Commit Graph

6455 Commits

Author SHA1 Message Date
a2d8b75f29 An API to get the internal estimation quality in the delay estimator has been added. Unit tests have been updated. There is no impact to other parts in WebRTC.
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1036004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3390 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-18 21:54:15 +00:00
2e2a4cff18 Remove <(library) from gyp file.
This is a corresponding change from Chome.
Review URL: https://webrtc-codereview.appspot.com/1053004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3389 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-18 17:13:47 +00:00
a3e6bec23a Posix Thread: Removes the setting of the run function to NULL which could cause data race.
BUG=http://code.google.com/p/chromium/issues/detail?id=103711
TESTED=Code analysis (no tools)

Review URL: https://webrtc-codereview.appspot.com/1008006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3388 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-18 16:39:21 +00:00
218c542c0b Make VoE handle longer delays
Review URL: https://webrtc-codereview.appspot.com/1047004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3385 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-17 22:25:49 +00:00
c7e935f5eb Adding timeEndPeriod to Synchronize function, see bug for details.
BUG=748
TEST=Win try bots.

Review URL: https://webrtc-codereview.appspot.com/1043005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3383 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-17 17:12:50 +00:00
efae5d5901 Extracted rtp receiver payload management to its own class, made video receiver depend on that instead.
Eliminated need for video receiver to talk to its parent. Also we will now determine if the packet is the first one already in the rtp general receiver. The possible downside would be that recovered video packets no longer can be flagged as the first packet, but I don't think that can happen. Even if it can happen, maybe the bit was set anyway at an earlier stage. The tests run fine.

BUG=
TEST=rtp_rtcp_unittests, vie_auto_test, voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/1022011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3382 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-17 16:10:45 +00:00
20ed36dada Break out RtpClock to system_wrappers and make it more generic.
The goal with this new clock interface is to have something which is used all
over WebRTC to make it easier to switch clock implementation depending on where
the components are used. This is a first step in that direction.

Next steps will be to, step by step, move all modules, video engine and voice
engine over to the new interface, effectively deprecating the old clock
interfaces. Long-term my vision is that we should be able to deprecate the clock
of WebRTC and rely on the user providing the implementation.

TEST=vie_auto_test, rtp_rtcp_unittests, trybots

Review URL: https://webrtc-codereview.appspot.com/1041004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-17 14:01:20 +00:00
3b7feb2a5d Convert psnr and ssim to strings before printing them.
Review URL: https://webrtc-codereview.appspot.com/1042004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3380 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-17 13:35:01 +00:00
a4b58860b7 Add a counter to the video rtp play output filename.
Review URL: https://webrtc-codereview.appspot.com/1040004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3379 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-17 09:27:17 +00:00
2fd947fb21 Removing outdated comment
Review URL: https://webrtc-codereview.appspot.com/1025007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3376 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-16 18:50:35 +00:00
acfdd96ee3 Reformatted rtp_rtcp_impl*.
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1035004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3374 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-16 10:27:33 +00:00
77a584be1d Made ViEToFileRenderer use a separate thread for rendering frames to file.
Review URL: https://webrtc-codereview.appspot.com/1021011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3373 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-15 16:34:34 +00:00
a22a9bd9ca Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional.
The audio receiver is now completely independent of rtp_receiver: video will hopefully be too in the next patch.

BUG=
TEST=vie & voe_auto_test full runs

Review URL: https://webrtc-codereview.appspot.com/1014006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3372 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-14 10:01:55 +00:00
49273ffa79 logical 'and' of mutually exclusive tests is always false in ViECodecImpl::CodecValid()
BUG = Issue1283
Review URL: https://webrtc-codereview.appspot.com/1013008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3371 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-14 01:52:26 +00:00
b119369cdc Fix android clang build.
no-builtin-cos|sin|cosf|sinf are not used for some files (g711.c, g711_interface.c, g722_encode.c, g722_decode.c, g722_interface.c, pcm16b.c).
Review URL: https://webrtc-codereview.appspot.com/1032006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3369 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-12 01:52:07 +00:00
3f9db3735e Fix android clang build.
Android clang build complains about unused private field.
Review URL: https://webrtc-codereview.appspot.com/1025006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3368 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-12 01:09:03 +00:00
bafdae3cfc Fix simulated analog gain in audioproc.
* It doesn't make much sense to apply at all when reading from the protobuf.
* Reduced the gain to be closer to actual mics.

