Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1945)

Unified Diff: content/renderer/media/webrtc_audio_device_impl.h

Issue 90743004: Add generic interfaces for the sinks of the media stream audio track (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: fixed the nits. Created 7 years ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
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.
« no previous file with comments | « content/renderer/media/webrtc_audio_capturer_unittest.cc ('k') | content/renderer/media/webrtc_audio_device_impl.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698