Commit graph

3 commits

Author SHA1 Message Date
henrika
90bace0958 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
(this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)

This SetAudioPlayout method lets applications disable audio playout while
still processing incoming audio data and generating statistics on the
received audio.

This may be useful if the application wants to set up media flows as
soon as possible, but isn't ready to play audio yet. Currently, native
applications don't have any API point to control this, unless they
completely implement their own AudioDeviceModule.

The SetAudioRecording works in a similar fashion but for the recorded
audio. One difference is that calling SetAudioRecording(false) does not
keep any audio processing alive.

TBR=solenberg

Bug: webrtc:7313
Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
Reviewed-on: https://webrtc-review.googlesource.com/16180
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20499}
2017-10-31 12:35:42 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/call/audio_state.h (Browse further)