BUG=1260

Review URL: https://webrtc-codereview.appspot.com/1027007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3366 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-11 23:11:29 +00:00
f908011eb4 Remove extra line.
TBR=elham

Review URL: https://webrtc-codereview.appspot.com/1024008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3365 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-11 22:39:55 +00:00
e7dc7f8553 Disable full stack PSNR/SSIM triggers on Mac and Win for now due to flakiness. Adding plots of PSNR and SSIM.
TBR=mflodman

BUG=1271

Review URL: https://webrtc-codereview.appspot.com/1032005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3360 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-11 12:55:19 +00:00
be86a6d968 Explicitly disable sincos optimization on Android.
I uploaded this CL before, now it turned out that although it's an
issue in compiler, but it will not be solved in short term, we have
to work around in our code termporally.

We can chat in person if you want to know more details.

BUG=
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/1026006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3358 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-10 22:15:51 +00:00
e468f08078 Disable PSNR/SSIM thresholds for the Gilber-Elliot test.
This is to avoid flakiness as the GE model can cause quite big freezes
from time to time. Will keep the test running to get the plots.

TBR=phoglund

BUG=1271

Review URL: https://webrtc-codereview.appspot.com/1030004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3357 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-10 15:17:36 +00:00
0af0d3d3f4 Address a build issue with Android-Clang compiler:
error: the value is truncated when put into register, use a modifier to specify the size [-Werror,-Wasm-operand-widths]
  __asm __volatile ("ssat %0, #16, %1" : "=r"(out16) : "r"(value32));
Review URL: https://webrtc-codereview.appspot.com/1029006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3352 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-10 00:46:37 +00:00
ef1a760446 Rounding error fix in media_opt_util.
Review URL: https://webrtc-codereview.appspot.com/1013006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3351 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-09 22:13:19 +00:00
a5e7e76def Use %d for signed value in trace.
BUG=1259
TBR=henrika

Review URL: https://webrtc-codereview.appspot.com/1028007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3349 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-09 17:11:50 +00:00
08d660f08e Allow for some error in volume testing.
BUG=616
TESTED=voe_auto_test:VolumeTest.* now passes on a MacBook

Review URL: https://webrtc-codereview.appspot.com/1028005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3348 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-09 17:07:02 +00:00
d005468e9b Generalized mechanism for excluding gtests on platforms, disabled broken tests on mac.
BUG=1268
TEST=vie_auto_test on mac and linux
TBR=mflodman, kjellander

Review URL: https://webrtc-codereview.appspot.com/1027006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3347 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-09 16:53:42 +00:00
2f225cadde Add logs when no RTCP RR has been received for three regular RTCP intervals.
BUG=1267
TEST=Unittest added.

Review URL: https://webrtc-codereview.appspot.com/1019006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3346 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-09 13:54:43 +00:00
d66eb8c4eb Disabled GQoS since it breaks ViE auto test.
BUG=1266
TEST=vie_auto_test.exe --automated --gtest_filter=-ViERtpFuzzTest* --capture_test_ensure_resolution_alignment_in_capture_device=false

Review URL: https://webrtc-codereview.appspot.com/1025005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3345 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-09 09:13:00 +00:00
fcd8585874 Enable external encoders with internal picture source.
CL enables registering of external encoder with internal picture
source on API by adding simple passthrough parameter that is already
supported within video engine.

Review URL: https://webrtc-codereview.appspot.com/1006006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3344 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-09 08:35:40 +00:00
658d423e81 Using Convert in lieu of ExtractBuffer: Less error prone (as we don't need to compute buffer sizes etc.). This cl is first in a series (doing all of WebRtc would make it quite a big cl). While at it, fixing a few headers.
BUG=988

Review URL: https://webrtc-codereview.appspot.com/995014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3343 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-08 19:19:59 +00:00
27cb3017f5 Updated version number to 3.20
Review URL: https://webrtc-codereview.appspot.com/1023008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3341 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-07 21:54:41 +00:00
c38eef896a Reformatted RTPReceiver.
This is a pure reformat patch, with the exception that I also fixed all comments and moved a constant. I did not change the types in this patch since I
though that is more risky, so I'll do that in a separate patch later (perhaps
we could purge the types from the whole module in one go?)

BUG=
TEST=Trybots, vie_ & voe_auto_test --automated

Review URL: https://webrtc-codereview.appspot.com/998007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3338 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-07 10:18:30 +00:00
df3a15f63b Removed spaces from full stack test labels, consolidated graphs
NOTE TO SELF: save history on master when deploying!

