Index: content/renderer/media/webrtc_audio_capturer.cc |
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc |
index 8340ae827f3c711a3f2e4239ffb147a72e296d3c..cca1421d5a2166fd270b50940c0518de938ad052 100644 |
--- a/content/renderer/media/webrtc_audio_capturer.cc |
+++ b/content/renderer/media/webrtc_audio_capturer.cc |
@@ -7,9 +7,11 @@ |
#include "base/logging.h" |
#include "base/metrics/histogram.h" |
#include "base/string_util.h" |
-#include "content/renderer/media/audio_device_factory.h" |
#include "content/renderer/media/audio_hardware.h" |
+#include "content/renderer/media/audio_input_message_filter.h" |
#include "content/renderer/media/webrtc_audio_device_impl.h" |
+#include "content/renderer/render_thread_impl.h" |
+#include "media/audio/audio_input_device.h" |
#include "media/audio/audio_util.h" |
#include "media/audio/sample_rates.h" |
@@ -95,7 +97,7 @@ void WebRtcAudioCapturer::RemoveCapturerSink(WebRtcAudioCapturerSink* sink) { |
} |
void WebRtcAudioCapturer::SetCapturerSource( |
- media::AudioCapturerSource* source) { |
+ const scoped_refptr<media::AudioCapturerSource>& source) { |
DVLOG(1) << "SetCapturerSource()"; |
scoped_refptr<media::AudioCapturerSource> old_source; |
{ |
@@ -155,7 +157,9 @@ bool WebRtcAudioCapturer::Initialize() { |
// Create and configure the default audio capturing source. The |source_| |
// will be overwritten if the client call the source calls |
// SetCapturerSource(). |
- SetCapturerSource(AudioDeviceFactory::NewInputDevice()); |
+ SetCapturerSource(new media::AudioInputDevice( |
+ RenderThreadImpl::current()->audio_input_message_filter(), |
+ RenderThreadImpl::current()->GetIOMessageLoopProxy())); |
UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputChannelLayout", |
channel_layout, media::CHANNEL_LAYOUT_MAX); |