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Author SHA1 Message Date
tommi@webrtc.org
a9da4c55ef Landing for thakis. Original review here: 
https://webrtc-codereview.appspot.com/667013/
Review URL: https://webrtc-codereview.appspot.com/701004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2522 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-20 11:17:23 +00:00
leozwang@webrtc.org
8495915442 Make loopback mode work properly 
Some minor changes and improvements are added into this cl

BUG=
TEST=vie_test
Review URL: https://webrtc-codereview.appspot.com/667005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2520 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-16 20:03:18 +00:00
andrew@webrtc.org
d41f59a23f Fix Mac-gcc warnings. 
Resolves:
- warning: allocating zero-element array
- warning: suggest a space before ‘;’ or explicit braces around empty
  body in ‘for’ statement

BUG=none
TEST=build on Mac-gcc, trybots

Review URL: https://webrtc-codereview.appspot.com/675006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2519 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-16 17:05:47 +00:00
turaj@webrtc.org
837bc7b44c ilbc: Make the decode input array const 
Review URL: https://webrtc-codereview.appspot.com/667009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2518 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-14 00:34:54 +00:00
mikhal@webrtc.org
73db8dbfc2 video conversion functions: switching from designated functions to a general one. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/686004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-14 00:03:55 +00:00
leozwang@webrtc.org
7760963d04 Make webrtc compile on android in chromium 
Message:
There probably is a better way, this cl is trying to seperate android
specific calls into android files, particular SetAndroidObject, by doing
this, webrtc can be built inside Chromium on android. Currently, Chromium
manages its own jvm, capturer and renderer, all webrtc code that manages
jvm, captuer and renderer should not be compiled. 

Description:
By re-organize android specific code, this cl will make webrtc build
in Chromium on android.

BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/668007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2516 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-13 22:00:43 +00:00
leozwang@webrtc.org
6c08f26c4e Terminate version string 
This cl doesn't directly solve b/6750185, but it's a potential bug
if string is not terminated correctly

BUG=
TEST=vie_auto_test
Review URL: https://webrtc-codereview.appspot.com/674009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2515 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-13 22:00:16 +00:00
marpan@webrtc.org
71707aaae8 Add the FEC mask type to FecProtectionParams and set the mask type in the VCM. 
Review URL: https://webrtc-codereview.appspot.com/682004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2514 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-13 16:27:51 +00:00
mikhal@webrtc.org
d96dcef422 vpm: Updating module to use CalcBufferSize 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/666008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2513 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-12 23:52:55 +00:00
bjornv@webrtc.org
08329f4a13 Added API to port internal speech probability in NS. 
Identical with CL652007 that's already been accepted for commit.

TBR=andrew@webrtc.org
BUG=None
TEST=None
Review URL: https://webrtc-codereview.appspot.com/670009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2511 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-12 21:00:43 +00:00
mikhal@webrtc.org
6182db10c8 vp8: Updating wrapper to use CalcBufferSize (includes odd size support). 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/685004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2510 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-11 18:43:36 +00:00
mikhal@webrtc.org
538f0ab96f I420: Updating computation of buffer size to use calcBufferSize (odd size support). 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/687004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2509 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-11 18:20:39 +00:00
wu@webrtc.org
262bdedfda Remove files that are not needed from direct_show_base_classes.gyp 
BUG=
TEST=try

Review URL: https://webrtc-codereview.appspot.com/689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2508 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-11 16:52:19 +00:00
wu@webrtc.org
13c09bc845 . 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2506 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-10 23:10:31 +00:00
kma@webrtc.org
ff2f861c71 Corrected one error for Android build. 
Also added iSAC in the default build in Android, to test any build errors in iSAC in platform build in buildbot.
Review URL: https://webrtc-codereview.appspot.com/684004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2505 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-10 21:37:49 +00:00
mikhal@webrtc.org
b95e9ca865 video_coding: Refatoring I420 wrapper. No functional updates. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/673009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2504 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-10 20:58:08 +00:00
mikhal@webrtc.org
0bb817dab0 1. Adding odd size support to LibYuv wrapper. 
2. Removing unused functionality.
3. Adding support for negative height (flips the image).

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/673008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2503 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-10 20:48:48 +00:00
leozwang@webrtc.org
475c26634e Re-enable WEBRTC_SVNREVISION script 
Message:
Another try to enable the script to get svn revision number. Most code borrowed from
lastchange.py, I simplified and modified to make it work with webrtc. The bottom line
of this script is 1. not breake any existing builds 2. get correct svn revision number
in a typical engineering setup, so it doesn't deal with some corner cases that lastchange.py
does, just simply returns "n/a" since these corner cases will most likely not happen, and
it also make this script simple.

Description:
This script runs "svn info" or "git svn info" to get svn revision number returns "n/a" if
both fail.

BUG=
TEST=try bots
Review URL: https://webrtc-codereview.appspot.com/671004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2502 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-10 20:36:29 +00:00
kma@webrtc.org
adf8ddf4aa Assembly coding for pitch filter in iSAC for ARMv6. 
Review URL: https://webrtc-codereview.appspot.com/631004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2501 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-10 19:30:57 +00:00
kma@webrtc.org
e2c16a83bc Optimized a filter bank function in iSAC/fix for ARM. 
Review URL: https://webrtc-codereview.appspot.com/631008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2500 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-10 17:59:44 +00:00
leozwang@webrtc.org
cf9855d9eb Update build.xml and api level 
Description:
This cl updates build.xml following the sdk_r20 release. Also upgrade api
level to 10. API level 9 is obsolete and we don't reply on level 9 particular
features, upgrade to 10 to make development more easier.

BUG=
TEST=local build
Review URL: https://webrtc-codereview.appspot.com/678005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2499 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-10 17:38:48 +00:00
kma@webrtc.org
d2f71003af correct one build error in linux. 
Review URL: https://webrtc-codereview.appspot.com/681005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2498 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-09 23:34:58 +00:00
kma@webrtc.org
72f8a6d77b Optimized PCorr2Q32() in iSAC with intrinsics in ARM Neon platform. 
Review URL: https://webrtc-codereview.appspot.com/634004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2497 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-09 23:27:02 +00:00
xians@webrtc.org
e9eb235bc1 Remove the useless dummy audio device impl which creates threads and high res timers on windows. 
BUG=630
Test=apprtc.appspot.com in chrome
Review URL: https://webrtc-codereview.appspot.com/667010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2494 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-06 08:33:13 +00:00
phoglund@webrtc.org
2eefb2242f Improved fuzzer. It will now throw in additional refreshes, which is known to mess with lifetime assumptions. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/679008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2492 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-04 12:29:09 +00:00
turaj@webrtc.org
01ad75888a ilbc: Mark untouched input arrays as const 
Review URL: https://webrtc-codereview.appspot.com/662004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2490 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-03 21:35:46 +00:00
stefan@webrtc.org
ddfdfed3b5 Pass capture time (wallclock) to the RTP sender to compute transmission offset 
- Change how the transmission offset is calculated, to
  incorporate the time since the frame was captured.
- Break out RtpRtcpClock and move it to system_wrappers.
- Use RtpRtcpClock to set the capture time in ms in the capture module.
  We must use the same clock as in the RTP module to be able to measure
  the time from capture until transmission.
- Enables the RTP header extension for packet transmission time offsets.

BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/666006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-03 13:21:22 +00:00
pwestin@webrtc.org
1853005f37 Change clock to be 64 bits in RTP module 
Review URL: https://webrtc-codereview.appspot.com/678011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2488 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-03 10:41:54 +00:00
tommi@webrtc.org
7b61049117 Land: https://webrtc-codereview.appspot.com/678010/ 
Add -Wno-unused-private-field until all violations are fixed.

This is currently in chromium's build/common.gypi, but I'd like
to remove it from there.
Review URL: https://webrtc-codereview.appspot.com/680006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2485 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-03 08:19:24 +00:00
tommi@webrtc.org
fb933bdb26 Landing: https://webrtc-codereview.appspot.com/680005/ 
Fix more -Wunused-private-field violations.
Review URL: https://webrtc-codereview.appspot.com/668010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2484 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-03 08:19:12 +00:00
vikasmarwaha@webrtc.org
e85c77bd7c Bump WebRTC version to 3.8.1 
Review URL: https://webrtc-codereview.appspot.com/665007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2479 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-02 18:11:06 +00:00
tommi@webrtc.org
cf21b9be87 Fix ChromeOS build by removing an unused variable. 
TBR=niklase
Review URL: https://webrtc-codereview.appspot.com/669008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2477 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-02 14:29:58 +00:00
phoglund@webrtc.org
ef8ca6a801 Wrote ClusterFuzz test for WebRTC GetUserMedia. 
This initial test is very simple since we are just releasing GetUserMedia in the next release.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/639006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2476 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-02 11:39:22 +00:00
vspasova@webrtc.org
b358bd8f87 A command-line tool based on libyuv to convert a set of RGBA files to a YUV video. 
BUG=
TEST=
tgbra_to_i420_converter --frames_dir=<directory_to_rgba_frames> --output_file=<output_yuv_file> --width=<width_of_input_frames> --height=<height_of_input_frames>

<output_yuv_file> should be an empty file because we open it in append mode

Review URL: https://webrtc-codereview.appspot.com/673006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2475 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-02 07:43:30 +00:00
marpan@webrtc.org
c5b392e9d6 Updates t resolution adaptation (cama): 
-set image type when QM is reset.
  -fix for undoing two stages of spatial downsampling.
  -some adjustments and code clean-up.
  -updates to control parameters and unittest.
Review URL: https://webrtc-codereview.appspot.com/641010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2473 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-29 21:44:55 +00:00
leozwang@webrtc.org
ea5b8b5903 Trival changes in gui layout based on feedback 
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/674006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-29 18:31:45 +00:00
leozwang@webrtc.org
fb59442c40 Change file path to make it work on android 
BUG=
TEST=try bots
Review URL: https://webrtc-codereview.appspot.com/672007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2471 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-29 18:28:12 +00:00
turaj@webrtc.org
8d59e70434 In this CL four pitch-filters are integrated into a single function. I have kept the interfaces unchanged so there was no need to modify any other file. A test is uploaded to show how this CL is tested. The test engages all the functions affected by this CL and compares their output with the version of iSAC before this change. This CL is bit-exact. Furthermore, I ran iSAC release test and diff results with previous version. The test file will not be commited, as running it requires a hack in old iSAC to. Hence you don't need to code-review it. 
test = bit-exact with previous version of iSAC verified by iSAC Release test and the test written specifically to test functions affected by this CL.
Review URL: https://webrtc-codereview.appspot.com/611004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2470 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-29 18:17:53 +00:00
mflodman@webrtc.org
e06ca3cef6 Removed nolint for include guards. 
BUG=
TEST=cpplint.py --filter=-build/header_guard src/video_engine

