Commit Graph

1416 Commits

Author SHA1 Message Date
punyabrata@webrtc.org
8fa31bc4e5 Truncated messages, need a %S instead of $s for a double byte TCHAR
Review URL: http://webrtc-codereview.appspot.com/333002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1321 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 22:34:15 +00:00
mflodman@webrtc.org
adec9271b0 Correcting VieChannelManager bug.
Review URL: http://webrtc-codereview.appspot.com/337010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1320 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 21:43:15 +00:00
amyfong@webrtc.org
de5a10a044 Added in setting the minimum bit rate of a codec to ViE Custom Call
Review URL: http://webrtc-codereview.appspot.com/333019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1319 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 21:06:26 +00:00
mikhal@webrtc.org
77c425b976 video_coding: Checking/updating seq num for an old packet regardless of size.
Review URL: http://webrtc-codereview.appspot.com/330028

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1318 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 20:35:25 +00:00
mikhal@webrtc.org
c00f91d62d Adding BGRA as a video type.
This CL is a prerequisite for the capture module update CL. 
Review URL: http://webrtc-codereview.appspot.com/329021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1317 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 18:49:15 +00:00
andrew@webrtc.org
877c54e230 Fix unused-variable warning in Release.
TBR=mflodman@webrtc.org
TEST=Build Debug/Release on Linux

Review URL: http://webrtc-codereview.appspot.com/338003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1316 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 18:32:01 +00:00
bjornv@webrtc.org
f175125e96 Refactoring vad_filterbank: Style changes.
Includes:
- Correct header guard
- Indentations and white spaces
- Changed to stdint
Review URL: http://webrtc-codereview.appspot.com/330030

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1315 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 15:07:42 +00:00
mflodman@webrtc.org
9c0aedc28b Removed constraint for changing resolution when using default encoder and added VP8 log.
Review URL: http://webrtc-codereview.appspot.com/330029

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1314 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 13:46:49 +00:00
henrik.lundin@webrtc.org
6c877363f7 Fix formatting for some NetEQ test tools
Format and lint for RTPchange.cc, RTPcat.cc and RTPanalyze.cc.

Review URL: http://webrtc-codereview.appspot.com/329024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1313 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 10:03:19 +00:00
perkj@webrtc.org
60c9bbd976 Fix GetReceivedRTCPStatistics and GetSendRTCPStatistics.
Comments where wrong and removed error message when trying to get RTT time from GetReceivedRTCPStatistics.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/335013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1312 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 09:54:29 +00:00
mflodman@webrtc.org
d5a4d9bce6 First refactoring of ViE interface.
Review URL: http://webrtc-codereview.appspot.com/337003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1311 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-02 13:04:05 +00:00
kjellander@webrtc.org
a643d5c4ef Integration test for videoprocessor
Added temporal layers number flag for video_quality_measurement tool.
This tool now also uses webrtc::VideoCodingModule::Codec() to get its
VideoCodec struct configuration instead of filling it in manually.

Updated paths for header files to use full directory paths.

Tested in Debug+Release on Linux, Mac and Windows. Passes Valgrind memcheck on Linux.

BUG=
TEST=video_codecs_test_framework_integrationtests. Also executed out/Debug/video_quality_measurement --input_filename=resources/foreman_cif.yuv  --width=352 --height=288

Review URL: http://webrtc-codereview.appspot.com/339001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1310 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-30 14:44:07 +00:00
mikhal@webrtc.org
62665b8cd3 video_coding: Adding a unit test to the decodableState class
Review URL: http://webrtc-codereview.appspot.com/315001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1309 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-29 18:09:58 +00:00
mikhal@webrtc.org
9eeafbef3c Updating the frame buffer return value in InsertPacket: Return NoError when a packet is inserted to a frame which is being decoded.
Review URL: http://webrtc-codereview.appspot.com/330027

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1308 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-29 17:38:56 +00:00
mikhal@webrtc.org
bed34a341a video_coding: Updating seq number for old zero size packets. Updating function name to reflect zero size packets and not empty packets.
Review URL: http://webrtc-codereview.appspot.com/333009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1307 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-29 17:37:39 +00:00
bjornv@webrtc.org
70adcd46b2 Delay estimator improvements.
Robustness improvements to the delay estimator used in AECM and AEC. In AEC only for logging. Faster convergence.

TEST=audioproc_unittest + offline file tests.

output_data_fixed.pb updated despite unverified changes in r1112.
Review URL: http://webrtc-codereview.appspot.com/337006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1306 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-29 14:51:21 +00:00
stefan@webrtc.org
efd0a48c61 Add error resilient mode options to the VP8 specific VideoCodec struct.
It is useful to disable error resilience when we know we won't decode
with errors.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/329015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1305 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-29 10:12:35 +00:00
mikhal@webrtc.org
67f294a48a Adding a return value to ConvertRotationMode
Review URL: http://webrtc-codereview.appspot.com/333023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1304 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-28 23:28:45 +00:00
andrew@webrtc.org
6d6a43d6e3 Use char as ring-buffer data type.
- Avoids a bunch of char* casts.
- Use enum type rather than char.

TEST=audioproc_unittest on Linux (float and fixed), build on Windows

Review URL: http://webrtc-codereview.appspot.com/336010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1303 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-28 22:40:15 +00:00
mikhal@webrtc.org
e2642494e4 libyuv: Updating API to use latest ConvertFrom/To functionality
Review URL: http://webrtc-codereview.appspot.com/333020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1302 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-28 21:21:40 +00:00
bjornv@webrtc.org
267d0133ff Fixed pointer operations on void.
This should fix the error on Win where pointer arithmetics are done on void pointers. Type cast to char to interpret a size.
Review URL: http://webrtc-codereview.appspot.com/329019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1300 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-28 10:26:17 +00:00
bjornv@webrtc.org
7270a6bcc2 Merged apm-buffer branch [r1293] back to trunk.
This CL includes a larger structural change in how we handle buffers in AEC. We now perform FFT at once and move within blocks to compensate for system delays.

TEST=audioproc_unittest(float and fix), voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/335012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1299 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-28 08:44:17 +00:00
mikhal@webrtc.org
e39de16fa5 Moving video type convert functionality to libyuv. deleting vplibConversions as it is no longer needed.
Review URL: http://webrtc-codereview.appspot.com/338002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1298 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-27 23:45:30 +00:00
stefan@webrtc.org
f6c6b1c5b5 Include the media packet FEC headers in the video bitrate.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/328014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1296 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-23 10:33:39 +00:00
stefan@webrtc.org
39670f6aa6 Only reset the last decoded sequence number after flushing until key frame.
We can't reset the complete last decoded state when we recycle until a
key frame because that will allow any delta frame to be decoded afterwards,
and since the decoder isn't reset we will get decode errors.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/330003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1295 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-23 09:08:51 +00:00
mflodman@webrtc.org
1ce66e4dfb Don't report error when failing to send RTCP BYE.
Review URL: http://webrtc-codereview.appspot.com/337002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1293 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 18:40:15 +00:00
amyfong@webrtc.org
ee2924cc56 Added vp8 codec temporal layer changing option to ViE AutoTest custom call.
Review URL: http://webrtc-codereview.appspot.com/330018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1292 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 18:24:28 +00:00
mflodman@webrtc.org
d32c44738a Changed constructor used for CriticalSectionScoped in ViE.
Only changed:
- Name of some of the critsects.
- All critsects (but one) are now scoped_ptr.
- Use of ptr constructor of CriticalSectionScoped instead of reference version.

BUG=184
TEST=vie_auto_test

Review URL: http://webrtc-codereview.appspot.com/330015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1291 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 14:17:53 +00:00
stefan@webrtc.org
6a4bef4e65 Implements selective retransmissions.
Default is set to not retransmit VP8 non-base layer packets or FEC packets.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/323010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1290 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 12:52:41 +00:00
mflodman@webrtc.org
51faeed6be Fixed REMB unit test on Windows.
TBR=pwestin

Review URL: http://webrtc-codereview.appspot.com/330022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1289 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 12:35:23 +00:00
pwestin@webrtc.org
f4d3b9d5a1 Cleaned up leaky symbols in NS.
Review URL: http://webrtc-codereview.appspot.com/337001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1288 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 12:33:08 +00:00
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
mflodman@webrtc.org
8da2417c9d Refactored ViERenderImpl and ViERTP_RTCPImpl.
Review URL: http://webrtc-codereview.appspot.com/329005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1232 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-19 14:18:41 +00:00
mflodman@webrtc.org
7752d11056 Fix test for external codec.
Review URL: http://webrtc-codereview.appspot.com/328007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1231 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-19 12:29:56 +00:00
phoglund@webrtc.org
86a9f9b946 Fixed build error.
Merge branch 'master' into voe_rewrites

Conflicts:
	src/voice_engine/main/test/auto_test/standard/after_streaming_fixture.cc

Revert "Rewrote network-before-streaming."

This reverts commit f1a07b813a90e4feef0a0737ebde9fb15acfd459.

Rewrote network-before-streaming.

Fixed strange build error.

Merge branch 'master' into voe_rewrites

Revert "Rewrote network-before-streaming."

This reverts commit f1a07b813a90e4feef0a0737ebde9fb15acfd459.

Rewrote network-before-streaming.

Nit fixes

Clarified some comments and method names.

Style fixes.

Removed tab characters.

Merge branch 'master' into voe_rewrites

Conflicts:
	src/voice_engine/main/test/auto_test/voe_standard_test.cc

Revert "Rewrote network-before-streaming."

This reverts commit f1a07b813a90e4feef0a0737ebde9fb15acfd459.

Rewrote network-before-streaming.

Rewrote the hold test.

Abstracted out resource handling and created a new fixture for starting and stopping playing.

Rewrote network-before-streaming.

Revert "Rewrote network-before-streaming."

This reverts commit f1a07b813a90e4feef0a0737ebde9fb15acfd459.

Rewrote network-before-streaming.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/333007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1230 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-19 10:24:46 +00:00
stefan@webrtc.org
b33f9dccd6 Correction to how the VP8 wrapper generates picture ids.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/329006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1229 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-19 10:24:02 +00:00
phoglund@webrtc.org
188fc35e07 Rewrote the hold and netw-before-streaming tests.
Rewrote the hold test.

Abstracted out resource handling and created a new fixture for starting and stopping playing.

Rewrote network-before-streaming.

BUG=
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/331001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1228 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-19 09:36:03 +00:00
tina.legrand@webrtc.org
398af2337b Solving issue 178, errorbuild warnings on Mac.
This CL continues the work of solving issue 178. A small change in one file.
Review URL: http://webrtc-codereview.appspot.com/330006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1227 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-19 07:36:07 +00:00
henrike@webrtc.org
cf5bcd1fd2 Removed usage of the deprecated critical section constructor in audio_conference_mixer.
Review URL: http://webrtc-codereview.appspot.com/320007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1226 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 23:00:30 +00:00
andrew@webrtc.org
8a44259ea8 Move static consts out of class.
Still causing a gtest error on non-Win platforms. This should fix it...

TBR=asapersson@webrtc.org
TEST=build on Linux

Review URL: http://webrtc-codereview.appspot.com/332006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1225 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 21:24:30 +00:00
andrew@webrtc.org
41192469f6 Switch enums to consts to fix gtest error.
TBR=asapersson@webrtc.org
TEST=build on Windows

Review URL: http://webrtc-codereview.appspot.com/330008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1224 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 20:55:46 +00:00
henrike@webrtc.org
105e07193e Removed usage of the deprecated critical section constructor in modules/utility.
Review URL: http://webrtc-codereview.appspot.com/321006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1223 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 19:53:46 +00:00
marpan@webrtc.org
57353a33f1 FEC Receiver: Fix to how old packets (e.g., re-tranmitted packets in hybird NACK-FEC mode) are treated.
This change avoids having old packets end up on the current packet list for FEC decoding, and so they are immediately sent out to jitter buffer.
The current list of packets for FEC decoding are sent out only when new packet arrives (with time-stamp greater than current).
Review URL: http://webrtc-codereview.appspot.com/322009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1222 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 17:21:09 +00:00
henrik.lundin@webrtc.org
e7d8c56c56 Fix for dual decoder in VCM receiver
In VCMReceiver::FrameForDecoding, one of the if-cases could sometimes
extract an incomplete frame without first copying the state to the
dual decoder.

Review URL: http://webrtc-codereview.appspot.com/328006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1221 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 15:40:52 +00:00
henrik.lundin@webrtc.org
a70f945086 Inject TickTimeInterface into VCM and tests
The purpose of this change is to introduce dependency injection
of the timer into the video coding module.

Review URL: http://webrtc-codereview.appspot.com/332003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1220 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 14:40:05 +00:00
asapersson@webrtc.org
5249cc8f77 Review URL: http://webrtc-codereview.appspot.com/295010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1219 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 14:31:37 +00:00
tina.legrand@webrtc.org
9775a30859 Added variable to catch return value.
Review URL: http://webrtc-codereview.appspot.com/329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1218 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 11:15:46 +00:00
kjellander@webrtc.org
08dec7f449 Now using fileutils.h OutputPath to write output to the right directory and ResourcePath to read resource files from the resource bundle.
Removed some Valgrind warnings by closing output files. There are still some Valgrind warnings left, that needs to be fixed by a developer with more insight.

Updated all include paths to contain full paths to header files.

Tested in Debug+Release on Linux, Mac and Windows.
All tests ran successfully except the VideoProcessingModuleTest.ContentAnalysis test that fails on Windows with the following error:
unknown file: error: SEH exception with code 0xc0000005
thrown in the test body.
Fixing that is out of scope for this CL.

Review URL: http://webrtc-codereview.appspot.com/266011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1217 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 10:31:38 +00:00
tina.legrand@webrtc.org
554ae1ad4e Changes to solve warnings on Mac, issue #178.
Review URL: http://webrtc-codereview.appspot.com/320005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1216 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 10:09:04 +00:00
mflodman@webrtc.org
605972edfd Refactored ViECaptureImpl.
Review URL: http://webrtc-codereview.appspot.com/332004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1215 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 08:59:24 +00:00
mflodman@webrtc.org
352dcd8b2d Refactored vie_file_image.
Review URL: http://webrtc-codereview.appspot.com/332002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1214 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 08:52:41 +00:00
andrew@webrtc.org
04f5cba069 Switch to new critsect interface for DataLog.
The introduction of the new interface broke DataLog in a release build
(with enable_data_logging=1).

TBR=henrike@webrtc.org
TEST=build Linux/Release with enable_data_logging=1

Review URL: http://webrtc-codereview.appspot.com/334001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1212 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 21:33:11 +00:00
henrike@webrtc.org
7136990a3f Removed usage of the deprecated critical section constructor in udp_transport.
Review URL: http://webrtc-codereview.appspot.com/321005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1211 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 19:17:28 +00:00
andrew@webrtc.org
986fab1496 Clean up file wrapper a bit further.
- Make error handling in Read, Write and WriteText consistent.
- Improve the interface comments a bit.

TEST=voe_auto_test, vie_auto_test

Review URL: http://webrtc-codereview.appspot.com/321012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1210 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 19:11:41 +00:00
leozwang@webrtc.org
0c839fe873 Add new source file to makefile
Review URL: http://webrtc-codereview.appspot.com/322015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1209 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 19:10:24 +00:00
henrike@webrtc.org
bfa80ce95e Removed usage of the deprecated critical section constructor in system_wrappers.
Review URL: http://webrtc-codereview.appspot.com/322004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1208 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 17:59:58 +00:00
henrik.lundin@webrtc.org
0a10e3c4b2 Fix order of include and guard in tick_time_interface.h
Review URL: http://webrtc-codereview.appspot.com/331002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1207 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 16:08:36 +00:00
mflodman@webrtc.org
091029ba26 Refactored ViEFileRecorder.
Types and arguments will be done in a  later CL.

Review URL: http://webrtc-codereview.appspot.com/317008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1206 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 15:31:47 +00:00
mflodman@webrtc.org
03c06505fb Refactored ViEChannel.
Pointers/references and types will be in a future CL.

Review URL: http://webrtc-codereview.appspot.com/322016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1205 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 15:23:08 +00:00
henrik.lundin@webrtc.org
c74b2861f3 Fix the include in fake_tick_timer_interface.h
The include was in error.

Review URL: http://webrtc-codereview.appspot.com/330002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1204 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 11:28:31 +00:00
phoglund@webrtc.org
610e90e910 Completed rewrite of codec test.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/324011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1203 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 10:40:19 +00:00
mflodman@webrtc.org
e8be22c192 Refactored ViEChannelManager ViEInputManager.
Pointers/references and types will come in a future CL.

Review URL: http://webrtc-codereview.appspot.com/317012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1202 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 10:19:29 +00:00
leozwang@webrtc.org
e0e07bbaa0 Change file name because of r1199
Review URL: http://webrtc-codereview.appspot.com/320013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1201 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 00:51:45 +00:00
kma@webrtc.org
ee36b9587d corrected android makefile for isac build.
Review URL: http://webrtc-codereview.appspot.com/321013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1200 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 00:18:45 +00:00
andrew@webrtc.org
59ccd5c71f Rename _windows.h -> _win.h in system_wrappers.
- Also rename _dummy -> no_op which states its purpose more clearly.
- Always use exclusion lists (i.e. sources! instead of sources)

TEST=builds and passes system_wrapper_unittest on Linux, Mac, Win

Review URL: http://webrtc-codereview.appspot.com/317007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1199 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 00:17:43 +00:00
kma@webrtc.org
6a17340db5 Review URL: http://webrtc-codereview.appspot.com/318014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1197 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 22:16:57 +00:00
leozwang@webrtc.org
5fddbeb7e5 Build libyuv for webrtc
Review URL: http://webrtc-codereview.appspot.com/322012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1196 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 20:03:26 +00:00
leozwang@webrtc.org
eda2da796e Fix compilation errors
Review URL: http://webrtc-codereview.appspot.com/322014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1195 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 20:03:09 +00:00
kma@webrtc.org
a30093bb85 Added one file associated with check in in r1192.
Review URL: http://webrtc-codereview.appspot.com/320012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1194 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 19:57:23 +00:00
leozwang@webrtc.org
9aa9f44ebc Add new source files because of r1174
Review URL: http://webrtc-codereview.appspot.com/320011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1193 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 19:08:33 +00:00
kma@webrtc.org
f0a964dc0a Optimized WebRtcIsacfix_NormLatticeFilterMa() function for iSAC fix for ARM Neon
architecture with intrinsics and assembly code. The total iSAC codec speech improved
about 3~5%.

