9a7c838ec4
Adding stddef.h to opus_inst.h.
...
This is to prevent size_t from undefined. This does not happen in current WebRTC since the sources that opus_inst.h gets used have proper definitions. But it would be good to add the definition in itself.
Review URL: https://codereview.webrtc.org/1446093003
Cr-Commit-Position: refs/heads/master@{#10653}
2015-11-16 16:07:04 +00:00
3cea256806
Reland "Prevent Opus DTX from generating intermittent noise during silence"
...
The original CL is reviewed at
https://codereview.webrtc.org/1415173005/
A silly mistake was made at the last patch set, and the CL was reverted. This CL is to fix and reland it.
BUG=
Review URL: https://codereview.webrtc.org/1422213003
Cr-Commit-Position: refs/heads/master@{#10574}
2015-11-10 11:49:32 +00:00
b4a753fdb5
Revert of Prevent Opus DTX from generating intermittent noise during silence (patchset #10 id:250001 of https://codereview.webrtc.org/1415173005/ )
...
Reason for revert:
Breaks voe_auto_test on all three "large tests bots".
https://build.chromium.org/p/client.webrtc/builders/Win32%20Release%20%5Blarge%20tests%5D/builds/5630/steps/voe_auto_test/logs/stdio
https://build.chromium.org/p/client.webrtc/builders/Mac32%20Release%20%5Blarge%20tests%5D/builds/5599/steps/voe_auto_test/logs/stdio
https://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/5645/steps/voe_auto_test/logs/stdio
Notice these bots are no longer a part of the default trybot set, so they have to be run manually when working with code that affects their tests (they were removed as they queued up all the time in the CQ, and usually don't catch breakages).
Original issue's description:
> Prevent Opus DTX from generating intermittent noise during silence.
>
> Opus may have an internal error that causes this. Here we make a workaround by adding some small disturbance to the input signals to break a long sequence of zeros.
>
> BUG=webrtc:5127
>
> Committed: https://crrev.com/f475add57eada116bc960fe2935876ec8c077977
> Cr-Commit-Position: refs/heads/master@{#10565}
TBR=tina.legrand@webrtc.org ,kwiberg@webrtc.org ,solenberg@webrtc.org ,minyue@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5127
Review URL: https://codereview.webrtc.org/1428613004
Cr-Commit-Position: refs/heads/master@{#10567}
2015-11-09 21:27:11 +00:00
f475add57e
Prevent Opus DTX from generating intermittent noise during silence.
...
Opus may have an internal error that causes this. Here we make a workaround by adding some small disturbance to the input signals to break a long sequence of zeros.
BUG=webrtc:5127
Review URL: https://codereview.webrtc.org/1415173005
Cr-Commit-Position: refs/heads/master@{#10565}
2015-11-09 18:08:20 +00:00
0ca768b131
Adding DTX to WebRTC Opus wrapper (relanding).
...
This is relanding of r7846, which failed since the unit test depended on whether Opus is in fixed-point or float-point.
See the review of r7846 here:
https://webrtc-codereview.appspot.com/13219004/
Patch set 1 is the same as r7846. Further fixes are found in patch set 2 and later.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7878 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 16:09:35 +00:00
19dd129c69
Revert 7846 "Adding DTX to WebRTC Opus wrapper"
...
> Adding DTX to WebRTC Opus wrapper
>
> This is a step toward adding Opus DTX support in WebRTC.
>
> Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See
>
> https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html
>
> We transmit the first 1-byte packet to let decoder be in-sync
>
> BUG=webrtc:1014
> R=henrik.lundin@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/13219004
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7848 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 15:11:15 +00:00
4321f175f1
Adding DTX to WebRTC Opus wrapper
...
This is a step toward adding Opus DTX support in WebRTC.
Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See
https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html
We transmit the first 1-byte packet to let decoder be in-sync
BUG=webrtc:1014
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7846 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 13:27:39 +00:00
33ccdfa1f5
Relanding r7807.
...
r7807 was reverted to be excluded from the cause of a failure.
It has been verified and can reland now.
BUG=
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7810 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 12:14:12 +00:00
52bc4f4797
Revert 7807 "Removing unused opus wrapper APIs."
...
> Removing unused opus wrapper APIs.
>
> WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().
>
> WebRtcOpus_DecodePlcMaster/Slave() are also removed.
>
> BUG=
> R=henrik.lundin@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/28139004
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7809 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 11:00:50 +00:00
e54a6342dd
Removing unused opus wrapper APIs.
...
WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().
WebRtcOpus_DecodePlcMaster/Slave() are also removed.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7807 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 08:47:25 +00:00
0040a6ef97
This is a setup to solve
...
https://code.google.com/p/webrtc/issues/detail?id=1906
In particular, we add an API to call Opus's set maximum bandwidth to prevent the encoder from coding audio content beyond this bandwidth so as to increase computation and transmission efficiency (without affecting sampling rate).
BUG=
R=henrik.lundin@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6817 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-04 14:41:57 +00:00
