| Index: content/renderer/media/webrtc_audio_device_impl.h
|
| diff --git a/content/renderer/media/webrtc_audio_device_impl.h b/content/renderer/media/webrtc_audio_device_impl.h
|
| index a3bbf6b8ee7f362b696f0911bc66509d78317cc7..de92afe4dda4d47ef36421178eecf030521cd41d 100644
|
| --- a/content/renderer/media/webrtc_audio_device_impl.h
|
| +++ b/content/renderer/media/webrtc_audio_device_impl.h
|
| @@ -7,6 +7,7 @@
|
|
|
| #include <stdint.h>
|
|
|
| +#include <list>
|
| #include <string>
|
| #include <vector>
|
|
|
| @@ -306,15 +307,12 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
|
| // Called on the main renderer thread.
|
| bool SetAudioRenderer(WebRtcAudioRenderer* renderer);
|
|
|
| - // Adds/Removes the capturer to the ADM.
|
| + // Adds/Removes the |capturer| to the ADM. Does NOT take ownership.
|
| + // Capturers must remain valid until RemoveAudioCapturer() is called.
|
| // TODO(xians): Remove these two methods once the ADM does not need to pass
|
| // hardware information up to WebRtc.
|
| - void AddAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer);
|
| - void RemoveAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer);
|
| -
|
| - // Gets the default capturer, which is the last capturer in |capturers_|.
|
| - // The method can be called by both Libjingle thread and main render thread.
|
| - scoped_refptr<WebRtcAudioCapturer> GetDefaultCapturer() const;
|
| + void AddAudioCapturer(WebRtcAudioCapturer* capturer);
|
| + void RemoveAudioCapturer(WebRtcAudioCapturer* capturer);
|
|
|
| // Gets paired device information of the capture device for the audio
|
| // renderer. This is used to pass on a session id, sample rate and buffer
|
| @@ -331,7 +329,7 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
|
| }
|
|
|
| private:
|
| - typedef std::list<scoped_refptr<WebRtcAudioCapturer> > CapturerList;
|
| + typedef std::list<WebRtcAudioCapturer*> CapturerList;
|
| typedef std::list<WebRtcPlayoutDataSource::Sink*> PlayoutDataSinkList;
|
| class RenderBuffer;
|
|
|
| @@ -364,7 +362,9 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
|
| mutable int ref_count_;
|
|
|
| // List of captures which provides access to the native audio input layer
|
| - // in the browser process.
|
| + // in the browser process. The last capturer in this list is considered the
|
| + // "default capturer" by the methods implementing the
|
| + // webrtc::AudioDeviceModule interface.
|
| CapturerList capturers_;
|
|
|
| // Provides access to the audio renderer in the browser process.
|
|
|