Notes
(1) The Neon version after this optimization is not bit-exact with the generic
C version. The out quality, however, is not worse as verified by test vectors ouput,
and undertandably in theory (32bit x 32bit in Neon is more accurate than the approximation
C code in the generic version).
(2) In Android, a isac neon library will be built. Along with some new function structures,
it is partly for preparation of introducing a run time detection of Neon architecture soon.
Review URL: http://webrtc-codereview.appspot.com/268016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1192 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 18:59:43 +00:00
mflodman@webrtc.org
02afbeaca5 Refactored ViERenderManager.
Will follow up with a new CL for pointer/references and functino arguments.

Review URL: http://webrtc-codereview.appspot.com/323013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1191 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 18:50:47 +00:00
kma@webrtc.org
6601902504 Introduced WebRtcSpl_SatW32ToW16 to iSAC fix, for Android platforms.
Review URL: http://webrtc-codereview.appspot.com/315005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1190 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 18:41:07 +00:00
leozwang@webrtc.org
f147bbc878 Change codec test app lib dependency from webrtc lib to codec library
Review URL: http://webrtc-codereview.appspot.com/317009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1189 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 18:22:41 +00:00
andrew@webrtc.org
7e5ddf5aa3 Restore behavior to FileWrapper::Read.
- Returning the number of bytes read was mistakenly removed in r1175 in
  an overzealous attempt to unify the interface.
- Now both Read and WriteText return the number of bytes/characters
  processed. Write unfortunately cannot be easily changed due to the
  inheritance from OutStream.
- Improve the interface comments.

TBR=henrika@webrtc.org
BUG=issue196, issue198
TEST=voe_auto_test passes at last...

Review URL: http://webrtc-codereview.appspot.com/326001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1188 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 18:02:02 +00:00
henrike@webrtc.org
7cdcde3460 Removed usage of the deprecated critical section constructor in media_file.
Review URL: http://webrtc-codereview.appspot.com/321004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1187 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 17:27:58 +00:00
stefan@webrtc.org
780a07a843 Fix infinite loop bug introduced in r1174.
Merges CleanUpSizeZeroFrames with CleanUpOldFrames, and changes the
behavior to go through all frames looking for empty frames.

TBR=mikhals

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/318013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1186 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 15:55:19 +00:00
pwestin@webrtc.org
9fe3d51372 Set the new layer sync bit in the VP8 info struct.
Review URL: http://webrtc-codereview.appspot.com/324010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1185 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 15:13:04 +00:00
phoglund@webrtc.org
667eca6290 Rewrote the hardware-before-streaming test.
Restructured the test hierarchy somewhat - there is now a fixture for before-voe-init time and one for after-voe-init time.

Rewrote the hardware-before-streaming test.
Separated unrelated tests out from the rtp_rtcp tests.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/323009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1184 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 13:55:34 +00:00
henrik.lundin@webrtc.org
fbf5af444b Adding a mockable wrapper class for TickTime in VCM
The class is called TickTimeInterface, with a fake implementation in FakeTickTime.

Review URL: http://webrtc-codereview.appspot.com/323012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1183 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 10:36:10 +00:00
stefan@webrtc.org
ef5247b5b1 Fix session_info_unittest error.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/324009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1182 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 10:25:38 +00:00
stefan@webrtc.org
0c40d3315f Fixes an assert triggered in jitter_buffer_test and disables deblocking.
When deblocking is enabled the first frames can include uninitialized
memory. Disabling for now.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/320010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1181 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 09:39:30 +00:00
mflodman@webrtc.org
7991c0501f Refactor ViEFilePlayer.
Types and arguments will be done in a  later CL.

Review URL: http://webrtc-codereview.appspot.com/324002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1180 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 08:38:37 +00:00
mflodman@webrtc.org
e6f64835a0 Refactored ViECapturer.
Types and function arguments will come in a later CL.

Review URL: http://webrtc-codereview.appspot.com/322011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1179 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 08:36:01 +00:00
mflodman@webrtc.org
9a8fa4e65d Refactored vie_manager_base.*.
The other files are only due to inheritance and will be refactored later. Same goes for pointer, references and function arguments.

Review URL: http://webrtc-codereview.appspot.com/318003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1178 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 08:18:42 +00:00
andrew@webrtc.org
6d609b59f3 Fix crashes due to static_instance.
- Initialize a needed critsect in the constructor of
  UdpSocket2ManagerWindows.
- Don't return NULL when creating a static instance.

TEST=voe_auto_test on Windows.

Review URL: http://webrtc-codereview.appspot.com/324008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1177 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 02:36:33 +00:00
andrew@webrtc.org
5a9c6f26ab Fix max size and read-only errors in Write().
- A size of zero is now correctly interpreted as unlimited.
- The read-only flag is correctly checked.

TBR=henrika@webrtc.org
TEST=vie_auto_test (for real this time...)

Review URL: http://webrtc-codereview.appspot.com/315007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1176 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 00:53:30 +00:00
andrew@webrtc.org
5ae19de3ec Fix error in RtpDump::Start due to r1156.
- r1156 fixed a check on the _text member of FileWrapper. Turns out this
  was incompatibile with the RTP dumps, which want to write both binary
  and text data. Writing text data to a file open as "b" isn't actually
  an error, so I simply removed the check.
- Also cleans up the interface, most notably removing all WebRtc types.

TEST=vie_auto_test

Review URL: http://webrtc-codereview.appspot.com/317005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1175 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 22:59:33 +00:00
mikhal@webrtc.org
832cacacff video-coding: Adding a decoded state to the JB logic (JB refactor).
This new class stores the last decoded info, including temporal info. 
Review URL: http://webrtc-codereview.appspot.com/300005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1174 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 21:15:05 +00:00
henrike@webrtc.org
65573f2922 Removed usage of the deprecated critical section constructor in rtp_rtcp.
Review URL: http://webrtc-codereview.appspot.com/315004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1173 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 19:17:27 +00:00
stefan@webrtc.org
f4c8286222 Pass NACK and FEC overhead rates through the ProtectionCallback to VCM.
These overhead rates are used by the VCM to compensate the source
coding rate for NACK and FEC.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/323003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1171 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 15:38:14 +00:00
henrik.lundin@webrtc.org
1ced840893 Fixing a nit in the unittest
This caused some of the build bots to fail.

Review URL: http://webrtc-codereview.appspot.com/324005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1170 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 14:59:00 +00:00
henrik.lundin@webrtc.org
eda86dc76b Adding a LayerSync bit to VP8 RTP header
Updated RtpFormatVp8, ModuleRTPUtility, VP8Encoder and VP8Decoder
to support a new LayerSync ("Y") bit. Note, in VP8Encoder the bit
must be used together with a non-negative value for temporalIdx.
Fixing the plumbing between RTP module and and from VP8 wrapper.
Updating unit tests; all pass.

The new bit is yet to be used by the VP8 wrapper.

Review URL: http://webrtc-codereview.appspot.com/323008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1169 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 14:11:06 +00:00
henrik.lundin@webrtc.org
4aae0e489f Shaping up formatting of rtp_utility_test.cc
Preparations for future work in this file.

Review URL: http://webrtc-codereview.appspot.com/318011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1168 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 09:15:21 +00:00
bjornv@webrtc.org
0edb25dcc9 Removed valgrind warnings in resampler_unittest.
Valgrind complained on uninitialized values in resampler_unittest. Added initialization of the member variable data_in_ in the tests.
Review URL: http://webrtc-codereview.appspot.com/322006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1167 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 09:06:54 +00:00
stefan@webrtc.org
076fa6e674 The second step towards a list based SessionInfo.
Added unittests for most of public functions of SessionInfo.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/301014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1166 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 07:54:56 +00:00
wjia@webrtc.org
c28e7980ef exclude trace_windows.cc and trace_posix.cc when building with Chromium.
BUG=none
TEST=compiles
Review URL: http://webrtc-codereview.appspot.com/324004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1165 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 02:16:11 +00:00
mikhal@webrtc.org
71d6391716 libyuv: fixing a bug in RotateI420 and updating test
Review URL: http://webrtc-codereview.appspot.com/324003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1164 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 00:56:11 +00:00
mikhal@webrtc.org
352ade7023 video_coding: Allocating encoded buffer based on length and not size
Review URL: http://webrtc-codereview.appspot.com/318010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1163 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 00:54:28 +00:00
phoglund@webrtc.org
fe61bc3607 Merge branch 'master' into voe_create_test
Fixed broken build.

Nit fix.

Fixed style issues.

Removed accidental comment-out.

Removed test that no longer makes sense.

Rewrote hardware-before-init and rtp-rtcp-before-streaming test code to gtest.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/320009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1162 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-12 17:02:16 +00:00
phoglund@webrtc.org
6418a24795 Rewrote hardware-before-init and rtp-rtcp-before-streaming test code to gtest.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/322003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1161 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-12 16:24:23 +00:00
stefan@webrtc.org
1480f02faf Fix VCM test build warnings on Mac with clang.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/318008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1160 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-12 13:45:59 +00:00
stefan@webrtc.org
7889a9b49a Remove use of CriticalSectionScoped(CriticalSectionWrapper& critsect) in VCM.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/318005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1159 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-12 08:18:24 +00:00
mikhal@webrtc.org
ea71440aec video_coding: Adding the non reference flag to the receive side logic.
Review URL: http://webrtc-codereview.appspot.com/323005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1157 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-12 02:29:34 +00:00
andrew@webrtc.org
114c790be7 Remove character limit in WriteText().
- vfprintf can be used directly here, removing the need for the interim
  buffer. This change allows us to remove the artificial character limit.
- Fix bugs with _text. It wasn't actually getting set earlier, and the
  check was wrong.
- Remove asserts that should use real error checks.

TEST=DataLog and VoECallReport (through voe_auto_test), the only users of WriteText().

Review URL: http://webrtc-codereview.appspot.com/323001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1156 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-10 02:33:33 +00:00
henrike@webrtc.org
2f47b5a70f Fixes a build error when disabling trace (which is done when building with chrome flag is set).
Review URL: http://webrtc-codereview.appspot.com/318006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1155 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-10 00:44:47 +00:00
wjia@webrtc.org
c6b286fc04 add correct include paths for both chrome build and standalone build.
BUG=none
TEST=compiles
Review URL: http://webrtc-codereview.appspot.com/320008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1154 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-10 00:20:54 +00:00
andrew@webrtc.org
bbea716117 Workaround for libyuv libjingle breakage.
libjingle depends on ConvertFromI420. This was previously available
through vplib. libjingle still has access to the vplib header, but the
implementation is no longer built.

Fortunately, the libyuv wrapper can supply the implementation, if we
hack the signature to return to the unsigned int types. We'll remove
this once libjingle has been updated to use libyuv directly.

Also, roll libyuv to r100 which fixes a gyp warning on Windows.

TEST=build

Review URL: http://webrtc-codereview.appspot.com/323004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1151 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 19:43:12 +00:00
henrike@webrtc.org
315282c01a Fixes a compiler warning related to dynamically allocated static memory. the fix is to leak the memory since the OS will clean it up anyways. This will not add noise to memory tools so it's ok. The issue is reported here: http://code.google.com/p/webrtc/issues/detail?id=147.
Review URL: http://webrtc-codereview.appspot.com/267023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1150 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 17:46:20 +00:00
mflodman@webrtc.org
d5651b98c5 Refactored ViEFrameProviderBase.
Only style changes, ointers/references and functions will come in a later CL.

vie_capturer.cc and vie_file_player.cc are only changed du to inheriting protected members from ViEFrameProviderBase.

Review URL: http://webrtc-codereview.appspot.com/324001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1148 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 15:20:58 +00:00
xians@webrtc.org
0744ee563d Disable API tests on ALSA since the tests don't work for all the alsa devices.
Review URL: http://webrtc-codereview.appspot.com/317004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1147 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 14:05:29 +00:00
henrik.lundin@webrtc.org
6198624815 Remove warnings on Mac (Issue 178)
Remove an if-else that can never execute the else statement.
Remove double parenthesis.

BUG=http://code.google.com/p/webrtc/issues/detail?id=178
TEST=

Review URL: http://webrtc-codereview.appspot.com/318004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1146 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 13:58:17 +00:00
mflodman@webrtc.org
5cc4dc9e0c Remove warnings in VideoEngine, capture module and render module.
BUG=164, 176, 180

Review URL: http://webrtc-codereview.appspot.com/303004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1145 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 10:12:57 +00:00
mflodman@webrtc.org
b19582b7dc Add pointer constructor to CriticalSectionScoped.
Mainly added to simplyfy the code, e.g. when having critsect as scoped_ptr in classes.

Review URL: http://webrtc-codereview.appspot.com/302005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1144 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 10:02:16 +00:00
henrikg@webrtc.org
af225d6bf6 The change http://webrtc-codereview.appspot.com/299001 (commit 1062) does not do what it intends (exclude codecs from Chromium build). This is a fix for that. webrtc.gyp is not pulled in Chromium, hence it has no effect putting a define there. Moving it to src/build/common.gypi.
Review URL: http://webrtc-codereview.appspot.com/315002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1143 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 09:58:39 +00:00
mflodman@webrtc.org
5885a4162a Refactored ViERenderer.
Only style changes, function and type changes will come in a later CL.

Review URL: http://webrtc-codereview.appspot.com/321001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1142 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 05:52:32 +00:00
mikhal@webrtc.org
2ab104e6be Switching WebRtc to LibYuv.
General Notes:
1. In general, API structure was not modified and is based on VPLIB. 
2. Modification to API: Return values are based on libyuv, i.e. 0 if ok, a negative value in case of an error (instead of length). 
3. All scaling (inteprolation) is now done via the scale interface. Crop/Pad is not being used.
4. VPLIB was completely removed. All tests are now part of the libyuv unit test (significantly more comprehensive and based on gtest).   
5. JPEG is yet to be implemented in LibYuv and therefore existing implementation remains.
Review URL: http://webrtc-codereview.appspot.com/258001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1140 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 02:46:22 +00:00
mikhal@webrtc.org
ffa0a9e9c9 updating libyuv to latest version (98).
This CL also includes some additional adaptations to the code due to the upgrade. 
Review URL: http://webrtc-codereview.appspot.com/306001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1139 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 22:45:53 +00:00
mallinath@webrtc.org
7766e2a82d - This issue raised by the chromium team when clang compiler is used. This was not an error as in this case we were accessing IPV6 address with IPV4 struct which is defined as 14 bytes in the header file, but we had the runtime check to determine the address space.
Now the solution is to use IPV6 structures instead of IPV4 when address space is determined.

I haven't put the new solution behind AF_INET6 flag, as i don't think it's necessary. 
Review URL: http://webrtc-codereview.appspot.com/291014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1138 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 21:37:19 +00:00
andrew@webrtc.org
b0be7aa7ae Remove deprecated OS X Core Audio APIs.
We no longer support the 10.4 SDK, so we can remove the weak-leaking
feature and exclusively use the added-in-10.5 APIs.

BUG=issue143
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/322001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1136 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 20:15:36 +00:00
marpan@webrtc.org
63b50f60d6 test_fec: Fix to valgrind warnings.
Review URL: http://webrtc-codereview.appspot.com/304002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1135 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 19:05:39 +00:00
mikhal@webrtc.org
f5ee1dc3e6 video_coding: Adding temporal layer info support to receive side
Review URL: http://webrtc-codereview.appspot.com/303005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1134 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 19:04:47 +00:00
xians@webrtc.org
832d7c6000 Disable typing detection for chromium since CGEventSourceKeyState is violating chromium sandbox.
Review URL: http://webrtc-codereview.appspot.com/320003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1132 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 16:45:46 +00:00
phoglund@webrtc.org
dd094fd6ae Started extracting methods out of the main test.
Started extracting methods out of the main test, which will hopefully make us able to make the tests independent.

Merge branch 'master' into voe_split_methods

Conflicts:
	src/voice_engine/main/test/auto_test/voe_extended_test.cc
	src/voice_engine/main/test/auto_test/voe_extended_test.h
	src/voice_engine/main/test/auto_test/voe_standard_test.cc
	src/voice_engine/main/test/auto_test/voe_standard_test.h

Extracted methods out of the standard test.

Added space before inheritance colons.

Rolled back some header file changes.

Fixed long lines.

Fixed long lines.

Fixed indentation. There is nothing but whitespace changes here, except for removing some extraneous semicolons in .h files and fixing a spelling error in a comment.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/313001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1131 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 15:07:59 +00:00
henrik.lundin@webrtc.org
d03718d1e4 Use ResourcePath in NetEQ unittest
Review URL: http://webrtc-codereview.appspot.com/320001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1130 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 11:42:43 +00:00
mflodman@webrtc.org
d2ee5d989d Changed sync bug introduced in refactoring.
Review URL: http://webrtc-codereview.appspot.com/319001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1129 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 09:25:11 +00:00
mflodman@webrtc.org
c78209c58b Add log when transport fails to send packet.
Review URL: http://webrtc-codereview.appspot.com/311002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1128 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 09:22:45 +00:00
kjellander@webrtc.org
7de6e10410 Fixing compilation error on Linux 64-bit
Problem was introduced in http://webrtc-codereview.appspot.com/311001/ because I had projects generated with Valgrind configuration, which is more forgiving about these implicit conversions.

BUG=
TEST=Compiling in Debug+Release on Linux, Mac and Windows.

Review URL: http://webrtc-codereview.appspot.com/318002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1127 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 08:39:13 +00:00
kjellander@webrtc.org
5b97b1216f Splitted FileHandler into FrameReader and FrameWriter classes and moved them to testsupport in test.gyp.
Fixed unit tests so they don't use ASSERT_DEATH since that doesn't work with Valgrind.

Fixed all Valgrind warnings except the one caused by CriticalSectionWrapper in system_wrappers.

Reworked all includes and GYP include paths to use full directory paths.

Removed util.h for logging, since it rendered warnings in Valgrind because of gflags. Replaced it with a verbose flag and a new function in video_quality_measurement.cc

BUG=
TEST=Passed test_support_unittests and video_codecs_test_framework_unittests on Linux, Mac and Windows.

