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author | justinlin@chromium.org <justinlin@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-05-31 17:58:09 +0000 |
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committer | justinlin@chromium.org <justinlin@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-05-31 17:58:09 +0000 |
commit | 6e37ba8c76c2240ec96bb95096cfe33e9c41eda6 (patch) | |
tree | 9006c83d118e5cde850fac75f8545a2caa024f8f /content/renderer/media/webrtc_audio_device_impl.h | |
parent | 9faccc4fecfc865078149ae258aa53452831ec9b (diff) | |
download | chromium_src-6e37ba8c76c2240ec96bb95096cfe33e9c41eda6.zip chromium_src-6e37ba8c76c2240ec96bb95096cfe33e9c41eda6.tar.gz chromium_src-6e37ba8c76c2240ec96bb95096cfe33e9c41eda6.tar.bz2 |
Revert 203427 "Hook up the device selection to the WebAudio live..."
Possibly caused content_browsertests to fail on Mac.
> Hook up the device selection to the WebAudio live audio.
> WebAudio live audio needs to pass the session_id to the browser process so that Chrome can open the correct input device for unitfied IO.
>
> This CL looks big because it touches quite some interfaces from the render to the browser. But the change is simple and basically adding a session_id/device_id to the classes. All the changes some together and it is very hard to break it down.
> It also makes the media output code more similar to the media input code as well, and it will be easier to merge them for the future.
>
>
> BUG=147327
> TEST=http://chromium.googlecode.com/svn/trunk/samples/audio/visualizer-live.html
> Change the device using the camera icon on the right of the omnibox, then reload. Verify the sound is coming from the correct input device.
>
> Review URL: https://chromiumcodereview.appspot.com/15721002
TBR=xians@chromium.org
Review URL: https://codereview.chromium.org/15725013
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@203434 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat (limited to 'content/renderer/media/webrtc_audio_device_impl.h')
-rw-r--r-- | content/renderer/media/webrtc_audio_device_impl.h | 14 |
1 files changed, 8 insertions, 6 deletions
diff --git a/content/renderer/media/webrtc_audio_device_impl.h b/content/renderer/media/webrtc_audio_device_impl.h index 960d0d0..e740a59 100644 --- a/content/renderer/media/webrtc_audio_device_impl.h +++ b/content/renderer/media/webrtc_audio_device_impl.h @@ -24,11 +24,13 @@ // webrtc::AudioDeviceModule which makes it possible for a user (e.g. webrtc:: // VoiceEngine) to register this class as an external AudioDeviceModule (ADM). // Then WebRtcAudioDeviceImpl::SetSessionId() needs to be called to set the -// session id that tells which device to use. The user can then call -// WebRtcAudioDeviceImpl::StartPlayout() and -// WebRtcAudioDeviceImpl::StartRecording() from the render process to initiate -// and start audio rendering and capturing in the browser process. IPC is -// utilized to set up the media streams. +// session id that tells which device to use. The user can either get the +// session id from the MediaStream or use a value of 1 (AudioInputDeviceManager +// ::kFakeOpenSessionId), the later will open the default device without going +// through the MediaStream. The user can then call WebRtcAudioDeviceImpl:: +// StartPlayout() and WebRtcAudioDeviceImpl::StartRecording() from the render +// process to initiate and start audio rendering and capturing in the browser +// process. IPC is utilized to set up the media streams. // // Usage example: // @@ -37,7 +39,7 @@ // { // scoped_refptr<WebRtcAudioDeviceImpl> external_adm; // external_adm = new WebRtcAudioDeviceImpl(); -// external_adm->SetSessionId(session_id); +// external_adm->SetSessionId(1); // VoiceEngine* voe = VoiceEngine::Create(); // VoEBase* base = VoEBase::GetInterface(voe); // base->Init(external_adm); |