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28 Commits

Author SHA1 Message Date
minyue@webrtc.org
adee8f9242 Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate 
This is to maintain the consistency with the Opus codec option "maxplaybackrate" defined in http://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03

BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7038 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 12:28:06 +00:00
kjellander@webrtc.org
34a865a038 Roll chromium_revision 288251:289723 
Mainly to pick up the libvpx.gyp change in r288724
to unblock https://webrtc-codereview.appspot.com/16229005/

Overview of changes in Chrome DEPS:
$ svn diff http://src.chromium.org/chrome/trunk/src/DEPS -r 288251:289723
which can be compared with the output of:
$ svn cat http://webrtc.googlecode.com/svn/trunk/DEPS | grep chromium_deps | sed 's/^ *//' | sort | uniq

In a WebRTC checkout, that sums up to the following relevant changes:
* src/buildtools 59b932:567f0a
* testing/gtest 643:692
* testing/gmock 410:485
* third_party/boringssl/src 533cbe:c3d796
* third_party/libvpx 287125:289332
* third_party/libyuv 1035:1038
* third_party/nss 287121:289430
* third_party/opus/src 256783:289085
* tools/gyp 1959:1964

BUG=2863, chromium:339647
TEST=Local testing as trybots currently cannot handle DEPS changes properly due to http://crbug.com/385594
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6913 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-16 18:49:55 +00:00
minyue@webrtc.org
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
minyue@webrtc.org
f563e85ab0 This is to re-open an earlier CL 
https://webrtc-codereview.appspot.com/16619005/

which is reverted due to an issue in audio conference mixer.

This issue has been solved in
https://webrtc-codereview.appspot.com/20779004/

BUG=webrtc:3155
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18819005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 21:11:27 +00:00
minyue@webrtc.org
d42da54768 Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..." 
> Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling.
> 
> TEST=passed_all_trybots
> R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/16619005

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6462 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 09:50:12 +00:00
minyue@webrtc.org
8f8503d947 Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling. 
TEST=passed_all_trybots
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6458 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 08:02:05 +00:00
solenberg@webrtc.org
a28c697d93 - Get rid of 'using' from .h 
- Add parenthesis to make order of evaluation clearer.

BUG=
R=minyue@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6304 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 15:22:33 +00:00
fischman@webrtc.org
2c89b5cb27 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. 
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done

(and then removed the talk/ impact)

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 20:08:03 +00:00
minyue@webrtc.org
46509c8d58 adding FEC support to WebRTC Opus wrapper and tests. 
BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-07 11:49:11 +00:00
minyue@webrtc.org
04546884bf This CL is to add Opus complexity knob and to test it. 
As a by-product, a generic tool for testing and comparing the complexity of codecs is added, and new audio files are added to the resources.

Three complexity tests are included
1. Default Opus complexity
2. Opus complexity knob
3. Default iSAC complexity (to compare with Opus)

The complexity tests are only meant for development reasons
and not to be run at bots.

The .isolate file is only needed for the APK packaging and test execution on Android.

TEST=passes all trybots

BUG=
R=kjellander@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5655 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-07 08:55:48 +00:00
pbos@webrtc.org
57eb858698 Remove ".." from include_dirs in build/common. 
BUG=1662
TEST=compile on trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2332004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-11 10:20:27 +00:00
minyue@webrtc.org
ecbe0aa543 Added Opus stereo support 
TESTED=git try
BUG=webrtc:1360
R=tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1868004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4521 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 06:48:09 +00:00
tina.legrand@webrtc.org
bd21fb5f8d Adding call to Opus PLC 
NetEq will call the PLC function in Opus only to set the decoder state. The actual PLC data will not be used.

BUG=https://code.google.com/p/webrtc/issues/detail?id=1181
R=tterribe@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1727004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4504 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-08 11:01:07 +00:00
tina.legrand@webrtc.org
45426eadf5 In call to Opus decoder: frame length too large 
BUG=https://code.google.com/p/webrtc/issues/detail?id=1201
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1752004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4292 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 13:32:04 +00:00
pbos@webrtc.org
3004c79c6a Fix clang errors in non-GYP_DEFINES=clang=1 build 
BUG=1623
R=stefan@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1368004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 12:36:21 +00:00
tina.legrand@webrtc.org
db11fab49e Adding Opus unit test 
This CL adds a unit test for Opus, as well as new APIs for true stereo decoding (skipping master/slave approach).

