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Author SHA1 Message Date
pwestin@webrtc.org
ebcb6421b1 Cleaned up leaky symbols in G722. 
Review URL: http://webrtc-codereview.appspot.com/333017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1287 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 12:20:06 +00:00
pwestin@webrtc.org
d8f8b32521 Cleaned up leaky symbols in iSAC. 
Review URL: http://webrtc-codereview.appspot.com/329014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1286 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 12:19:43 +00:00
stefan@webrtc.org
2ae4c8cf44 Disable temporal toggling by default. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/329012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1285 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 10:30:42 +00:00
mflodman@webrtc.org
84dc3d134d Add REMB functionality to ViE. 
This CL only adds support for encoding one stream, but receiving multiple streams.

BUG=
TEST=video_engine_core_unittest + auto_test/loopback

Review URL: http://webrtc-codereview.appspot.com/333011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 10:26:13 +00:00
pwestin@webrtc.org
093ffad26b Removed unused function messing up the symbols. 
Review URL: http://webrtc-codereview.appspot.com/336006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 09:48:48 +00:00
pwestin@webrtc.org
43761beb47 Bugfix get thread ID for linux. 
Review URL: http://webrtc-codereview.appspot.com/331015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1282 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 09:40:28 +00:00
mflodman@webrtc.org
a4863dbdf0 Moved video_engine/main/interface to video_engine/include. 
Only changed include paths in files, gyp-files and Android.mk.

TEST=vie_auto_test and peerconnection_client builds.

Review URL: http://webrtc-codereview.appspot.com/330017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 08:51:52 +00:00
henrik.lundin@webrtc.org
1e28d3c2e1 Change VP8 packetizer to use a single max payload size 
The packetizer class is changed so that the max payload size is
provided on construction of the class rather than for each packet.
The tests are re-written to comply with the new design.

Also fixing a few errors in the tests.

Review URL: http://webrtc-codereview.appspot.com/335010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 08:49:31 +00:00
stefan@webrtc.org
f5edb923b1 Remove unused variable. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/333016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1279 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 08:34:31 +00:00
tina.legrand@webrtc.org
5c43b1b861 Updated resampler unit test with stereo. 
I needed to run valgrind on this particular test, to exclude from valgrind warnings in ACM. Test passes valgrind without problems.
Review URL: http://webrtc-codereview.appspot.com/332010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1278 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 08:28:05 +00:00
pwestin@webrtc.org
8edb39db30 Prevent sending empty RTCP packet. 
Review URL: http://webrtc-codereview.appspot.com/331009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 07:40:33 +00:00
henrik.lundin@webrtc.org
4a19030131 New VCM robustness API 
This CL defines and starts to implement a new robustness API for
video coding module. The API is partly implemented. Some of the
modes and methods are still TBD.

Also including a new unittest with mocking of decoder and callbacks,
and faking of system clock.

Review URL: http://webrtc-codereview.appspot.com/333006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1276 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 20:38:37 +00:00
andrew@webrtc.org
697bc43b67 Restore item deletions in Windows UDP. 
TEST=voe_auto_test on Windows.

Review URL: http://webrtc-codereview.appspot.com/331013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1275 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 19:58:11 +00:00
andrew@webrtc.org
71571c5446 Remove unneeded variables from windows UDP. 
TEST=build on Windows.

Review URL: http://webrtc-codereview.appspot.com/329013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1274 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 18:30:59 +00:00
andrew@webrtc.org
3192d655bd Fix for devices lacking stereo support. 
The number of capture channels can only be determined upon receiving the
first captured frame. We now assume stereo capture by default and set the
number of AudioProcessing input channels based on captured frames.

TEST=Windows mono-only device now runs AudioProcessing correctly (NS etc.), voe_auto_test (though some new, seemingly unrelated, tests are failing)

Review URL: http://webrtc-codereview.appspot.com/330013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1273 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 18:00:59 +00:00
andrew@webrtc.org
003044a6df Enable warnings-as-errors on Mac. 
TEST=build on Mac (make and XCode)

Review URL: http://webrtc-codereview.appspot.com/335007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1272 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 16:36:56 +00:00
kjellander@webrtc.org
173b7bbc16 Integration test that tracks dropped frames and compares video output. 
The recorded frame timestamps are used to modify the output video on a frame-per-frame so it can be compared with the reference video using PSNR. This code will make it possible to use vie_auto_test for full stack comparisons with network interference and similar interesting simulations.

There's some refactoring done in vie_comparison_test.cc to make it fit to the new test.

Compiled and executed in Debug+Release on Linux, Mac and Windows.

