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authorhenrika@chromium.org <henrika@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98>2013-03-11 08:38:14 +0000
committerhenrika@chromium.org <henrika@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98>2013-03-11 08:38:14 +0000
commit15c0e988468004e7cb3de746084e65b3c9a32c5d (patch)
treed650475e78ca311c9290c62cb50e0bbcb2527536 /media
parentdad804c692d856e33a85c9ecb530f5c071411401 (diff)
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Adds support of local audio streams for more than one media player.
Main goal is to allow the user to use a local stream as source for more than one audio element. Adding the stream to one extra element will result in an increased output volume and the possibility to tune the volume on each element. If two elements are used, both elements must be muted to silence the audio. The design is now more clear since more work is done in the local renderer and not divided between the capturer (which used to own the FIFO) and the local renderer. The capturer only dispatches the recorded audio out to its sinks and they deal with the rest. - Removes auto-mute on local media streams when rendered by a media player. - Moves local FIFO from WebRtcAudioCapturer to WebRtcLocalAudioRenderer. - Now allows multiple WebRtcLocalAudioRenderer objects where each instance creates its own output stream. - Adds support for the enabled attribute on the MediaStreamTrack object. BUG=none TEST=All existing WebRTC demos and a new test at: https://www.corp.google.com/~henrika/webrtc/multi_local_rendering.html TBR=wjia@chromium.org Review URL: https://codereview.chromium.org/12599002 git-svn-id: svn://svn.chromium.org/chrome/trunk/src@187252 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat (limited to 'media')
-rw-r--r--media/audio/win/audio_low_latency_input_win.cc2
1 files changed, 2 insertions, 0 deletions
diff --git a/media/audio/win/audio_low_latency_input_win.cc b/media/audio/win/audio_low_latency_input_win.cc
index 1e5464f..9b617da 100644
--- a/media/audio/win/audio_low_latency_input_win.cc
+++ b/media/audio/win/audio_low_latency_input_win.cc
@@ -142,6 +142,7 @@ void WASAPIAudioInputStream::Start(AudioInputCallback* callback) {
void WASAPIAudioInputStream::Stop() {
DCHECK(CalledOnValidThread());
+ DVLOG(1) << "WASAPIAudioInputStream::Stop()";
if (!started_)
return;
@@ -167,6 +168,7 @@ void WASAPIAudioInputStream::Stop() {
}
void WASAPIAudioInputStream::Close() {
+ DVLOG(1) << "WASAPIAudioInputStream::Close()";
// It is valid to call Close() before calling open or Start().
// It is also valid to call Close() after Start() has been called.
Stop();