Review URL: https://webrtc-codereview.appspot.com/676008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2469 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-29 13:20:14 +00:00
mflodman@webrtc.org
ab2610ffd9 Removed the last lint warnings in video_engine. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/670006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2468 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-29 10:05:28 +00:00
henrike@webrtc.org
a5fcf7ab41 Fixes broken Chromium build. 
BUG=brakes chrome build
TEST=Manually on Linux

Review URL: https://webrtc-codereview.appspot.com/679006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2462 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-28 12:49:35 +00:00
mflodman@webrtc.org
c802e0ed0c Changed max codec resolution. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/674008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2457 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-28 07:57:39 +00:00
asapersson@webrtc.org
d2e6779565 Fix for negative transmission time offset. 
Review URL: https://webrtc-codereview.appspot.com/671006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2456 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-28 07:53:15 +00:00
stefan@webrtc.org
5f28498149 First step in refactoring audio/video synchronization. Adds unittests. 
BUG=
TEST=stream_synchronization_unittest

Review URL: https://webrtc-codereview.appspot.com/669005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-28 07:51:16 +00:00
mflodman@webrtc.org
cee447a5bb cpplint passes for vie_performance_monitor, vie_manager_base, vie_impl, vie_renderer, vie_defines and vie_render_manager. 
NOLINT is used where API changes would be needed, for include guards and include files in WebRTC root.

Lots of changes, but no real logical changes.

BUG=627
TEST=vie_auto_test + compiles on all platforms.

Review URL: https://webrtc-codereview.appspot.com/679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2454 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-28 07:29:46 +00:00
asapersson@webrtc.org
100463e828 Added initial nack configuration for rtp module. 
Review URL: https://webrtc-codereview.appspot.com/677007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2453 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-28 07:21:51 +00:00
mflodman@webrtc.org
1b1cd78dd2 Made cpplint pass for vie_remb, vie_ref_count, vie_sender and vie_receiver. 
NOLINT is used for include guards. I took a shortcut for vie_ref_count, the class will be deleted very soon anyway.

BUG=627
TEST=cpplint and compiles

Review URL: https://webrtc-codereview.appspot.com/677008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2452 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-28 06:34:08 +00:00
andrew@webrtc.org
e22beabaf1 [MIPS] Adding support for MIPS architecture for WebRTC. 
Small change to typedefs.h to enable MIPS Little Endian port.

TBR=niklas.enbom@webrtc.org
BUG=https://code.google.com/p/chromium/issues/detail?id=130022
TEST=make chrome

Review URL: https://webrtc-codereview.appspot.com/679005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2451 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-27 22:24:43 +00:00
mflodman@webrtc.org
f5e99db10b Made cpplint pass for vie_channel.* and vie_encoder.*. NOLINT is used for API changes, include guards and include files in WebRTC root. 
WebRTC types and webrtc:: will be removed in a follow up.

BUG=627
TEST=vie_auto_test + compiles

Review URL: https://webrtc-codereview.appspot.com/677005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2450 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-27 09:49:37 +00:00
tina.legrand@webrtc.org
3ddc974039 Handle VAD/DTX in a correct way if running stereo ACM. 
BUG=issue573
TEST=audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/669006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2449 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-27 09:25:50 +00:00
andrew@webrtc.org
4ecea3e105 Downmix before resampling in capture and render paths. 
We previously had an error when a mono capture device was used with
a stereo codec. This is prevented by avoiding any remixing in
AudioProcessing. Instead, capture side downmixing is now done before
resampling. Upmixing can now be handled properly by AudioCoding,
since the AudioProcessing error condition has been removed.

On the render side, downmixing now occurs before resampling. Ideally
this would be handled still earlier in the chain. Similarly, downmixing
for the AudioProcessing reference data occurs before resampling. This
code has been refactored into RemixAndResample, with a comprehensive
unittest added in output_mixer_unittest.cc.

BUG=issue624
TEST=manually through voe_cmd_test, by using mono and stereo capture
and render devices with mono and stereo codecs. voice_engine_unittest,
voe_auto_test.

Review URL: https://webrtc-codereview.appspot.com/676004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2448 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-27 03:25:31 +00:00
andrew@webrtc.org
7a281a5634 Fix Android build after test/ -> src/test/ 
TBR=leozwang@webrtc.org
BUG=none
TEST=Android trybot

Review URL: https://webrtc-codereview.appspot.com/677006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2447 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-27 03:22:37 +00:00
andrew@webrtc.org
81cf5e4752 Move test to src/test. 
- Refer to top-level directories by <(DEPTH), e.g. <(DEPTH)/testing.
- Remove now unneeded third_party_root.

TBR=henrike@webrtc.org
BUG=none
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/669007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2446 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-27 01:41:54 +00:00
leozwang@webrtc.org
253912c188 Disable a few features to save CPU cycles on android 
BUG=
TEST=try bots
Review URL: https://webrtc-codereview.appspot.com/677004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2445 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-26 17:08:41 +00:00
marpan@webrtc.org
5567ebfd1f VPM: Assign correct required size for odd size destination frame. 
Updates to spatial resampler unittest.
Review URL: https://webrtc-codereview.appspot.com/660006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2444 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-26 16:47:36 +00:00
astor@webrtc.org
bd7aeba8fb Expose a set of options to the OveruseDetector supporting experiments 
Updated overuse_detector.* to use google style naming convention
Removed OveruseDetector::Reset
Review URL: https://webrtc-codereview.appspot.com/666005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2443 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-26 10:47:04 +00:00
hta@webrtc.org
f494fd0954 Use system-independent sleep in video_capture_unittest. 
Another ifdef bites the dust!

BUG=603
TEST=unittest

Review URL: https://webrtc-codereview.appspot.com/674004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2441 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-25 11:33:34 +00:00
hta@webrtc.org
626dccc85b Use one OS-independent sleep function in a video test 
Sleep using no compile flags

BUG=603
TEST=

Review URL: https://webrtc-codereview.appspot.com/668004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2440 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-25 11:30:33 +00:00
henrike@webrtc.org
643be71700 Adds variable for third party directory. 
BUG=348
TEST=Manual testing in Chrome and WebRTC workspace.

Review URL: https://webrtc-codereview.appspot.com/674005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2439 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-25 10:48:58 +00:00
tnakamura@webrtc.org
b9c1833c2c Add channel info to the Actions->Codec Changes menu in the VoE test app. 
Review URL: https://webrtc-codereview.appspot.com/665005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2438 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-22 16:29:38 +00:00
braveyao@webrtc.org
77e18124f9 Fix the flakiness in FileBeforeStreamingTest 
BUG = 619
TEST = voe_auto_test
Review URL: https://webrtc-codereview.appspot.com/658006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2437 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-22 10:41:11 +00:00
mflodman@webrtc.org
64f86fba19 Fix test app render bug. 
Review URL: https://webrtc-codereview.appspot.com/669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2435 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-21 12:32:39 +00:00
mflodman@webrtc.org
8baed51f6e This CL is part of enabling cpplint check for video_engine uploads. 
BUG=627
TEST=vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/653006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2434 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-21 12:11:50 +00:00
mflodman@webrtc.org
9ba151bdf9 Removed cpplint warnings from all impl-files to be able to add this check as presubmit step. I don't want to change the API right now, will come later, so there are several NOLINT comments added to get around this for now. 
BUG=627
TESTS=vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/661005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2433 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-21 10:02:13 +00:00
hta@webrtc.org
2bd8d62d3b Sleep using no compile flags 
BUG=603
TEST=

Review URL: https://webrtc-codereview.appspot.com/665004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2432 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-21 09:57:24 +00:00
mflodman@webrtc.org
67f98ec63a Removed flaky REMB test. This test is now covered by: 
- RemoteBitrateEstimatorTest
- BitrateControllerTest
- RtcpFormatRembTest
- ViERembTest

BUG=477
TEST=See above.

Review URL: https://webrtc-codereview.appspot.com/667004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2431 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-21 09:29:53 +00:00
kma@webrtc.org
173538faa3 Refactored function WebRtcIsacfix_GetLpcCoef() in iSAC-fix. 
One reason behind it is for further optimization of it in ARM.
Review URL: https://webrtc-codereview.appspot.com/646012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2429 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-20 17:17:15 +00:00
hta@webrtc.org
3168e5349c Working unit test for critical sections. 
This extends unittest coverage, and allows to add more tests if these functions
ever are found to behave strangely.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/632005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2427 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-20 06:45:56 +00:00
kjellander@webrtc.org
5608fe9861 Disabling FileBeforeStreamingTest due to flakiness. 
BUG=619
TBR=xians1
TEST=Tested on Linux, Mac and Windows.

Review URL: https://webrtc-codereview.appspot.com/654006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2426 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-20 06:14:31 +00:00
wu@webrtc.org
2259f855ea Remove unused member variables found by clang's -Wunused-private-field. 
No intended behavior change.

On behavior of thakis@chromium.org.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/641011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2425 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-19 14:56:50 +00:00
hta@webrtc.org
72e3a89b52 Created a wrapper class for condition_variable that lets me write (hopefully) reliable tests for some of its properties. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/600005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2424 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-19 13:49:48 +00:00
bjornv@webrtc.org
b38fca1ec2 VAD Refactoring: API change of return value from int16_t to int. 
This CL change the return int on Process() to meet Google Style. The change affects audio_coding and neteq.

Tests have been changed accordingly and the code has been tested on trybots, vad_unittests, audioproc_unittest, audio_coding_unittests, audio_coding_module_test and neteq_unittests.

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/663005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2423 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-19 11:03:32 +00:00
vspasova@webrtc.org
f477aac844 Removed gflags header from vie_auto_test. 
Removed gflags include file from src/video_engine/test/automated/
vie_video_verification.cc as it is no longer needed.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/645005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-19 09:20:33 +00:00
braveyao@webrtc.org
dfa6b697e2 Refine the error handling made in rev2373 
Review URL: https://webrtc-codereview.appspot.com/644005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-19 06:38:59 +00:00
wu@webrtc.org
67f256fab4 Use 32 as the alignment if possible in VP8 wrapper. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/663004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2420 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-18 21:15:32 +00:00
bjornv@webrtc.org
df596ae444 VAD Refactoring of GMM test section 
The CL is organized w.r.t. patch sets as follows:
1) Comments on functionality added.
2) Renamed local variable n to channel for clarity.
3) Dropped the extension _vector of variable |feature_vector| since it doesn't add anything new.
4) Minor comment update w.r.t. |feature|
5) Replaced an else if scheme with two if statements. This way we can use the same calculation for all sub cases which could be a source of error.
6) Moved two code lines to where they are used and rearranged such that avoiding tmp variable.
7) Instead of performing a bit-wise OR operation within an if statement we could perform the bit-wise OR at once.
8) Name change of |shifts0| to |shifts_h0| for clearer reading. Likewise for H1.
9) Renamed |nr| to |gaussian| for clearer reading.
10) Removed multiplication macro.
11) Re-organized local arrays to have the same structure as constants and member arrays used elsewhere in the code.
12) Changed locally declared variable to function declared.
13) Added array initialization at declaration.