BUG=

Review URL: https://webrtc-codereview.appspot.com/1222006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3860 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 10:39:41 +00:00
tina.legrand@webrtc.org
46d90dcd74 Adding three frame sizes to Opus 
Adding support for 10, 40 and 60 ms packet sizes for Opus.

BUG=issue1015

Review URL: https://webrtc-codereview.appspot.com/1086004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3454 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-01 14:20:06 +00:00
henrik.lundin@webrtc.org
5dfb1f2cd3 Bug fix in WebRtcOpus_DurationEst 
The function WebRtcOpus_DurationEst returned the number of samples
per packet in the native 48 kHz sample rate, while the decoder
function returns data in 32 kHz sample rate. This creates a discrepancy
that makes NetEQ's lip-sync functionality add too little delay.

BUG=1334
TEST=try bots

Review URL: https://webrtc-codereview.appspot.com/1069006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3403 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-23 11:57:03 +00:00
tina.legrand@webrtc.org
d0d41498a3 Adding AUDIO application as default for Opus stereo 
The Opus audio codec targets applications for pure conversations as well as other types of audio (e.g. music), and there are two different settings to use for this (VoIP and AUDIO). In the current implementation of Opus in WebRTC we use VoIP only.

I this CL I have changed default setting to AUDIO in the case of stereo, and kept VoIP as default in case of mono.

Next step is to add an API to choose application mode.

BUG=issue1239

Review URL: https://webrtc-codereview.appspot.com/1007006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3319 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-20 09:23:10 +00:00
leozwang@webrtc.org
ac77084583 Roll opus to 172355 and delete opus_demo from webrtc opus 
opus_demo has been inlucded in opus in chromium.

BUG=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/973013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 17:24:30 +00:00
tina.legrand@webrtc.org
4275ab1ca0 Implement NetEq duration estimation for Opus. 
Review URL: https://webrtc-codereview.appspot.com/983004
Patch from Ralph Giles <giles@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3314 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 09:52:45 +00:00
tina.legrand@webrtc.org
c4590580e8 Opus mono/stereo on the same payloadtype, and fix of memory bug 
During call setup Opus should always be signaled as a 48000 Hz stereo codec, not depending on what we plan to send, or how we plan to decode received packets.
The previous implementation had different payload types for mono and stereo, which breaks the proposed standard.

While working on this CL I ran in to the problem reported earlier, that we could get a crash related to deleting decoder memory. This should now be solved in Patch Set 3.

BUG=issue1013, issue1112

Review URL: https://webrtc-codereview.appspot.com/933022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3177 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-28 12:23:29 +00:00
andrew@webrtc.org
4b97793f91 Ensure opus_demo has a targets block. 
TBR=leozwang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/933023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3147 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-21 20:16:53 +00:00
leozwang@webrtc.org
cfcadab8d4 Build opus_demo 
BUG=1082
TEST=trybots

With this cl we can build opus_demo.
Review URL: https://webrtc-codereview.appspot.com/936029

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3144 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-21 18:13:46 +00:00
tina.legrand@webrtc.org
0ad3c1af0a Adding Opus stereo support to WebRTC 
This CL adds support for sending and receiving stereo using the Opus codec.

BUG=issue1013

Review URL: https://webrtc-codereview.appspot.com/930008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 08:07:29 +00:00
andrew@webrtc.org
a3ae6dc846 Remove opus include path from opus.gypi. 
It's now exported by opus.gyp.

Review URL: https://webrtc-codereview.appspot.com/928007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2997 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-25 17:51:09 +00:00
tina.legrand@webrtc.org
1f8c02a4de Getting Opus from Chrome 
This CL replaces http://review.webrtc.org/921006/

BUG=issue982

TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/933006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2993 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-25 12:37:08 +00:00
andrew@webrtc.org
14b43beb7c Move src/ -> webrtc/ 
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-22 18:19:23 +00:00