
   void TearDown() override {
     renderer_proxy_ = nullptr;
     renderer_ = nullptr;
@@ skipping 143 matching lines @@
   loop.Run();
   EXPECT_EQ(kDefaultOutputDeviceId,
             mock_sink_->GetOutputDeviceInfo().device_id());
 
   EXPECT_CALL(*mock_sink_.get(), Stop());
   EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
   renderer_proxy_->Stop();
 }
 
 }  // namespace content