Tested with trybots, vad_unittests, audioproc_unittest

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/595006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2417 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-18 18:22:53 +00:00
tina.legrand@webrtc.org
50d5ca5bf2 Refactoring of TestAllCodecs 
ACM testing consists of seven individual tests. Up til now we haven't used gtest everywhere, and many of the tests needs some rewriting to follow the style guide.

I've started with this tests, doing formatting, adding the test as a separate test which can now either succeed of fail in a proper way.

Still to do in this test is handling of input file, but that will be changed in a separate CL, because all tests uses the  PCMFile class that will be affected by the change.

BUG=none
TEST=audio_coding_module_test, ACM_AUTO_TEST and ACM_TEST_ALL_CODECS.

Review URL: https://webrtc-codereview.appspot.com/646011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2416 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-18 13:35:52 +00:00
hta@webrtc.org
db2f6cf878 Added missing define guard to sleep.h 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/656006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2415 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-18 13:23:48 +00:00
hta@webrtc.org
86a6aacaee Unittest utilities - starting out with an encapsulated trace-to-screen. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/655005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2414 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-18 13:22:08 +00:00
mflodman@webrtc.org
e3a0712f04 Deregister RTP module before deleting it. 
BUG=617
TEST=

Review URL: https://webrtc-codereview.appspot.com/661004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2413 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-18 12:43:41 +00:00
hta@webrtc.org
41adcdbf13 An OS-independent sleep function, and one usage thereof. 
BUG=603
TEST=none

Review URL: https://webrtc-codereview.appspot.com/659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2412 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-18 11:24:57 +00:00
henrika@webrtc.org
37198007ea GetRecPayloadType now logs a warning instead of and error when the user asks for the payload type while no packets have been received. 
BUG=605
TEST=

Review URL: https://webrtc-codereview.appspot.com/660004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2411 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-18 11:00:12 +00:00
stefan@webrtc.org
190541578a Correct gypi files to match the actual filenames. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/656005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-18 10:57:05 +00:00
niklas.enbom@webrtc.org
d63d06a289 bump version to 3.8 
Review URL: https://webrtc-codereview.appspot.com/657004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-18 08:36:36 +00:00
braveyao@webrtc.org
4de777ba2b Refine the error processing of StopRecordingMicrophone. 
BUG = 
TEST = 
Review URL: https://webrtc-codereview.appspot.com/636007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2406 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-15 02:37:53 +00:00
turaj@webrtc.org
bdf7ee5bab This simple change should adress issue 471. 
Previously I uploaded patch 640007 to address issue 471. Today, while discussing that patch with Andrew, we noticed this patch should do the job. Leo is not here to verify it, but Andrew did some test to verify it. I'll ask Leo to do some testing. 

We don't want to abandon patch 640007 as it will save some complexity. 
Review URL: https://webrtc-codereview.appspot.com/648004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2405 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-14 23:46:35 +00:00
marpan@webrtc.org
352d09ab28 Updates to videoprocessor_integration test: 
-added metric for expected key frame size mismatch
   -fix to start bitrate
   -updates to some expected values in tests
Review URL: https://webrtc-codereview.appspot.com/641007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2404 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-14 18:35:00 +00:00
marpan@webrtc.org
f088448c37 Libyuv Scalerunittest: Added PSNR check to some tests in scaler unittest: 
-for downsampling to 1/2x1/2
    -for the odd frame sizes cases
Review URL: https://webrtc-codereview.appspot.com/642009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2403 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-14 17:00:45 +00:00
mflodman@webrtc.org
139c4678c1 Fixed a/v sync issue. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2402 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-14 11:08:51 +00:00
leozwang@webrtc.org
46d83fa26c Use digital mode on mobile 
Use fixed digital mode in test app on android

BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/636010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-12 23:47:20 +00:00
marpan@webrtc.org
c35f1d26c5 FEC: Fix to coverity issue 14448: unintended sign extension. 
Review URL: https://webrtc-codereview.appspot.com/647004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2400 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-12 20:12:13 +00:00
mflodman@webrtc.org
d41851480c Bumped version number to 3.7. 
Review URL: https://webrtc-codereview.appspot.com/642007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2397 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-12 08:31:36 +00:00
bjornv@webrtc.org
b1c3276f5a VAD Refactoring: WebRtcVad_Process() 
Code style: Indentation, braces

Tested with trybot, vad_unittests, audioproc_unittest

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/579012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2396 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-12 08:19:24 +00:00
tina.legrand@webrtc.org
5e7ca608d5 Use new fileutil functions for trace in ACM 
I this CL I have changed to use filutil functions in the ACM tests. I have also reformated the file Tester.cc, and fixe one minor bug in TestAllCodecs.cc.

BUG=issue195
TEST=audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/636006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2394 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-12 07:16:24 +00:00
leozwang@webrtc.org
6724c4239b Add VoiceEngine apm settings to test application 
Implement apm settings and add a small bug fix

BUG=
TEST=build and test on android
Review URL: https://webrtc-codereview.appspot.com/632008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-11 21:23:16 +00:00
andrew@webrtc.org
be581640c1 Add a variable for the libjpeg include directory. 
- Also clean up the use of libjpeg_gyp_path. Both the Chromium and
  standalone builds can use it.

BUG=none
TEST=build with all combinations of use_libjpeg_turbo and build_libjpeg

Review URL: https://webrtc-codereview.appspot.com/640004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2389 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-11 20:38:48 +00:00
bjornv@webrtc.org
eec739f846 VAD Refactoring: Changed pointer structure in WebRtcVad_FindMinimum(). 
For easier code reading, a couple of structural changes together with name changes have been performed in the function WebRtcVad_FindMinimum():
- Removed temporary pointers
- Updated comments
- Pointer name changes
- Changed pointer nomenclature to array index
- Made local variable const

Tested with trybots, vad_unittests, audioproc_unittest

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/594005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2386 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-11 07:57:57 +00:00
tina.legrand@webrtc.org
fa7138f889 Change CriticalSectionScoped to use pointer constructor 
BUG=issue183
TEST=audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/638005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2384 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-08 10:51:28 +00:00
leozwang@webrtc.org
276dc1872a Add libremote_bitrate_estimator to android makefile 
The order of libraries is bit messy, will clean up later.

BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/646007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-07 18:58:12 +00:00
kma@webrtc.org
f85b35a2f4 Refactored Neon code for AECM module, by using pure assembly code. 
Bit exact.
Review URL: https://webrtc-codereview.appspot.com/447008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2382 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-07 16:17:17 +00:00
stefan@webrtc.org
d81ab1397b abs() was used instead of fabsf(), which returns int and not float and therefore truncated the return value. 
Also fixes problems with the remote_bitrate_estimator_unittest.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/641006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2380 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-07 13:48:04 +00:00
tina.legrand@webrtc.org
90af7f841c Changing Celt to run on 20 msec frames 
BUG=none
TEST=-

Review URL: https://webrtc-codereview.appspot.com/641004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2377 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-07 08:57:27 +00:00
stefan@webrtc.org
9354cc965c Refactoring the receive-side bandwidth estimation into its own module. 
Each REMB group has one remote_bitrate_estimator object. For now the
estimator keeps one estimate for every SSRC. In a later commit this will
be unified and one estimate will be used for all SSRC in one group.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/637009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2375 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-07 08:10:14 +00:00
braveyao@webrtc.org
b0bcf13dd4 Trival fix to relative paths of audio files in voe_ui_win_test 
BUG  = 
TEST = voe_ui_win_test
Review URL: https://webrtc-codereview.appspot.com/635005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-07 02:21:44 +00:00
marpan@webrtc.org
5f97232cac Removing a TODO in the FEC: renaming the exisiting packets mask to indicate random mode, 
and refactored and renamed corresponding table file.
Review URL: https://webrtc-codereview.appspot.com/632007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-06 22:34:38 +00:00
wu@webrtc.org
cac603f390 Fix for the alignment problems/mismatch in ViECapture and VP8Encoder. 
BUG=576
TEST=unittest

Review URL: https://webrtc-codereview.appspot.com/637010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2371 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-05 23:52:59 +00:00
marpan@webrtc.org
f4c2de9e2f Added some tests to videoprocessor_integrationtest, for testing: 
-encooder response to changing bit rate and frame rate
   -frame dropper and spatial resize
   -temporal layers
Review URL: https://webrtc-codereview.appspot.com/613006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2370 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-05 21:07:28 +00:00
marpan@webrtc.org
8866bb1132 FEC: Added another set of packet masks for the FEC. 
These FEC codes perform better for bursty (consecutive loss) 
than the existing set (which were designed for random loss). 
Updates to the unittests and test_fec accordingly.
Review URL: https://webrtc-codereview.appspot.com/581005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2369 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-05 16:42:20 +00:00
bjornv@webrtc.org
20e13edede New attempt to revert r2362, since drover failed. 
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/640005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2368 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-05 13:07:56 +00:00
bjornv@webrtc.org
cb89c6f914 Revert 2363 - Refactoring the receive-side bandwidth estimation into its own module. 
Each REMB group has one remote_bitrate_estimator object. For now the
estimator keeps one estimate for every SSRC. In a later commit this will
be unified and one estimate will be used for all SSRC in one group.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/605007

TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/634006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-05 12:25:35 +00:00
stefan@webrtc.org
f72881406f Refactoring the receive-side bandwidth estimation into its own module. 
Each REMB group has one remote_bitrate_estimator object. For now the
estimator keeps one estimate for every SSRC. In a later commit this will
be unified and one estimate will be used for all SSRC in one group.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/605007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-05 10:44:00 +00:00
bjornv@webrtc.org
d2acea6b30 Minor style changes 
Original CL=577007

Tested on trybots.

