Commit Graph

1992 Commits

Author SHA1 Message Date
leozwang@webrtc.org
a11299648c Retrieve data from input
Espeically on tablet, we have to read data dirtectly from input text edit rather than
track key input to let text edit get updated automatically

BUT=None
TEST=local test
Review URL: https://webrtc-codereview.appspot.com/705010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2599 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-11 04:33:02 +00:00
andrew@webrtc.org
030d752dce Removing unneeded baseclass headers.
TBR=wu@webrtc.org
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/714009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2598 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-11 03:36:24 +00:00
kma@webrtc.org
de91bf7cdc Changed compiler assert code so that it can be used by both C and C++ source files.
Review URL: https://webrtc-codereview.appspot.com/733005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2597 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-11 00:13:25 +00:00
andrew@webrtc.org
cdfa63f94f Fix mismatched signature (due to const) error.
TBR=mikhal@webrtc.org
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/717013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2596 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-10 22:56:17 +00:00
henrike@webrtc.org
7742479428 Fixes build bot breakage. Resizing was enabled which some tests assumed wouldn't be the case. Changed the default so that it is now disabled.
Review URL: https://webrtc-codereview.appspot.com/731006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2595 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-10 19:31:24 +00:00
marpan@webrtc.org
b18261ce5f Set error concealment off for a test in videoprocessor_integrationtest.
Review URL: https://webrtc-codereview.appspot.com/714008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2594 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-10 17:42:16 +00:00
henrike@webrtc.org
268a24fa56 Reverts changes to auto test.
Review URL: https://webrtc-codereview.appspot.com/724006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2593 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-10 15:15:51 +00:00
mflodman@webrtc.org
ba853c9970 Fixes for RTP extension time offset.
Review URL: https://webrtc-codereview.appspot.com/720008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2592 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-10 14:30:53 +00:00
astor@webrtc.org
c0496e66f6 Expose a function for setting bandwidth estimation parameters in ViERTP_RTCP.
Review URL: https://webrtc-codereview.appspot.com/678007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2591 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-10 10:14:43 +00:00
henrike@webrtc.org
3c286747ce Makes it possible to disable automatic resizing.
Review URL: https://webrtc-codereview.appspot.com/710010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2589 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-10 02:44:30 +00:00
andrew@webrtc.org
54da26d2d2 Move include_tests to a higher variable nesting.
TBR=wu@webrtc.org
BUG=none
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/734004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2587 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-10 00:41:18 +00:00
leozwang@webrtc.org
b20916e336 Change libaries path because of recent file structure changes
Description:
1. Changed file path.
2. Because of optimization code changes, a new neon library is created, add it to finial build.

BUG=None
TEST=local build
Review URL: https://webrtc-codereview.appspot.com/731005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2586 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-09 22:17:26 +00:00
andrew@webrtc.org
6a16e74f9f Move iSAC -> isac.
BUG=none
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/722007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2584 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-09 18:35:39 +00:00
andrew@webrtc.org
cb53410877 Make some dependencies more flexible.
BUG=none
TEST=trybot

Review URL: https://webrtc-codereview.appspot.com/728005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2583 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-09 18:28:40 +00:00
phoglund@webrtc.org
48cf686933 Removed v4l2_file_player code, which is checked into the signal repo.
BUG=

Review URL: https://webrtc-codereview.appspot.com/718007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2581 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-09 14:57:58 +00:00
kma@webrtc.org
ccb7cc6175 Added unittest for iSAC-Fix.
Only one function in this version.
Review URL: https://webrtc-codereview.appspot.com/713006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2575 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-08 18:26:19 +00:00
perkj@webrtc.org
c2fde804fa Fix VideoCaptureModule and unit test valgrind errors on linux.
BUG= 302

Review URL: https://webrtc-codereview.appspot.com/717008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2574 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-08 14:01:09 +00:00
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
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