Review URL: http://webrtc-codereview.appspot.com/311001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1126 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 07:42:18 +00:00
henrike@webrtc.org
441b3fe2a1 Made some global statics have function scope so that the global static count is 0 for the rtp_rtcp module.
Review URL: http://webrtc-codereview.appspot.com/316001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1125 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 02:03:49 +00:00
stefan@webrtc.org
cc7b649474 Add trace for the situation when the min bitrate > available bandwidth.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/312001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1123 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-07 13:22:06 +00:00
phoglund@webrtc.org
693240f2d9 Fixed many formatting and indentation problems in voe_auto_test.
Fixed indentation. There is nothing but whitespace changes here, except for removing some extraneous semicolons in .h files and fixing a spelling error in a comment.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/305004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1122 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-07 12:32:58 +00:00
henrik.lundin@webrtc.org
598ad06432 Fixing compiler warning in NetEQ
With some compiler settings, a warning was issued for NetEQ,
saying that pw16_randVec was accessed out of bounds.
This did never happen in practice, but this change makes the
compiler understand this.

Review URL: http://webrtc-codereview.appspot.com/309001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1121 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-07 11:52:09 +00:00
stefan@webrtc.org
b3bd1cd5f1 Fixes Valgrind warnings in the default VCM tests.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/299010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1120 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-07 11:48:09 +00:00
henrik.lundin@webrtc.org
bf86c33b0e Removing OutputDebugString from rtp_rtcp module
This is in response to WebRTC issue 167.

BUG=http://code.google.com/p/webrtc/issues/detail?id=167

Review URL: http://webrtc-codereview.appspot.com/301013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1119 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-07 10:44:05 +00:00
henrik.lundin@webrtc.org
44ef3774ce Fixing a compiler error in NetEQ
This error would only arise when compiling without support for
DTMF (which is not the default config).

Review URL: http://webrtc-codereview.appspot.com/310001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1118 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-07 10:43:25 +00:00
phoglund@webrtc.org
5b343aedcc Added missing .h files to .gypi files so they will show up in xcode / vc projects.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/304008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1117 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-07 10:34:35 +00:00
stefan@webrtc.org
58927e8d8f Disable deblocking temporarily due to Valgrind warnings.
Also corrects the copying of the decoded image data for frames
with odd width or height.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/307002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1116 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-07 08:13:31 +00:00
marpan@webrtc.org
1d34212a45 FEC: Update to packets masks (FEC generator matrix) in fec_private_tables.h
A set of the packet masks (up 10x10 size) are modified for the following reasons:

1) have more even column and row degree (number of 1 bits), when possible.

2) if cases where the column degree cannot be constant across source packets, placed the extra 1 bit in the first packet column (so little more protection on 1st partition), as opposed to having some ~middle source packet have the extra bit.

3) in some cases, made the mask a little more sparse/reduced the overlap.

Overall the average recovery is a little better with these masks.

Mask sizes above 10 will be updated in future changelist.
Review URL: http://webrtc-codereview.appspot.com/305001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1113 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-06 18:50:11 +00:00
kma@webrtc.org
4a8b1eaf6e In NS, replaced a divide calculatoin by shifting, and thus saved the MIPS by 5%(ARMv7) and 10%(ARMv7-Neon). Bit is not exact with the original. Quality is similar.
Review URL: http://webrtc-codereview.appspot.com/298004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1112 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-06 18:04:48 +00:00
henrik.lundin@webrtc.org
b6e58eb5a1 Fix formatting of rtp_format_vp8*
Sorting out all lint issues and fixing indentation.

Review URL: http://webrtc-codereview.appspot.com/301011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1111 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-06 15:56:18 +00:00
stefan@webrtc.org
c7e2bffb66 Fix header/lib mismatch caused by a constant not defined for header file.
BUG=http://code.google.com/p/webrtc/issues/detail?id=170
TEST=

Review URL: http://webrtc-codereview.appspot.com/300008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1110 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-06 13:44:25 +00:00
phoglund@webrtc.org
048b037342 Fixed vie_auto_test shutdown race conditions.
Fixed a race condition crash in vie_auto_test shutdown. Certain tests did not clean up the voice engine properly which caused crashes during certain uncommon timing conditions.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/307001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1109 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-06 10:42:05 +00:00
xians@webrtc.org
eff3c8905f this patch fixes the valgrind warnings in the adm api test for pulseaudio in linux.
Review URL: http://webrtc-codereview.appspot.com/301012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1108 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-06 10:02:56 +00:00
mikhal@webrtc.org
cae01010bd libyuv unit test: adding check for fread return value
Review URL: http://webrtc-codereview.appspot.com/303007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1107 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-06 00:44:16 +00:00
mikhal@webrtc.org
a5e980a906 Updating jitter buffer test following latest changes.
Review URL: http://webrtc-codereview.appspot.com/294002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1106 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-05 18:27:31 +00:00
phoglund@webrtc.org
23e1c0a0b1 File handling in vie_auto_test now uses fileutils so input and output file end up in a good place.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/301003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1103 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-05 15:27:04 +00:00
perkj@webrtc.org
ec7759a8c4 Fix broken vie_capture_module_test on mac.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/303006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1101 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-05 12:17:10 +00:00
perkj@webrtc.org
8627adc158 Refactored Video capture Unit test to use gtest.
Fix Valgrind warnings on Linux.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/296009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1100 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-05 09:58:55 +00:00
stefan@webrtc.org
0ae71b9ccb Disable temporal layers when building with Chromium.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/301010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1099 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-05 08:42:52 +00:00
henrika@webrtc.org
af71f0e5d9 Fixes two minor issues reported by the Coverty Integration Manager.
BUG=none
TEST=voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/302002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1098 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-05 07:02:22 +00:00
andrew@webrtc.org
c9cc3750cf Add missing system_wrappers dependency.
TBR=kma@webrtc.org

Review URL: http://webrtc-codereview.appspot.com/301009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1097 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-03 20:51:20 +00:00
kma@webrtc.org
b59c031660 For Android ARMv7 platforms, added a feature of dynamically detecting the existence of Neon,
and when it's present, switch to some functions optimized for Neon at run time.
Review URL: http://webrtc-codereview.appspot.com/268002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1096 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-03 18:34:50 +00:00
andrew@webrtc.org
ae7017d588 Fix missing dependency in audioproc.
TBR=bjornv@webrtc.org

Review URL: http://webrtc-codereview.appspot.com/300006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1095 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-03 01:43:29 +00:00
andrew@webrtc.org
7bf2646e4d Make protobuf use optional.
- By default, disable the AudioProcessing protobuf usage in the Chromium
  build. The standalone build is unaffected.
- Add a test for the AudioProcessing debug dumps.

TEST=audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/303003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1094 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-03 00:03:31 +00:00
mflodman@webrtc.org
626fbfd4cd Correcting vie_encoder nits.
Review URL: http://webrtc-codereview.appspot.com/302004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1093 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 23:39:11 +00:00
perkj@webrtc.org
6b1bfd6c5e Changed webrtc::ACMCodecDB::neteq_decoders_ to a const array.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/304003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1092 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 12:48:19 +00:00
pwestin@webrtc.org
db221d2b81 Fixes to temporal layers, Henrika please review src/common_types.h
Review URL: http://webrtc-codereview.appspot.com/286001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1091 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 11:31:08 +00:00
phoglund@webrtc.org
6aed73d218 Fixed release compilation error.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/298003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1090 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 11:14:12 +00:00
henrik.lundin@webrtc.org
e26aad4a9e Disable NetEQ unittest for Windows
Disable NetEqDecodingTest::TestNetworkStatistics for Windows.
It was never tested for Windows. Something is causing it to
fail, probably need different set of test vectors.

Review URL: http://webrtc-codereview.appspot.com/302003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1089 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 10:27:14 +00:00
stefan@webrtc.org
9cb2b56b65 Corrected a fread verification.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/301006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1088 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 10:22:29 +00:00
phoglund@webrtc.org
b956b4856a vie_auto_test may now be run in automated mode on all three platforms.
Fixed chrash bug on Mac, but there are still crash bugs since a couple weeks back. These will have to be fixed separately.

Removed dialogs from capture tests on Windows.

Removed some dead code related to answer files.

Added the last Windows fixes.

Fixed the Mac vie_auto_test runner - it will now run on Mac again. It will still crash randomly on codec and rtcp tests though.

Fixed compilation error.

Got patch to commit on Mac.

Temp commit on mac

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/292011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1087 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 10:19:27 +00:00
perkj@webrtc.org
38ca4f2953 Fix code review comments.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1086 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 09:34:10 +00:00
perkj@webrtc.org
d3eac4158c Fixed webrtc::perm variable.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1085 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 09:34:01 +00:00
perkj@webrtc.org
1b72fcd27b Fix symbol RTPFILE_VERSION.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1084 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 09:33:51 +00:00
stefan@webrtc.org
772d70bcd2 Fix release build error.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/304005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1083 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 09:18:49 +00:00
stefan@webrtc.org
a4a88f90c4 Implemented NACK based reference picture selection.
This CL implements NACK based reference picture selection for VP8. A separate
class is used for keeping track of the references and managing the VP8 encode
flags. Appropriate tests have also been added.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/284002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1082 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 08:34:05 +00:00
henrike@webrtc.org
4b00560a6e Fixes build error in rtp_rtc module introduced in r1076.
Review URL: http://webrtc-codereview.appspot.com/301005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1081 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 00:32:24 +00:00
punyabrata@webrtc.org
c1ed87602a Adding some error handling functionality in the windows audio core implementation to
stop rendering automatically and throw a playout-error callback when RequestPlayoutData
fails
Review URL: http://webrtc-codereview.appspot.com/300003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1080 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-01 17:55:35 +00:00
mflodman@webrtc.org
c6182915a3 Fix vie_encoder.cc.
TBR=ajm

Review URL: http://webrtc-codereview.appspot.com/301004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1079 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-01 17:45:28 +00:00
mflodman@webrtc.org
84d17838ac Refactored ViEEncoder.
Style changes + QT Metrics class from h-file to cc-file, type changes will be in another CL.

Review URL: http://webrtc-codereview.appspot.com/303001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1078 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-01 17:02:23 +00:00
kjellander@webrtc.org
5f4f69ac57 Removing sleeps from vp8_test.
These sleeps were remains from earlier tests that required them to work with some codecs. Removing these sleep calls cut the execution time from 90s to 30s on my machine.

Review URL: http://webrtc-codereview.appspot.com/304004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1077 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-01 15:50:04 +00:00
pwestin@webrtc.org
0644b1dc35 Introduce a mockable RtpRtcpClock interface replacing ModuleRTPUtility time functions
A new RtpRtcpClock interface has been added to rtp_rtcp_defines.h
and provides time facilities used by an RTP/RTCP module. Also,
NTP constants have been made public in the
webrtc::ModuleRTPUtility namespace to make implementation of
external clocks easier.

An overloaded version of CreateRtpRtcp() accepts a clock argument. By
default, if no clock is provided, the module uses the system clock
(old ModuleRTPUtility implementation).

Throughout the RTP/RTCP module code, calls to TickTime and
ModuleRTPUtility time functions have been replaced with calls to time
methods on a clock object.

The following classes take a clock object in their constructor and
hold a _clock field (either directly, or inherited from a parent):

Bitrate
ModuleRtpRtcpImpl
RTCPReceiver
RTCPSender
RTPReceiver
RTPSender
RTPSenderAudio
RTPSenderVideo

Methods from other classes that do not derive any of those and
require a time take an additional nowMS parameter, that should be
the result of calling GetTimeInMS() on a clock object.
Review URL: http://webrtc-codereview.appspot.com/268017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1076 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-01 15:42:31 +00:00
bjornv@webrtc.org
132feb1270 Made tables static.
In this CL global tables have been moved to where they are actually used. If for some reason they need to be available in a larger scope we can add them again at that point.
Review URL: http://webrtc-codereview.appspot.com/303002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1075 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-01 15:40:50 +00:00
kjellander@webrtc.org
4c4b7f500f Converting vp8_test to use fileutils and gtest
Review URL: http://webrtc-codereview.appspot.com/289012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1074 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-01 15:24:36 +00:00
tina.legrand@webrtc.org
f64162c335 Adding const to a number of constant tables. Setting some tables to static.
Patch set 2: Renaming static const tables. They no longer need the prefix WebRtc_Isac...
Review URL: http://webrtc-codereview.appspot.com/301001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1073 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-01 13:01:39 +00:00
bjornv@webrtc.org
bedabb25bf Added const on const tables.
Builds on Linux.

Tommi: Can you try on Windows?
Review URL: http://webrtc-codereview.appspot.com/300002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1072 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-01 07:44:32 +00:00
henrike@webrtc.org
c2ac8953d5 Fixes Valgrind warnings in system wrappers unittest.
Review URL: http://webrtc-codereview.appspot.com/293006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1071 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 22:46:59 +00:00
zakkhoyt@webrtc.org
a7e70b43e2 When entering fullscreen mode, the CocoaRenderView is attached as a subview to a new full screen window.
When the class is torn down, the view was not being attached back to it's original NSView. I added a 
new class variable to remember the original superview and then reattach it at the appropriate time.
Review URL: http://webrtc-codereview.appspot.com/290009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1070 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 22:30:50 +00:00
mikhal@webrtc.org
b9db43e1b6 video_coding/jitter buffer: Reduce delay on a complete frame: No need for the next frame when current frame is already complete.
Review URL: http://webrtc-codereview.appspot.com/289007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1069 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 18:38:01 +00:00
mflodman@webrtc.org
511f82eee9 Refactored ViESyncModule.
Only style changes, will follow up with references/ptrs.

Review URL: http://webrtc-codereview.appspot.com/291007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1068 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 18:31:36 +00:00
perkj@webrtc.org
68f2168978 Remove global voe::Channel::numSocketThreads.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1067 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 18:11:23 +00:00
mflodman@webrtc.org
27a82a65ca Refactored ViEBaseImpl.
Only style changes, will follow up with references/ptrs.

Review URL: http://webrtc-codereview.appspot.com/290008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1066 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 18:04:26 +00:00
andrew@webrtc.org
587c844741 Query the capture volume immediately on Win Core.
This allows the capture volume to be queried immediately at capture
startup, rather than waiting the usual one second interval.

Review URL: http://webrtc-codereview.appspot.com/297003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1064 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 17:43:05 +00:00
henrik.lundin@webrtc.org
524eb48081 Removing deprecated NetEQ APIs
Removing WebRtcNetEQ_GetPreferredBufferSize and
WebRtcNetEQ_GetCurrentDelay and all dependent APIs.

Review URL: http://webrtc-codereview.appspot.com/289006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1063 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 16:21:22 +00:00
xians@webrtc.org
0dffc6449a To be able to get webrtc into chrome, we need to reduce the size of the binary and the usage of memory.
This patch disbale some codecs which are not considered necessary. 
Review URL: http://webrtc-codereview.appspot.com/299001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1062 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 15:35:44 +00:00
stefan@webrtc.org
0c2adf0b75 Fix bug introduced when enabling VP8 frame dropping.
Also fixes two unit test mismatches.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/299002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1061 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 14:41:58 +00:00
stefan@webrtc.org
ac2c677bf6 Make all video_coding tests use the resources and output directories.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/298001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1060 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 14:23:39 +00:00
andrew@webrtc.org
d2daa5c13e Use clang by default on Mac.
But disable Chrome clang plugins for the time being.

TEST=build

Review URL: http://webrtc-codereview.appspot.com/297005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1059 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 01:16:06 +00:00
andrew@webrtc.org
268257475b Fix one more Objective-C clang error.
(Analogous to r1056).

BUG=issue78

Review URL: http://webrtc-codereview.appspot.com/297004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1058 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 00:54:04 +00:00
zakkhoyt@webrtc.org
2687b261d5 Since the CocoaRenderView is forward declared with @class instead of imported,
instance must be cast to NSView* when passed to NSView's addSubView method.
Review URL: http://webrtc-codereview.appspot.com/288001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1056 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 23:55:19 +00:00
punyabrata@webrtc.org
c9801465b6 Adding a check to ensure that the memcpy does not exceed bounds of the arrays.
Review URL: http://webrtc-codereview.appspot.com/290007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1055 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 18:49:54 +00:00
andrew@webrtc.org
1e91693fe2 Move stream_delay check to ProcessStream().
- was_stream_delay_set_ was being incorrectly reset in
AnalyzeReverseStream().
- Added tests to catch this case.

BUG=
TEST=audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/291011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1054 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 18:28:57 +00:00
henrike@webrtc.org
0bf2ca2eed Fixes broken unit test http://code.google.com/p/webrtc/issues/detail?id=154
Review URL: http://webrtc-codereview.appspot.com/292007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1053 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 18:21:46 +00:00
mikhal@webrtc.org
5fef05b529 libyuv: Updating paths for test files
Review URL: http://webrtc-codereview.appspot.com/289010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1052 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 17:50:07 +00:00
mflodman@webrtc.org
ffabb59f6e Refactored ViERefCount.
In a coming CL: Use ref count in system_wrappers instead of this class.

Review URL: http://webrtc-codereview.appspot.com/291010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1051 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 17:31:21 +00:00
henrik.lundin@webrtc.org
fc9b903fbe Enable NetEQ statistics unit testing
Adding test target NetEqDecodingTest::TestNetworkStatistics.
Update neteq_unittest to get files from resources folder.
Update DEPS file to get resources revision 2.

Review URL: http://webrtc-codereview.appspot.com/291013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1050 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 15:38:27 +00:00
henrik.lundin@webrtc.org
2d8125dd1a Testing NetEQ network statistics
Implementing helper function for new unit test
NetEqDecodingTest::TestNetworkStatistics. The test itself
remains to be defined. (Will be added in a coming CL.)
This change required some refactoring of the test code
to avoid excessive code duplication.

Review URL: http://webrtc-codereview.appspot.com/295009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1049 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 14:30:28 +00:00
kjellander@webrtc.org
c625c1010a Updated system_wrappers_unittests to use the test_support_main target.
Review URL: http://webrtc-codereview.appspot.com/291012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1048 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 12:11:06 +00:00
stefan@webrtc.org
932ab18d32 Default to always NACKing residual losses when having both FEC and NACK.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/296002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1047 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 11:33:31 +00:00
bjornv@webrtc.org
4b80eb4fcd Name change resampler.c/h to aec_resampler.c/h.
To avoid possible conflict with resampler in common_audio.
Review URL: http://webrtc-codereview.appspot.com/296006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1046 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 08:44:01 +00:00
mflodman@webrtc.org
611e4c3253 Refactored ViEPerformanceMonitor.
Only style changes, will follow up with references/ptrs.

