admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
6,752 Commits 1 Branch 0 Tags
Commit Graph

2815 Commits

Author SHA1 Message Date
bjornv@webrtc.org
78ea06dd34 audio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
Removed usage of trivial macro.

BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7480 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 07:17:24 +00:00
henrik.lundin@webrtc.org
913f7b8d5e Fix for glitches in ACM when switching desired output sample rate 
The problem was that if the output sample rate is changed such from one
where no resampling is needed to a rate that requires resampling, the
first output from the resampler will contain an onset period. The
solution provided in this CL is to keep a copy of the last output frame
in ACM, and if the resampler is engaged, it will be primed with this
old frame before resampling the current frame.

BUG=3919
R=bjornv@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7479 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 06:54:23 +00:00
bjornv@webrtc.org
b69ea9a35a common_audio: Replaced invalid operand in min_max_operations_neon.S" 
Vector Move immediate can not load #0x7FFF. Changed to us vdup from already loaded register.

BUG=N/A
TESTED=ios and android trybots
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7477 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 14:08:35 +00:00
pbos@webrtc.org
b35b136480 Make avg_{psnr,ssim}_threshold_ const. 
Triggered warning on next clang version being rolled as these variables
are annotated to be protected by crit_.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/24949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7475 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 09:14:38 +00:00
bjornv@webrtc.org
2abebe7baf audio_coding/codecs/isac/main: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7474 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 08:26:41 +00:00
bjornv@webrtc.org
a5ce7bbe17 audio_coding/neteq: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7473 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 08:24:54 +00:00
henrike@webrtc.org
28100cb388 Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." 
BUG=N/A
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 22:03:39 +00:00
henrike@webrtc.org
b1dac33cac Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..." 
BUG=3932
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/27779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7470 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 18:54:46 +00:00
houssainy@google.com
0371a37f85 Moving creating TURN configration to the host machine instead of the bots - rtcBot 
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7468 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 16:43:50 +00:00
glaznev@webrtc.org
f7030d4ed7 Query Android device orientation on every camera frame received. 
Remove orientation listener from Android camera, since device
orientation change events are not well synchronized with actual
device display orientation. Plus these event may not be delivered
at all if device is in stationary position causing initial camera
frames appear rotated.

BUG=
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7467 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 16:25:06 +00:00
houssainy@google.com
c221db6165 Test names changed from e.g) testOneWayVideo/chrome=>chrome to testOneWayVideo/chrome-chrome. 
Because the symbol ">"  is interpreted as special command for output to file in bash commands.

TBR= andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7465 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 09:13:43 +00:00
henrik.lundin@webrtc.org
264e66f7a5 Add encoded_timestamp to AudioEncoder base class 
BUG=3926
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7464 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-16 21:16:07 +00:00
henrik.lundin@webrtc.org
9ea6f8a84d New interface class AudioEncoder 
This class will be the base for new C++ wrapper classes for all
encoders.

BUG=3926
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7463 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-16 11:26:24 +00:00
andresp@webrtc.org
458c2c3b06 Improve rtcbot to load all test files at start and allow them to registerTests 
via: registerBotTest. After loading all tests main.js starts running the
requested one on the command arguments.

R=houssainy@google.com

Review URL: https://webrtc-codereview.appspot.com/29779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7461 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-16 07:36:37 +00:00
asapersson@webrtc.org
9aed002090 Add ability to include a larger time span (in addition to encode time) for measuring the processing time of a frame. 
Controlled by setting enable_extended_processing_usage. Enabled by default.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7460 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-16 06:57:12 +00:00
henrike@webrtc.org
d1ba6d9cbf Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. 
BUG=3379
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27709005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 17:30:28 +00:00
houssainy@google.com
3e2f8ff36c Selecting bot_type changed to be specified in the test file 
Selecting bot_type changed to be specified in the test file instead of
specify it in the running command.

Now we can write test for rtcBot that run one bot on chrome for android
and the other bot on chrome for desktop.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7458 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 15:01:11 +00:00
pbos@webrtc.org
e93cbd13d5 Fix data races in ThreadTest.ThreeThreadsInvoke. 
R=henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/26819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7457 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 14:54:56 +00:00
bjornv@webrtc.org
f87c0aff7f audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
Also includes a typo in a comment.
Affects
* aecm
* hpf

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7456 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 12:51:23 +00:00
bjornv@webrtc.org
f02ba9be54 audio_processing/agc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
Affects AGC only.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 11:16:48 +00:00
bjornv@webrtc.org
8dc00d76af audio_processing/ns: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
Affects fixed point version of Noise Suppression.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7454 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 09:31:40 +00:00
henrik.lundin@webrtc.org
99e561f6a6 Extend AcmSwitchingOutputFrequencyOldApi with more frequencies 
Also reducing test duration, since the issue is triggered anyway.
The tests that are not failing are now enabled.