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/637007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2362 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-05 08:09:23 +00:00
marpan@webrtc.org
da7fdf4af8 Fix to scaler in libyuv for odd size frames. 
Review URL: https://webrtc-codereview.appspot.com/633004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2360 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-04 21:56:13 +00:00
turaj@webrtc.org
ba108aee21 This CL contains some refactoring. Spectrum coding is main place that is affected. Therefore, I have bit-exactness test, test_spectrum_ 
coding.c, to be sure about the changelist. You can go through the test to be sure the changes are tested. However, I don't intend to commi
t the test, as it would be a source of confusion and requires hack to iSAC to be able to run the test. It is basically a one-time test. 

The part which not covered in this test is where we limit payload for super-wideband bit-stream. I'll add a test for that as well. 

I kept format changes at minimum in all files except isac.c, which was in bad shape, and coding changes were minimum. I'm planning to uplo
ad following patches to this CL where I try to address formatting issues. But I don't intend to change variable names, for the moment. 

The refactoring is not yet finished, so you would find part of the code which could be cleaned up, especially KLT transforms in entropy_co
ding.c
Review URL: https://webrtc-codereview.appspot.com/580004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2359 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-04 20:04:58 +00:00
andrew@webrtc.org
2cc55096d5 Fix syntax error in jpeg.gypi. 
TBR=mflodman@webrtc.org
BUG=none
TEST=build on Linux.

Review URL: https://webrtc-codereview.appspot.com/639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2358 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-04 20:01:23 +00:00
mflodman@webrtc.org
ad6083f414 Added condition for which jpeg lib to use. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/638004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-04 19:10:43 +00:00
tina.legrand@webrtc.org
77fd39aa99 ACM PCM16B, fixing a copy-and-paste error. 
Review URL: https://webrtc-codereview.appspot.com/631006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2355 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-04 11:47:49 +00:00
phoglund@webrtc.org
e6f235cfa5 Attempt to fix broken encoding. 
TBR=niklase@webrtc.org
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/637004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2353 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-04 11:04:05 +00:00
niklas.enbom@webrtc.org
9cf4d72d5d git-svn-id: http://webrtc.googlecode.com/svn/trunk@2352 4adac7df-926f-26a2-2b94-8c16560cd09d 2012-06-04 10:58:43 +00:00
niklas.enbom@webrtc.org
82bf033380 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2351 4adac7df-926f-26a2-2b94-8c16560cd09d 2012-06-04 10:57:51 +00:00
niklas.enbom@webrtc.org
265e38c336 Fixing test gypi for bit rate controller 
Review URL: https://webrtc-codereview.appspot.com/636004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2350 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-04 10:12:44 +00:00
braveyao@webrtc.org
ab12990b1b In the past we support calling StartPlayingFileLocally() before StartPlayout(), then when playout is started, the file would be played out immediately. 
Now we would get a failure if we do the same thing and the file would not be played out. Then GTalk/Hangout also reported this failure to us. 
This CL is to restore the original function. 

BUG = Issue 490
TEST = Manual test and voe_auto_test->FileBeforeStreamingTest
Review URL: https://webrtc-codereview.appspot.com/569016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2347 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-04 03:26:39 +00:00
marpan@webrtc.org
899baa821b Temporarily disable first partition packet counting to avoid a bug in ProducerFec which doesn't properly handle important packets. 
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/631005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-01 18:32:16 +00:00
leozwang@webrtc.org
354b0ed015 Check return result of fwrite [Audio Module] 
Description:
On ChromeOS/ARM, compiler enforces to check return result of a function.
Currently, we don't check return result of fwrite, it causes building errors.

The following files need to patch. The patch should be similar, before I patch all
of them, I will start with 2 files, please take a quick look, if the patch is OK,
I will continue and upload a new patch that covers all of them.
it to all of them.
Review URL: https://webrtc-codereview.appspot.com/566016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2345 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-01 17:46:21 +00:00
kma@webrtc.org
c3b2683bf4 Refactored the pitch filter function in iSAC-fix. One important purpose is to prepare the function for assembly optimization in ARM platforms. 
Note that,
(1) The main change is a new function PitchFilter() replacing a couple of common code blocks. Next step will be the assembly coding of this function in ARM.
(2) Resulted code is not bit exact with the original. The only reason is replacing two saturation blocks (lines 197 and 208) for the case of "type == 2" with the general case (line 147 and 159). The change makes the code more consistent, and I think the original code might just be a bug. I raised the issue in an email to Turaj and Bjorn last week.
Listening test might be needed. I will send the resulted files to Turaj for this purpose.
(3) I used Astyle to make the code more stylish, but didn't try extra effort to correct all the code style details.  Local code style consistency was considered for new code. So this is not a full and final refactor project (will leave that to future refactoring).
Review URL: https://webrtc-codereview.appspot.com/573009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-01 17:00:07 +00:00
tina.legrand@webrtc.org
5b4f36db88 ACM: Too short char vector 
Revision 2340 failed on Linux Release, and the problem was that we allocated a too short vector for the output file name.

BUG=r2340 failed on Linux release
TEST=audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/624006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2343 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-01 14:51:28 +00:00
tina.legrand@webrtc.org
4517585db5 Adding separate payload types for stereo modes 
BUG=Issue 452
TEST=audio_coding_test, voe_auto_test, voe_cmd_test

Edit: adding Patrik to review:
src/modules/rtp_rtcp/source/rtp_receiver.cc
...and Shijing to review:
src/voice_engine/main/source/channel.cc
src/voice_engine/main/test/cmd_test/voe_cmd_test.cc

Review URL: https://webrtc-codereview.appspot.com/540004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2340 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-01 09:27:35 +00:00
pwestin@webrtc.org
c2722a0e68 Fixed compiler warning 
Review URL: https://webrtc-codereview.appspot.com/624005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2339 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-01 08:56:42 +00:00
stefan@webrtc.org
f5d934dfd8 Upgrade libvpx to cab6ac16 (v. 1.1.1 pre-point-release). 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2337 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-01 07:43:02 +00:00
andrew@webrtc.org
7d8c567982 Ignore return value of fwrites. 
The removed error return was of course failing in the void ProcessBlock.
Ignored the returns of the remaining fwrites as well for consistency.

TBR=leozwang@webrtc.org
BUG=none
TEST=run audioproc with debug enabled

Review URL: https://webrtc-codereview.appspot.com/623004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2336 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-01 02:41:14 +00:00
kjellander@webrtc.org
2e84c112f5 Updating bitrate controller tests to test naming conventions. 
The test is now named 'bitrate_controller_unittests'.
This CL also enables it on the bots. The test is excluded on ASAN since
it fails when compiled with projects generated with GYP_DEFINES='asan=1' (see issue 555).

BUG=None
TEST=bitrate_controller_unittests was tested in Debug+Release on Linux, Mac and Windows + TSAN and memcheck.

Review URL: https://webrtc-codereview.appspot.com/612004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2333 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-31 13:55:01 +00:00
phoglund@webrtc.org
baaf2434a7 Extracted a method for sending padded data. 
BUG=
TEST=Ran vie_auto_test and voe_auto_test standard tests.

Review URL: https://webrtc-codereview.appspot.com/605004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-31 10:47:35 +00:00
andrew@webrtc.org
36ccce4f58 Remove documentation folders. 
Review URL: https://webrtc-codereview.appspot.com/606007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2329 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-30 17:28:24 +00:00
andrew@webrtc.org
16fcb247b2 Disable flaky VolumeTests only on Linux. 
BUG=issue367
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/611005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-30 17:26:32 +00:00
leozwang@webrtc.org
e7e64e3468 Fix compilation errors on ChromeOS 
Description:
This cl fixes two compilation errors on ChromeOS/ARM, it could
also be reproduced by gcc 4.5+.

I also add comments about error message and how I solve them.

BUG=webrtc issue 554
TEST=try bots and build on chromeos arm
Review URL: https://webrtc-codereview.appspot.com/611006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2327 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-30 16:46:09 +00:00
niklas.enbom@webrtc.org
0cb79cc851 Fixing gyp bug in https://webrtc-codereview.appspot.com/599006 
Review URL: https://webrtc-codereview.appspot.com/609006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2325 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-30 14:32:42 +00:00
stefan@webrtc.org
dc257b5781 Add option to configure error concealment and disable by default. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/597005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2324 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-30 11:25:00 +00:00
mflodman@webrtc.org
327ada1cb0 Refactored IncomingVideoStream and VideoRenderFrame, to get code in better shape when hunting BUG=481. 
BUG=481
TEST=Compiles on all platformas and autotest passes.

Review URL: https://webrtc-codereview.appspot.com/608005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2323 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-30 10:45:18 +00:00
bjornv@webrtc.org
281b7983db Resolved Coverity warnings. 
This CL includes changes to resolve Coverity warnings 14086, 14110 and 14111.

Tested with trybots and audioproc_unittests.

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/606004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2321 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-30 07:41:57 +00:00
leozwang@webrtc.org
b5ea03adbb Add print out stats summary to integrationtest.cc 
Stats summary prints out cpu usage.

BUG=
TEST=test on linux
Review URL: https://webrtc-codereview.appspot.com/602004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2320 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-30 00:34:50 +00:00
andrew@webrtc.org
459955f821 Move audio_frame_operations to the utility module. 
TBR=henrika@webrtc.org
BUG=issue551
TEST=voe_auto_test, webrtc_utility_unittest, trybots

Review URL: https://webrtc-codereview.appspot.com/599006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2318 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-29 22:13:14 +00:00
andrew@webrtc.org
aafa49bb85 Disable flaky VolumeTest.DefaultSpeakerVolumeIsAtMost255. 
This test failed on six CLs in a row recently.

TBR=xians@webrtc.org
BUG=issue367
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/595007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-29 22:05:15 +00:00
andrew@webrtc.org
5f23d64cf2 Set the stream delay to zero if too low. 
- Return a stream warning instead of an error.
- Add a few delay offset tests.

BUG=
TEST=audioproc_unittest

Review URL: https://webrtc-codereview.appspot.com/607004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2316 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-29 21:14:06 +00:00
leozwang@webrtc.org
2fc6e388c0 Check return value of fwrite. [Video Module] 
Description:
On ChromeOS/ARM, compiler enforces to check return result of a function.
Currently, we don't check return result of fwrite, it causes building errors.

The following files need to patch. The patch should be similar, before I patch all
of them, I will start with 3 files, once we agree upon the solution, we will expand
it to all of them.