BUG=
TEST=Ran locally

Review URL: https://webrtc-codereview.appspot.com/1021007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3337 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-07 10:06:52 +00:00
1ea4b502ef Refactor receiver.h/.cc.
TEST=video_coding_unittests, vie_auto_test --automated

Review URL: https://webrtc-codereview.appspot.com/994008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3336 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-07 08:49:41 +00:00
1926d33344 Change Sleep() comment in test fixture.
TBR=phoglund

Review URL: https://webrtc-codereview.appspot.com/1023006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3335 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-05 03:30:11 +00:00
bcb717428f .gitignore: Add *.mk, created as part of ChromiumOS build
Contributed by Josh Triplett <josh.triplett@intel.com>

BUG=None
TEST=Build Chromium and ChromiumOS from source, and run "repo status",
     with and without this change.

Review URL: https://webrtc-codereview.appspot.com/1000006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3334 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-04 21:25:42 +00:00
f545cf8f10 Addressing webrtc issue 1237, http://code.google.com/p/webrtc/issues/detail?id=1237.
Code compared to C. Bit-exact.
Review URL: https://webrtc-codereview.appspot.com/1021004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3333 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-04 17:40:21 +00:00
6f62836ccf Reverting two mixing test patches: seems to introduce a persistent problem for win voe_auto_test (wrapping problem?)
Revert "Further relax thresholds in mixing test."

This reverts commit 53c7e973a02d65e0b4981129e7ccfc145d955eda.

Revert "Fix implicit conversion error in mixing test."

This reverts commit 68d7e2258082d7d2b9461061e03e2f2d6ae78c4f.

TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1018005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3331 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-03 14:33:00 +00:00
5c8d9d30e2 Reformatted tick_util.
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1014004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3330 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-03 09:50:17 +00:00
daabfd25a6 Reformatted trace* files.
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1015004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3329 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-03 09:37:03 +00:00
201d4b61d1 Fix implicit conversion error in mixing test.
TBR=mikhal

Review URL: https://webrtc-codereview.appspot.com/1020004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3328 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-02 19:59:53 +00:00
b2b628d5cd Further relax thresholds in mixing test.
TBR=mikhal

Review URL: https://webrtc-codereview.appspot.com/1019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3327 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-02 18:50:13 +00:00
00c7c4315b Replace voice engine utility functions with system wrapper variants.
SLEEP -> SleepMs
GET_TIME_IN_MS -> TickTime::MillisecondTimestamp

These could cause unused function errors on some compilers.

BUG=1228

Review URL: https://webrtc-codereview.appspot.com/1013004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3326 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-02 16:06:39 +00:00
ec9c942e45 Reformatted thread and static_instance.
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1006005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3324 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-02 08:45:03 +00:00
1b6da28047 Bugfix for NACK behavior. Current code sends a number of duplicate NACK requests.
Landing of 573005 On behalf of an1kumar@gmail.com

TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1002008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3322 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-21 17:46:24 +00:00
f556890844 Added possibility to repeat frames. Also added unittest for that feature.
BUG=

Review URL: https://webrtc-codereview.appspot.com/994005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3321 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-20 11:42:45 +00:00
d73527ccab Changed assert to log.
Review URL: https://webrtc-codereview.appspot.com/1010004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3320 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-20 09:26:17 +00:00
d0d41498a3 Adding AUDIO application as default for Opus stereo
The Opus audio codec targets applications for pure conversations as well as other types of audio (e.g. music), and there are two different settings to use for this (VoIP and AUDIO). In the current implementation of Opus in WebRTC we use VoIP only.

I this CL I have changed default setting to AUDIO in the case of stereo, and kept VoIP as default in case of mono.

Next step is to add an API to choose application mode.

BUG=issue1239

Review URL: https://webrtc-codereview.appspot.com/1007006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3319 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-20 09:23:10 +00:00
ad0ed582b5 Fixed a missed initialization (found by valgrind FYI bot).
http://webrtc-cb-linux-master.cbf.corp.google.com:8011/builders/LinuxLargeTests/builds/327/steps/memory%20test%3A%20memcheck_voe_auto_test/logs/stdio

BUG=
TEST=Reproduced valgrind error, verified gone after fixing.

Review URL: https://webrtc-codereview.appspot.com/1008005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3318 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-20 09:14:36 +00:00
ac77084583 Roll opus to 172355 and delete opus_demo from webrtc opus
opus_demo has been inlucded in opus in chromium.

BUG=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/973013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3317 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-19 17:24:30 +00:00