Review URL: http://webrtc-codereview.appspot.com/289009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1045 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 02:39:28 +00:00
mikhal@webrtc.org
a85590d383 libyuv: Adding Android.mk
Review URL: http://webrtc-codereview.appspot.com/291009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1044 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 01:42:57 +00:00
mflodman@webrtc.org
ad4ee3659e Refactored ViEReceiver.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1043 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-28 22:39:24 +00:00
marpan@webrtc.org
9d8bec6f76 FEC: Fix to valgrind warning.
Review URL: http://webrtc-codereview.appspot.com/292009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1042 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-28 22:10:05 +00:00
andrew@webrtc.org
400ad6928e Fix compile warning in NS.
BUG=issue151
TEST=audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/290005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1041 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-28 21:33:42 +00:00
marpan@webrtc.org
d1b7932adf VP8: Setting non-zero (conservative) threshold for frame dropper.
Review URL: http://webrtc-codereview.appspot.com/291001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1040 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-28 19:20:31 +00:00
mikhal@webrtc.org
2cdb2d3833 Adding Libyuv to Webrtc:
- Adding library to DEPS file
 - Adding Wrapper implementation and tests. 

This is an interim state, as these files are not being linked at this stage.
Review URL: http://webrtc-codereview.appspot.com/259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1039 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-28 18:09:41 +00:00
xians@webrtc.org
e07247af8d Valgrind reports a racing condition on _sending because it is accessed by
both TransmitMixer::PrepareDemux() and StartSend()/StopSend().
Put a lock to resolve it.
Review URL: http://webrtc-codereview.appspot.com/293005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1038 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-28 16:31:28 +00:00
andrew@webrtc.org
1e39bc80dc Handle debug files from multiple AEC instances.
Review URL: http://webrtc-codereview.appspot.com/295006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1036 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-27 23:46:23 +00:00
andrew@webrtc.org
a919d3a643 Don't return a zero delay with insufficient data.
For estimating a delay over a long segment (e.g. a file) this can
throw off the estimate. Better not to add values to the AEC's histogram
until they're reliable.

BUG=
TEST=audiproc, audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/292004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1035 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-27 23:40:58 +00:00
stefan@webrtc.org
94a8c03141 Slightly increased bandwidth adaptation at both receive- and send-side.
The send-side increase factor is increased to better follow the pace
of the receive-side estimate, while the receive-side factor is
increased to speed up adaptation.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/297002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1030 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 14:09:37 +00:00
xians@webrtc.org
8738d277a1 Valgrind detects that there are racing conditions in RTPReceiver::PacketTimeout and RTPSender
This CL fixes two of them.
Review URL: http://webrtc-codereview.appspot.com/295005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1029 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 13:43:53 +00:00
henrik.lundin@webrtc.org
0fcc2eb368 Cleaning up neteq_unittest
- Conforming to testing standards.
- Fixing a way of generating new reference output files.
- ifdef the test to run only on linux 64-bit
- Renaming unittest source file.
- Renaming test vectors

Review URL: http://webrtc-codereview.appspot.com/296007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1028 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 13:43:42 +00:00
henrik.lundin@webrtc.org
789da89d37 Fix a valgrind warning in NetEQ
The special cases for packet sizes <= 10 ms (one case for each
sample rate) resulted in reading outside of the pw16_decoded
vector. This is now fixed by making sure that WebRtcSpl_DownsampleFast
gets correct input and output vector lengths.

Review URL: http://webrtc-codereview.appspot.com/295008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1027 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 12:35:31 +00:00
stefan@webrtc.org
0ee8ba1929 Remove WebRTC dependency on libvpx_lib and libvpx_include.
Removes dependencies on libvpx_lib and libvpx_include targets when
building with Chromium.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/293004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1026 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 12:12:43 +00:00
xians@webrtc.org
83661f534e fixing the racing conditions
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1025 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 10:58:15 +00:00
henrik.lundin@webrtc.org
859626570a VP8 RTP work
Fixing the plumbing to get the KEYIDX between VP8 wrapper and
rtp_rtcp module. Also fixing a missing pipe for temporalIdx

Review URL: http://webrtc-codereview.appspot.com/295004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1024 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 10:17:00 +00:00
braveyao@webrtc.org
0a18522e1b Add support to 96kHz sampling rate to Windows CoreAudio interface.
Review URL: http://webrtc-codereview.appspot.com/295003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1018 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 02:45:39 +00:00
mflodman@webrtc.org
26b9777e62 Only trigger one call to OnNetworkChanged for each incoming RTCP packet.
Review URL: http://webrtc-codereview.appspot.com/289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1016 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-24 15:22:33 +00:00
mflodman@webrtc.org
471e83e592 Refactored ViESharedData.
Only vie_shared_data.* are refactored, all *_impl.cc are only changed due to changed names of members in ViESharedData. These files will be refactored later, so the indentation in these files might be corrupt at this stage.

References are not changed to pointers at this stage.

Review URL: http://webrtc-codereview.appspot.com/292006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1015 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-24 15:16:00 +00:00
henrik.lundin@webrtc.org
9af365d3c5 Fixing VP8 RTP parser bug
Missing one initialization of new struct variable hasKeyIdx.

TBR=stefan@webrtc.org

Review URL: http://webrtc-codereview.appspot.com/296004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1014 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-24 13:28:29 +00:00
henrik.lundin@webrtc.org
6f2c0168f0 Updating to VP8 RTP spec rev -02
Updating the VP8 packetizer class (RtpFormatVp8) and VP8 parser
(in class RTPPayloadParser) to follow the -02 revision of the spec.
See http://tools.ietf.org/html/draft-ietf-payload-vp8-02.

Updating the unit tests, too. Finally, updating the tests to
follow the recommendations from the test team; specifically
including the test code in the webrtc namespace, and omitting
the main function at the end of each test file.

Review URL: http://webrtc-codereview.appspot.com/296003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1013 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-24 12:52:40 +00:00
mflodman@webrtc.org
6d26ef76ea Refactored ViESender.
In a later CL:
- References -> const or ptr.

Review URL: http://webrtc-codereview.appspot.com/291003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1011 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-24 08:31:06 +00:00
kjellander@webrtc.org
d492f72e43 Added empty unit tests to get code coverage measured.
In order to get code coverage recorded, there must be an executing test that is linked to the code to measure.
These projects are currently not showing up in the code coverage.

Review URL: http://webrtc-codereview.appspot.com/293002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1010 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-24 07:20:00 +00:00
amyfong@webrtc.org
55d81ea517 ViE Custom Call observer now using pointers, fixed protection method and miscellaneous TODO cleanup
Review URL: http://webrtc-codereview.appspot.com/282004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1009 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-24 01:15:10 +00:00
andrew@webrtc.org
ba028a31c9 Fix sample rate printout in process_test.
TBR=bjornv

Review URL: http://webrtc-codereview.appspot.com/292005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1008 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 20:37:12 +00:00
phoglund@webrtc.org
f3d10d3dfd Fixed release compilation error-warnings.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/290004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1006 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 15:56:27 +00:00
phoglund@webrtc.org
c4c56ed20b Rewrote vie_auto_test to use googletest macros.
Removed error counting entirely - that's completely managed by googletest now, except for custom call, loopback and simulcast call.

Rewrote remaining tests to use GTest asserts.

Rewrote more tests to use GTest macros. The External Codec module is now in the build by default.

Merge branch 'master' into macro_improvements

Rewrote some more code to use GTest asserts.

The manual standard tests now also go through gtest.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/287002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1004 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 15:23:11 +00:00
bjornv@webrtc.org
48b68c0c24 Added support for 96 kHz sampling frequency.
Updated resampler_unittests with the new valid combinations.
Verified audio quality on files.

TEST=resampler_unittests, voe_auto_test
BUILDTYPE=Debug, Release
PLATFORM=Linux
Review URL: http://webrtc-codereview.appspot.com/294001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1002 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 13:50:41 +00:00
henrik.lundin@webrtc.org
4257790d2d NetEQ-related bug in ACM
Fixing a bug when creating new NetEQ slave instances in ACM.
The old code called WebRtcNetEQ_GetCurrentDelay() for the
master instance to get a delay value for WebRtcNetEQ_SetExtraDelay().
This is wrong, since WebRtcNetEQ_GetCurrentDelay() reports on the
current total buffer length, while WebRtcNetEQ_SetExtraDelay() is
the extra delay that is desired to in order to sync with video.

The fix includes keeping the extra delay value in a member variable
in the ACMNetEQ class.

Review URL: http://webrtc-codereview.appspot.com/295001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1001 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 13:04:05 +00:00
kjellander@webrtc.org
543c3eaa46 Fixing Release compilation errors
Review URL: http://webrtc-codereview.appspot.com/267026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1000 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 12:20:35 +00:00
henrik.lundin@webrtc.org
89ab652250 Cleaning up NetEQ statistics
Removed struct MCUStats_t and all references to it.
Removed totalDiscardedPackets and totalFlushedPackets
from the PacketBuf_t struct.

Review URL: http://webrtc-codereview.appspot.com/293001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@999 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 11:06:05 +00:00
henrik.lundin@webrtc.org
df10de4b27 Removing statistics API from NetEQ
Removing WebRtcNetEQ_GetJitterStatistics(),
WebRtcNetEQ_ResetJitterStatistics(), and the associated
struct WebRtcNetEQ_JitterStatistics. The change ripples
through all the way to the VoiceEngine API.

Review URL: http://webrtc-codereview.appspot.com/285002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@998 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 09:36:23 +00:00
braveyao@webrtc.org
7d3e9498bc This CL is to support certain audio devices which don't offer volume control. Try to be more compatible to those rare cases.
Review URL: http://webrtc-codereview.appspot.com/276011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@997 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 03:35:42 +00:00
mikhal@webrtc.org
2b838b4121 video_coding: updating the session info unit test following recent changes
Review URL: http://webrtc-codereview.appspot.com/290002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@996 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 00:20:19 +00:00
mikhal@webrtc.org
425b377973 video_coding: Updating internal_defines to resolve latest build error. Refers to JB flush update.
Review URL: http://webrtc-codereview.appspot.com/289001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@995 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-22 23:41:29 +00:00
mikhal@webrtc.org
f13388f134 video_coding: Requesting a key frame after a JB flush
Review URL: http://webrtc-codereview.appspot.com/280006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@994 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-22 22:57:51 +00:00
mikhal@webrtc.org
6b9a7f8704 video_coding: Allowing for a decodable state independent of selective nacking
Review URL: http://webrtc-codereview.appspot.com/263001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@993 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-22 22:48:20 +00:00
andrew@webrtc.org
828af1b4b9 Add lookahead to the delay estimator.
TEST=audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/279014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@992 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-22 22:40:27 +00:00
andrew@webrtc.org
5a529395aa Make DMO init safe when not supported.
BUG=issue133
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/284001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@990 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-22 18:04:26 +00:00
mflodman@webrtc.org
dfe89e337e Move ViE main/test/AutoTest to test/auto_test.
Only paths in gyp and mk files are changed, source files are only moved.

Review URL: http://webrtc-codereview.appspot.com/267027

git-svn-id: http://webrtc.googlecode.com/svn/trunk@988 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-22 14:03:15 +00:00
andrew@webrtc.org
8594f7688b Add a gyp variable for AEC debug dumps.
TEST=process_test.cc

Review URL: http://webrtc-codereview.appspot.com/276012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@987 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-22 00:51:41 +00:00
kma@webrtc.org
a249f35203 Correct several makefile errors for Android build.
Review URL: http://webrtc-codereview.appspot.com/267024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@986 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-21 22:16:10 +00:00
mflodman@webrtc.org
6830bdd929 Fix xcode build.
Review URL: http://webrtc-codereview.appspot.com/280007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@985 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-21 15:53:15 +00:00
mflodman@webrtc.org
94ea32ef60 Move video_engine/source* to video_engine/. No code changes except paths in gyp-files.
Review URL: http://webrtc-codereview.appspot.com/283002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@984 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-21 14:49:31 +00:00
kjellander@webrtc.org
274c2efbc1 Adding empty test method required to get code coverage
Review URL: http://webrtc-codereview.appspot.com/279008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@983 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-21 09:36:28 +00:00
marpan@webrtc.org
3caa327af0 VP8 wrapper: Turn on some mild amount of deblocking in post-processing.
Review URL: http://webrtc-codereview.appspot.com/268015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@982 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-19 01:08:09 +00:00
henrike@webrtc.org
ce9d89d892 Fixes linux build error introduced in r980.
Review URL: http://webrtc-codereview.appspot.com/279012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@981 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-19 00:14:37 +00:00
henrike@webrtc.org
ad98a3eed0 Fixes TEST crash triggered by webrtc-codereview.appspot.com/268014.
Review URL: http://webrtc-codereview.appspot.com/280005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@980 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 23:55:12 +00:00
henrike@webrtc.org
31d30700d6 Addressed review comments from http://webrtc-codereview.appspot.com/256004/
Review URL: http://webrtc-codereview.appspot.com/256007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@979 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 19:59:32 +00:00
kma@webrtc.org
ced118636d Changed keyword __restrict__ to __restrict.
Review URL: http://webrtc-codereview.appspot.com/279011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@978 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 17:51:19 +00:00
henrike@webrtc.org
3798ecb25b Made CPU initialization on Windows lazy to prevent long startup time.
Review URL: http://webrtc-codereview.appspot.com/268014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@977 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 16:25:54 +00:00
kjellander@webrtc.org
543611a77a Reverting r972 due to compilation error on Windows Release build.
TBR=kma
Review URL: http://webrtc-codereview.appspot.com/282003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@976 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 13:25:13 +00:00
bjornv@webrtc.org
2f047ccede Removed unnecessary variable to avoid compiler error on Win.
Review URL: http://webrtc-codereview.appspot.com/267021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@975 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 12:03:25 +00:00
henrik.lundin@webrtc.org
ba74924043 Remove use of exceptions in NetEQ test code
Replaced the exceptions thrown when codec instance creation failed
with simple exit(EXIT_FAILURE). There is no point in continuing
if creating the codec fails.

Review URL: http://webrtc-codereview.appspot.com/282002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@974 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 09:55:01 +00:00
bjornv@webrtc.org
6a9835d59c Delay estimator structural changes.
Improved the way we handle different data types (float vs fixed) and reduced the complexity by nearly 50%.
We now have a generic struct for both float and fixed delay estimators and a core struct for the binary spectrum based delay estimator. All wrapper codes (for both fixed and float) are gathered in delay_estimator_wrappers.*.
Moved out the far end history buffer to AEC(M).
Added a union to handle difference types when create.
Review URL: http://webrtc-codereview.appspot.com/277004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@973 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 08:30:34 +00:00
kma@webrtc.org
fa9b016fb5 Optimized WebRtcIsacfix_AutocorrFix() function for iSAC fix.
(1) For generic platforms, code was changed to remove the shifting within loops.
Basically, it's just change a loop from
  for() {
    sum += (a*b) >> scale;
  }
to:
  for() {
    sum += (a*b);
  }
  sum >> scale;

Type int64_t is used for sum to make sure no information is not lost.
Performance is about the same as before the change. Bits are not exact,
although in theory the change should have preserved more information. The purpose
of this change is to make the generic code and ARM code bit exact, simpify the code,
while keep the speech quality at least not lower. (Some speech tests might be good.)

(2) For ARM platform, used assembly to optimize the performance. iSAC runs faster
with this change. (Reduced run time of an offline file test from 10.16ms to 8.81ms)
Review URL: http://webrtc-codereview.appspot.com/267014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@972 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 02:50:55 +00:00
braveyao@webrtc.org
f556b9d1f4 This modification is supposed to fix the webrtc issue 144/145. With this fix, people could set/get mic volume before StartSend().
Review URL: http://webrtc-codereview.appspot.com/277007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@971 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 02:17:28 +00:00
amyfong@webrtc.org
917fa6b923 ViE Custom Call added SetImageScaleStatus toggle option and other changes.
1. added SetImageScaleStatus for testing purposes
2. added getting the codec information from the incoming/outgoing stream of a videochannel to print call information
3. fixed problem with toggling the one of the observers
4. did more clean up of the code style (mostly spacing)
5. renamed the GetVideo* functions properly to SetVideo* to reflect what the function does

Currently only tested on mac.  Need to test on win7 & linux before final commit.
Review URL: http://webrtc-codereview.appspot.com/267017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@969 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-17 21:29:39 +00:00
kjellander@webrtc.org
cd7b57ef9e Fixing release compilation error
Review URL: http://webrtc-codereview.appspot.com/279007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@968 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-17 14:26:21 +00:00
kjellander@webrtc.org
3f1cb8e546 Restructuring and adding dummy unit test target.
Empty test added to get code coverage recorded.

Review URL: http://webrtc-codereview.appspot.com/269018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@967 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-17 13:56:54 +00:00
kjellander@webrtc.org
cc2ecb3c2e Restructuring and adding dummy unit test target.
Empty test added to get code coverage recorded.

Review URL: http://webrtc-codereview.appspot.com/267019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@966 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-17 13:48:36 +00:00
kjellander@webrtc.org
b72268e147 Restructuring and adding dummy unit test target.
Empty test added to get code coverage recorded.

Review URL: http://webrtc-codereview.appspot.com/280004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@965 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-17 13:39:15 +00:00
kjellander@webrtc.org
64a897a772 Restructuring and adding dummy unit test target.
Empty test added to get code coverage recorded.

Review URL: http://webrtc-codereview.appspot.com/282001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@964 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-17 13:33:11 +00:00
phoglund@webrtc.org
8f89f09626 Note: this patch may seem intimidating but it mostly moves code around and renames things. There are quite few actual changes.
Separated new-style tests from old-style tests. Abstracted code for reuse.

Fully separated the new automated tests from the old-style tests. We now have old-style tests running in manual mode, old-style tests running in automated mode and new-style tests that uses input files and make actual video comparisons.

Introduced a small "library" of helper functions in order to move a lot
of stuff out of the original base and codec tests, which have been made
dependent on the new "library" (which is a header file and a source
file). The new-style tests also depends on this "library".
The comparison test flags are now required only when the comparison tests actually runs.

