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120 Commits

Author SHA1 Message Date
hclam@chromium.org
1a7b9b94be Cleanup WebRTC tracing 
The goal of this change is to:
1. Remove unused tracing events.
2. Organize tracing events to facilitate measurement of end to end latency.

The major change in this CL is to use ASYNC_STEP such that operation
flow can be traced for the same frame.

R=marpan@webrtc.org, pwestin@webrtc.org, turaj@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1761004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4308 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-08 21:31:18 +00:00
henrike@webrtc.org
a950300b0e Disables unit tests that don't work on Android for Android. 
BUG=N/A
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1747004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4306 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-08 18:53:54 +00:00
pbos@webrtc.org
d900e8bea8 Proper spacing for end-of-namespace comments. 
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 15:12:26 +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
tina.legrand@webrtc.org
f6f033f8bd Possible divide by 0 in ACM. 
BUG=https://code.google.com/p/webrtc/issues/detail?id=1551
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1757004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4291 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 12:00:14 +00:00
tina.legrand@webrtc.org
b1698ab827 Error in update of read index in ACM 
Fixing a bug where we increase read index with too few samples when the input is stereo.

BUG=https://code.google.com/p/webrtc/issues/detail?id=714
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1753004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 09:25:34 +00:00
kjellander@webrtc.org
63e988856e Merge more tests into modules_{unit,integration}tests. 
A new test target named 'modules_integrationtests' is created
and the following test targets were merged into it:
* audio_coding_module_test
* test_fec
* video_coding_integrationtests
* vp8_integrationtests

A couple of other targets were merged into modules_unittests:
* audio_coding_unittests
* audioproc_unittest
* common_unittests
* video_coding_unittests
* video_processing_unittests
* vp8_unittests

I wasn't able to merge audio_decoder_unittests and neteq_unittests due to
conflicts with different defines in these tests.

Some tests that have special requirements aren't merged into
modules_integrationtests yet. I took the opportunity to rename them
since the bot configs will need to be update anyway:
* audio_device_test_api -> audio_device_integrationtests
* video_capture_module_test -> video_capture_integrationtests
* video_render_module_test -> video_render_integrationtests

Exclude files were added for modules_integrationtests to make sure
the memcheck and tsan bots doesn't tests that are too slow
(audio_coding_module_test and vp8_integrationtests were previously
disabled on those bots).

Suppressions for AudioCodingModuleTest needed to be added to get
modules_integrationtests to pass memcheck (even if the test is
excluded from execution).

BUG=1843
TEST=local execution on Linux and trybots (passing except the merged tests of course)
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1656004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4228 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-14 20:09:44 +00:00
turaj@webrtc.org
fee739c224 Risk of division by zero. 
bug=b9338699

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1634004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4223 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 20:10:06 +00:00
tina.legrand@webrtc.org
b097670264 G722_1/G722_1C codecs won't instantiate 
BUG=issue1890
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1650004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4215 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 07:41:42 +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
3942f3a985 Issue 1847, memcopy is wrong and unnecessary, it is sufficient to store the pointer before clearing the instance, and write back the pointer. 
bug=issue1847

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1585006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4178 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 21:31:22 +00:00
turaj@webrtc.org
9238de9d49 resolve b9050210. Avoid pushing sync packet before any packet received. Do not turn on AV-sync if initial delay is zero. 
Also solve DTMF playout with Opus. 

issue=b9050210
Test=Manual by QA Team.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1583004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4176 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 19:18:39 +00:00
kjellander@webrtc.org
5156c94f89 Disable neteq_unittests on Win x64 in code. 
Having this failing test being disabled in code will make it
possible to add it on the bots again, and make thus no bot
configuration update needs to be communicated when it's fixed.

BUG=1460
TEST=Compiled with GYP_DEFINES=target_arch=x64 and ran the
test successsfully on Windows. Also ran regular trybots.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1595004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4165 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 05:47:24 +00:00
kjellander@webrtc.org
b6e49aa3f2 Disable audio_decoder_unittests on Win x64 in code. 
Having this failing test being disabled in code will make it
possible to add it on the bots again, and make thus no bot
configuration update needs to be communicated when it's fixed.

