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45 Commits

Author SHA1 Message Date
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
turaj@webrtc.org
4ce838934c Address sanitizer out of bounds read in iSAC 
BUG=issue1770
TBR=tlegrand@google.com

Review URL: https://webrtc-codereview.appspot.com/1472006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4030 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 17:42:22 +00:00
pwestin@webrtc.org
03efc89151 Fix when SetMinimumPlayoutDelay is configured to 0 
BUG=1720
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1386005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3942 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 21:19:12 +00:00
andrew@webrtc.org
342353780d Consolidate common_audio into a single target. 
In principle should reduce gyp processing time, but the difference was not measurable. In any case, it's a good simplification that aligns with having a single common_video target.

R=bjornv@webrtc.org, kma@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1375004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3928 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 23:43:26 +00:00
turaj@webrtc.org
a942692725 Buf fix for r3883. 
Review URL: https://webrtc-codereview.appspot.com/1319012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3889 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 16:08:29 +00:00
turaj@webrtc.org
28d54ab18f Improve AV-sync when initial delay is set and NetEq has long buffer. 
Review URL: https://webrtc-codereview.appspot.com/1324006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3883 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-22 18:53:35 +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
turaj@webrtc.org
f1a3b4bc0c Issue 1647. Avoid unsequenced modification. 
issue=1647
test=trybots,manual

Review URL: https://webrtc-codereview.appspot.com/1327004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3858 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 17:01:35 +00:00
pbos@webrtc.org
6e788df19e Remove vim/emacs modelines from .gypi files 
BUG=1655

Review URL: https://webrtc-codereview.appspot.com/1326005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3857 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 12:40:34 +00:00
turaj@webrtc.org
92d1f07551 Elevate NetEq short-term activity statistics to ACM level for logging. 
Review URL: https://webrtc-codereview.appspot.com/1313004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3850 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-15 16:52:04 +00:00
kjellander@webrtc.org
4b8de90dce Disable -Wunsequenced warning in audio_coding_module 
BUG=1647
TEST=Compile locally on Linux with clang enabled.
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1316005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-15 06:38:56 +00:00
pbos@webrtc.org
ab9202b673 Removing remaining WebRtc_Word32 not in typedefs.h 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1306006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 17:59:17 +00:00
hclam@chromium.org
806dc3b0e6 More trace events 
The goal of this change is to unify tracing events styles
and add trace events for all RTP traffic.

BUG=1555
Review URL: https://webrtc-codereview.appspot.com/1290007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 19:54:10 +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
turaj@webrtc.org
2e6b7e938f In streaming mode it is preferable to fade to silence when sender stops sending, or long period of packet loss. 
test=try bots.
Review URL: https://webrtc-codereview.appspot.com/1272004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3771 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-06 00:08:11 +00:00
edjee@google.com
79b0289bfc Adds event traces and counters for WebRTC receive side. 
Review URL: https://webrtc-codereview.appspot.com/1279005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3766 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 19:43:34 +00:00
henrika@webrtc.org
bb8ada686e TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC 
BUG=226044
TEST=content_unittests in Chrome with TSan v2 enabled

Review URL: https://webrtc-codereview.appspot.com/1201010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3760 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 08:39:09 +00:00
justinlin@chromium.org
f81fad6267 Fix opus bitrate truncated to 16-bit int. This prevented setting bitrates higher 
than 2^16kbps.
Review URL: https://webrtc-codereview.appspot.com/1275004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3748 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-01 22:25:11 +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
kjellander@webrtc.org
fa53d8717c Fixing/disabling Windows x64 warnings 
Disabled MSVC #4267 warnings in common.gypi to enable x64 builds
for Windows.
Fixed MSVC #4267 warnings in test/testsupport.
Added third_party/directxsdk to .gitignore.

With http://review.webrtc.org/1070008 landed, this should make it possible
to build for x64 on Windows.

BUG=1348
TEST=Compiling with http://review.webrtc.org/1070008 applied:
set GYP_DEFINES="target_arch=x64"
set GYP_GENERATORS=ninja
gclient sync
ninja -C out\Debug_x64

Review URL: https://webrtc-codereview.appspot.com/1060008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3464 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-04 10:07:17 +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
63464a9354 Enabling unit tests for NetEq4 in the bots 
The unit tests for NetEq4 are made a part of audio_coding_unittests.