Separated comparison tests into its own test since it seems we will be running classic vie_auto_test using a fake video driver on Linux.

Made tbInterfaces follow Google conventions.
Merge branch 'render_to_file' into vivi_driver

Resolution alignment testing is now optional behind a flag.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/269011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@962 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-17 10:46:59 +00:00
kjellander@webrtc.org
c05b56a38b Fixing compilation error
Review URL: http://webrtc-codereview.appspot.com/276010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@961 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-17 08:59:48 +00:00
kjellander@webrtc.org
0403ef419f Restructuring and adding unit test targets on project level instead of in common_audio.
Review URL: http://webrtc-codereview.appspot.com/280001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@959 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-17 08:35:47 +00:00
phoglund@webrtc.org
337dc68992 Included modules in webrtc.gyp and fixed build errors.
Removed TODO from webrtc.gyp since it is done.

Tabs -> spaces.

Tabs -> spaces.

Tabs -> spaces.

Fixed compilation on Windows.

Added missing file.

Merge branch 'master' into fix_mac_modules

Fixed compilation errors for the modules.gyp on Mac. This included some pretty large refactorings.

 Please enter the commit message for your changes. Lines starting

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/269005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@957 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-16 15:36:44 +00:00
niklas.enbom@webrtc.org
af26f64616 Inband DTMF stereo support
Review URL: http://webrtc-codereview.appspot.com/267011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@956 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-16 12:41:36 +00:00
niklas.enbom@webrtc.org
e33a102eee Resubmitting http://webrtc-codereview.appspot.com/269007/
Review URL: http://webrtc-codereview.appspot.com/268012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@955 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-16 10:33:53 +00:00
stefan@webrtc.org
fcf33eb7e0 Limit number of send-side BWE increases to one per second.
Also report 0 losses if not enough expected packets since
previous receiver report.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/270009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@954 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-16 07:58:31 +00:00
punyabrata@webrtc.org
81d4499dee Microphone volume on Mac not being printed properly due
to a mismatch in variable type. Additionally, now printing
a volume that will range from 0 - 255
Review URL: http://webrtc-codereview.appspot.com/267016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@951 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-16 02:06:49 +00:00
andrew@webrtc.org
755b04a06e Add RMS computation for the RTP level indicator.
- Compute RMS over a packet's worth of audio to be sent in Channel, rather than the captured audio in TransmitMixer.
- We now use the entire packet rather than the last 10 ms frame.
- Restore functionality to LevelEstimator.
- Fix a bug in the splitting filter.
- Fix a number of bugs in process_test related to a poorly named
  AudioFrame member.
- Update the unittest protobuf and float reference output.
- Add audioproc unittests.
- Reenable voe_extended_tests, and add a real function test.
- Use correct minimum level of 127.

TEST=audioproc_unittest, audioproc, voe_extended_test, voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/279003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@950 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 16:57:56 +00:00
andrew@webrtc.org
6a85b17a0a Potential fix for crash after Mac sleep.
When a Mac goes to sleep, the OS pauses the IO threads. If a
subsequent StopSend/Playout happens, we time out waiting for the IO
threads, but didn't ensure they were shut down.

BUG=
TEST=voe_cmd_test, voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/269013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@949 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 16:23:41 +00:00
kjellander@webrtc.org
85596d5bf4 Setting completeFrame to true for all created encoded images.
Review URL: http://webrtc-codereview.appspot.com/276008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@948 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 13:45:25 +00:00
tommi@webrtc.org
cde1e7f42a Use a TraceNoop instance when tracing disabled (to be used in Chromium).
I'm also adding an empty implementation for static methods in the Trace
interface since the default implementation relies on TraceImpl.
Review URL: http://webrtc-codereview.appspot.com/267013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@946 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 12:23:36 +00:00
henrik.lundin@webrtc.org
bc91d5af86 NetEQ tests
Adding capability to parse RED payloads to the RTPanalyze tool.
Also adding a method to scramble an RTP payload (currently not
used).

Review URL: http://webrtc-codereview.appspot.com/276006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@945 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 10:16:01 +00:00
mflodman@webrtc.org
a02ef1ace2 Fix broken tree.
Review URL: http://webrtc-codereview.appspot.com/267015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@943 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 07:50:50 +00:00
mflodman@webrtc.org
1f69c03739 Added size sanity check for copying app specific RTCP data.
Similar check as done in RTCPUtility::RTCPParserV2::ParseAPPItem.

Review URL: http://webrtc-codereview.appspot.com/277002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@942 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 06:12:39 +00:00
henrik.lundin@webrtc.org
33df5335bf Change luminance of all pixels by a specified value.
Modeled on color_enhancement.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/269004
Patch from SriRam <tvnsriram@google.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@941 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-14 15:30:26 +00:00
stefan@webrtc.org
7de07652ad Disables a flaky metric test.
This is a duplication of issue 255008 since I wasn't able to commit that one
from the computer on which it was created.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/276007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@940 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-14 15:16:16 +00:00
tommi@webrtc.org
ded85f14ef Enable WEBRTC_NO_TRACE for Chromium builds.
I'm also fixing WEBRTC_TRACE so that it won't break the build but on Linux I had to do something non traditional as is explained in the comments.
Review URL: http://webrtc-codereview.appspot.com/269012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@939 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-14 09:39:31 +00:00
andrew@webrtc.org
0db7dc6e18 Add file-playing channels to voe_cmd_test.
Fix file reading and writing.

TEST=voe_cmd_test

Review URL: http://webrtc-codereview.appspot.com/279001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@938 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-13 01:34:05 +00:00
andrew@webrtc.org
cd8243807e Unpack the full set of audioproc data.
Review URL: http://webrtc-codereview.appspot.com/276004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@937 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-11 19:13:36 +00:00
kma@webrtc.org
d71d480487 Fixed a build error of audio conference mixer in Android.
Review URL: http://webrtc-codereview.appspot.com/267009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@936 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-11 17:14:23 +00:00
stefan@webrtc.org
b351d6a8d8 Reverting rev 929 due to failing assert on Linux.
Failing at: audio_buffer.cc:159

TBR=mflodman
Review URL: http://webrtc-codereview.appspot.com/270008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@935 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-11 13:26:05 +00:00
mflodman@webrtc.org
fd3a0efd15 RTP bw estimate fix.
Review URL: http://webrtc-codereview.appspot.com/279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@932 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-11 10:55:26 +00:00
phoglund@webrtc.org
1144ba2268 Base and codec tests now run verify output and render to file instead of to screen.
Rewrote the codec test to render to file and do video comparisons.

Refactored the coded tests somewhat. I still need to figure out how to do comparison in the automated case.

Added video analysis to the test. This will make sure that the system output roughly the right thing.

Moved the video metrics library into the test_support library. Made the metrics library available in the automated tests.

Made sure no one passes in too large YUV videos into the autotest.

The standard test's output now gets captured for both the left and right windows.

Wrote a rendering device which just writes the raw frames to file, for analysis. Updated the base standard test to dump its left window output to file. We don't do anything with it yet though.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/249001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@931 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-11 09:01:03 +00:00
niklas.enbom@webrtc.org
50b3cbe979 First pass. You can now enable a stereo codec and send and receive. This does not include more advances use cases (DTMF etc), but I'd rather keep the CLs manageable.
Review URL: http://webrtc-codereview.appspot.com/269007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@929 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-11 08:31:32 +00:00
kma@webrtc.org
b61c410347 Fixed a couple of Android makefiles to let voe and vie build properly.
Review URL: http://webrtc-codereview.appspot.com/278001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@928 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-10 18:10:25 +00:00
kma@webrtc.org
13318ef422 (1) Corrected the makefile for testing iLBC in Android, and changed the location of the test makefile to make it consistent with audio_processing.
(2) Added a makefile for testing fiexed point iSAC in Android.
Review URL: http://webrtc-codereview.appspot.com/266005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@927 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-10 18:00:22 +00:00
mflodman@webrtc.org
7a4eb2837a Calculate the available bandwidth before sending a TMMBR
Also changed the way TMMBR was processed since it did not match the new bandwidth estimator.

Review URL: http://webrtc-codereview.appspot.com/270003
Patch from pwestin1 <pwestin@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@925 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-10 12:54:46 +00:00
mflodman@webrtc.org
637a59e68e jitter buffer update: waiting for key frame when Nack is enabled and continuity cannot be determined.
Review URL: http://webrtc-codereview.appspot.com/266010
Patch from mikhals <mikhal@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@924 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-10 12:50:48 +00:00
tina.legrand@webrtc.org
855a77c972 Audio Coding Module: Fixing a bug that prevented the encoder from being re-initialized when changing codec from mono to stereo.
Solving issue 130 reported by Niklas.

Reviewer: Turaj
Review URL: http://webrtc-codereview.appspot.com/268007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@921 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-10 08:17:08 +00:00
andrew@webrtc.org
c4f129f97c Improve the mixing saturation protection scheme.
A single participant is not processed at all. With multiple
participants, we divide-by-2 as before when mixing. Afterwards,
the mixed signal is limited by the AGC to -7 dBFS and then doubled to
restore the original level.

This preserves the level while guaranteeing good saturation protection.

Add a test to voe_auto_test. Hijack and improve the existing mixing test
for this.

TEST=voe_auto_test, voe_cmd_test

Review URL: http://webrtc-codereview.appspot.com/241013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@920 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-10 03:41:22 +00:00
andrew@webrtc.org
d30b688751 Remove TraceScan executable.
Review URL: http://webrtc-codereview.appspot.com/270002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@918 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 22:23:20 +00:00
andrew@webrtc.org
4b13fc9c09 Add delay modification to process_test.
Review URL: http://webrtc-codereview.appspot.com/266007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@916 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 19:27:11 +00:00
henrike@webrtc.org
2f32b5c8a7 Fixes an issue where file playing could happen at a lower sampling frequency than the file.
Details:
The mixer looks at all the participants desired frequency and concludes the highest desired mixing frequency. This is the frequency that the mixer will mix at. Participants that are always mixed are in a separate list and the function concluding the highest desired mixing frequency did not look at that list and therefore always conclude that the lowest mixing frequency is sufficient.
Review URL: http://webrtc-codereview.appspot.com/277003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@915 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 19:02:17 +00:00
mikhal@webrtc.org
eb4ef17bbd Removing vplib include and VideoInterpolator when not needed
Review URL: http://webrtc-codereview.appspot.com/268004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@914 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 18:11:02 +00:00
kjellander@webrtc.org
488ed92c3b Removing exceptions since not used
Review URL: http://webrtc-codereview.appspot.com/267003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@912 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 16:12:40 +00:00
kjellander@webrtc.org
c3a4dcd101 Removing exceptions since not used
Review URL: http://webrtc-codereview.appspot.com/266008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@911 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 16:11:44 +00:00
kjellander@webrtc.org
ad79d6f164 Removing exceptions since not used
Review URL: http://webrtc-codereview.appspot.com/267002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@910 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 16:11:14 +00:00
mflodman@webrtc.org
03a9eb1526 RTP module: Make sure payloadName is null terminated.
Review URL: http://webrtc-codereview.appspot.com/268006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@908 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 14:51:18 +00:00
niklas.enbom@webrtc.org
f3c1b87f00 my eyes started bleeding when I saw this...
Review URL: http://webrtc-codereview.appspot.com/268005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@907 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 12:43:48 +00:00
kjellander@webrtc.org
9dcab8fb14 Restoring Android.mk
This is the last file left from 256006 that I forgot to restore according to your comments.
The other Android.mk you fixed in 266004.

Review URL: http://webrtc-codereview.appspot.com/268003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@905 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 08:59:13 +00:00
niklas.enbom@webrtc.org
4cd841e9a6 Fix win compile error for interpolator_test
Review URL: http://webrtc-codereview.appspot.com/269003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@904 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 08:02:16 +00:00
phoglund@webrtc.org
cff98ca6ff Made it possible to run the voe_auto_test standard test in GTest behind a flag. The purpose is to run the whole test without any manual intervention since we want to run the test on a build bot in automated mode.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/267001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@903 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-08 13:08:25 +00:00
henrikg@webrtc.org
c58ef08da2 Removes system CPU measurement for Chrome build.
It does not work on Chrome Windows, and is anyway not needed for Chrome.
Review URL: http://webrtc-codereview.appspot.com/243006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@902 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-08 08:44:17 +00:00
henrik.lundin@webrtc.org
f15fbc379d Change in RTP module SendVP8
Changing how the max payload length is calculated. Instead
of handling RTP and FEC header overhead explicitly, call the
MaxDataPayloadLength method which already does it. Avoid redundant code. Had to move MaxDataPayloadLength to the
RTPSenderInterface.

Review URL: http://webrtc-codereview.appspot.com/269002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@901 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-08 08:23:47 +00:00
kma@webrtc.org
9b813510eb Changes for building audio coding in anroid. Only makefiles are touched.
Review URL: http://webrtc-codereview.appspot.com/266004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@899 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-07 23:30:01 +00:00
henrike@webrtc.org
26d3667a26 Fix for broken test after r897
Review URL: http://webrtc-codereview.appspot.com/274001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@898 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-07 23:24:40 +00:00
henrike@webrtc.org
e2a34f8275 Removes the API for setting RX VAD since the RX vad should always be on anyways.
Review URL: http://webrtc-codereview.appspot.com/264001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@897 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-07 21:33:24 +00:00
mflodman@webrtc.org
5ae9f5ed6c Adding logs in RTPSender::ReSendToNetwork.
Review URL: http://webrtc-codereview.appspot.com/273001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@896 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-07 20:03:00 +00:00
kjellander@webrtc.org
bf483844af Restructuring and removing neteq_tests.gypi according to project structure discussed with Andrew. We want to flatten out the hierarchy and minimize the number of GYP files.
I also fixed compilation on Mac (by enabling exceptions for the NetEqTestTools target). Executing the test fails on Mac, but I assume this is because it checks bit exactness, similar to the issue we had with audio_coding_module (see issue 114)

Review URL: http://webrtc-codereview.appspot.com/255004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@895 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-07 16:05:19 +00:00
kjellander@webrtc.org
36e1ad9b5d Restructuring and removing ilbc_test.gypi.
According to project structure discussed with Andrew. We want to flatten out the hierarchy and minimize the number of GYP files.

No changes at all are being made in the source files; they are just moved.
The only modified files are the GYP file and Android.mk

Kevin: I updated relative paths in Android.mk so please verify it is correct, since I don't know how to build that.

Review URL: http://webrtc-codereview.appspot.com/256006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@894 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-07 15:27:11 +00:00
andrew@webrtc.org
b353d21560 ...and now fix the Debug build.
Review URL: http://webrtc-codereview.appspot.com/272001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@892 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-05 00:57:33 +00:00
andrew@webrtc.org
369766ed29 Fix Release mode errors in common_video tests.
Review URL: http://webrtc-codereview.appspot.com/271001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@891 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-04 23:59:56 +00:00
vikasmarwaha@webrtc.org
a5c4c1f1d4 Fix for WebRTC issue 64, removed the screenupdate thread and events from start render as they are already created in the ctor.
Review URL: http://webrtc-codereview.appspot.com/253008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@890 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-04 23:22:51 +00:00
marpan@webrtc.org
040cb71e0a Fix windows compilation errors and warning for test_fec. Disabled VERBOSE_OUTPUT.
Review URL: http://webrtc-codereview.appspot.com/253005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@889 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-04 22:57:56 +00:00
tina.legrand@webrtc.org
731e9aea79 Fixes ACM API test to build on 32-bits machines.
Changing counters from unsigned int64 to int.
Review URL: http://webrtc-codereview.appspot.com/256010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@887 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-04 07:34:22 +00:00
kjellander@webrtc.org
20a370e875 Changing the namespace of TestSuite to webrtc::test.
Adding gmock initialization into main test runner class

Review URL: http://webrtc-codereview.appspot.com/254004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@885 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-04 01:19:16 +00:00
kjellander@webrtc.org
1a8d08ad76 Changing usage of gtest_main target, to use test_support_main instead.
Review URL: http://webrtc-codereview.appspot.com/252002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@884 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-03 23:28:47 +00:00
andrew@webrtc.org
89088b963e Fix the path to protoc.gypi.
It was mistakenly pointing to the trunk/build rather than the
trunk/src/build copy, causing the Chrome build to fail.

TEST=./build/gyp_chromium in Chrome

Review URL: http://webrtc-codereview.appspot.com/253006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@883 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-03 20:43:45 +00:00
tina.legrand@webrtc.org
2475a1953a Committing a file that was part of CL 175002, but for wome reason weren't uploaded correctly.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@882 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-03 17:54:27 +00:00
tina.legrand@webrtc.org
fb389e3b02 This CL is divided in several patches, to make review easier.
Patch Set 1: Removing blanks at end of lines.
Patch Set 2: Removing tabs.
Patch Set 3: Fixing include-guards.
Patch Set 4-7: Formatting files in the list.
Patch Set 8: Formatting CNG.

Patch Set 9: 
* Fixing comments from code review
* Fixing formating in acm_dtmf_playout.cc
* Started fixing formating of acm_g7221.cc. More work needed, so don't spend too much time reviewing.
* Refactored constructor of ACMGenericCodec. Rest of file still to be fixed.
* Fixing break; after return ...; in several files.

Patch Set 10:
* Chaning from reintepret_cast to static_cast in three files, acm_amr.cc, acm_cng.cc and acm_g722.cc
NOTE! Not all files have the right format. That work will continue in separate CLs.

Review URL: http://webrtc-codereview.appspot.com/175002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@881 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-03 17:20:10 +00:00
andrew@webrtc.org
a4b9660372 Add mistakenly removed VAD enabling function.
This resolves the unknown VAD status warnings introduced in r845.

