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Author SHA1 Message Date
henrike@webrtc.org
89c674053e Adds all unittests to android NDK-APK framework. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1872004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4474 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-02 16:53:47 +00:00
turaj@webrtc.org
a305e9612a Nack for audio. 
R=stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1507004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-06 19:00:09 +00:00
turaj@webrtc.org
e46c8d3875 API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. 
TEST=unit-test, manual, trybots.
R=henrik.lundin@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1384005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4087 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-22 20:39:43 +00:00
andresp@webrtc.org
185bae4b6f Replace ExtraCodecOptions with new Config class that supports multiple settings at once. 
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1452004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4017 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 08:02:25 +00:00
tina.legrand@webrtc.org
d5726a1286 Formatting ACM tests 
Pure formatting of all files located in /webrtc/modules/audio_coding/main/test/

Smaller manual modifications done after using Eclipse formatting tool, like wrapping long lines (mostly comments).

BUG=issue1024

Review URL: https://webrtc-codereview.appspot.com/1342004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3946 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-03 07:34:12 +00:00
pbos@webrtc.org
0946a56023 WebRtc_Word32 => int32_t etc. in audio_coding/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1271006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 00:28:06 +00:00
tina.legrand@webrtc.org
73222cff1a Adding Opus frame length test 
BUG=issue1015

Review URL: https://webrtc-codereview.appspot.com/1193005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3672 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-15 13:29:17 +00:00
tina.legrand@webrtc.org
7a7a008031 Changing non-const reference arguments to pointers, ACM 
Part of refactoring of ACM, and recent lint-warnings.
This CL changes non-const references in the ACM API to pointers.

BUG=issue1372

Committed: https://code.google.com/p/webrtc/source/detail?r=3543

Review URL: https://webrtc-codereview.appspot.com/1103012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3555 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-21 10:27:48 +00:00
tina.legrand@webrtc.org
eb7ebf20ed Revert 3543 
> Changing non-const reference arguments to pointers, ACM
> 
> Part of refactoring of ACM, and recent lint-warnings.
> This CL changes non-const references in the ACM API to pointers.
> 
> BUG=issue1372
> 
> Review URL: https://webrtc-codereview.appspot.com/1103012

TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1116004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3544 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 15:57:31 +00:00
tina.legrand@webrtc.org
374aa49e1a Changing non-const reference arguments to pointers, ACM 
Part of refactoring of ACM, and recent lint-warnings.
This CL changes non-const references in the ACM API to pointers.

BUG=issue1372

Review URL: https://webrtc-codereview.appspot.com/1103012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3543 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 15:22:23 +00:00
tina.legrand@webrtc.org
a092cbf9b7 Fixing lint warnings from previous commit 
In this CL I have removed (almost) all lint warnings I got for this commit:
https://code.google.com/p/webrtc/source/detail?r=3454.

The only warning not fixed is a warning about usage of  non-const reference. This will be fixed in a separate CL.

BUG=issue1372

Review URL: https://webrtc-codereview.appspot.com/1091006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3510 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-14 09:28:10 +00:00
turaj@webrtc.org
6388c3e2fd Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM. 
TEST=ACM unit test is added, also a manual integration test is writen. 
Review URL: https://webrtc-codereview.appspot.com/1097009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3506 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-12 21:42:18 +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
turaj@webrtc.org
c454fab03b Reformatting ACM. All changes are bit-exact in this CL. 
TEST=VoE auto-test, audio_coding_module_test; 

only 15 ms of teststereo_out_1.pcm is not bit-exact with output file of the head revision
Review URL: https://webrtc-codereview.appspot.com/937035

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3287 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 22:46:43 +00:00
turaj@webrtc.org
226db898f7 Dual-stream implementation, not including VoE APIs. 
Review URL: https://webrtc-codereview.appspot.com/933015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3230 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 22:13:31 +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
henrika@webrtc.org
5ba3decc94 Ensures that we can build using VS 2012 on Windows. 
See more details at https://code.google.com/p/webrtc/issues/detail?id=1146&

TBR=Niklas
BUG=1146

Review URL: https://webrtc-codereview.appspot.com/939028

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-26 09:12:02 +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
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