Commit Graph

1825 Commits

Author SHA1 Message Date
stefan@webrtc.org
d7b17e436a Enable denoising by default.
BUG=

Review URL: https://webrtc-codereview.appspot.com/716005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2573 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-08 10:02:51 +00:00
phoglund@webrtc.org
54e22eb977 Made it possible to run video_capture tests on mac.
Abstracted out a suitable main from vie_auto_test and put it into testsupport.
Cleaned up unused vie_auto_test mac code.

BUG=

Review URL: https://webrtc-codereview.appspot.com/723004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2572 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-08 08:27:46 +00:00
braveyao@webrtc.org
743e5cf6b7 remove flaky test case in FileBeforeStreamingTest
BUG = Issue 719
TEST = VoE standard test
Review URL: https://webrtc-codereview.appspot.com/718006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2571 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-08 06:07:39 +00:00
kma@webrtc.org
da236dfde5 Added more unit tests for min-max operations in signal processing module.
Review URL: https://webrtc-codereview.appspot.com/668009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2570 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-07 19:35:00 +00:00
perkj@webrtc.org
5d6be542be Make sure the video capture delay is set to an initial value on Mac.
BUG=

Review URL: https://webrtc-codereview.appspot.com/719006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2569 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-07 09:03:07 +00:00
kma@webrtc.org
f388fcc67e Added dynamic Neon detect in isac-fix for Android NDK build, and thus fixed a build error in the last version.
Review URL: https://webrtc-codereview.appspot.com/726004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2567 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-07 01:20:51 +00:00
kma@webrtc.org
715509890c Added run time detection of Neon architecture in iSAC-fix.
Review URL: https://webrtc-codereview.appspot.com/715004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2563 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-06 21:04:34 +00:00
kma@webrtc.org
8fe5f32ccc Refactor three signal processing library files. WebRTC issue 545 is solved by the way.
Review URL: https://webrtc-codereview.appspot.com/692007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2562 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-06 20:19:56 +00:00
mflodman@webrtc.org
1e1a250413 Wrong RTP module used when calling RegisterReceiveRtpHeaderExtension in ViE channel.
Review URL: https://webrtc-codereview.appspot.com/717010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2561 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-06 08:49:41 +00:00
kma@webrtc.org
2d4c4ae553 Optimization of function CalculateResidualEnergy() for iSAC-fix in ARM Neon platforms.
Bit not exact with the previous version, but result quality is not worse.
Review URL: https://webrtc-codereview.appspot.com/687005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2559 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-03 21:46:05 +00:00
andrew@webrtc.org
07ebdb9432 Handle 96 kHz when downmixing the capture path.
BUG=issue721
TEST=96 kHz capture on Windows works.

Review URL: https://webrtc-codereview.appspot.com/722004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2558 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-03 18:03:02 +00:00
elham@webrtc.org
c0348fb349 bump version to 3.9.0
Review URL: https://webrtc-codereview.appspot.com/708007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2556 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-03 17:47:52 +00:00
marpan@webrtc.org
4889120a84 Fix integer divisin truncation error.
Patch fix from: thakis@chromium.org
https://webrtc-codereview.appspot.com/717006/

TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/721006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2555 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-03 17:33:13 +00:00
mflodman@webrtc.org
10a31520a5 Disabled FileBeforeStreamingTest.TestStartPlayingFileLocallyWithStartPlayout.
BUG=719

TBR=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/710007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2554 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-03 09:50:15 +00:00
mikhal@webrtc.org
15264364a1 Removing RawImage.
Last cl in the series.

Removing RawImage

BUG=

Review URL: https://webrtc-codereview.appspot.com/709006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2553 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-02 22:18:47 +00:00
andrew@webrtc.org
686a731c18 Fix error when receiving an already sent timestamp from VoE.
BUG=issue715
TEST=automatic rapid switching between 32 kHz stereo and 16 kHz mono codecs in voe_cmd_test does not repro.