BUG=
TEST=voe_cmd_test

Review URL: http://webrtc-codereview.appspot.com/252004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@879 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-03 01:36:27 +00:00
mikhal@webrtc.org
e203de7ba2 jitter_buffer updates:
1. Determining continuity based on pictureId and not seq. numbers when available.
2. Hybrid bug fix: Don't set to decodable when the nack list is empty.
Review URL: http://webrtc-codereview.appspot.com/255001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@878 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-03 00:42:52 +00:00
pwestin@webrtc.org
7232ad78b2 reverted back the sanity and changed the test
Review URL: http://webrtc-codereview.appspot.com/254006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@877 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-03 00:36:32 +00:00
pwestin@webrtc.org
cfc1070586 Fixed sanity for min length
Review URL: http://webrtc-codereview.appspot.com/259003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@876 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-03 00:15:44 +00:00
pwestin@webrtc.org
075e91fa27 Added parsing of width and height from VP8 header
Review URL: http://webrtc-codereview.appspot.com/241012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@875 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 23:14:58 +00:00
henrik.lundin@webrtc.org
679cb07980 Fix build error for release build
Review URL: http://webrtc-codereview.appspot.com/252003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@874 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 19:52:27 +00:00
henrik.lundin@webrtc.org
baf6db5ead Making dual decoder work again in VCM
Changing the assignment operator in VCMJitterBuffer to a named
function (CopyFrom) instead, since it is not a straight
assignment. Also fixing two bugs in the jitter copy function.

Bug fix in VCMEncodedFrame: The copy constructor did not
copy _length.

In VCM codec database, make sure that the callback object is
preserved when copying back from secondary to primary decoder.

In VP8 wrapper, adding code to copy the _decodedImage to the
Copy() method.

Bugfix in video_coding_test rtp_player:
The retransmissions where made in reverse order. Now new items are
appended to the end of the LostPackets list, which makes the order
correct when retransmitting.

Handling the case when cloning an unused decoder state:
When the decoder has not successfully decoded a frame yet,
it cannot be cloned. A NULL pointer will be returned all
the way out to VideoCodingModuleImpl::Decode(). When this
happens, the VCM will call Reset() for the dual receiver,
in order to reset the state to kPassive.

Review URL: http://webrtc-codereview.appspot.com/239010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@873 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 18:58:39 +00:00
kma@webrtc.org
4bb141078f A change to Android makefile for building voe auto test.
Review URL: http://webrtc-codereview.appspot.com/255007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@872 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 16:38:22 +00:00
kjellander@webrtc.org
d292b9c9da Unit tests now compile and run at all platforms.
Cosmetic changes to mocks.h.

Review URL: http://webrtc-codereview.appspot.com/253003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@871 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 16:34:52 +00:00
niklas.enbom@webrtc.org
0ba31331a8 Aligning license file with file header
git-svn-id: http://webrtc.googlecode.com/svn/trunk@868 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 09:31:39 +00:00
henrik.lundin@webrtc.org
895870b68f Adding marker bit to RTPanalyze results
Review URL: http://webrtc-codereview.appspot.com/254005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@867 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 08:44:42 +00:00
mikhal@webrtc.org
bb8dfbdee2 updating vpm unit_test following r858
Review URL: http://webrtc-codereview.appspot.com/255005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@865 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 22:07:16 +00:00
turaj@webrtc.org
7395d3d8e9 Addressing issue 115 http://code.google.com/p/webrtc/issues/detail?id=115
Review URL: http://webrtc-codereview.appspot.com/261002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@864 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 17:33:06 +00:00
turaj@webrtc.org
fac5316856 Address the problem that iSAC could not go 16 kHz. It was addressed in P4 but not moved to svn.
Review URL: http://webrtc-codereview.appspot.com/261001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@863 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 17:32:46 +00:00
turaj@webrtc.org
9116cf7c9b Have a guard on computing nrg to avoid wrap-around. This is discovered in a release test. During entropy coding of spectrum the value of "nrg" was too large and after shifting it became negative, resulting in decoder error.
Review URL: http://webrtc-codereview.appspot.com/239016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@862 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 17:29:34 +00:00
mflodman@webrtc.org
29d75b3f7d Only allow increasing capture time.
Review URL: http://webrtc-codereview.appspot.com/259001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@861 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 17:10:49 +00:00
andrew@webrtc.org
18ee6ec8e9 Use __inline in NS-fixed.
The use of "inline" was failing to build on Windows.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/255003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@860 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 17:07:46 +00:00
andrew@webrtc.org
3119ecfec8 Fix audioproc build errors on Windows.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/254003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@859 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 17:00:18 +00:00
mikhal@webrtc.org
c4ab8706f4 video_processing: Adding logic to avoid a memcpy when not required
Review URL: http://webrtc-codereview.appspot.com/255002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@858 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 16:44:24 +00:00
punyabrata@webrtc.org
0ab521f754 Resolving a crash related to strncopy followed by a strcat
call. strncopy will not explicity copy or add a "\0" therefore
strcat did not know where to append the "\n" which was causing
an out of bounds crash.
Because we are checking the length, strcpy should be good enough
as it also copies the "\0". Please note that that I am pre-emptively
adding 2 instead of 1 to the length to take into account of the \n
that will be added later.
Review URL: http://webrtc-codereview.appspot.com/253004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@857 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 15:19:44 +00:00
kjellander@webrtc.org
d6837709cf Fixing VPMUnitTest compilation error on Windows.
It tried to include Visual Leak Detector which is not a tool that is installed/configured by default in the build.

Review URL: http://webrtc-codereview.appspot.com/257002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@854 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 01:51:10 +00:00
henrike@webrtc.org
b37c628ae4 Fixes crash due to r841.
Review URL: http://webrtc-codereview.appspot.com/256004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@853 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 23:53:04 +00:00
kma@webrtc.org
e9f909b575 Move the SetAndroidObjects to VideoCaptureFactory so that ViE can get access to it.
Review URL: http://webrtc-codereview.appspot.com/244002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@852 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 22:24:57 +00:00
andrew@webrtc.org
f1a45d77fb Add missing <stdlib.h> to data_log test.
BUG=
TEST=system_wrappers_unittests

Review URL: http://webrtc-codereview.appspot.com/256002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@851 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 21:44:54 +00:00
andrew@webrtc.org
3134aacd6b Use fileutils for the audio file in voe_auto_test.
BUG=
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/250010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@850 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 21:31:07 +00:00
kma@webrtc.org
27957508a3 Changed Android makefile to make the lastest video render code run.
Review URL: http://webrtc-codereview.appspot.com/247005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@849 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 21:29:50 +00:00
kjellander@webrtc.org
84736882ad Fixing system_wrappers unittests.
Not complete, but enough to include them in the build again.

Review URL: http://webrtc-codereview.appspot.com/241008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@848 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 20:44:24 +00:00
andrew@webrtc.org
2c74bab8b9 Remove unneeded assert and tracing.
This is related to r840.

BUG=
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/239019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@845 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 19:54:20 +00:00
amyfong@webrtc.org
299e2c9ea4 vie_autotest_custom_call.cc - fixed VieAutotestDevcoderObserver to use const int for videoChannel for IncomingCodecChanged, RequestNewKeyFrame
- this caused vie_auto_test to fail for Windows (but fine for Linux & Mac).
Review URL: http://webrtc-codereview.appspot.com/253001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@844 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 19:10:26 +00:00
kjellander@webrtc.org
177bb523bd Fixing system_wrappers unittests.
Not complete, but enough to include them in the build again.

Review URL: http://webrtc-codereview.appspot.com/241008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@842 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 17:10:01 +00:00
henrike@webrtc.org
066f9e5a2f Ray, please verify that this cl fixes the issue. Once the verification has been made, please review:
Henrik A: VoE
Andrew: audio_conference_mixer

Thanks!
Review URL: http://webrtc-codereview.appspot.com/241010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@841 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 23:15:47 +00:00
henrike@webrtc.org
731ecba47d Review URL: http://webrtc-codereview.appspot.com/251002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@840 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 22:49:24 +00:00
braveyao@webrtc.org
1f6d740571 This CL is about to manually reset the ShutdownRenderEvent at StopPlayout().
It could happen that if you want to restart playout, the new sponsored Render thread would catch this event
if the previous Render thread quits before this event is set.
With this modification, the device plugging out/in during talking would be supported well.
Review URL: http://webrtc-codereview.appspot.com/248002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@839 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 21:30:30 +00:00
wu@webrtc.org
88e0a34815 Remove duplicated code.
Review URL: http://webrtc-codereview.appspot.com/251001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@838 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 17:29:44 +00:00
stefan@webrtc.org
f960211f8b Fixes two jitter buffer bugs related to NACK.
Avoid decoding delta frames after a Flush() and after requesting
a key frame due to full NACK list.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/247011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@837 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 16:00:49 +00:00
bjornv@webrtc.org
250cd6f41b Added a VAD unit test to common_audio. At this stage it runs through the API calls, but should later be complemented with tests on a file.
Review URL: http://webrtc-codereview.appspot.com/243002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@832 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 12:45:58 +00:00
stefan@webrtc.org
eb65860720 Reverts the workaround in r823 and solves a macro bug.
The macro bug caused frames to be dropped after being grabbed
for decoding.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/248004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@831 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 12:25:34 +00:00
tina.legrand@webrtc.org
8b1f621e3a Updated gypi for tests to work on osx.
Review URL: http://webrtc-codereview.appspot.com/250002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@830 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 08:57:34 +00:00
amyfong@webrtc.org
ca4666b75c vie wintest added hybrid protection mode
also fixed Max Framerate to reflect its actually the min framerate
Review URL: http://webrtc-codereview.appspot.com/244010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@828 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 21:16:40 +00:00
amyfong@webrtc.org
1e7e60b739 Fixed issue build failling due to vie_autotest_custom_call calling GetBandwidthUsage, which was
changed in r822.
Review URL: http://webrtc-codereview.appspot.com/240014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@827 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 20:53:30 +00:00
amyfong@webrtc.org
51e1bb4e1a vie_autotest_customcall added encoder/decoder observer, maxBitrate set, print call statistics, enable kTraceAll
When creating a new custom call, now able to set start bit rate (default is 1000)

The following modify call options were added
  9. Toggle Encoder Observer
 10. Toggle Decoder Observer
 12. Print Call Statistics

Also set the trace filter to kTraceAll

File defaults new call VGA (640x480)
Review URL: http://webrtc-codereview.appspot.com/239012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@826 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 18:16:35 +00:00
mikhal@webrtc.org
5200a05500 video_coding/jitter_buffer Updating condition on which we return a frame.
Review URL: http://webrtc-codereview.appspot.com/240011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@825 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 16:54:54 +00:00
mikhal@webrtc.org
30f6376802 VP8: Updating codec version: VP8 version will now return the libvpx version used.
Review URL: http://webrtc-codereview.appspot.com/247009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@824 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 16:45:00 +00:00
stefan@webrtc.org
2d28aff785 Workaround for an issue where frames are grabbed for decoding prematurely.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/240013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@823 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 16:13:18 +00:00
stefan@webrtc.org
fbea4e555d Solves two bandwidth estimation issues and measures the sent video bitrate.
Issues solved:
1. Possible overflow when reducing the bandwidth estimate at the send-side
2. A burst of loss reports could make us reduce the rate way too far since
   we reduced the rate relative the current estimate and not the actual
   rate sent.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/244011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@822 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 16:08:29 +00:00
mflodman@webrtc.org
7e4269e9ee Changed VP8 qp min and added noise reduction.
Review URL: http://webrtc-codereview.appspot.com/248003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@821 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 12:59:47 +00:00
mflodman@webrtc.org
8fc663b3ae Don't trigger false ViE SetReceiveCodec warning.
Review URL: http://webrtc-codereview.appspot.com/250001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@820 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-26 11:30:52 +00:00
kjellander@webrtc.org
6b7799021c Fixing build errors on Windows platform. Minor changes...
Review URL: http://webrtc-codereview.appspot.com/241004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@819 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-26 02:38:09 +00:00
andrew@webrtc.org
fdde8b3fb7 Add references to src/ copies for LICENSE etc.
Review URL: http://webrtc-codereview.appspot.com/246007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@818 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-26 01:05:07 +00:00
andrew@webrtc.org
cb18121990 Add an unpacker tool for audioproc debug files.
- It only unpacks audio data at the moment.
- Also switch to Chrome's protoc.gypi for protobuf targets. This reduces
  the complexity of our targets.

Review URL: http://webrtc-codereview.appspot.com/241009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@817 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-26 00:27:17 +00:00
frkoenig@google.com
fc9bcef8c7 Data alignment fix for SSIM.
WebRtc_UWord64[2] wasn't always aligned to 128 bytes, which
is necessary for _mm_store_si128.  By declaring the 
variable as __m128i it will always be 128 bytes aligned.

Incorrect include files.

__m128i is defined in emmintrin.h for visual studio.  Extra include on mac and linux is not a problem.
Review URL: http://webrtc-codereview.appspot.com/239013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@816 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-26 00:07:32 +00:00
phoglund@webrtc.org
78c767f9ba Rewrote codec test to use fake camera.
Tests now fail more cleanly if the input video file is incorrect. Fixed some of the style issues in vie_autotest_codec.

Rewrote the automated standard codec test to use the new fake camera.

Started sketching on a new test case. Wrote a new abstraction called ViEFakeCamera which hides the details of how to thread a file capture device in the typical test case.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/242008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@815 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 12:54:38 +00:00
stefan@webrtc.org
d855c1a4e8 Reverts r807 and fixes the real issue in the VCM.
This fixes an issue in the VCM where we don't wait for a packet to arrive
if the jitter buffer is empty. This also fixes an issue where an old
packet can trigger a packet event signal for a future frame.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/248001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@814 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 11:52:48 +00:00
henrika@webrtc.org
bdb55c806f This CL is an attempt to remove a crash we can see when closing down VoiceEgine.
It can happen that the capture thread tries to access an invalid object after StopPlayout has been called.

I have also extended the usage of the new ScopedCOMInitializer to all threads. See this step as code cleanup.
Review URL: http://webrtc-codereview.appspot.com/239014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@813 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 11:03:28 +00:00
henrika@webrtc.org
a6c23357c0 Solves crash in ADM API unit test for Core Audio on Windows
Review URL: http://webrtc-codereview.appspot.com/244009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@812 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 08:31:33 +00:00
henrika@webrtc.org
5423bc2d0b Adds correct absolute paths to all input files in ADM functional unit tests.
Files are now read and played out correctly.
Review URL: http://webrtc-codereview.appspot.com/246006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@811 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 08:24:20 +00:00
kma@webrtc.org
ca325ececd Corrected a linux build error introduced in issue 246005.
Review URL: http://webrtc-codereview.appspot.com/246008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@809 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 02:36:09 +00:00
wjia@webrtc.org
f0cd394a2e Put fwrite calls under corresponding macros since they shouldn't show up in release build.
This also make chromeos build happy.
BUG=none
TEST=compile
Review URL: http://webrtc-codereview.appspot.com/247006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@808 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 00:40:43 +00:00
mikhal@webrtc.org
f31826e17b adding a wait on the decode thread when no frames are available
Review URL: http://webrtc-codereview.appspot.com/246009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@807 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 00:20:54 +00:00
mikhal@webrtc.org
a412924c0e VP8:Setting number of cores based on image size
Review URL: http://webrtc-codereview.appspot.com/242010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@806 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 00:02:43 +00:00
kma@webrtc.org
913644b92d For commiting changes in CL 277002, due to file structure changes introduced during
the review of the code.
Review URL: http://webrtc-codereview.appspot.com/246005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@805 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-24 21:36:33 +00:00
henrike@webrtc.org
0d0037c2fd Return cached data instead of sleeping in CpuWrapperMac (shaves 2s off WebrtcMediaEngine creation time on Mac).
Review URL: http://webrtc-codereview.appspot.com/226005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@804 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-24 15:48:14 +00:00
phoglund@webrtc.org
0a9c318c9f The fread result is no longer ignored.
Changed unsigned longs into uint64_t to be a bit more portable.

Merge branch 'master' into fake_camera

Conflicts:
	src/video_engine/main/test/AutoTest/source/vie_autotest_base.cc

Removed unnecessary use of WebRTC types. Fixed style issues.

Fixed style issues. Added comments where needed.

(After review) Made the standard base test not mirror the render stream since that is assumed to be tested in the render module. Renamed functions accordingly.

Fixed merge errors.
Merge branch 'master' into fake_camera

Conflicts:
	src/video_engine/main/interface/vie_capture.h
	src/video_engine/main/test/AutoTest/automated/vie_standard_integration_test.cc
	src/video_engine/main/test/AutoTest/interface/vie_autotest.h
	src/video_engine/main/test/AutoTest/interface/vie_autotest_defines.h
	src/video_engine/main/test/AutoTest/source/vie_autotest_base.cc
	src/video_engine/main/test/AutoTest/source/vie_autotest_linux.cc
	src/video_engine/main/test/AutoTest/vie_auto_test.gypi

Merge branch 'extended_tests' into fake_camera

Conflicts:
	src/video_engine/main/test/AutoTest/automated/vie_standard_integration_test.cc
	src/video_engine/main/test/AutoTest/interface/vie_autotest.h
	src/video_engine/main/test/AutoTest/source/vie_autotest_base.cc
	src/video_engine/main/test/AutoTest/source/vie_autotest_linux.cc
	src/video_engine/main/test/AutoTest/vie_auto_test.gypi

More updates after review.

Updates after review.

Added new automated test. - Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

Added comments to the new test.

- Fixed a bug which caused test error messages to not get shown.

- Added extended and API tests.
- Abstracted out an integration test base class since all integration
tests set up the exact same way.

- The ViETest::TestError static method will now assert using GTest
asserts if we are running in GTest mode. This gets rid of the hard
asserts that get run otherwise. The hard asserts are still in when using
"classic" mode. TestError will use neither GUnit nor hard asserts if
VIE_ASSERT_ERROR is not defined.
- Formatted vie_autotest_defines.h according to Google style rules.

- Extracted a method for finding a capture device on the system. This
removes a fair bit of logic from the huge test method (mostly straight
statements remain there now).

Rebase from svn.

- Whitespace fixes after review.

Fixed presubmit warning.

- Fixed cpplint.py warnings.

Fixed merge error.

Merge branch 'extended_tests' into fake_camera

Conflicts:
	src/video_engine/main/test/AutoTest/automated/vie_extended_integration_test.cc
	src/video_engine/main/test/AutoTest/automated/vie_standard_integration_test.cc
	src/video_engine/main/test/AutoTest/helpers/vie_window_creator.cc
	src/video_engine/main/test/AutoTest/interface/vie_autotest.h
	src/video_engine/main/test/AutoTest/source/vie_autotest_base.cc
	src/video_engine/main/test/AutoTest/source/vie_autotest_linux.cc
	src/video_engine/main/test/AutoTest/vie_auto_test.gypi

More updates after review.