BUG=
TEST=vie_auto_test --automated --gtest_filter=ViEVideoVerificationTest.*

Review URL: http://webrtc-codereview.appspot.com/320002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1269 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 16:11:25 +00:00
mallinath@webrtc.org
03532b5f41 Fixing the double delete problem in UdpSocket2ManagerWindow. PopFront deletes the items, to there is no need to delete item explicitly. 
Review URL: http://webrtc-codereview.appspot.com/333014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1268 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 15:36:44 +00:00
henrik.lundin@webrtc.org
7d8c72e2db Re-implement dependency injection of TickTime into VCM and tests 
This change basicly re-enables the change of r1220, which was
reverted in r1235 due to Clang issues.

The difference from r1220 is that the TickTimeInterface was
renamed to TickTimeClass, and no longer inherits from TickTime.

Review URL: http://webrtc-codereview.appspot.com/335006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1267 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 15:24:01 +00:00
kjellander@webrtc.org
5490c71a1b Converted to gtest, writing output files properly and no longer uses exceptions. 
This test now runs and fails as a gtest should (previously it always
exited with 0 even if the tests failed).
The audio_coding_module_test target no longer uses exceptions in the generated project.
Output files are written to our global output folder, using
testsupport/fileutils.h.

BUG=
TEST=audio_coding_module_test on all platforms, in Debug+Release

Review URL: http://webrtc-codereview.appspot.com/334004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1266 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 13:34:18 +00:00
mflodman@webrtc.org
1fe2ada38d Fixed Win bug introduced when refactoring ViECodecImpl. 
TBR=perkj

Review URL: http://webrtc-codereview.appspot.com/328013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1264 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 12:23:15 +00:00
mflodman@webrtc.org
c12686c2d0 Refactored ViEEncryptionImpl, ViECodecImpl and removed unused SRTP hooks/APIs in ViEEncrption. 
Review URL: http://webrtc-codereview.appspot.com/331004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1262 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 09:29:28 +00:00
stefan@webrtc.org
898f881e32 Make sure the next frame to be decoded is cleaned up if it's empty. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/332001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1261 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 09:13:14 +00:00
niklas.enbom@webrtc.org
6c9be123ef Letting strncpy do its job. Landing and extending http://webrtc-codereview.appspot.com/329010/ on behalf of tbreisacher. 
Review URL: http://webrtc-codereview.appspot.com/335009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 08:59:31 +00:00
stefan@webrtc.org
8c5d24266e Fix VP8 layer 2 sync dependencies. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/333010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1259 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 08:56:04 +00:00
henrik.lundin@webrtc.org
00e730730e Refactoring RtpFormatVp8Test 
This is the first change in a series of changes to get new functionality
into the VP8 packetizer.

This first refactors the RtpFormatVp8Test class, without changing the
operation of the tested RtpFormatVp8 class. A test helper class
RtpFormatVp8TestHelper is introduced to reduce code duplication.

Review URL: http://webrtc-codereview.appspot.com/304009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1258 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 08:51:36 +00:00
niklas.enbom@webrtc.org
b2c115c460 Forcing external transport to be on in Chrome. 
Review URL: http://webrtc-codereview.appspot.com/330010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1257 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 07:38:54 +00:00
mikhal@webrtc.org
61045a4a03 video_coding/jitter_buffer: Account for layer info when searching for the next frame 
Review URL: http://webrtc-codereview.appspot.com/328003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1256 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 23:24:19 +00:00
andrew@webrtc.org
a38ce09919 Fix last Mac/clang compile error. 
Fixes "receiver is a forward class and corresponding @interface may
not exist" error.

TEST=build on Mac with -Werror enabled.
TBR=zakkhoyt@webrtc.org

Review URL: http://webrtc-codereview.appspot.com/333012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1255 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 22:23:21 +00:00
andrew@webrtc.org
e858d13ac6 Add a NOOP target for merge libs. 
Also allow certain components to not be built.

TEST=build merged_lib

Review URL: http://webrtc-codereview.appspot.com/328001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1254 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 22:07:48 +00:00
mikhal@webrtc.org
6f7fbc7fbe libyuv: Adding psnr/ssim to libyuv and updating unit tests according to latest conventions. 
Review URL: http://webrtc-codereview.appspot.com/331007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1253 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 17:38:28 +00:00
pwestin@webrtc.org
061fa5b828 Changed handling of padding data. 
Review URL: http://webrtc-codereview.appspot.com/331008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1252 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 15:56:17 +00:00
henrik.lundin@webrtc.org
dbba1f969f Packet waiting-time statistics 
Adding new statistics API to NetEQ, reporting the waiting time
for each frame. The output is raw waiting time for the frames
that have been decoded since the last statistics report (or
maximum 100 frames). The statistics are reset on each query.