The question is should we do 
1. if (error) { return -1;} 
or 
2. if (error) { /*ignor the error*/ }

I took "return -1" in this patch, but I'm OK with either. Please let me know your
thoughts and I will upload a new patch.
Review URL: https://webrtc-codereview.appspot.com/583010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2315 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-29 17:33:13 +00:00
pwestin@webrtc.org
1eef9c16ff Bitrate bugfixes 
Review URL: https://webrtc-codereview.appspot.com/609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2313 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-29 09:28:43 +00:00
stefan@webrtc.org
5abab0b1b5 Revert 2311 - Disable error concealment. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/609004

TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/604006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2312 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-29 09:04:40 +00:00
stefan@webrtc.org
3348b2990b Disable error concealment. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2311 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-29 08:44:00 +00:00
mflodman@webrtc.org
ca8d788362 Fix a bug where a RAII object was created for just one line instead of a block. 
Found by clang:

../../third_party/webrtc/video_engine/vie_render_manager.cc:157:3: error: expression result unused [-Werror,-Wunused-value]
  ViEManagerWriteScoped(*this);
  ^~~~~~~~~~~~~~~~~~~~~~~~~~~~
1 error generated.

Review URL: https://webrtc-codereview.appspot.com/599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2309 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-28 18:56:20 +00:00
phoglund@webrtc.org
dbaa893525 Completed rewrite of APM extended test. 
Removed NS tests since they are already covered by audio_processing_test.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/603004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2308 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-28 14:36:59 +00:00
bjornv@webrtc.org
1747427861 VAD Refactoring: Replaced pointer operation with array index 
This CL contains a change of pointer nomenclature to array index. In addition, one place with two hard coded Gaussians has been generalized with a for loop.

Tested with trybots, vad_unittests, audioproc_unittest

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/592004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2307 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-28 12:50:05 +00:00
bjornv@webrtc.org
4e12d3065e VAD Refactoring: Removed assign calls 
These calls are not used anywhere in WebRTC and there is no plan on using them.
Removed them and updated corresponding unit tests.

Tested on trybots, vad_unittests, audioproc_unittest

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/608004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2306 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-28 12:25:07 +00:00
tina.legrand@webrtc.org
0de1ee3830 NetEQ: Remove an unnecessary condition, to fix a clang warning 
This is a duplicate of issue 606005: https://webrtc-codereview.appspot.com/606005/

BUG=
TEST=neteq_unittests

Review URL: https://webrtc-codereview.appspot.com/605005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2305 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-28 11:37:50 +00:00
kma@webrtc.org
0d321da7e1 Refactored ARM specific code in Noise Suppression. Bit exact. 
Review URL: https://webrtc-codereview.appspot.com/459006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2303 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-26 01:05:27 +00:00
leozwang@webrtc.org
1755a57cbc Check return result of fwrite, [APM] 
Description:
This cl added checking return result of fwrite which makes it compile
on ChromeOS/ARM.

BUG=issue:541
TEST=Build on all platforms
Review URL: https://webrtc-codereview.appspot.com/583009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2302 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-25 19:20:35 +00:00
leozwang@webrtc.org
f14575fd8e Dynamically load codec list 
Description:
This cl adds a feature that can query video engine and voice engine and load code list in
gui settings. Currently, codec lists are fixed in resource file, it caused confusion and
problems.

TBR=ronghua
BUG=
TEST=test on android
Review URL: https://webrtc-codereview.appspot.com/583006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2301 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-25 16:47:56 +00:00
leozwang@webrtc.org
351fb6d3b4 Exclude code that don't work on android in voe_cmd_test 
Description:
Ths cl makes voe_cmd_test work on android by excluding some code
that are availabel on android today, some highlights
1. change maxnumofchannles
2. disable audio device selection
3. disable set/get volume

BUG=
TEST=test on try bots
Review URL: https://webrtc-codereview.appspot.com/584009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2300 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-25 16:47:35 +00:00
turaj@webrtc.org
10d3b5239b I haven't done any refactoring here. 
Resolve coverity warnings.

14305.

The warning is not really valid. The 'decode' function should be called with a 'mode' variable, where inside the function it is assumed that mode is either zero or one. If mode is taking other values some varibles are used uninitialized. However, this is an internal function and it is always called with either ZERO or ONE. Therefore, the code operates correctly. I made small changes as I beleive it is a bit nicer way. 

In ACM:
- Conditions on 'mode' is changed.


Tested with trybots.

BUG=None
TEST=None
Review URL: https://webrtc-codereview.appspot.com/564014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2297 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-24 21:20:25 +00:00
andrew@webrtc.org
f45d47ad7d Remove mixing tests from voe_extended_test.cc 
These have been moved to:
src/voice_engine/main/test/auto_test/standard/mixing_test.cc

BUG=
TEST=build voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/588005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-24 17:27:59 +00:00
andrew@webrtc.org
51b4f3e6a8 Try to fix MixingTest on the Win bots. 
- Relax the constraints on recording duration.
- Remove unneeded file deletes. (These files will be properly
  overwritten anyway).

TBR=henrike@webrtc.org
BUG=issue534
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/600006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2295 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-24 17:26:05 +00:00
stefan@webrtc.org
42e78ac087 Disable normal_async_test and rename tests to vp8_integrationtests. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/598004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2294 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-24 14:36:34 +00:00
mflodman@webrtc.org
6af9594d71 Added gyp variable to include/exclude all tests. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/597004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2292 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-24 13:23:35 +00:00
niklas.enbom@webrtc.org
ee646c37d4 I know this is ugly, but it helps a lot to quickly update webRTC in Chrome and libJingle. 
Review URL: https://webrtc-codereview.appspot.com/596004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-24 11:41:02 +00:00
pwestin@webrtc.org
2d1fc9bf17 Added critical section to prevent race. 
Review URL: https://webrtc-codereview.appspot.com/595004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2289 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-24 11:28:31 +00:00
pwestin@webrtc.org
5c3a400fae Re-added ChangeUniqueId temporary for chrome. 
Review URL: https://webrtc-codereview.appspot.com/594004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2287 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-24 09:52:19 +00:00
andrew@webrtc.org
7fbfc4ce79 Use correct variable in trace. 
TBR=leozwang@webrtc.org
TEST=build

Review URL: https://webrtc-codereview.appspot.com/593004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-23 22:22:36 +00:00
andrew@webrtc.org
f98b6cc10e Remove noise during build on Win32 
TBR=niklas.enbom@webrtc.org
BUG=chromium:126483

Review URL: https://webrtc-codereview.appspot.com/590006
Patch from Scott Graham <scottmg@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-23 18:49:47 +00:00
hta@webrtc.org
40300131dc No more TSAN errors on start_stop 
A very pedestrian approach to making TSAN stop complaining about the state variables: Wrap them in a critical section.
More creative approaches can be considered.

(not sure if diffbase works here, but...)
DIFFBASE=583008

BUG=webrtc:300
TEST=

Review URL: https://webrtc-codereview.appspot.com/584008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2282 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-23 15:49:48 +00:00
turaj@webrtc.org
ea0aa13aa8 I haven't done any refactoring here. 
Resolve coverity warnings.

14240, 14241.

In ACM:
- NULL pointer sanity checks corrected.

Tested with trybots.

BUG=None
TEST=None
Review URL: https://webrtc-codereview.appspot.com/571012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-23 15:43:51 +00:00
andrew@webrtc.org
9dc45dad1b Move trunk/test/data -> trunk/data 
BUG=
TEST=all trybot test failures passed locally

Review URL: https://webrtc-codereview.appspot.com/583007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-23 15:39:01 +00:00
hta@webrtc.org
1484ac07cb Replaced 2 time functions with thread-safe equivalents 
This does not fix bug 544, but changes where it reports issues.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/578008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2279 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-23 15:39:17 +00:00
stefan@webrtc.org
78d8d99180 Fixes the vp8_test on Linux. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/579015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-23 14:18:56 +00:00
bjornv@webrtc.org
cb0a86e913 VAD refactoring: Added function for repeated code. 
Added WeightedAverage() to calculate global mean. This removes hard coded Gaussian model size and repeated code.

Tested with vad_unittests, audioproc_unittest and trybots.

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/571006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2275 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-23 07:56:51 +00:00
andrew@webrtc.org
a1a34d675f Avoid flakiness by skipping more output verification. 
- Add a SCOPED_TRACE in case it flakes out again.
- The test doesn't need to be very long, so shorten it to save the bots some time.

TBR=henrike@webrtc.org
BUG=
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/588006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2273 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-23 00:45:00 +00:00
hta@webrtc.org
6ed617be22 Fixing memory leak error in test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/571013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2272 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-22 16:55:16 +00:00
marpan@webrtc.org
747cd87da1 FEC: For multi-frame FEC, allow for the size of the code to be increased, 
under certain conditions. This generally improves the FEC recovery for 
     bursty loss under medium-high protection level.
Review URL: https://webrtc-codereview.appspot.com/566012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2271 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-22 16:50:00 +00:00
hta@webrtc.org
e1919f41b7 Unittest for thread_wrapper. 
Added an explanation of the state variables of posix, wrote a simple test.
This might help in fixing the TSAN issue with thread.

BUG=webrtc:300
TEST=

Review URL: https://webrtc-codereview.appspot.com/583008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2270 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-22 15:57:34 +00:00
pwestin@webrtc.org
0d71c1cfe0 Cleanup code. 
Review URL: https://webrtc-codereview.appspot.com/569021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2269 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-22 14:23:25 +00:00
astor@webrtc.org
e231c61ba7 Style guide reformatting of overuse_detector.{h,cc} 
Review URL: https://webrtc-codereview.appspot.com/575006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2268 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-22 12:24:00 +00:00
mflodman@webrtc.org
7597a85bc4 Changed merge-name. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/584007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2266 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-22 08:32:20 +00:00
andrew@webrtc.org
294be77c2e Permit mixing mono and stereo streams. 
Add mixing tests based on older ones from the extended tests.

BUG=issue534
TEST=manual, voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/576014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2265 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-22 03:28:41 +00:00
pwestin@webrtc.org
b2179c20f0 Bugfix issue 533. Client does not handle NACK or PLI requests received from far end if a RTCP report from it has not been processed when RFC 5506 is enabled. 
Review URL: https://webrtc-codereview.appspot.com/569020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2263 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-21 12:00:49 +00:00
stefan@webrtc.org
1bca6d2437 Fixes coverity warnings. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/566014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2262 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-21 07:42:12 +00:00
andrew@webrtc.org
b3bea2eb3e Fix build errors on OS X Lion. 
TBR=henrika@webrtc.org
TEST=build on Lion, trybots
Review URL: https://webrtc-codereview.appspot.com/583005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2261 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-18 19:02:42 +00:00
leozwang@webrtc.org
e3ba738ba3 Set OpenGL as the default renderer 
Description:
This cl sets OpenGL as the default renderer which is widely used and should
be test first

BUG=
TEST=test on android
Review URL: https://webrtc-codereview.appspot.com/573007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-18 17:07:49 +00:00
leozwang@webrtc.org
ada5db4e75 Enable aecm neon optimized code in android build 
Description:
This CL is a follow up of http://review.webrtc.org/575008/ and r2241.
Because of r2243, r2241 is messed up and reverted, I'm going to commit it again.
This is exact same to the last patch in 575008, just want to inform you and have
your double check before I commit it.