BUG=1459
TEST=Compiled with GYP_DEFINES=target_arch=x64 and ran the
test successsfully on Windows. Also ran regular trybots.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1594004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 05:47:04 +00:00
kjellander@webrtc.org
6eba2774c9 Disable audio_coding_unittests on Win x64 in code. 
Having this failing test being disabled in code will make it
possible to add it on the bots again, and make thus no bot
configuration update needs to be communicated when it's fixed.

BUG=1458
TEST=Compiled with GYP_DEFINES=target_arch=x64 and ran the
test successsfully on Windows. Also ran regular trybots.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1593004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4163 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 05:46:37 +00:00
andrew@webrtc.org
c1eb560a5c Replace the old resampler with SincResampler in the voice engine signal path. 
* The old resampler was found to have a wraparound bug.
* Remove support for the old resampler from PushResampler.
* Use PushResampler in AudioCodingModule.
* The old resampler must still be removed from the file utility.

BUG=webrtc:1867,webrtc:827
TESTED=unit tests, Chrome using apprtc and voe_cmd_test to verify wrap-around is corrected, voe_cmd_test running through all supported codec sample rates and channels to verify good quality audio
R=henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1590004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4156 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-03 19:00:29 +00:00
andrew@webrtc.org
31c5f1c91a Remove ancient and unused CNG test. 
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1585005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4154 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-03 16:07:07 +00:00
pbos@webrtc.org
9aca5b34e1 Remove #pragma once 
BUG=1830
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1568004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4130 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 13:19:09 +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
turaj@webrtc.org
8630cfe016 Guarding certain operations, e.g. bandwidth estimation, RTCP statistics update etc., not to be run on sync RTPS. 
BUG=issue1770
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1485004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 23:54:54 +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
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
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
sergeyu@chromium.org
bd4a2feddb Fix off-by-one buffer overflow in WebRtcNetEQ_PacketBufferInsert(). 
BUG=1725
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1395004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3953 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-03 18:11:36 +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
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
e4b6064f8e Replace legacy G_CONST with const. 
BUG=1608

Review URL: https://webrtc-codereview.appspot.com/1310005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 18:06:57 +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
kma@webrtc.org
cf8e108158 Fixed a bug in isac-fix's entropy coding function: out of bounds acces to array. 
BUG=227286
Review URL: https://webrtc-codereview.appspot.com/1293005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3781 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 16:37:53 +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
turaj@webrtc.org
4b1cd5c5c0 G722-stereo has been missing when creating AudioDecoder. 
Review URL: https://webrtc-codereview.appspot.com/1266004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 20:42:48 +00:00
turaj@webrtc.org
4d06db557a NetEq4 fails if the first packets inserted in are out-of-band DTMFs. 
I had to take few steps to solve this issue. I have comments on places I made cahanges to clarify why I did the change.

   
Review URL: https://webrtc-codereview.appspot.com/1195004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 18:31:42 +00:00
pbos@webrtc.org
ae4e2b352b WebRtc_Word -> stdint in audio_coding/g711/ 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1223004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3699 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-21 13:38:29 +00:00
pbos@webrtc.org
01b507a406 WebRtc_Word -> stdint in audio_coding/cng/ 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1222004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3697 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-21 11:28:42 +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
kma@webrtc.org
d6cd64ac6a Change intrinsic code in isac fix to let it pass chrome clang compiler. 
Compiler complains about variables not initialized in instructions veor_s32() and vset_lane_s32().
Review URL: https://webrtc-codereview.appspot.com/1187006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3660 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-12 17:45:41 +00:00
kma@webrtc.org
e5a81ed793 Fixed issue 1497 in iSAC fixed point. 
Bit exact.
Review URL: https://webrtc-codereview.appspot.com/1177005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3653 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-12 00:23:21 +00:00