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author | xians@chromium.org <xians@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2012-11-16 18:11:43 +0000 |
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committer | xians@chromium.org <xians@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2012-11-16 18:11:43 +0000 |
commit | 4bc6545870f3d086c9a9aadd1dd6310a01b5ebe5 (patch) | |
tree | 2fd31bc4a457ae3f4df3c50548f17bf14a709656 /content/content_renderer.gypi | |
parent | 021b42b445d17573b4c53522f553e3558add3797 (diff) | |
download | chromium_src-4bc6545870f3d086c9a9aadd1dd6310a01b5ebe5.zip chromium_src-4bc6545870f3d086c9a9aadd1dd6310a01b5ebe5.tar.gz chromium_src-4bc6545870f3d086c9a9aadd1dd6310a01b5ebe5.tar.bz2 |
Break down the webrtc code and AudioInputDevice into a WebRtcAudioCapturer.
This capturer contains a source (AudioInputDevice) and a sink (WebRtcAudioDeviceImpl) by default. What it does is to get data from AudioInputDevice via CaptureCallback::Capture() callback,
and forward the data to WebRtcAudioDeviceImpl.
The source can be over written by API:
SetCapturerSource(media::AudioCapturerSource* source)
This capture currently only support one sink (WebRtcAudioDeviceImpl), but this can be extended to multiple sinks for the future.
Design doc: https://docs.google.com/a/google.com/document/d/1FmiXtk1pxFlAw_CWwbfG-EQ4Syi4vpnZm6GWMyJ1UfU/edit
TBR=tommi@chromium.org
BUG=157306
TEST=manual test
Review URL: https://codereview.chromium.org/11366244
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@168242 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat (limited to 'content/content_renderer.gypi')
-rw-r--r-- | content/content_renderer.gypi | 4 |
1 files changed, 3 insertions, 1 deletions
diff --git a/content/content_renderer.gypi b/content/content_renderer.gypi index 516dfae..efc9e4a 100644 --- a/content/content_renderer.gypi +++ b/content/content_renderer.gypi @@ -343,7 +343,9 @@ 'renderer/media/rtc_video_capture_delegate.cc', 'renderer/media/rtc_video_capture_delegate.h', 'renderer/media/rtc_video_capturer.cc', - 'renderer/media/rtc_video_capturer.h', + 'renderer/media/rtc_video_capturer.h', + 'renderer/media/webrtc_audio_capturer.cc', + 'renderer/media/webrtc_audio_capturer.h', 'renderer/media/webrtc_audio_device_impl.cc', 'renderer/media/webrtc_audio_device_impl.h', 'renderer/media/webrtc_audio_renderer.cc', |