The original description
This cl enables cpu detection and aecm optimized code in android build.

BUG=
TEST=build on all platforms
Review URL: https://webrtc-codereview.appspot.com/568006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2259 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-18 16:48:45 +00:00
phoglund@webrtc.org
1ad477de3e Added a bit flip fuzz test to the voice engine. 
Extracted encryption classes to a new test library.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/564009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2256 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-18 08:02:37 +00:00
leozwang@webrtc.org
00c07e66a7 Fix a random crash because of NULL point 
Description:
This cl fixes NULL point crash problem which was not detected runtime.
Also, small reformats are added too.

BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/579009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2254 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-16 16:57:28 +00:00
leozwang@webrtc.org
4d7d23cde4 Refactor OpenGL Code 
Description:
This CL refactored OpenGL java code. Most of code referenced sample code.
And bug-fix and reformats are added into this cl too.

BUG=
TEST=build and test on android
Review URL: https://webrtc-codereview.appspot.com/583004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2253 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-16 16:57:15 +00:00
hta@webrtc.org
3a698651d7 Coverity issue 14317 (uninitialized status may influence execution flow) 
Solution: restructure the flow when AquireSocket fails

BUG=coverity:14317
TEST=trybot

Review URL: https://webrtc-codereview.appspot.com/573006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2252 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-16 14:28:52 +00:00
kjellander@webrtc.org
7e4a72a78a Revert 2241 - Enable aecm neon optimized code in android build 
Description:
This cl enables cpu detection and aecm optimized code in android build.

BUG=
TEST=build on all platforms
Review URL: https://webrtc-codereview.appspot.com/575008

TBR=leozwang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/566013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2249 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-16 11:49:45 +00:00
kjellander@webrtc.org
0290a7a827 Revert 2243 - Re-enable embedding svn revision into code 
Description:
By using a python scription, this cl tries to get svn revision properly. It
current support git-svn and svn, if it fails, "n/a" will be defined as
svn revision.


BUG=500
TEST= test cases: w/o svn, w/o git-svn. test platforms, linux/windows. Passed all trybots.
Review URL: https://webrtc-codereview.appspot.com/564010

TBR=leozwang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/577008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2248 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-16 11:39:14 +00:00
hta@webrtc.org
d429086f62 Deleted udp_socket_windows and all references to it. 
This fixes a couple of Coverity issues, and doesn't impact any tested
platforms.

BUG=Coverity:14423
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/564012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2247 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-16 09:32:14 +00:00
tina.legrand@webrtc.org
86da94ea69 Remove functions for unregistering decoder 
This cl removes unused functions in the ACMGenericCodec class.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/568005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2245 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-16 07:11:53 +00:00
leozwang@webrtc.org
d2d54c72b6 Improve WebRTCDemo 
Description:
This cl basically bring video and audio alive and add couple features to
existing demo application,
1. Remove "stats" tab, add on screen stats display
2. Add a button to select surfaceview or opengl render
Also some reformat and minor bug-fixes are included

BUG=
TEST=build on android
Review URL: https://webrtc-codereview.appspot.com/579010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2244 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-16 03:18:23 +00:00
leozwang@webrtc.org
f6e27f5e03 Re-enable embedding svn revision into code 
Description:
By using a python scription, this cl tries to get svn revision properly. It
current support git-svn and svn, if it fails, "n/a" will be defined as
svn revision.


BUG=500
TEST= test cases: w/o svn, w/o git-svn. test platforms, linux/windows. Passed all trybots.
Review URL: https://webrtc-codereview.appspot.com/564010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2243 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-15 19:25:37 +00:00
kma@webrtc.org
f1ccdb9fb5 Aligned video buffer to 32 bytes boundary, when using vpx_img_alloc() in vp8. 
Review URL: https://webrtc-codereview.appspot.com/570009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2242 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-15 18:16:46 +00:00
leozwang@webrtc.org
fe742200d5 Enable aecm neon optimized code in android build 
Description:
This cl enables cpu detection and aecm optimized code in android build.

BUG=
TEST=build on all platforms
Review URL: https://webrtc-codereview.appspot.com/575008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2241 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-15 16:11:24 +00:00
marpan@webrtc.org
cf2cd7e4c4 Enable VP8 deblocker. 
Review URL: https://webrtc-codereview.appspot.com/569011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2237 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-14 16:50:36 +00:00
hta@webrtc.org
93116ba4fc Added an unittest for udp_socket_wrapper. 
This involves checking what the reasonable call sequences for deleting a
socket is; documented the API for this by making the destructor protected.
Checked out that the behaviour of undeleted sockets is inconsistent across
platforms, and changed the udp_socket_manager_unittest accordingly.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/578007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2236 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-14 16:03:37 +00:00
kma@webrtc.org
4e7840d943 Revert 2233 - Aligned video buffer to 32 bytes boundary, when using vpx_img_alloc() in vp8. 
M    vp8.cc



TBR=kma@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/579013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-11 18:16:35 +00:00
kma@webrtc.org
6d47c08e14 Aligned video buffer to 32 bytes boundary, when using vpx_img_alloc() in vp8. 
M    vp8.cc



git-svn-id: http://webrtc.googlecode.com/svn/trunk@2233 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-11 17:29:59 +00:00
pwestin@webrtc.org
2853dde520 Refactor the internal API to the rtp/rtcp module. 
Combination of previous CLs in revisions 2211, 2212, 2214, 2215, 2216.
Review URL: https://webrtc-codereview.appspot.com/570008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-11 11:08:54 +00:00
phoglund@webrtc.org
5dbe568417 Disabled flaky tests. Standard tests will no longer run within extended tests. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/578006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2230 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-11 08:59:11 +00:00
bjornv@webrtc.org
61d0745e86 Resolve coverity warnings. 
14050, 14051, 14243, 14314

In APM:
- Uninitialized variable initialized.
- NULL pointer sanity checks corrected.

Tested with trybots and audioproc_unittest.

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/571009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2229 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-11 07:51:44 +00:00
phoglund@webrtc.org
7eadad6d95 Fixed valgrind errors so the fuzz test can be run under valgrind. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/576008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2228 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-11 07:36:53 +00:00
turaj@webrtc.org
3c383abd27 Revert 2211 - Refactor the internal API to the rtp/rtcp module. 
Review URL: https://webrtc-codereview.appspot.com/568004

A series of CL:s by Patrik W. is breaking the auto-test. It started with CL 2211, but the later CL:s seems dependent on another. So I decided to go in reverse order and revert all of them.

TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/563011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 23:01:04 +00:00
turaj@webrtc.org
980d6be535 Revert 2212 - Bug fix 
Review URL: https://webrtc-codereview.appspot.com/576009

A series of CL:s by Patrik W. is breaking the auto-test. It started with CL 2211, but the later CL:s seems dependent on another. So I decided to go in reverse order and revert all of them.

TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/575009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2225 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 22:57:21 +00:00
turaj@webrtc.org
449b525453 Revert 2214 - Bugfix 
A series of CL:s by Patrik W. is breaking the auto-test. It started with CL 2211, but the later CL:s seems dependent on another. So I decided to go in reverse order and revert all of them.

TBR
Review URL: https://webrtc-codereview.appspot.com/570007

TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/571010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2223 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 22:54:39 +00:00
turaj@webrtc.org
f02ee061ab Revert 2215 - Fixed destroy order. 
A series of CL:s by Patrik W. is breaking the auto-test. It started with CL 2211, but the later CL:s seems dependent on another. So I decided to go in reverse order and revert all of them.

TBR
Review URL: https://webrtc-codereview.appspot.com/564007

TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/562008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2222 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 22:53:33 +00:00
turaj@webrtc.org
f448ccd16c Revert 2216 - Fix for receive only channels. 
A series of CL:s by Patrik W. is breaking the auto-test. It started with CL 2211, but the later CL:s seems dependent on another. So I decided to go in reverse order and revert all of them.

TBR
Review URL: https://webrtc-codereview.appspot.com/564008

TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/579011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2221 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 22:52:19 +00:00
leozwang@webrtc.org
8aaf31d14f Create makefile for video engine 
Description:
This build.xml will enable video engine build on android by ant. As we discussed before,
in order to make video engine build based on current webrtc file structure which is not
normal, I created this customized build.xml which is based on build.xml in android SDK.

BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/571008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2219 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 18:16:27 +00:00
leozwang@webrtc.org
f29d3fa177 Fix vp8 complexity parameter on android 
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/575007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2217 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 18:12:01 +00:00
pwestin@webrtc.org
a69634aa92 Fix for receive only channels. 
TBR
Review URL: https://webrtc-codereview.appspot.com/564008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2216 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 16:17:05 +00:00
pwestin@webrtc.org
c309c7c52b Fixed destroy order. 
TBR
Review URL: https://webrtc-codereview.appspot.com/564007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2215 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 15:49:45 +00:00
pwestin@webrtc.org
fd3fef514b Bugfix 
TBR
Review URL: https://webrtc-codereview.appspot.com/570007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2214 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 15:14:41 +00:00
pwestin@webrtc.org
e1c97a2723 Bug fix 
Review URL: https://webrtc-codereview.appspot.com/576009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2212 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 13:55:23 +00:00
pwestin@webrtc.org
0774838f3d Refactor the internal API to the rtp/rtcp module. 
Review URL: https://webrtc-codereview.appspot.com/568004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 12:33:50 +00:00
bjornv@webrtc.org
cd54e56d72 Delay estimation performance test. 
Added a test to verify estimation performance. Tested with audioproc_unittest and trybots.

BUG=issue435
TEST=None

Review URL: https://webrtc-codereview.appspot.com/569006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2209 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 07:41:23 +00:00
andrew@webrtc.org
270e9db039 Clarify the impact of disabling VAD on DTX. 
TBR=henrika@webrtc.org
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/566009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2207 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-09 19:09:03 +00:00
leozwang@webrtc.org
b213cd55ef Remove bitmap implementation from surfaceview render 
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/571004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2205 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-09 18:24:39 +00:00
leozwang@webrtc.org
de6a6b40fa Rename ViEAndroidDemo to WebRTCDemo 
Desription:
This CL will change VideoEngine application name to ViEAndroidDemo. This
is the first step to refactor videoengine test application on android.