Updates after review.

Added new automated test. - Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

Added comments to the new test.

- Fixed a bug which caused test error messages to not get shown.

- Added extended and API tests.
- Abstracted out an integration test base class since all integration
tests set up the exact same way.

- The ViETest::TestError static method will now assert using GTest
asserts if we are running in GTest mode. This gets rid of the hard
asserts that get run otherwise. The hard asserts are still in when using
"classic" mode. TestError will use neither GUnit nor hard asserts if
VIE_ASSERT_ERROR is not defined.
- Formatted vie_autotest_defines.h according to Google style rules.

- Extracted a method for finding a capture device on the system. This
removes a fair bit of logic from the huge test method (mostly straight
statements remain there now).

Rebase from svn.

- Whitespace fixes after review.

Fixed presubmit warning.

- Fixed cpplint.py warnings.

Fixed merge error.

Fixed cpplint.py warnings.

Merge branch 'extended_tests' into fake_camera

Conflicts:
	src/video_engine/main/test/AutoTest/automated/vie_api_integration_test.cc
	src/video_engine/main/test/AutoTest/automated/vie_extended_integration_test.cc
	src/video_engine/main/test/AutoTest/automated/vie_integration_test_base.cc
	src/video_engine/main/test/AutoTest/automated/vie_standard_integration_test.cc
	src/video_engine/main/test/AutoTest/helpers/vie_window_creator.cc
	src/video_engine/main/test/AutoTest/interface/vie_autotest.h
	src/video_engine/main/test/AutoTest/source/vie_autotest_base.cc
	src/video_engine/main/test/AutoTest/source/vie_autotest_linux.cc
	src/video_engine/main/test/AutoTest/source/vie_autotest_main.cc
	src/video_engine/main/test/AutoTest/vie_auto_test.gypi

Revert "Revert "- Whitespace fixes after review.""

This reverts commit 3da2a148814e8dea78f73d3feeb32dce690dc2d4.

Revert "- Whitespace fixes after review."

This reverts commit fac670ca313580fb883191ae919091a2637ad0af.

- Whitespace fixes after review.

- Wrote a "file capture device" which is a kind of fake capture device. It reads a YUV file from disk and pretends that it is what the "camera" is seeing. This makes is possible to run tests based on video input without having an actual physical camera. This is good because physical cameras are quite unreliable. - Rewrote the standard mirrored preview loopback test so it can use the new file capture device. The old "classic" test is preserved. I tried to minimize duplication between the classic test case and the new one, which turned out to be quite painful. - There are some rough edges left in in the code. Suggested improvements is to get rid of the error counting mechanism since the code seems to assume that TestError invocations cause hard asserts anyway. The code will segfault for certain errors if the hard asserts doesn't happen, which means the error counting mechanism is unnecessary. This, by the way, could be a problem for the new test since it doesn't cause hard asserts. - Fixed comments for the thread wrapper and the external capture device interface.

- Extracted a method for finding a capture device on the system. This removes a fair bit of logic from the huge test method (mostly straight statements remain there now).

- The ViETest::TestError static method will now assert using GTest asserts if we are running in GTest mode. This gets rid of the hard asserts that get run otherwise. The hard asserts are still in when using "classic" mode. TestError will use neither GUnit nor hard asserts if VIE_ASSERT_ERROR is not defined. - Formatted vie_autotest_defines.h according to Google style rules.

- Added extended and API tests. - Abstracted out an integration test base class since all integration tests set up the exact same way.

- Fixed a bug which caused test error messages to not get shown.

Added comments to the new test.

- Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

- Fixed cpplint.py warnings.

Fixed presubmit warning.

- Whitespace fixes after review.

Rebase from svn.

- Extracted a method for finding a capture device on the system. This removes a fair bit of logic from the huge test method (mostly straight statements remain there now).

- The ViETest::TestError static method will now assert using GTest asserts if we are running in GTest mode. This gets rid of the hard asserts that get run otherwise. The hard asserts are still in when using "classic" mode. TestError will use neither GUnit nor hard asserts if VIE_ASSERT_ERROR is not defined. - Formatted vie_autotest_defines.h according to Google style rules.

- Added extended and API tests. - Abstracted out an integration test base class since all integration tests set up the exact same way.

- Fixed a bug which caused test error messages to not get shown.

Added comments to the new test.

- Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/247004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@803 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-24 15:33:07 +00:00
andrew@webrtc.org
537096a5c1 Remove unnecessary objective-c compiler flags.
Review URL: http://webrtc-codereview.appspot.com/239011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@802 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-24 15:07:08 +00:00
phoglund@webrtc.org
c63f788e0f Added fake camera, rewrote one test to use it.
Wrote a "file capture device" which is a kind of fake capture device. It reads a YUV file from disk and pretends that it is what the "camera" is seeing. This makes is possible to run tests based on video input without having an actual physical camera. This is good because physical cameras are quite unreliable.

Rewrote the standard mirrored preview loopback test so it can use the new file capture device. The old "classic" test is preserved. I tried to minimize duplication between the classic test case and the new one, which turned out to be quite painful.

There are some rough edges left in in the code. Suggested improvements is to get rid of the error counting mechanism since the code seems to assume that TestError invocations cause hard asserts anyway. The code will segfault for certain errors if the hard asserts doesn't happen, which means the error counting mechanism is unnecessary. This, by the way, could be a problem for the new test since it doesn't cause hard asserts.

Fixed comments for the thread wrapper and the external capture device interface.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/224003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@801 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-24 13:20:09 +00:00
henrika@webrtc.org
bf478faebb Ensures that ADM unit tests builds on all platforms.
Review URL: http://webrtc-codereview.appspot.com/240009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@800 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-24 10:31:02 +00:00
andrew@webrtc.org
f1a605cad6 Update DEPS to support Mac clang build.
Review URL: http://webrtc-codereview.appspot.com/244003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@797 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 15:29:16 +00:00
stefan@webrtc.org
5eb64f06be Fix BitrateSent() API when having a default RTP module.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/242004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@796 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 13:42:50 +00:00
stefan@webrtc.org
158f496030 Fixes a rate control bug in the VP8 wrapper.
Changes how we signal frame rate and frame durations to the encoder. Rather
than changing the time base, we now only modify the frame durations, while
keeping the timebase constant. The frame duration is currently calculated
from the average input frame rate. Ideally, the frame duration should
be calculated as the timestamp diff, which is the real duration of a
frame, but the encoder doesn't seem to like too varying durations.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/247001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@795 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 13:15:16 +00:00
stefan@webrtc.org
ead87b5051 Fix potential issue where frame buffers might be freed while being decoded.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/243004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@791 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 06:46:37 +00:00
stefan@webrtc.org
2b0f094c8f Avoid reallocating the decodedImage for every decoded frame.
Also made sure the right size is allocated.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/240004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@790 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 06:39:03 +00:00
mikhal@webrtc.org
ee3dfa6f43 Review URL: http://webrtc-codereview.appspot.com/241007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@789 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 00:46:09 +00:00
mikhal@webrtc.org
1af915d8ae video_coding: vp8: Updating error propagation threshold
Review URL: http://webrtc-codereview.appspot.com/246002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@788 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-20 18:19:18 +00:00
kma@webrtc.org
d75889e2eb Change of Android makefiles to build latest video coding code.
Review URL: http://webrtc-codereview.appspot.com/239008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@786 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-20 16:28:56 +00:00
henrika@webrtc.org
7cf893743a git-svn-id: http://webrtc.googlecode.com/svn/trunk@785 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-10-20 12:30:35 +00:00
henrika@webrtc.org
cedbb036d1 [Issue 101] Solves memory leak on Windows
git-svn-id: http://webrtc.googlecode.com/svn/trunk@784 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-20 12:11:45 +00:00
stefan@webrtc.org
c4d1983b7b Changes in rtp_format_vp8_unittest to match the changes in CL 774.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/241006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@782 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-20 08:19:34 +00:00
mflodman@webrtc.org
ae499a2ac8 Set correct codec info before sending frame to VCM.
Review URL: http://webrtc-codereview.appspot.com/240003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@780 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-20 05:55:46 +00:00
kjellander@webrtc.org
81f25f9ff8 Fixing build errors on Windows platform. Minor changes...
Review URL: http://webrtc-codereview.appspot.com/241004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@779 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 20:06:56 +00:00
wu@webrtc.org
f3f2f6abdb * Add include_internal_video_capture and include_internal_video_render to include/exclude the internal VCM and VRM.
* Split the WEBRTC_VIDEO_EXTERNAL_CAPTURE_AND_RENDER into WEBRTC_INCLUDE_INTERNAL_VIDEO_CAPTURE and WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER.
* Add DummyDeviceInfo for the case when WEBRTC_INCLUDE_INTERNAL_VIDEO_CAPTURE is not defined.
Review URL: http://webrtc-codereview.appspot.com/224005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@778 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 18:42:17 +00:00
henrike@webrtc.org
509c9c5d09 operator + is evaluated before ?:
Parenthesis ensures the intended behavior.
Review URL: http://webrtc-codereview.appspot.com/239003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@777 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 18:31:01 +00:00
henrike@webrtc.org
4df8c9a2ed Review URL: http://webrtc-codereview.appspot.com/243001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@776 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 18:30:25 +00:00
andrew@webrtc.org
7ecdf585cb Enable chromium_code:1 in the Chrome build.
Review URL: http://webrtc-codereview.appspot.com/240001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@775 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 17:53:56 +00:00
stefan@webrtc.org
ffd28f95c5 Request key frames to battle error propagation.
The VP8 decoder wrapper will request key frames 30 frames after seeing
a packet loss, if it hasn't received a state refresh (only possible
through key frames in this version).

For this to be possible the jitter buffer has been made aware of
picture ids to be able to detect frame losses. Legacy JB code to
handle streams without marker bits was also removed since it
conflicts with streams with FEC.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/239002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@774 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 15:55:39 +00:00
mikhal@webrtc.org
d0752c370d video_coding: Update to hybrid mode: Set FEC values for zero below a threshold.
Review URL: http://webrtc-codereview.appspot.com/245001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@773 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 15:48:30 +00:00
mflodman@webrtc.org
c693bac6e7 Only start ViEPerformanceMonitor when needed.
Tested by taking the added part in base extended test and running in Standard test with cpu threashold in ViEPeroformanceMonitor manually changed to 0.

Review URL: http://webrtc-codereview.appspot.com/240005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@772 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 13:40:58 +00:00
phoglund@webrtc.org
b5475d0076 vie_auto_test will now obey the Mac .mm rules for files including objective-c code.
Fixed the Windows build.

Fixed whitespace.

Split the platform-specific code for creating a window manager into separate source files since the mac one must be suffixed .mm and not .cc when we happen to use objective-c code. Tested on Linux.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/214009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@771 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 10:59:39 +00:00
bjornv@webrtc.org
4c636764b7 Updated the AEC delay logging to output values in ms. PB output updated.
Review URL: http://webrtc-codereview.appspot.com/223003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@770 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 08:47:40 +00:00
mflodman@webrtc.org
cc412c1735 Remove second instance of ViE PerformanceMonitor.
Review URL: http://webrtc-codereview.appspot.com/244001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@769 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 08:27:30 +00:00
mflodman@webrtc.org
ce8813da4e Using id instead of name when setting Mac/QTKit capture device.
Review URL: http://webrtc-codereview.appspot.com/241002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@768 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 06:45:16 +00:00
andrew@webrtc.org
4d5d5c1267 Reorganize the audio_processing source.
- Remove main and source directories.
- Change .gyp, .gypi and Android.mk files correspondingly. No other
  source changes.

Review URL: http://webrtc-codereview.appspot.com/241001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@767 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 01:40:33 +00:00
andrew@webrtc.org
5d3bdf71ab Fix clang warnings in ViE autotest.
Review URL: http://webrtc-codereview.appspot.com/239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@766 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 01:09:41 +00:00
wu@webrtc.org
8fd93d4d96 Move DeliverCapturedFrame from private to protected.
Review URL: http://webrtc-codereview.appspot.com/246001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@765 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 00:16:36 +00:00
bjornv@webrtc.org
52eddf7378 Made Tina, Andrew and Jan as OWNERS to entire common_audio and removed the sub-OWNERS files. Let me know if that's fine.
Review URL: http://webrtc-codereview.appspot.com/225006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@763 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-18 07:57:04 +00:00
stefan@webrtc.org
5b15cfc6dd Fix BWE unit test build issue
git-svn-id: http://webrtc.googlecode.com/svn/trunk@762 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-18 07:22:33 +00:00
kjellander@webrtc.org
61f07c3184 I have made a small fix so it will execute properly from the default working directory location (trunk), finding its resource files.
The ApmTest.Process test is still failing and needs to be resolved.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/194002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@761 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-18 06:54:58 +00:00
henrik.lundin@webrtc.org
5dedd0ee38 Handling of white-space in DataLog::Combine
The Combine method cannot handle white-space. Adding a comment to
the header file saying this, and modifying the unittests. Also,
adding a new unittest to test the method.

Review URL: http://webrtc-codereview.appspot.com/217001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@760 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-18 05:45:08 +00:00
amyfong@webrtc.org
929789b528 vie_auto_test - moved custom call specific functions to be static, added video protect method to custom call
- moved all of the custom call specific functions out of vie_autotest.h and into vie_autotest_custom_call.cc
- added option to modify a running call's video protection method
Review URL: http://webrtc-codereview.appspot.com/234001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@759 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-17 21:57:08 +00:00
wu@webrtc.org
76aea651ff When _audioConfigured, should not try to use the _video.
Review URL: http://webrtc-codereview.appspot.com/224004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@758 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-17 21:40:32 +00:00
bjornv@webrtc.org
3765bd2cc2 Added AEC delay logging metrics to VoE. Echo metrics and delay logging metrics are enabled simultaneously through the SetEcMetricsStatus(). Updated standard and extended VoE tests.
class VoEAudioProcessing
-API renaming:
  SetEchoMetricsStatus() to SetEcMetricsStatus()
  GetEchoMetricsStatus() to GetEcMetricsStatus()
  since delay logging is not strictly an echo metric.
-New API:
  GetEcDelayMetrics()
-Implementations
  --SetEcMetricsStatus() sets same status to all EC related metrics, currently Echo Metrics and Delay Logging.
  --GetEcMetricsStatus() gets an error if all EC related metrics don't have the same status.
  --GetEcDelayMetrics() gets the median and standard deviation of AEC internal delay (on a block by block basis).

class VoECallReport
The changes above leads to changes in the Call Report.
-New API:
  GetEcDelaySummary()
-API updates:
  ResetCallReportStatistics()
  WriteReportToFile()

auto_tests updates:
-Standard test, with new Call Report calls and APM calls
-Extended test, with new Call Report calls and APM calls
Review URL: http://webrtc-codereview.appspot.com/187004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@754 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-17 08:49:23 +00:00
wu@webrtc.org
f10ea31211 Add IncomingFrameI420 to ViEExternalCapture interface to take captured video frame buffer as 3 planes.
Review URL: http://webrtc-codereview.appspot.com/219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@753 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-14 17:16:04 +00:00
marpan@webrtc.org
14aaaf116a Some re-organization of the fec-uep code: updated protection modes, comments, and some variable/function re-naming.
Review URL: http://webrtc-codereview.appspot.com/231001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@752 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-14 16:28:02 +00:00
wu@webrtc.org
55c39f0940 Add mallinath@webrtc.org and wu@webrtc.org as the capture owner for US office.
Review URL: http://webrtc-codereview.appspot.com/230001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@751 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-14 15:34:19 +00:00
wu@webrtc.org
58691ebb97 Remove the DestroyDeviceInfo for mac video capture. (This is missed in r731.)
Review URL: http://webrtc-codereview.appspot.com/229001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@750 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-14 15:13:16 +00:00
stefan@webrtc.org
d0bdab0128 Adding API to get sent total bitrate, FEC bitrate and NACK bitrate.
Also adding tests for this in vie_auto_test.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/199001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@749 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-14 14:24:54 +00:00
phoglund@webrtc.org
26c041673f Added more tests, fixed a bug and refactored.
Fixed merge error.

Fixed cpplint.py warnings.

Fixed presubmit warning.

Whitespace fixes after review.

Rebase from svn.

Extracted a method for finding a capture device on the system. This removes a fair bit of logic from the huge test method (mostly straight statements remain there now).

The ViETest::TestError static method will now assert using GTest asserts if we are running in GTest mode. This gets rid of the hard asserts that get run otherwise. The hard asserts are still in when using "classic" mode. TestError will use neither GUnit nor hard asserts if VIE_ASSERT_ERROR is not defined. - Formatted vie_autotest_defines.h according to Google style rules.

Added extended and API tests. - Abstracted out an integration test base class since all integration tests set up the exact same way.

Fixed a bug which caused test error messages to not get shown.

Added comments to the new test.

Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/188002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@747 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-14 13:00:20 +00:00
bjornv@webrtc.org
2111d3b0b0 Removed the vad_const files and added the constants to the files where they are
used. Having them in a separate file did not add anything in readability or
conceptual overview.
Review URL: http://webrtc-codereview.appspot.com/230004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@746 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-14 12:58:34 +00:00
wu@webrtc.org
4ee906d297 When WEBRTC_VIDEO_ENGINE_FILE_API is not defined, disable the code in vie_file_impl.cc and vie_file_image.cc so that we can remove the libjpeg dependency. Also disable the auto test for the vie file api.
Review URL: http://webrtc-codereview.appspot.com/227001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@739 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-13 17:56:38 +00:00
marpan@webrtc.org
5a3e20f678 Removed unused variables (build error) for test_fec.
Review URL: http://webrtc-codereview.appspot.com/223001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@738 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-13 16:59:24 +00:00
pwestin@webrtc.org
1da1ce0da5 First implementation of simulcast, adds VP8 simulcast to video engine.
Changed API to RTP module
Expanded Auto test with a test for simulcast
Made the video codec tests compile
Added the vp8_simulcast files to this cl
Added missing auto test file
Review URL: http://webrtc-codereview.appspot.com/188001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@736 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-13 15:19:55 +00:00
stefan@webrtc.org
4c059d87b3 Add metric for number of packets discarded by JB due to not being decodable
Also fixes a couple of bugs related to sequence number wrap found while
testing.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/218001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@732 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-13 07:35:37 +00:00
wu@webrtc.org
77d7d5455e Replace the DestroyDeviceInfo with a virtual destructor.
Review URL: http://webrtc-codereview.appspot.com/212005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@731 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-12 16:57:53 +00:00
amyfong@webrtc.org
e5542a0af5 Add file record and play functions to voe_cmd_test, fix Play local file (path was incorrect)
Fixed:
	24. Play local file (audio_long16.pcm) 
New:
	34. Record a PCM file 
	35. Play a previously recorded PCM file locally 
	36. Play a previously recorded PCM file as microphone 
Review URL: http://webrtc-codereview.appspot.com/209001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@729 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-11 20:30:56 +00:00
amyfong@webrtc.org
6330cf2a14 Fixed ViE AutoTest trace file names to be consistent
Fixed some space issues in vie_autotest_custom_call.cc
Fixed incorrect default codec W&H for I420 in vie_autotest_custom_call.cc
Added functionality to modify a running custom call.  The following options were added:
0. Finished modifying custom call
1. Change Video Codec
2. Change Video Size by Common Resolutions
3. Change Video Size by Width & Height
4. Change Video Device
5. Record Incoming Call
6. Record Outgoing Call
7. Play File on Video Channel(Assumes you recorded incoming & outgoing call)
8. Print Call information

Tested with r670, builds fine on Ubuntu & Win7.  Mac is not building due to changes in r666, but this patch should be fine on top of it mac as well (compiles fine with r661).
Review URL: http://webrtc-codereview.appspot.com/188003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@728 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-11 18:17:22 +00:00
wu@webrtc.org
ea89922b56 Add VideoCaptureFactory so that we don't need to expose VideoCaptureImpl.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/213002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@727 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-11 17:13:51 +00:00
andrew@webrtc.org
199f4defd3 Rename all .cc files which include Objective-C headers to .mm.
This allows the Mac Make build to pass. We were hacking it in XCode with "-x objective-c++", but gyp/Make doesn't seem to accept that flag.

Also switch Objective-C #includes to #imports.

There is one file missing from this: vie_autotest_main.cc, because it's required on multiple platforms. I'm not immediately sure what the best approach is there, but the Objective-C headers should be somehow hidden.
Review URL: http://webrtc-codereview.appspot.com/153005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@726 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-11 15:43:35 +00:00
henrike@webrtc.org
a0258defd4 Fixes test build errors (warnings treated as errors) in system_wrappers.
Review URL: http://webrtc-codereview.appspot.com/212003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@725 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-11 14:49:27 +00:00
henrik.lundin@webrtc.org
26c9ff983e Add dummy implementation of DataLog::Combine method
The dummy implementations of class methods are needed when
building without support for data logging (i.e., when
enable_data_logging != 1). The Combine method was missing
from data_log_dummy.cc.

Review URL: http://webrtc-codereview.appspot.com/220003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@724 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-11 14:43:41 +00:00
stefan@webrtc.org
791eec7424 Add API to get the number of packets discarded by the video jitter buffer due to being too late.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/200001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@723 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-11 07:53:43 +00:00
stefan@webrtc.org
06887aebae Fixes two bugs when decoding with packet losses.
Disable _missingFrame bit since we can't set it correctly with FEC.

No longer return more than one decoded frame per Decode() call.
This is a work-around for a bug where the frame info map was popped more often than items were added to the map.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/215001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@722 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-10 14:17:46 +00:00
henrik.lundin@webrtc.org
1843664f2a DataLog: Changing from common_types to typedefs
The file common_types.h cannot be used in data_log_c.h, since
the latter is a pure C header file, and common_types.h is
not. Changing to typedefs.h instead.

Review URL: http://webrtc-codereview.appspot.com/216001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@719 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-10 09:56:52 +00:00
tommi@webrtc.org
c0b2250b20 Fix the Windows build.
Review URL: http://webrtc-codereview.appspot.com/213004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@717 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-10 08:43:33 +00:00
henrik.lundin@webrtc.org
d855bd4d6f C wrapper for DataLog class
A pure C wrapper for the DataLog class was created. Since templates
are not supported in C, the InsertCell method of the DataLog class
must be wrapped using one wrapper function for each data type. So far,
the wrapper includes int, float, double, Word32, UWord32, and Word64.

Unittests were created for the wrapper. A separate helper file was
included in the tests. This helper file was implemented as a C file,
in order to actually test the C linkage of the wrapper.
The unittests for DataLog were cloned to make versions that do the same
things but through the C wrapper interface. Restructured the code
so that the log file verification was not duplicated.

Review URL: http://webrtc-codereview.appspot.com/195003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@715 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-10 08:06:17 +00:00
tommi@webrtc.org
6364d128a1 Fix a couple of build warnings.
Review URL: http://webrtc-codereview.appspot.com/214004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@714 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-10 08:04:59 +00:00
phoglund@webrtc.org
e95458c30a Started rewriting video_engine tests to use GUnit.
- Added comments to the new test.
- Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/168002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@713 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-10 07:23:51 +00:00
kjellander@webrtc.org
25e0b8e3a0 Python output flag and keyframe interval flags.
Refactored main method into using 6 helper methods for better overview.

Review URL: http://webrtc-codereview.appspot.com/207001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@710 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-07 07:52:00 +00:00
kjellander@webrtc.org
a31b254084 Python output flag and keyframe interval flags.
Refactored main method into using 6 helper methods for better overview.

Review URL: http://webrtc-codereview.appspot.com/207001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@709 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-07 06:50:22 +00:00
mikhal@webrtc.org
80dd19be0a vplib tests: Removing old and unused file and directories.
Note that the convert_test and scale_test directories are also removed. 
Review URL: http://webrtc-codereview.appspot.com/208001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@708 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-06 22:57:06 +00:00
henrike@webrtc.org
bf54ef9bb7 Removed code under a non-existing define.
Review URL: http://webrtc-codereview.appspot.com/193006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@706 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-06 18:14:25 +00:00
henrike@webrtc.org
1a2933c71a Fixes a Valgrind warning triggering when the number of pending messages hit the limit.
Review URL: http://webrtc-codereview.appspot.com/200002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@705 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-06 17:55:56 +00:00
andrew@webrtc.org
b2d4921f3b Remove trailing whitespace in AudioDevice.
(That I introduced...)
Review URL: http://webrtc-codereview.appspot.com/198002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@703 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-06 16:34:36 +00:00
mikhal@webrtc.org
d6132f54d2 Review URL: http://webrtc-codereview.appspot.com/193007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@702 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-06 16:23:38 +00:00
kjellander@webrtc.org
35a1756502 First version of video quality measurement program and test framework.
See https://docs.google.com/a/google.com/document/d/1w6Nrxw6yTg_sDu18Ux8oZPEMo5F_R-zt62udrmmTeOc/edit?hl=en_US
for background, details and additional instructions on usage.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/175001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@700 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-06 06:44:54 +00:00
andrew@webrtc.org
3ce62fcfe4 Move merge_libs targets to their own gyp.
The main reason is to depend on all ("*") targets in voice_engine.gyp and video_engine.gyp. We don't want the merge_lib targets building by default, since they do funny stuff like delete some libraries.
Review URL: http://webrtc-codereview.appspot.com/191003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@699 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-06 01:03:18 +00:00
kma@webrtc.org
af57de006a Some code style changes in audio_processing/ns/main/source/ by Astyle,
with a little manual modification.
Review URL: http://webrtc-codereview.appspot.com/201002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@698 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 23:36:01 +00:00
henrik.lundin@webrtc.org
01ca01f6e6 Adding neteq_tests to modules tests
Also moving neteq_tests.gyp and renaming to gypi. Cleaning up a
little in neteq_tests.gypi.

Review URL: http://webrtc-codereview.appspot.com/191004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@696 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 20:38:19 +00:00
kma@webrtc.org
bbc1f10187 Changed modules/audio_processing/utility/Android.mk, to correct a build error in
Android with the change from version r674.
Review URL: http://webrtc-codereview.appspot.com/197003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@694 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 18:09:02 +00:00
kma@webrtc.org
bf39ff4271 Some general optimization in NS.
No big effort in introducing new style.
Speed improved ~2%.
Bit exact.
Will introduce mulpty-and-accumulate and sqrt_floor next, which increase speed another 2% or so.

Note: In function WebRtcNsx_DataAnalysis, did the block separation because I found one "if" case is more frequent than "else" within a for loop; rest is kind of code re-aligning.
Review URL: http://webrtc-codereview.appspot.com/181002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@692 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 17:10:06 +00:00
kma@webrtc.org
a58224f9f0 Introduced a SPL inline function (multiple-accumulate), for preformance in ARMv7.
It's used in quite some occations over many modules.
Review URL: http://webrtc-codereview.appspot.com/178004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@691 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 16:44:11 +00:00
stefan@webrtc.org
4b6f747373 Fixes a newly introduced bug in the jitter buffer where buffer reallocation
causes corrupt pointers.
Review URL: http://webrtc-codereview.appspot.com/186003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@688 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-04 06:58:39 +00:00
stefan@webrtc.org
93d216c23f Fixed bug in jitter buffer which caused the missingFrames bit to never be set.
Also updated the VP8 wrapper to return fully concealed frames (for rendering).

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/190003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@687 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-04 06:48:11 +00:00
stefan@webrtc.org
61b4abf1f8 Proper use of frame rate argument in generic_codec_test.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/181005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@686 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-04 06:40:21 +00:00
mikhal@webrtc.org
e06be4f678 video coding tests: Adding ssimFrame to interface
Review URL: http://webrtc-codereview.appspot.com/188004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@685 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-03 22:54:43 +00:00
mikhal@webrtc.org
ae7a0522c5 video_coding robustness: Updating hybrid mode's settings
1. Disabling adjustment factor - temporary update. 
2. Enabling a windowed filtered loss for the hybrid mode.  
Review URL: http://webrtc-codereview.appspot.com/192003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@684 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-03 22:54:34 +00:00
marpan@google.com
f1f3fb33b5 Update to rate-mismatch factor in media_opt_util.
Review URL: http://webrtc-codereview.appspot.com/193003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@678 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-03 19:09:45 +00:00
andrew@webrtc.org
f458916145 Returning errors if any of the Init() settings in VoE fail.
There's no reason to try to continue if these simple settings fail; better to know about it immediately.

Also, readjusting the indentation to avoid breaking strings over several lines. This bends GStyle a bit, but it's well worth it to avoid the common "forgot to add a space" error.
Review URL: http://webrtc-codereview.appspot.com/173003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@676 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-03 15:22:28 +00:00
stefan@webrtc.org
5b91464edf Allow an aggregated partition to spill over to a new packet.
Adds support for the case where the partition 0 and parts of partition 1
are transmitted in packet 1, and the end of partition 2 is transmitted
in packet 2.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/181003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@675 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-03 10:26:12 +00:00
bjornv@google.com
1ba3dbecbb Adds possibility to log delay estimates in AEC.
Review URL: http://webrtc-codereview.appspot.com/178001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@674 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-03 08:18:10 +00:00
stefan@webrtc.org
f72c36763f Reverting changelist 666 since it broke the build on Mac.
TBR=mflodman
Review URL: http://webrtc-codereview.appspot.com/187003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@673 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-03 07:37:41 +00:00
andrew@webrtc.org
6d169f2474 Fix Mac build error in vie_auto_test introduced in r666.
COCOA_RENDERING was undefined. Committing without review.
Review URL: http://webrtc-codereview.appspot.com/191002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@672 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-03 06:00:42 +00:00
mflodman@webrtc.org
5eec6cf29a Started rewriting video_engine tests to use GUnit.
- Added comments to the new test.
- Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/168002
Patch from Patrik Hoglund <phoglund@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@666 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-29 12:24:13 +00:00
punyabrata@webrtc.org
6b6d08164f Remove assert "currentVoEMicLevel <= kMaxVolumeLevel". We ran into an issue on a Linux system where the currentVoEMicLevel was in fact greater than the kMaxVolumeLevel. Therefore we are removing this assert and capping the currentMicLevel to the maxVolumeLevel when this case is detected.
Review URL: http://webrtc-codereview.appspot.com/180001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@661 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-28 17:45:03 +00:00
kma@google.com
c611b1a950 Bit-exact with non-Neon version.
Review URL: http://webrtc-codereview.appspot.com/180002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@660 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-28 16:03:38 +00:00
bjornv@google.com
0beae6798d Removed level estimator calls, since it is not supported. There are still one place left; used within SetRTPAudioLevelIndicationStatus(). The error return value of level_estimator() has no effect there.
The VoE auto tests have been updated as well.
Review URL: http://webrtc-codereview.appspot.com/178003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@658 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-28 14:08:19 +00:00
andrew@webrtc.org
18421f2063 Remove unnecessary include from NS interface.
http://code.google.com/p/webrtc/issues/detail?id=46
Review URL: http://webrtc-codereview.appspot.com/183001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@656 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-27 19:50:52 +00:00
amyfong@webrtc.org
6a23ad5702 Fixed the CameraCap button to say Version, also change the function name inside ChannelDlg.cpp
Review URL: http://webrtc-codereview.appspot.com/182001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@655 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-27 19:19:10 +00:00
amyfong@webrtc.org
2d08d43206 * Added modification of Start Bit Rate to vie_auto_test_custom_call
* Added minor spacing and ":" for user input during vie_auto_test_custom_call
* Changed the default Video Port to 11111 and Audio Port to be 11113 to bring it inline with the WindowsTest application for ViE
Review URL: http://webrtc-codereview.appspot.com/181001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@654 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-27 17:46:45 +00:00
mikhal@webrtc.org
848fad23c6 video_coding: Updating media opt test - fixing call to protection callback.
Review URL: http://webrtc-codereview.appspot.com/179003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@653 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-27 16:30:59 +00:00
xians@google.com
49d025f262 Get the right guid str for GetRecordingDeviceName
Bug=http://code.google.com/p/webrtc/issues/detail?id=99
Test=none
Review URL: http://webrtc-codereview.appspot.com/183002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@652 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-27 14:43:06 +00:00
bjornv@google.com
a2c6ea09b0 Removed a segmentation fault error when processing near_file only.
Review URL: http://webrtc-codereview.appspot.com/174001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@650 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-27 08:04:45 +00:00
kma@google.com
961885a8bb In spl, introduced function WebRtcSpl_Sat32To16(), and changed file resample_by_2.c, both for optimization in ARMv7.
Review URL: http://webrtc-codereview.appspot.com/140010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@649 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-26 16:35:25 +00:00
mikhal@webrtc.org
e185e9f68a video_coding: updates to jitter buffer logic: Make sure that every frame is inserted only once to the list.
Review URL: http://webrtc-codereview.appspot.com/165001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@648 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 22:02:40 +00:00
turajs@google.com
cf136186f5 Deleting matlab files
git-svn-id: http://webrtc.googlecode.com/svn/trunk@647 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 21:49:25 +00:00
turajs@google.com
13335ccd7a Deleting matlab files
git-svn-id: http://webrtc.googlecode.com/svn/trunk@646 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 21:47:25 +00:00
turajs@google.com
610f478705 Deleting matlab files
git-svn-id: http://webrtc.googlecode.com/svn/trunk@645 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 21:45:34 +00:00
turajs@google.com
53439d9982 Deleting matlab files
git-svn-id: http://webrtc.googlecode.com/svn/trunk@644 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 21:44:00 +00:00
amyfong@webrtc.org
713f91e12b Fixed vie_autotest_custom_call.cc minor issues.
1. mirror of local render removed
2. the video device the user selected wasn't what was actually being used when the call is being made
3. fixed mentions of loopback calls
Review URL: http://webrtc-codereview.appspot.com/171001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@643 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 16:41:26 +00:00
mikhal@webrtc.org
105ff39dec video coding: updating offline tests.
Additional clean-up to the offline test: Placing test callbacks in a designated file. 
Review URL: http://webrtc-codereview.appspot.com/167002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@642 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 16:41:11 +00:00
turajs@google.com
496ef8aca8 To fix warnings in test files.
Review URL: http://webrtc-codereview.appspot.com/169001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@641 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 15:45:48 +00:00
bjornv@google.com
8e9e83b530 This CL adds guards against division by zero, that should fix http://b/issue?id=5278531
In addition a read outside memory event has been detected and removed.
Also an improper noise weighting has been corrected.
Review URL: http://webrtc-codereview.appspot.com/152001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@640 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 12:39:47 +00:00
kjellander@webrtc.org
9e7774f163 Added compare methods for TickInterval class.
This is useful to be able to sort them using the STL algorithm library.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/173002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@639 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 11:33:31 +00:00
bjornv@google.com
dc743a8bba Replaces a use of log2.
I've replaced a log2 operation so it works on Windows.
Review URL: http://webrtc-codereview.appspot.com/171002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@637 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 08:13:53 +00:00
leozwang@google.com
90eff6c7c6 Fix compilation error in build-in AEC test
Review URL: http://webrtc-codereview.appspot.com/164001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@636 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-21 18:02:03 +00:00
wu@webrtc.org
221b522118 Return the number of /dev/video* without trying to open it.
Consider the case when there're /dev/video0 and /dev/video1. But for somereason the video0 is not in a correct state and can't be open. As a result, current NumberOfDevices will return 1, which is fine. However, we will then never be able to get the device we really want - /dev/video1. Consider the code below, the GetCaptureDevice will fail because it calls into DeviceInfoLinux::GetDeviceName(0, ...) which will again try to open the /dev/video0. So the root cause is the mismatching of the NumberOfDevices and GetDeviceName.

Since we will open the device in DeviceInfoLinux::GetDeviceName anyway, I think we should return the number of /dev/video* in DeviceInfoLinux::NumberOfDevices without trying to open it. Otherwise the DeviceInfoLinux::NumberOfDevices should return more information like which /dev/video* is valid which is not.

bool found = false;
for (int i = 0; i < vie_capture->NumberOfCaptureDevices(); ++i) {
  if (vie_capture->GetCaptureDevice(i, ...) == 0) {
    found = true;
    break;
  }
}
Review URL: http://webrtc-codereview.appspot.com/148004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@635 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-21 16:57:15 +00:00
bjornv@google.com
65e6ab31eb Temporary log2 remove to build in chrome
git-svn-id: http://webrtc.googlecode.com/svn/trunk@633 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-21 11:56:46 +00:00