Implemented functionality in ACM to query NetEQ for the raw
waiting times, and process it to produce max, average and
median.

Updating common_types.h and VoiceEngine tests to include the
new metrics.

Unit tests are also added for NetEQ and AcmNetEq.

Review URL: http://webrtc-codereview.appspot.com/328011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 15:45:05 +00:00
henrik.lundin@webrtc.org
219acc6cec Including Brighten function in namespace VideoProcessing 
This change is in response to Issue 173.

BUG=http://code.google.com/p/webrtc/issues/detail?id=173

Review URL: http://webrtc-codereview.appspot.com/328012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 15:33:49 +00:00
bjornv@webrtc.org
c68f80a70a Refactoring vad_gmm.[c/h]. 
- Changed to stdint.
- Replaced SHIFT macros.
- Variable name changes.
- Style changes.
- Comments updates.
- Added a unit test.
Review URL: http://webrtc-codereview.appspot.com/323011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1249 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 14:08:34 +00:00
mflodman@webrtc.org
42d07f0c58 Render impl fix from refactoring. 
Review URL: http://webrtc-codereview.appspot.com/329009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1248 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 13:20:36 +00:00
mflodman@webrtc.org
1bdf1dffb4 Refactored ViEImageProcess, ViEImpl and ViENetworkImpl. 
Review URL: http://webrtc-codereview.appspot.com/331005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1247 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 11:57:47 +00:00
mflodman@webrtc.org
813b4ef2ea Refactored ViEFileImpl and ViEExternalCodec. 
Review URL: http://webrtc-codereview.appspot.com/330007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1246 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 10:39:30 +00:00
phoglund@webrtc.org
f3cea2336b Added an empty voice engine unit test binary in order to get correct coverage measurements. This will make the voice engine show up in the coverage measurements. The empty test is necessary to get the coverage tool to pick it up (and it will be easier to start writing unit tests for the voice engine later). 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/334003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1245 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 10:14:53 +00:00
stefan@webrtc.org
62fdc42e9c Fix build issue with clang. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/330009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1244 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 09:27:09 +00:00
stefan@webrtc.org
8dc9e4760e Fixes for selective NACKing. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/332007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1243 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 09:12:50 +00:00
phoglund@webrtc.org
fda17c2b00 Rewrote NetEQ test, made standard suite run googletestified tests too. 
The standard suite will now also run the googletestified tests.

Removed NetEQ tests from the standard test.

Initial version of new neteq test. Moved fixtures to own folder.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/328010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1242 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 09:07:37 +00:00
tina.legrand@webrtc.org
5efcad1758 We used the wrong syntax for "new", which generated a warning/error building with clang. 
Review URL: http://webrtc-codereview.appspot.com/336003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1241 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 09:05:55 +00:00
mikhal@webrtc.org
9e4f3830a7 Removing vplib: Following the switch to Libyuv, this CL removes all vplib files. 
Review URL: http://webrtc-codereview.appspot.com/321003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1239 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 19:24:32 +00:00
mikhal@webrtc.org
0e7d9d862a Adding layer info consideration when applying FEC protection. In this first version, we hard code protection zero for non-base layer frames. As a future enhancement, an array should be passed from mediaOpt to set the protection per layer. A TODO was added in MediaOpt. 
Review URL: http://webrtc-codereview.appspot.com/330005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1238 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 19:04:49 +00:00
mikhal@webrtc.org
190e88a6d3 video_coding: When in hybrid mode, don't NACK non-base layer packets 
Review URL: http://webrtc-codereview.appspot.com/334002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1237 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 18:57:14 +00:00
mikhal@webrtc.org
884d8e7f4b video_coding: Updating sync state based on the layer flag 
Review URL: http://webrtc-codereview.appspot.com/333004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1236 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 18:53:05 +00:00
henrik.lundin@webrtc.org
303158588b Revert "Inject TickTimeInterface into VCM and tests" 
This CL reverts r1220.

Review URL: http://webrtc-codereview.appspot.com/336002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1235 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 17:55:45 +00:00
henrika@webrtc.org
e32c08a5a6 Removes usage of default parameters and fixes a bug which was found 
using Clang on Linux.

BUG=none
TEST=none
TBR=pwestin

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 17:39:48 +00:00
henrike@webrtc.org
4158c35820 Removed the WEBRTC_NO_TRACE macro since the style guide wants us to stear clear of macros and this one doesn't seem to have a purpose at this point. 
Review URL: http://webrtc-codereview.appspot.com/315006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1233 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 17:28:25 +00:00