Commit Graph

1731 Commits

Author SHA1 Message Date
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