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Author SHA1 Message Date
andrew@webrtc.org
6f8db36e04 Reorganize voice_engine/. 
The usual changes:
voice_engine/main/source -> voice_engine/
voice_engine/main/interface -> voice_engine/include
voice_engine/main/test -> voice_engine/test
Include path changes.

BUG=none
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/705004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2535 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-27 21:49:28 +00:00
andrew@webrtc.org
c1354bd768 Make handling of libyuv more flexible. 
- Use gyp variable for libyuv path.
- Rename internal libyuv.h to webrtc_libyuv.h to avoid conflicts.
- Update affected includes.

BUG=none
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/711004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2534 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-27 18:21:16 +00:00
braveyao@webrtc.org
ad69ca73d4 webrtc crashes with virtual cameras on Windows. 
In the market, there are several Virtual Camera apps which could allow multi-apps share one camera at same time. We have several reports that at least two of them would cause Webrtc crash because of 0 maxFPS read from their drivers. Finally I can reproduce and verify with one of them, 'Asus Virtual Camera' as in feedback. So I submit the fix as we discussed before.

BUG = Issue 464 & 675
TEST = ViE_Win_Test with virtual camera app installed
Review URL: https://webrtc-codereview.appspot.com/698005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2526 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-25 03:02:15 +00:00
andrew@webrtc.org
f5a91fdfab Make some build settings more flexible. 
BUG=issue676
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/700006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2524 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-23 16:28:02 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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