"ViE" is not a proper name because the app also supports audio. And it's good
to put WebRTC in application name.

BUG=
TEST=build on all trybots
Review URL: https://webrtc-codereview.appspot.com/576006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2204 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-09 15:23:52 +00:00
niklas.enbom@webrtc.org
f6edfeff63 Adding one parameter to typing detection tuning 
Review URL: https://webrtc-codereview.appspot.com/569009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2203 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-09 13:16:12 +00:00
leozwang@webrtc.org
9f49af9cea Add libbitrate_controller which is introduced recently 
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/571007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2202 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-09 04:46:24 +00:00
leozwang@webrtc.org
d63cf71413 Fix broken build because gyp doesn't filter out platform specific file with suffixes of .c 
BUG=
TEST=test on all trybots, failed only on win_rel for weird error.
Review URL: https://webrtc-codereview.appspot.com/563009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2200 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-08 21:33:04 +00:00
andrew@webrtc.org
710eac763a Update DEPS comment and remove tabs from common.gypi. 
TBR=wu@webrtc.org
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/575005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2198 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-08 18:00:57 +00:00
andrew@webrtc.org
22f6f85fcc Remove redundant initialization. 
TBR=xians@webrtc.org
BUG=6140661
TEST=build on Linux

Review URL: https://webrtc-codereview.appspot.com/576005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2197 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-08 17:18:28 +00:00
andrew@webrtc.org
e59a0aca6a Fix AudioFrame types. 
volume_ is not set anywhere so I'm removing it.

BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/556004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2196 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-08 17:12:40 +00:00
leozwang@webrtc.org
f5fe1000de Enable cpu auto detection and ns optimized code on android 
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/547006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2195 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-08 15:20:02 +00:00
hta@webrtc.org
9cc186405f Removed CleanUp call in Windows, since I couldn't find a call to it. 
This solves Coverity issue 14424 by deleting the code.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2194 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-08 11:40:06 +00:00
mflodman@webrtc.org
e553031580 Refactore base, Capture, Codec and Custom Call parts of autotest. The CL doesn't contain any real functional changes, only style changes, cast changes and changing reference to pointer as input argument to functions. 
Custome call still doesn't pass cpplint, but I'll take that in another CL to not change the structure in the style change CL.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/523001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2193 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-08 10:38:36 +00:00
braveyao@webrtc.org
113f851cc3 Merge Chromium issue 95797 into WebRTC. 
Bug = 450
Test = Manual test
Review URL: https://webrtc-codereview.appspot.com/551004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2192 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-08 09:28:39 +00:00
pwestin@webrtc.org
7415f371ac Revert VP8 Deblocker. 
Review URL: https://webrtc-codereview.appspot.com/563007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2191 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-08 09:06:31 +00:00
pwestin@webrtc.org
5019c9583c Enable VP8 deblocker. 
Review URL: https://webrtc-codereview.appspot.com/578004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2190 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-08 08:40:28 +00:00
andrew@webrtc.org
589673f1cb Fix volume setting while not playing on PulseAudio. 
We now only set the volume when starting playout if the user has called
SetSpeakerVolume while we weren't playing. This now also ensures it will
actually get set to what the user requested rather than being overridden
by a default value.

Add tests to voe_auto_test.

BUG=6140661
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/566006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-07 21:42:49 +00:00
leozwang@webrtc.org
20a05cd778 Disable WEBRTC_SVNREVISION 
Description:
This CL will temporally define WEBRTC_SVNREVISION as "n/a" because it
could break Chromium if svn is not installed.
The long term solution is a have a script that could deal with it, and
have it support git-svn which is used by most developers.

BUG=496
TEST=build on Linux
Review URL: https://webrtc-codereview.appspot.com/569007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2187 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-07 21:16:33 +00:00
turaj@webrtc.org
fe4cfa7e5e Hi Tina, 
I have uploaded this patch for your review. I have done an extensive test to be sure that removing of tables does not create any problem. 

The test file, is called test_lpc.c which requires a hack to standard iSAC. The test computes LPC coefficients, then encodes and decodes with old and new (size-reduced) tables. It compares the results is all steps. I have ran the test over large set of files, more then 51 hours of audio, and there was no error. 

I tried to do no formatting so the review to be easier, but I know it can be a tricky CL. Hopefully, the test file helps you to be more confident on the CL. 

Thanks,... Turaj  

In this change list the LPC tables associated with mode 1 & 2 are remoded, and necessary cahnges are made to other files. 

The only model allowed is model number 0. Therefore, this CL breaks compatibility with iSAC released prior to 2.4.3. To avoid changing the bit-stream, we still keep the model number in the bit-stream. 

entropy_coding.c is cleaned up, especially encoding of LAR had KLT transform of LPC gains which are removed now. 
Review URL: https://webrtc-codereview.appspot.com/548004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2186 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-07 20:36:22 +00:00
leozwang@webrtc.org
d46fe7034b Two bug fixs in android surface render 
Descritption:
This CL addresses two issues in android surface view render,
1. Uninitlized class members _javaByteBufferObj and _directBuffer which
   could cause crash.
2. Using ConvertI420ToRGB565. We should use high level libyuv apis
   to help libyuv maintainer.

BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/566005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2185 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-07 20:29:43 +00:00
braveyao@webrtc.org
ba0f9fe10b Trival fix to voe_auto_test according to the main source codes 
BUG = NULL
TEST = voe_auto_test
Review URL: https://webrtc-codereview.appspot.com/554004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-07 10:06:43 +00:00
phoglund@webrtc.org
f5657efd31 Rolled back r2177 since it breaks vie_auto_test. 
I'm not sure what is the right thing to do here. That would probably be to call Release() the right amount of times and ensuring that the last call returns 0 (e.g. all references have been released), but I'll leave it up to the CL author to investigate that.

TBR=elham@webrtc.org
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/569005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2183 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-07 07:59:32 +00:00
andrew@webrtc.org
07bf9a07f5 Add test to verify identical input channels result in identical output channels. 
BUG=issue411
TEST=audioproc_unittest, trybots

Review URL: https://webrtc-codereview.appspot.com/553009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2182 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-05 00:32:00 +00:00
leozwang@webrtc.org
329fcbba29 Reduce PSNR because I420ToARGB888 return lowers number on windows 
TBR=stefan, marpan

BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/562005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2178 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-04 21:31:22 +00:00
elham@webrtc.org
c2f8832a32 Fix for FileTest failure in vie_auto_test 
Review URL: https://webrtc-codereview.appspot.com/570004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2177 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-04 21:17:30 +00:00
leozwang@webrtc.org
1ea25b4c3d Change PSNR for I420ToARGB888 and I420ToRGB565 
TBR=stefan, marpan

BUG=
TEST=test on linux
Review URL: https://webrtc-codereview.appspot.com/569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2176 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-04 17:55:57 +00:00
leozwang@webrtc.org
83958dfe06 Add ARGB and RGB565 unit test 
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/563004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2175 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-04 17:07:45 +00:00
leozwang@webrtc.org
3ebff4c2e9 Add ConvertToARGB and enable RGB565 
Review URL: https://webrtc-codereview.appspot.com/566004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2174 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-04 17:07:30 +00:00
leozwang@webrtc.org
e62fec2285 Bug fix and refactor video capture code on android 
BUG=
TEST=test on android
Review URL: https://webrtc-codereview.appspot.com/541009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2173 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-04 17:06:32 +00:00
hta@webrtc.org
b6f2417f37 Renamed all _test.cc files to _unittest.cc, to conform to convention 
for webrtc.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/560004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2172 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-04 08:13:57 +00:00
leozwang@webrtc.org
8b6f749b0a Rewrite makefile to link with stl statically 
By doing this way, webrtc.so doesn't depend on stlport_shared

BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/546005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2170 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-03 16:07:15 +00:00
hta@webrtc.org
54536bb6d4 Refactoring of the TMMBRSet class, giving it a reasonably tight interface. 
The CL also fixes a number of line length and tab issues in touched files.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/553005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2168 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-03 14:07:23 +00:00
pwestin@webrtc.org
3bc38c86e8 Fix color enhancement test. 
Review URL: https://webrtc-codereview.appspot.com/553007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2167 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-03 13:15:35 +00:00
pwestin@webrtc.org
209442a560 Missing file 
Review URL: https://webrtc-codereview.appspot.com/556005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2166 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-03 11:36:06 +00:00
pwestin@webrtc.org
e9727cdbaa Fixed some memory leaks. 
Review URL: https://webrtc-codereview.appspot.com/558004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2165 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-03 11:32:25 +00:00
andrew@webrtc.org
63a509858d Rename AudioFrame members. 
BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/542005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 23:56:37 +00:00
leozwang@webrtc.org
7fdb909339 Reformat and add more debug info into ViESurfaceRenderer 
BUG=
TEST=test on android
Review URL: https://webrtc-codereview.appspot.com/546004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2163 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 16:45:55 +00:00
bjornv@webrtc.org
b286bfb13e VAD refactoring: Replaced hard coded array sizes with enum. 
Further replaced hard coded calculations with a for loop for better understanding.

Tested with vad_unittests and audioproc_unittest.

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/519002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 12:14:57 +00:00
hta@webrtc.org
404843e6e5 Timeout tests for TMMBR 
Added a test that injects 3 packets and then times out.

Rewrote the packet construction in test to use a builder format.
This makes tests a lot more readable.

Odd behaviour of timeout found; documented as test.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/553004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2161 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 09:56:45 +00:00
hta@webrtc.org
3c0df7d376 Fixed a build break: I'd forgotten to remove a DELETE statement. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/555004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2160 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 08:22:21 +00:00
hta@webrtc.org
47059b5dfb Adds unit tests for RTCP receiver, focusing on TMMBR handling. 
This is the first part of a plan:

- Get basic unit tests for rtcp_receiver.
- Get an unit test for some code inside rtcp_receiver
  that touches the TMMBRSet class in hard-to-decipher ways
  (rtcp_receiver_help, GetTMMBRSet function, use of memmove()).
- Refactor the TMMBRSet class.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/547005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2159 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 07:46:22 +00:00
phoglund@webrtc.org
719dba7e79 Further cleaned up voe_standard_test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/522003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2157 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 07:32:37 +00:00
andrew@webrtc.org
ecac9b721e Add tests for downmixing and no processing. 
BUG=issue419
TEST=audioproc_unittest

Review URL: https://webrtc-codereview.appspot.com/532001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2154 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 00:04:10 +00:00
leozwang@webrtc.org
63ea5ef5da Regenerate jni files and bring audio alive 
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/550004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2153 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-01 22:13:49 +00:00
andrew@webrtc.org
d5548f5d04 Disable clang Chrome plugins on all platforms. 
(Will fix Linux-clang bot).

