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-rw-r--r--content/renderer/media/webrtc_local_audio_renderer.cc5
1 files changed, 2 insertions, 3 deletions
diff --git a/content/renderer/media/webrtc_local_audio_renderer.cc b/content/renderer/media/webrtc_local_audio_renderer.cc
index cc5ba77..da68006 100644
--- a/content/renderer/media/webrtc_local_audio_renderer.cc
+++ b/content/renderer/media/webrtc_local_audio_renderer.cc
@@ -9,8 +9,8 @@
#include "base/message_loop_proxy.h"
#include "base/synchronization/lock.h"
#include "content/renderer/media/audio_device_factory.h"
-#include "content/renderer/media/renderer_audio_output_device.h"
#include "content/renderer/media/webrtc_audio_capturer.h"
+#include "media/audio/audio_output_device.h"
#include "media/base/audio_bus.h"
namespace content {
@@ -158,12 +158,11 @@ void WebRtcLocalAudioRenderer::Start() {
source_params.sample_rate(),
source_params.bits_per_sample(),
2 * source_params.frames_per_buffer());
- sink_ = AudioDeviceFactory::NewOutputDevice();
+ sink_ = AudioDeviceFactory::NewOutputDevice(source_render_view_id_);
// TODO(henrika): we could utilize the unified audio here instead and do
// sink_->InitializeIO(sink_params, 2, callback_.get());
// It would then be possible to avoid using the WebRtcAudioCapturer.
sink_->Initialize(sink_params, this);
- sink_->SetSourceRenderView(source_render_view_id_);
// Start the capturer and local rendering. Note that, the capturer is owned
// by the WebRTC ADM and might already bee running.