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Author SHA1 Message Date
kjellander@webrtc.org
1bc6d3238c Only status from interesting bots are reported to the Dashboard 
TBR=phoglund
BUG=none
TEST=Tested with local modifications on the live installation.

Review URL: https://webrtc-codereview.appspot.com/675005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2496 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-09 09:43:30 +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
mflodman@webrtc.org
2a45209a6d Adde video_engine to cpplint check. 
BUG=627
TEST=Manual test with style violations in src/video_engine and in other folders.

Review URL: https://webrtc-codereview.appspot.com/399014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2474 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-01 05:55:23 +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
mflodman@webrtc.org
efe20b39db Only add Mac compiler warning for clang, not gcc. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/676007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2466 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-28 17:28:06 +00:00
andrew@webrtc.org
1115fdbe69 Remove tab from DEPS. 
TBR=kjellander@webrtc.org
BUG=none
TEST=gclient sync

Review URL: https://webrtc-codereview.appspot.com/670005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2465 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-28 15:55:01 +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
kjellander@webrtc.org
44361abb20 Moving from Chromium cygwin dependency to our own 
BUG=
TEST=gclient sync on Windows followed by Debug build from scratch.

Review URL: https://webrtc-codereview.appspot.com/677009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2461 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-28 12:30:27 +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
andrew@webrtc.org
7450896a12 Roll Chromium 140240:143348. 
TBR=kjellander@webrtc.org
BUG=none
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/673005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2436 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-22 00:25:43 +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