TBR=wu@webrtc.org
BUG=issue163
TEST=build on Linux-clang

Review URL: https://webrtc-codereview.appspot.com/545005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2152 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-01 21:39:34 +00:00
leozwang@webrtc.org
85b089a0ca Fix ConvertI420ToRGB565 bug 
BUG=
TEST=test on android
Review URL: https://webrtc-codereview.appspot.com/541005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2150 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-30 17:59:08 +00:00
leozwang@webrtc.org
e7ac5fde72 Minor changes to remove dead code in opensl es 
BUG=
TEST=build on all platforms
Review URL: https://webrtc-codereview.appspot.com/539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2149 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-30 14:42:17 +00:00
hta@webrtc.org
65a4e4ed56 Minor refactoring: RTCPReceiver::BoundingSet 
Remove ability to modify a pointer argument.

Added a test for transmitting a non-empty TMMBN

BUG=
TEST=unittest

Review URL: https://webrtc-codereview.appspot.com/542004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2148 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-30 11:23:41 +00:00
hta@webrtc.org
c2d985257b untabify 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-30 08:25:10 +00:00
hta@webrtc.org
9d54cd12ab TMMBN sender test passes, fixed receiver flag bug 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2144 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-30 08:24:55 +00:00
andrew@webrtc.org
5c0c18d823 Fix coverity issues in ACM. 
Fixes: Big parameter passed by value (PASS_BY_VALUE)
Passing parameter codec of size 52 bytes by value.

BUG=
TEST=audio_coding_module_tests, trybots

Review URL: https://webrtc-codereview.appspot.com/529002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2142 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 17:06:48 +00:00
andrew@webrtc.org
a88cb6fce0 Add HighPassFilter and StereoChannelSwapping tests. 
BUG=issue451
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/531001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2141 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 17:00:20 +00:00
marpan@webrtc.org
2d0223286b VPM: fix to coverity issues 10255-10258 (unintended sign extension). 
Review URL: https://webrtc-codereview.appspot.com/532002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2140 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 15:56:02 +00:00
phoglund@webrtc.org
ca08c41701 Replacing virtual camera on linux video bot: adaptings tests accordingly. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/537001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2139 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 12:38:17 +00:00
pwestin@webrtc.org
b1313aac7c Fix missing h file change. 
Review URL: https://webrtc-codereview.appspot.com/535001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2136 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 05:42:05 +00:00
pwestin@webrtc.org
49888ce428 Breaking out send side bitrate controll cont. 
Review URL: https://webrtc-codereview.appspot.com/475004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 05:25:53 +00:00
mallinath@webrtc.org
e611619f60 Fixing the header file path in gypi file. 
BUG=473
Review URL: https://webrtc-codereview.appspot.com/529001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2134 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 23:03:15 +00:00
andrew@webrtc.org
9c4f6a5ff9 Add an AudioFrameOperations unittest. 
Additionally, reformat audio_frame_operations to Goog style.

BUG=issue451
TEST=voice_engine_unittests

Review URL: https://webrtc-codereview.appspot.com/528001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2133 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 22:32:03 +00:00
tommi@webrtc.org
e49d908baf Fix how we were using TbInterfaces and disallow operator=() and the copy constructor. 
The reason is that this will cause a crash:

TbInterfaces foo = TbInterfaces("blah");


It relies on the generated copy constructor (or assignment operator), which copies the
pointer values from a temporary object.  After |foo| in this case has been initialized
with values from the temporary object, the temp goes out of scope and is deleted.
The result is that |foo| has been initialized with pointers do a deleted object.

Also fixing expectations for the return value of VoE Release() calls after I checked
in my change that makes the VoiceEngine per-object ref counted and not per-interface.
Review URL: https://webrtc-codereview.appspot.com/509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2128 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 17:05:32 +00:00
tommi@webrtc.org
a990e122c4 * Change the reference counting implementation for VoE to be per object and 
not per interface. This simplifies things a bit, reduces code and makes it
  possible to implement reference counting (if we ever want) without the
  static Delete() method.  (Reference counted objects are traditionally
  implicitly deleted via the last Release())

* Since the reference counting code is now simpler, there's no need for the
  RefCount class so I'm removing it.

* Also removing the optional |ignoreRefCounters| variable from the VoiceEngine::Delete
  method.  The justification is that it's no longer used and the reason it was there
  in the first place was to avoid bugs in third party code, so it's an indication that
  something is wrong elsewhere.

* Fix a crash in voe_network_impl and voe_extended_test that would happen on machines with IPv6 support.

* Fix an assert (handle the case) in the extended audio tests when SetCNAME is called with a NULL pointer.

* As a side effect of this change, hopefully the footprint of VoE will be slightly smaller :)

BUG=10445 (Coverity)
Review URL: https://webrtc-codereview.appspot.com/521003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2127 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 15:28:22 +00:00
phoglund@webrtc.org
497fb4fda9 Fixed vie_auto_test on mac so it will exit when the test completes instead of hanging like it used to. 
Also applied an old patch I had which rewrites the window handling to do ui stuff on the main thread instead of the worker thread, which as far as I know is a very bad idea.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/502001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2126 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 12:58:02 +00:00
tina.legrand@webrtc.org
bc1b43b297 Refactoring of audio_coding_module_impl 
First patch set: pure formatting.

Review URL: https://webrtc-codereview.appspot.com/522001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2125 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 08:53:45 +00:00
tina.legrand@webrtc.org
a6ecd1ebb5 Refactoring one of the ACM tests: TestStereo, to follow the style guide. 
(First patch: formatting the test file)

TEST=audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/507001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2124 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 07:54:30 +00:00
mflodman@webrtc.org
1868780c81 Disabled UnremovedSocketsGetCollectedAtManagerDeletion in UdpSocketManager unittest. 
TBR= hta@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/520004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2122 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 06:40:00 +00:00
andrew@webrtc.org
1c7bfe02f7 Fail silently when swapping mono. 
TBR=tina.legrand@webrtc.org
BUG=issue451
TEST=forthcoming unittest

Review URL: https://webrtc-codereview.appspot.com/527003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2121 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 00:20:28 +00:00
hta@webrtc.org
ad929899c7 Tests for udp_socket_manager. 
These tests basically check that socket allocation does not leak memory.

BUG=
TEST=unittested, ran under valgrind.

Review URL: https://webrtc-codereview.appspot.com/519003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2118 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-25 14:06:50 +00:00
asapersson@webrtc.org
d18dd6dc7e Made OnPacketLossStatisticsUpdate function virtual (needed for ViCE). 
Review URL: https://webrtc-codereview.appspot.com/520002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2115 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-25 07:19:02 +00:00
andrew@webrtc.org
02d7174722 Add API to swap stereo channels. 
BUG=issue451
TEST=manually with voe_cmd_test, using stereo and mono codecs

Review URL: https://webrtc-codereview.appspot.com/519001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2106 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-24 19:47:00 +00:00
andrew@webrtc.org
369166a179 Add API for disabling the high pass filter. 
BUG=issue419
TEST=manually with voe_cmd_test

Review URL: https://webrtc-codereview.appspot.com/509003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2105 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-24 18:38:03 +00:00
leozwang@webrtc.org
48a5df6481 Embed svn revision number into code 
BUG=
TEST=build on linux
Review URL: https://webrtc-codereview.appspot.com/516001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2104 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-24 14:50:50 +00:00
elham@webrtc.org
5f49dba1a1 Hi Magnus, I added some of the changes that you suggested before. Let me know what you think. 
Review URL: https://webrtc-codereview.appspot.com/507004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2101 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 21:24:02 +00:00
andrew@webrtc.org
4e423b3b1e Handle master/slave timestamp wrap. 
BUG=issue410
TEST=neteq_unittests, audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/506001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2098 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 18:59:00 +00:00
vikasmarwaha@webrtc.org
99ac3f7be5 Fixed trunacated trace problem in WebRTC. http://b.corp.google.com/issue?id=5607856 
Review URL: https://webrtc-codereview.appspot.com/508004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2096 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 17:04:35 +00:00
pwestin@webrtc.org
ddab60be56 Wire up pading. 
Review URL: https://webrtc-codereview.appspot.com/509002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2094 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 14:52:15 +00:00
bjornv@webrtc.org
11654c2344 VAD refactoring: Local variable name changes 
Changes tested with vad_unittests and audioproc_unittest

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/485004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2093 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 14:50:11 +00:00
mflodman@webrtc.org
5284d6e905 Minor change to trigger REMB packets in RTCP RR if there is no sending channel. 
BUG=
TEST=video_engine_core_unittest

Review URL: https://webrtc-codereview.appspot.com/508007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2092 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 13:22:26 +00:00
hta@webrtc.org
bf9f469a13 Lifetime management for UdpSocketManager 
Make tests use Create/Destroy *or* new/delete for UdpSocketManager.
Move responsibility for calling Destroy on UdpSocketManager from transport
destructor to transport Destroy function.

This all ends up in not leaking memory in InitializeSourcePorts test.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/512001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 13:19:30 +00:00
asapersson@webrtc.org
92591adc67 Fixes link issues in google3 (change by tomasl). 
Review URL: https://webrtc-codereview.appspot.com/509001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 13:10:55 +00:00
asapersson@webrtc.org
83ed0a4359 Try to resend next packet in nack list even if previous packet is not found. 
Review URL: https://webrtc-codereview.appspot.com/515001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 12:43:05 +00:00
pwestin@webrtc.org
fcbbe1f341 Removed unused callback code from video coding test. 
Review URL: https://webrtc-codereview.appspot.com/504003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2086 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 08:33:18 +00:00
pwestin@webrtc.org
a2cd732139 Fix for calling OnNetworkChanged too often. 
Review URL: https://webrtc-codereview.appspot.com/508006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 08:32:47 +00:00
marpan@webrtc.org
88ad06b999 VCM: Added condition(s) for setting FEC protection factor to zero at low bitrates. 
Review URL: https://webrtc-codereview.appspot.com/494001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-20 16:05:24 +00:00
asapersson@webrtc.org
63a34f4f29 Fix in SendPadData. Do not increase sequence number if packet is "empty" and not sent. 
Review URL: https://webrtc-codereview.appspot.com/508001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2083 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-20 13:20:27 +00:00