zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
12,074 Commits 1 Branch 0 Tags
07df20b51ab50078263f43a9a516b7ac7130d42e
Commit Graph

19 Commits

Author SHA1 Message Date
pbos
1ba8d39a9c Remove webrtc/stream.h and unutilized inheritance. 
Removes inheritance and a virtual call. Also removes a root header that
would have needed to be moved into a subdirectory otherwise to prevent
circular dependencies.

BUG=webrtc:4243
R=kjellander@webrtc.org, solenberg@webrtc.org
TBR=mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/1924793002
Cr-Commit-Position: refs/heads/master@{#12586}
 2016-05-02 03:18:36 +00:00
mflodman
3d7db263b9 Switch voice transport to use Call and Stream instead of VoENetwork. 
VoENetwork is kept for now, but is not really used anylonger.

webrtcvoiceengine is changed to have the same behavior for unsignaled
ssrc as video has, which is reflected by disabling one test case and
this will be discussed and followed up.

BUG=webrtc:5079

TBR=tommi

Review-Url: https://codereview.webrtc.org/1909333002
Cr-Commit-Position: refs/heads/master@{#12555}
 2016-04-29 07:57:21 +00:00
kwiberg
fffa42b57e Replace scoped_ptr with unique_ptr in webrtc/audio/ 
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1706183002

Cr-Commit-Position: refs/heads/master@{#11723}
 2016-02-23 18:46:39 +00:00
stefan
ba4c0e45ff Add send-side BWE to WebRtcVoiceEngine under a finch experiment. 
This adds negotiation of both transport sequence number and transport
feedback. Only offers transport seq num if the
WebRTC-Audio-SendSideBwe finch experiment is enabled.

TBR=mflodman@webrtc.org
BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1604563002

Cr-Commit-Position: refs/heads/master@{#11487}
 2016-02-04 12:12:31 +00:00
deadbeef
884f58523a Storing raw audio sink for default audio track. 
BUG=webrtc:5250

Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
Cr-Commit-Position: refs/heads/master@{#11230}

Review URL: https://codereview.webrtc.org/1551813002

Cr-Commit-Position: refs/heads/master@{#11275}
 2016-01-15 17:20:08 +00:00
deadbeef
2d110be77f Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) 
Reason for revert:
tommi pointed out that using a refptr for the sink may cause issues. Will reland with a slightly different approach.

Original issue's description:
> Storing raw audio sink for default audio track.
>
> BUG=webrtc:5250
>
> Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
> Cr-Commit-Position: refs/heads/master@{#11230}

TBR=pthatcher@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5250

Review URL: https://codereview.webrtc.org/1588693002

Cr-Commit-Position: refs/heads/master@{#11241}
 2016-01-13 20:00:29 +00:00
deadbeef
e591f9377f Storing raw audio sink for default audio track. 
BUG=webrtc:5250

Review URL: https://codereview.webrtc.org/1551813002

Cr-Commit-Position: refs/heads/master@{#11230}
 2016-01-13 00:45:33 +00:00
Stefan Holmer
3842c5c7f7 Wire-up BWE feedback for audio receive streams. 
Also wires up receiving transport sequence numbers.

BUG=webrtc:5263
R=mflodman@webrtc.org, pbos@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1535963002 .

Cr-Commit-Position: refs/heads/master@{#11220}
 2016-01-12 12:55:11 +00:00
Tommi
f888bb58da Support for unmixed remote audio into tracks. 
BUG=chromium:121673
R=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1505253004 .

Cr-Commit-Position: refs/heads/master@{#10995}
 2015-12-12 00:37:14 +00:00
Fredrik Solenberg
a4527c89e7 Add comments about the Audio parts of the public Call API being WIP. 
BUG=webrtc:4690
R=kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1493933003 .

Cr-Commit-Position: refs/heads/master@{#10882}
 2015-12-03 12:06:31 +00:00
Fredrik Solenberg
4f4ec0a927 Re-Land: Implement AudioReceiveStream::GetStats(). 
R=tommi@webrtc.org
BUG=webrtc:4690

Committed: a457752f4a

Review URL: https://codereview.webrtc.org/1390753002 .

Cr-Commit-Position: refs/heads/master@{#10369}
 2015-10-22 08:49:39 +00:00
solenberg
43e83d44f0 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ ) 
Reason for revert:
webrtc_perf_tests started failing on Win32 Release, Mac32 Release and Linux64 Release (all running large tests). These were not caught by try bots.

Original issue's description:
> Implement AudioReceiveStream::GetStats().
>
> R=tommi@webrtc.org
> TBR=hta@webrtc.org
> BUG=webrtc:4690
>
> Committed: a457752f4a

TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1411083006

Cr-Commit-Position: refs/heads/master@{#10340}
 2015-10-20 13:41:06 +00:00
Fredrik Solenberg
a457752f4a Implement AudioReceiveStream::GetStats(). 
R=tommi@webrtc.org
TBR=hta@webrtc.org
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1390753002 .

Cr-Commit-Position: refs/heads/master@{#10338}
 2015-10-20 13:01:55 +00:00
solenberg
cf18b34cf3 Align new VoE API with design. 
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1376153003

Cr-Commit-Position: refs/heads/master@{#10136}
 2015-10-01 15:13:46 +00:00
pbos
6bb1b6e7fe Control combined_audio_video_bwe with config bool. 
Permits setting RTP extensions for AudioReceiveStream without enabling
combined A/V BWE. This prevents spamming the log with "Failed to find
extension id:".

BUG=webrtc:4870
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1256803004

Cr-Commit-Position: refs/heads/master@{#9633}
 2015-07-24 14:10:25 +00:00
Jelena Marusic
cd6702282a Define Stream base classes 
BUG=webrtc:4690

Defined classes Stream, SendStream and ReceiveStream. Inherited existing stream classes from either SendStream or ReceiveStream.
This is a step towards having a Transport associated with streams instead of a Call. It will allow a lot of code in the Call to be media type agnostic.

R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1226123005 .

Cr-Commit-Position: refs/heads/master@{#9591}
 2015-07-16 07:30:20 +00:00
pbos
8fc7fa798f Base A/V synchronization on sync_labels. 
Groups of streams that should be synchronized are signalled through
SDP. These should be used rather than synchronizing the first-added
video stream to the first-added audio stream implicitly.

BUG=webrtc:4667
R=hta@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1181653002

Cr-Commit-Position: refs/heads/master@{#9586}
 2015-07-15 15:03:04 +00:00
Fredrik Solenberg
04f4931ef0 VoE2 API draft 
BUG=4690
R=jmarusic@webrtc.org, kwiberg@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50029004

Cr-Commit-Position: refs/heads/master@{#9392}
 2015-06-08 11:05:07 +00:00
Fredrik Solenberg
23fba1ffa0 Add AudioReceiveStream to Call API. 
BUG=4574
R=kwiberg@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51749004

Cr-Commit-Position: refs/heads/master@{#9114}
 2015-04-29 13:24:10 +00:00