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 9337756466598745357db83f5d09cca21061eaf4..0533ce708b858ee9a2b4288e79ece8770815b065 100644 |
--- a/content/renderer/media/webrtc_audio_capturer.cc |
+++ b/content/renderer/media/webrtc_audio_capturer.cc |
@@ -175,8 +175,8 @@ bool WebRtcAudioCapturer::Initialize() { |
// Create and configure the default audio capturing source. |
SetCapturerSourceInternal( |
- AudioDeviceFactory::NewInputDevice(render_frame_id_), channel_layout, |
- device_info_.device.input.sample_rate); |
+ AudioDeviceFactory::NewAudioCapturerSource(render_frame_id_), |
+ channel_layout, device_info_.device.input.sample_rate); |
// Add the capturer to the WebRtcAudioDeviceImpl since it needs some hardware |
// information from the capturer. |
@@ -351,9 +351,9 @@ void WebRtcAudioCapturer::EnablePeerConnectionMode() { |
// Create a new audio stream as source which will open the hardware using |
// WebRtc native buffer size. |
- SetCapturerSourceInternal(AudioDeviceFactory::NewInputDevice(render_frame_id), |
- input_params.channel_layout(), |
- input_params.sample_rate()); |
+ SetCapturerSourceInternal( |
+ AudioDeviceFactory::NewAudioCapturerSource(render_frame_id), |
+ input_params.channel_layout(), input_params.sample_rate()); |
} |
void WebRtcAudioCapturer::Start() { |