Review URL: https://webrtc-codereview.appspot.com/712006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2547 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-02 03:26:14 +00:00
wu@webrtc.org
792e974949 Refactor the public interfaces to use the full path in include.
BUG=

Review URL: https://webrtc-codereview.appspot.com/708006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2546 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-01 22:14:51 +00:00
mallinath@webrtc.org
42033b4a43 This change will allow us to set proper frame rate for the camera on Linux. Earlier we were setting based on the resolution irrespective of input frame rate.
Review URL: https://webrtc-codereview.appspot.com/692006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2545 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-01 06:31:34 +00:00
andrew@webrtc.org
d7a71d0719 Prepare to roll Chromium to 149181.
- This roll brings in VS2010 by default. The buildbots
  need updating (issue710).
- We'll roll to 149181 later (past current Canary) to fix
  a Mac gyp issue:
  https://chromiumcodereview.appspot.com/10824105
- Chromium is now using a later libvpx than us. We should
  investigate rolling our standalone build.
- Fix set-but-unused-warning
- Fix -Wunused-private-field warnings on Mac.

TBR=kjellander@webrtc.org
BUG=issue709,issue710
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/709007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2544 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-01 01:40:02 +00:00
mikhal@webrtc.org
bf853918eb Fix issue introduced in r2540
Setting render time of decoded frame

BUG=

Review URL: https://webrtc-codereview.appspot.com/719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2543 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-31 22:15:11 +00:00
mikhal@webrtc.org
4147562088 Fixing error introduced in r2540.
The error was in the test framework - did not copy the frame appropriately.
TESTED = test_framework unittest

Memory free

BUG=

Review URL: https://webrtc-codereview.appspot.com/713005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2541 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-31 18:10:14 +00:00
mikhal@webrtc.org
a2031d58f6 Replacing RawImage with VideoFrame in video_coding and related engine code.
This is the first step of replacing RawImage with VideoFrame in all WebRtc modules.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/672010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2540 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-31 15:53:44 +00:00
andrew@webrtc.org
5fe91a89ac Add license header to no_op.cc.
TBR=niklas.enbom@webrtc.org
BUG=chromium98592
TEST=none

Review URL: https://webrtc-codereview.appspot.com/717004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2539 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-31 05:50:21 +00:00
marpan@webrtc.org
8639fd9341 Use correct rtp header size for FEC packets.
This is needed as of r2489 which introduced the header extension.

This also is a fix for issue 701.
Review URL: https://webrtc-codereview.appspot.com/708005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2537 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-30 18:17:02 +00:00
andrew@webrtc.org
d1f3b1a113 Reorganize the vp8 directory.
The usual changes:
vp8/main/source -> vp8/
vp8/main/test -> vp8/test
vp8/main/interface -> vp8/include

All include paths etc. updated as needed.

BUG=none
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/704004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2536 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-27 22:19:16 +00:00
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
f5f69c7f5d Resource Preprocessor Definitions which contain spaces are handled incorrectly in Visual Studio 2010
BUG=ISSUE687
TEST=Building with VS2010
Review URL: https://webrtc-codereview.appspot.com/710004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2533 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-27 08:07:49 +00:00
mikhal@webrtc.org
7cbb5a05c4 JPEG: Replacing RawImage with VideoFrame.
Replacing RawImage with VideoFrame in JPEG related code

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/703004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2530 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-25 20:38:14 +00:00
leozwang@webrtc.org
8d95a700e9 Change libvp8 library patch in makefile
It's caused by recent file structure changes in vp8

TBR=ronghua,kma
BUG=
TEST=local build
Review URL: https://webrtc-codereview.appspot.com/707004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2528 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-25 18:37:00 +00:00
andrew@webrtc.org
bc934ccc4e Temporarily disable version.py.
TBR=leozwang@webrtc.org
BUG=issue687
TEST=build on Mac

Review URL: https://webrtc-codereview.appspot.com/706004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2527 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-25 15:45:33 +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
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