Implicit audio output device selection for getUserMedia.

content/renderer/media/webrtc_audio_device_unittest.cc

 // Copyright (c) 2012 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 <vector>
 
 #include "base/environment.h"
 #include "base/file_util.h"
 #include "base/files/file_path.h"
 #include "base/path_service.h"
@@ skipping 502 matching lines @@
 
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("created"))).Times(1);
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamPlaying(_, 1, true)).Times(1);
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("closed"))).Times(1);
   EXPECT_CALL(media_observer(),
       OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
+  media::AudioHardwareConfig* hardware_config =
+      RenderThreadImpl::current()->GetAudioHardwareConfig();
+  int sample_rate = hardware_config->GetOutputSampleRate();
+  int frames_per_buffer = hardware_config->GetOutputBufferSize();
+
   scoped_refptr<WebRtcAudioRenderer> renderer =
-      new WebRtcAudioRenderer(kRenderViewId);
+      new WebRtcAudioRenderer(kRenderViewId, -1, sample_rate,
+                              frames_per_buffer);
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get()));
 
   WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
   ASSERT_TRUE(engine.valid());
 
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
   ASSERT_TRUE(base.valid());
   int err = base->Init(webrtc_audio_device.get());
@@ skipping 159 matching lines @@
 
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("created"))).Times(1);
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamPlaying(_, 1, true)).Times(1);
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("closed"))).Times(1);
   EXPECT_CALL(media_observer(),
       OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
+  media::AudioHardwareConfig* hardware_config =
+      RenderThreadImpl::current()->GetAudioHardwareConfig();
+  int sample_rate = hardware_config->GetOutputSampleRate();
+  int frames_per_buffer = hardware_config->GetOutputBufferSize();
+
   scoped_refptr<WebRtcAudioRenderer> renderer =
-      new WebRtcAudioRenderer(kRenderViewId);
+      new WebRtcAudioRenderer(kRenderViewId, -1, sample_rate,
+                              frames_per_buffer);
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get()));
 
   WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
   ASSERT_TRUE(engine.valid());
 
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
   ASSERT_TRUE(base.valid());
   int err = base->Init(webrtc_audio_device.get());
@@ skipping 55 matching lines @@
   if (!HardwareSampleRatesAreValid())
     return;
 
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamPlaying(_, 1, true));
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("closed")));
   EXPECT_CALL(media_observer(),
       OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
+  media::AudioHardwareConfig* hardware_config =
+      RenderThreadImpl::current()->GetAudioHardwareConfig();
+  int sample_rate = hardware_config->GetOutputSampleRate();
+  int frames_per_buffer = hardware_config->GetOutputBufferSize();
+
   scoped_refptr<WebRtcAudioRenderer> renderer =
-      new WebRtcAudioRenderer(kRenderViewId);
+      new WebRtcAudioRenderer(kRenderViewId, -1, sample_rate,
+                              frames_per_buffer);
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get()));
 
   WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
   ASSERT_TRUE(engine.valid());
 
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
   ASSERT_TRUE(base.valid());
   int err = base->Init(webrtc_audio_device.get());
@@ skipping 123 matching lines @@
   EXPECT_CALL(media_observer(),
               OnSetAudioStreamStatus(_, 1, _)).Times(AnyNumber());
   EXPECT_CALL(media_observer(),
               OnSetAudioStreamPlaying(_, 1, true));
   EXPECT_CALL(media_observer(),
               OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
   base::WaitableEvent event(false, false);
   scoped_ptr<MockWebRtcAudioRendererSource> renderer_source(
       new MockWebRtcAudioRendererSource(&event));
+
+  media::AudioHardwareConfig* hardware_config =
+      RenderThreadImpl::current()->GetAudioHardwareConfig();
+  int sample_rate = hardware_config->GetOutputSampleRate();
+  int frames_per_buffer = hardware_config->GetOutputBufferSize();
+
   scoped_refptr<WebRtcAudioRenderer> renderer =
-      new WebRtcAudioRenderer(kRenderViewId);
+      new WebRtcAudioRenderer(kRenderViewId, -1, sample_rate,

[Jói, 2013/09/06 14:49:26]

These parameters and the boilerplate to get the sample rate and frames per
buffer are repeated a few times in the file; maybe make a factory for
WebRtcAudioRenderer that does that work?

[tommi (sloooow) - chröme, 2013/09/06 16:56:54]

Done.  Added a factory method to WebRTCAudioDeviceTest.

+                              frames_per_buffer);
   renderer->Initialize(renderer_source.get());
 
   // Start the timer and playout.
   base::Time start_time = base::Time::Now();
   renderer->Play();
   EXPECT_TRUE(event.TimedWait(TestTimeouts::action_timeout()));
   int delay = (base::Time::Now() - start_time).InMilliseconds();
   PrintPerfResultMs("webrtc_playout_setup_c", "t", delay);
 
   renderer->Stop();
@@ skipping 25 matching lines @@
 #endif
 
 TEST_F(MAYBE_WebRTCAudioDeviceTest,
        MAYBE_WebRtcLoopbackTimeWithSignalProcessing) {
   int latency = RunWebRtcLoopbackTimeTest(audio_manager_.get(), true);
   PrintPerfResultMs("webrtc_loopback_with_signal_processing (100 packets)",
                     "t", latency);
 }
 
 }  // namespace content