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 86f8716911d84bb9e6a6c8b88174334f611c3cc5..f515c6e24cbc2bcea6d2bcb7fb027e6cc39b23e0 100644 |
--- a/content/renderer/media/webrtc_audio_device_impl.h |
+++ b/content/renderer/media/webrtc_audio_device_impl.h |
@@ -202,7 +202,7 @@ class WebRtcAudioRendererSource { |
virtual ~WebRtcAudioRendererSource() {} |
}; |
-class WebRtcAudioCapturerSink { |
+class PeerConnectionAudioSink { |
public: |
// Callback to deliver the captured interleaved data. |
// |channels| contains a vector of WebRtc VoE channels. |
@@ -216,31 +216,31 @@ class WebRtcAudioCapturerSink { |
// audio processing. |
// The return value is the new microphone volume, in the range of |0, 255]. |
// When the volume does not need to be updated, it returns 0. |
- virtual int CaptureData(const std::vector<int>& channels, |
- const int16* audio_data, |
- int sample_rate, |
- int number_of_channels, |
- int number_of_frames, |
- int audio_delay_milliseconds, |
- int current_volume, |
- bool need_audio_processing, |
- bool key_pressed) = 0; |
+ virtual int OnData(const int16* audio_data, |
+ int sample_rate, |
+ int number_of_channels, |
+ int number_of_frames, |
+ const std::vector<int>& channels, |
+ int audio_delay_milliseconds, |
+ int current_volume, |
+ bool need_audio_processing, |
+ bool key_pressed) = 0; |
// Set the format for the capture audio parameters. |
// This is called when the capture format has changed, and it must be called |
// on the same thread as calling CaptureData(). |
- virtual void SetCaptureFormat(const media::AudioParameters& params) = 0; |
+ virtual void OnSetFormat(const media::AudioParameters& params) = 0; |
protected: |
- virtual ~WebRtcAudioCapturerSink() {} |
+ virtual ~PeerConnectionAudioSink() {} |
}; |
// Note that this class inherits from webrtc::AudioDeviceModule but due to |
// the high number of non-implemented methods, we move the cruft over to the |
// WebRtcAudioDeviceNotImpl. |
class CONTENT_EXPORT WebRtcAudioDeviceImpl |
- : NON_EXPORTED_BASE(public WebRtcAudioDeviceNotImpl), |
- NON_EXPORTED_BASE(public WebRtcAudioCapturerSink), |
+ : NON_EXPORTED_BASE(public PeerConnectionAudioSink), |
+ NON_EXPORTED_BASE(public WebRtcAudioDeviceNotImpl), |
NON_EXPORTED_BASE(public WebRtcAudioRendererSource) { |
public: |
// The maximum volume value WebRtc uses. |
@@ -327,21 +327,21 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl |
// Make destructor private to ensure that we can only be deleted by Release(). |
virtual ~WebRtcAudioDeviceImpl(); |
- // WebRtcAudioCapturerSink implementation. |
+ // PeerConnectionAudioSink implementation. |
// Called on the AudioInputDevice worker thread. |
- virtual int CaptureData(const std::vector<int>& channels, |
- const int16* audio_data, |
- int sample_rate, |
- int number_of_channels, |
- int number_of_frames, |
- int audio_delay_milliseconds, |
- int current_volume, |
- bool need_audio_processing, |
- bool key_pressed) OVERRIDE; |
+ virtual int OnData(const int16* audio_data, |
+ int sample_rate, |
+ int number_of_channels, |
+ int number_of_frames, |
+ const std::vector<int>& channels, |
+ int audio_delay_milliseconds, |
+ int current_volume, |
+ bool need_audio_processing, |
+ bool key_pressed) OVERRIDE; |
- // Called on the main render thread. |
- virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE; |
+ // Called on the AudioInputDevice worker thread. |
+ virtual void OnSetFormat(const media::AudioParameters& params) OVERRIDE; |
// WebRtcAudioRendererSource implementation. |