The bit-exactness tests are disabled due to problems in iLBC. See
https://code.google.com/p/webrtc/issues/detail?id=281.

A few smaller fixes for valgrind errors and bot failures are included.
Some of the fixes are adpted from
http://webrtc-codereview.appspot.com/1072008/.

Review URL: https://webrtc-codereview.appspot.com/1063012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3432 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-30 09:41:56 +00:00
niklas.enbom@webrtc.org
cd2f1356ee Revert 3405 
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1074004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3407 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-24 22:05:30 +00:00
niklas.enbom@webrtc.org
05e7bfeeea Mainly hlundin's patch. 
Review URL: https://webrtc-codereview.appspot.com/1052004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3405 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-24 18:53:43 +00:00
wjia@webrtc.org
a3c82bf667 Remove '<(library)' in gyp files. 
This will remove all usage of '<(library)' in all webrtc gyp files. 
Review URL: https://webrtc-codereview.appspot.com/1049005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3392 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-18 23:42:21 +00:00
roosa@google.com
b8ba4d8109 Add number of inserted samples to NetEq statistics. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/964030

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3289 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 00:06:18 +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
kma@webrtc.org
fa5b6bf4f4 Optimized WebRtcIsacfix_Spec2Time() for iSAC-Fix in ARM Neon processor. Speed doubled. 
Review URL: https://webrtc-codereview.appspot.com/930033

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3274 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 23:00:52 +00:00
roosa@google.com
b718619f0a Expose NetEq playout mode off through VoiceEngine. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/971016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3272 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 21:59:14 +00:00
turaj@webrtc.org
36965b1803 Bug fix for iSAC fixed-point. The bug was the result of changes in iSAC floating-point to add 48 kHz extension. 
TBR=tlegrand@google.com

TEST=voe_cmd_test, ACM unittest.
Review URL: https://webrtc-codereview.appspot.com/974011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3256 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-10 23:52:43 +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
turaj@webrtc.org
277ec8e3f5 Fix a bug when iSAC-48kHz was added. 
I discovered this by running extended VoE test on "Codecs."

TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/973010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3229 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 21:16:23 +00:00
turaj@webrtc.org
b0dff12d2b 48 kHz extension to iSAC. 
Test:
-manual test with voe_cmd_test.
-manual test with RTPEncode & NetEqRTPPlay.
-manual test with simpleKenny.
-Bit-exact test of iSAC-swb and iSAC-wb with head revision of trunk. The bit-exactness is confirmed on all files generated by running webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACLongtest.txt
Review URL: https://webrtc-codereview.appspot.com/937025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 17:43:52 +00:00
tina.legrand@webrtc.org
5b4fe494e7 Changing default bitrate to 64000 bps for Opus. 
Default settings for Opus in WebRtc is stereo, but we had default rate to 32 kbps. This is too low for stereo, where we need to encode using 64 kbps to get the quality we like.

BUG=

Review URL: https://webrtc-codereview.appspot.com/974008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3223 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 12:08:53 +00:00
kjellander@webrtc.org
ad0f3baf90 Removing redundant codec unittest targets. 
The following targets have been merged into audio_coding_unittests:
* cng_unittests
* g711_unittests
* g722_unittests
* isacfix_unittests
* pcm16b_unittests

Some of them were empty and were created with the assumption they were
needed in order to get code coverage (which was actually not needed).

The following test has been removed since it was empty:
* audio_conference_mixer_unittests

BUG=none
TEST=trybots passing (well, except for the unittests that are removed, they're failing until the try server configuration is updated)

Review URL: https://webrtc-codereview.appspot.com/971008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3222 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 10:52:29 +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
tina.legrand@webrtc.org
7577ddf27b Refactoring acm_generic_codec 
First patch: updating comments.

BUG=1024

Review URL: https://webrtc-codereview.appspot.com/936019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-13 14:09:35 +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
tina.legrand@webrtc.org
f7fa6276e2 Reformating files in audio coding module. 
This CL format the ramaining files on the audio coding module. No other changes are done, except for fixing a few long lines and TODOs without owner.

BUG=issue1024

Review URL: https://webrtc-codereview.appspot.com/928012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-05 09:35:51 +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