Commit Graph

1474 Commits

Author SHA1 Message Date
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
mikhal@webrtc.org
e58112adec Updating libyuv version to latest (121)
Review URL: http://webrtc-codereview.appspot.com/330024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1301 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-28 21:21:31 +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
amyfong@webrtc.org
9b377aa1da Added interface-changes@webrtc.org group to WATCHLIST for monitoring changes to interfaces in WebRTC. If you wish to subscribe to this group, please contact
Niklas or Jan L. 
Review URL: http://webrtc-codereview.appspot.com/330025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1297 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-27 18:43:18 +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
andrew@webrtc.org
d3131ed049 Roll Chrome to canary (114939)
- Keep up to date with the clang rev Chrome is using.
- tools/clang/update.sh now updates clang on Linux if it has been
  manually run once.
- We can remove clang_revision from DEPS; now maintained in update.sh.
- Also keep the gyp rev in sync.

TEST=build on Linux, Mac, Win

Review URL: http://webrtc-codereview.appspot.com/331010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1271 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 16:25:30 +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