BUG=3919
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7453 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 08:50:00 +00:00
bjornv@webrtc.org
fab5439112 common_audio: Removed version API from signal_processing 
The Signal Processing version API is not used anymore.

BUG=3353
R=kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7451 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 04:38:42 +00:00
pbos@webrtc.org
a73a678e25 Remove -1 from Call::Config::start_bitrate_bps. 
Instead initialize it to a good default value. The code does the same,
but we don't have to check explicitly for -1.

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/23989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7445 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 11:52:10 +00:00
stefan@webrtc.org
eb24b04f16 Add periodic logging of received RTP headers and estimated clock offsets for e2e delay. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7444 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 11:40:13 +00:00
henrik.lundin@webrtc.org
81a78930ee New ACM test to trigger audio glitch when switching output sample rate 
This CL implements a new unit test. The test is designed to trigger
a problem in ACM where switching the desired output frequency creates
a short discontinuity in the output audio. The problem itself is not
solved in this CL, but the failing test is disabled for now.

BUG=3919
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7443 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 10:49:58 +00:00
stefan@webrtc.org
c216b9aeaf Add a packet loss full stack test to the new API. 
Remove all full stack tests for the old API.

BUG=3750
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7442 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 10:38:49 +00:00
kwiberg@webrtc.org
a57678a70e Workarounds for a bug in VS2013.3 linker when PGO is turned on. 
See crbug.com/421607 for more details about this. This CL solve a linker bug when the PGO is turned on, without changing the behaviour or the performances.

BUG=crbug.com/421607
R=kwiberg@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26789005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7441 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 09:40:04 +00:00
aluebs@webrtc.org
b6af4283ca Adjust speech probability in NS when echo 
The average speech probability for the higher band is multiplied by the quotient of the process and analyze powers, to avoid thinking that suppressed echo is speech. In order to do this both magnitudes, alanyze and process, needed to be stored. This also was used to calculate different previous STSA estimates for analyze and process.
This CL was tested on two long team member recordings (bjornv and kwiberg) and the noisiest (5) recordings from the QA set.

BUG=webrtc:3763
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7437 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 20:48:05 +00:00
bjornv@webrtc.org
bc1a4578e0 common_audio: Removed macro WEBRTC_SPL_RSHIFT_W16 
Replaced the trivial right shift macro at remaining 4 places and removed from signal_processing.

Affected components:
* vad
* aecm

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7434 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 14:00:43 +00:00
kwiberg@webrtc.org
a3722b643d iSAC tests: Type buffers as uint8_t[] to avoid casts 
The iSAC interface functions now expect uint8_t arrays, so change some
arrays to be of that type instead of casting at each point of use.

R=bjornv@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7433 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 13:29:04 +00:00
bjornv@webrtc.org
d4fe824862 audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >> 
The implementation of WEBRTC_SPL_RSHIFT_W16 is simply >>. This CL removes the macro usage in audio_processing and signal_processing.

Affected components:
* aecm
* agc
* nsx

Indirectly affecting (through signal_processing changes)
* codecs/cng
* codecs/isac/fix
* codecs/isac/main

BUG=3348,3353
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28699005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7432 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 13:01:13 +00:00
kwiberg@webrtc.org
396a5e0001 WebRtcIsac_Decode et al.: Type encoded data as uint8[], not uint16[] 
This patch changes WebRtcIsac_Decode, WebRtcIsac_DecodeRcu, and
WebRtcIsacfix_Decode so that they read the encoded data from a uint8
array instead of a uint16 array.

BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7431 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 11:23:24 +00:00
kwiberg@webrtc.org
3f7f899a15 WebRtcIsac_UpdateBwEstimate et al.: Type byte streams as uint8, not uint16 
This patch changes the signature of WebRtcIsac_UpdateBwEstimate,
WebRtcIsacfix_UpdateBwEstimate, and WebRtcIsacfix_UpdateBwEstimate1 so
that they expect the encoded data to be uint8 arrays, not uint16,
which is more natural. The implementations of the functions are left
unchanged for now.

BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7430 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 11:07:06 +00:00
kwiberg@webrtc.org
1172988c79 Some WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[] 
The affected functions are

  WebRtcIsacfix_ReadFrameLen
  WebRtcIsacfix_GetNewBitStream
  WebRtcIsacfix_ReadBwIndex

and

  WebRtcIsac_ReadFrameLen
  WebRtcIsac_GetNewBitStream
  WebRtcIsac_ReadBwIndex
  WebRtcIsac_GetRedPayload

BUG=909
R=aluebs@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7429 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 10:53:42 +00:00
braveyao@webrtc.org
c502df54f8 Merge the supporting to UYVY on Linux video capture in crbug/410202 to webrtc standalone. 
BUG=3765
TEST=Manual
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7427 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 02:13:00 +00:00
braveyao@webrtc.org
651c05e4fc Release _inputSendPin & _outputCapturePin before _captureFilter & _sinkFilter since they should depend on the filters. 
The previous steps work fine for all the webcam, but have problem on SplitCam driver as in the issue report.
Anyway it's always good to de-initial with the reversing order to initialization.

BUG=3845
TEST=Manual
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7426 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 02:11:55 +00:00
henrike@webrtc.org
7f7b0a1cdd Re-enable ThreadCheckerDeathTest.MethodNotAllowedOnDifferentThreadInDebug (missed when enabling other base tests). 
BUG=3836
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/24909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7425 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-10 21:41:55 +00:00
marpan@webrtc.org
4ddbbed16e Disable SendsAndReceivesVP9 test for now. 
Fails on linux memcheck and DrMemory.
Will re-enable on next libvpx roll.

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7424 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-10 21:25:20 +00:00
marpan@webrtc.org
c87b74717b Adjust/increase rate control thresold for a vp9 test. 
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7423 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-10 17:55:57 +00:00
marpan@webrtc.org
573c78e31c Add VP9 codec to VCM and vie_auto_test. 
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.
Passes trybots.

R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-10 16:44:47 +00:00
xians@webrtc.org
3cefbc99f4 Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. 
This also marks all virtual overrides of other classes in the same files. 

This will make a subsequent change I intend to do safer, where I'll change the 
argument types of the base Transport functions, by breaking the compile if I 
miss any overrides. 

This also highlighted a number of unused functions. I've removed some of these. 

TBR=mflodman@webrtc.org, pkasting@chromium.org
BUG=none 
TEST=none

Review URL: https://webrtc-codereview.appspot.com/28709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-10 09:42:53 +00:00
xians@webrtc.org
e46bc77e94 Reland 28629004: adding new AEC dump start interface for chrome. 
This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
http://msdn.microsoft.com/en-us/library/ms235460.aspx

R=andresp@webrtc.org, andrew@webrtc.org, bjornv@webrtc.org, henrike@webrtc.org, henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7418 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-10 08:36:56 +00:00
henrike@webrtc.org
c732a3e511 Re-enable allmost all base tests. 
BUG=3836
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/22989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7416 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 22:08:15 +00:00
henrike@webrtc.org
4a73519690 Re-enables a bunch of base unittests part II. 
BUG=3836
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/30709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7415 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 20:27:13 +00:00
henrike@webrtc.org
e30dab77df base/thread_unittest: wrap test was setting current thread to NULL. 
This broke unittests following ThreadTest.Wrap

BUG=3836
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7413 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 15:41:40 +00:00
henrike@webrtc.org
17f8ddd6c4 Make pbos and kjellander only owners of tsan2 suppressions. 
R=pbos@webrtc.org
TBR=kjellander@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/29709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7412 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 15:40:18 +00:00
henrik.lundin@webrtc.org
8768f161cd Fix comments in common_types.h 
Two of the metrics in NetworkStatistics were desribed as being in
percent, while they are in fact fractions between 0 and 1, scaled
to Q14 domain.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7411 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 12:58:45 +00:00
pbos@webrtc.org
3ff788cf73 Increase timeout for AsyncWriteTest.TestWrite. 
Having a 10ms timeout for something meant to run on DrMemory is insane.

TBR=henrike@webrtc.org
BUG=3490

Review URL: https://webrtc-codereview.appspot.com/23959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 12:47:15 +00:00
kwiberg@webrtc.org
4bd2db9a55 Opus wrapper: Use const for inputs and uint8[] for byte streams 
About half of the functions already followed the desired pattern; this
patch fixes the other half.

BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7409 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 11:21:10 +00:00