Commit Graph

2222 Commits

Author SHA1 Message Date
braveyao@webrtc.org
ab12990b1b In the past we support calling StartPlayingFileLocally() before StartPlayout(), then when playout is started, the file would be played out immediately.
Now we would get a failure if we do the same thing and the file would not be played out. Then GTalk/Hangout also reported this failure to us. 
This CL is to restore the original function. 

BUG = Issue 490
TEST = Manual test and voe_auto_test->FileBeforeStreamingTest
Review URL: https://webrtc-codereview.appspot.com/569016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2347 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-04 03:26:39 +00:00
marpan@webrtc.org
899baa821b Temporarily disable first partition packet counting to avoid a bug in ProducerFec which doesn't properly handle important packets.
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/631005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2346 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-01 18:32:16 +00:00
leozwang@webrtc.org
354b0ed015 Check return result of fwrite [Audio Module]
Description:
On ChromeOS/ARM, compiler enforces to check return result of a function.
Currently, we don't check return result of fwrite, it causes building errors.

The following files need to patch. The patch should be similar, before I patch all
of them, I will start with 2 files, please take a quick look, if the patch is OK,
I will continue and upload a new patch that covers all of them.
it to all of them.
Review URL: https://webrtc-codereview.appspot.com/566016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2345 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-01 17:46:21 +00:00
kma@webrtc.org
c3b2683bf4 Refactored the pitch filter function in iSAC-fix. One important purpose is to prepare the function for assembly optimization in ARM platforms.
Note that,
(1) The main change is a new function PitchFilter() replacing a couple of common code blocks. Next step will be the assembly coding of this function in ARM.
(2) Resulted code is not bit exact with the original. The only reason is replacing two saturation blocks (lines 197 and 208) for the case of "type == 2" with the general case (line 147 and 159). The change makes the code more consistent, and I think the original code might just be a bug. I raised the issue in an email to Turaj and Bjorn last week.
Listening test might be needed. I will send the resulted files to Turaj for this purpose.
(3) I used Astyle to make the code more stylish, but didn't try extra effort to correct all the code style details.  Local code style consistency was considered for new code. So this is not a full and final refactor project (will leave that to future refactoring).
Review URL: https://webrtc-codereview.appspot.com/573009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2344 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-01 17:00:07 +00:00
tina.legrand@webrtc.org
5b4f36db88 ACM: Too short char vector
Revision 2340 failed on Linux Release, and the problem was that we allocated a too short vector for the output file name.

BUG=r2340 failed on Linux release
TEST=audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/624006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2343 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-01 14:51:28 +00:00
kjellander@webrtc.org
343301ff73 Fixing release compilation on Linux and Mac trybots
BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/620005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2342 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-01 11:10:24 +00:00
kjellander@webrtc.org
c03df177c6 Enabling audio_coding_module_test on trybots
BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/615005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2341 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-01 11:09:39 +00:00
tina.legrand@webrtc.org
4517585db5 Adding separate payload types for stereo modes
BUG=Issue 452
TEST=audio_coding_test, voe_auto_test, voe_cmd_test

Edit: adding Patrik to review:
src/modules/rtp_rtcp/source/rtp_receiver.cc
...and Shijing to review:
src/voice_engine/main/source/channel.cc
src/voice_engine/main/test/cmd_test/voe_cmd_test.cc

Review URL: https://webrtc-codereview.appspot.com/540004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2340 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-01 09:27:35 +00:00
pwestin@webrtc.org
c2722a0e68 Fixed compiler warning
Review URL: https://webrtc-codereview.appspot.com/624005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2339 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-01 08:56:42 +00:00
kjellander@webrtc.org
29c5a2335c Renamed to Network Emulator and improved error handling.
Changed default start port of the port-range to 32768.

BUG=None
TEST=Tested locally.

Review URL: https://webrtc-codereview.appspot.com/627004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2338 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-01 08:42:17 +00:00
stefan@webrtc.org
f5d934dfd8 Upgrade libvpx to cab6ac16 (v. 1.1.1 pre-point-release).
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2337 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-01 07:43:02 +00:00
andrew@webrtc.org
7d8c567982 Ignore return value of fwrites.
The removed error return was of course failing in the void ProcessBlock.
Ignored the returns of the remaining fwrites as well for consistency.

TBR=leozwang@webrtc.org
BUG=none
TEST=run audioproc with debug enabled

Review URL: https://webrtc-codereview.appspot.com/623004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2336 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-01 02:41:14 +00:00
kjellander@webrtc.org
595749f7ec Network simulation script based on Dummynet.
This script only intends to support Mac and Linux so far. Additional coding and conditional checking will be required to support Windows.

BUG=None.
TEST=Tested on Linux and Mac.

Review URL: https://webrtc-codereview.appspot.com/606006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2335 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-31 20:19:05 +00:00
andrew@webrtc.org
ad0f05bdf5 Remove empty directories.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2334 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-31 18:40:10 +00:00
kjellander@webrtc.org
2e84c112f5 Updating bitrate controller tests to test naming conventions.
The test is now named 'bitrate_controller_unittests'.
This CL also enables it on the bots. The test is excluded on ASAN since
it fails when compiled with projects generated with GYP_DEFINES='asan=1' (see issue 555).

BUG=None
TEST=bitrate_controller_unittests was tested in Debug+Release on Linux, Mac and Windows + TSAN and memcheck.

Review URL: https://webrtc-codereview.appspot.com/612004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2333 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-31 13:55:01 +00:00
phoglund@webrtc.org
baaf2434a7 Extracted a method for sending padded data.
BUG=
TEST=Ran vie_auto_test and voe_auto_test standard tests.

Review URL: https://webrtc-codereview.appspot.com/605004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2332 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-31 10:47:35 +00:00
kjellander@webrtc.org
bb24b14c8a Libvpx waterfall additional changes. The CL http://review.webrtc.org/595005/ was not complete since it didn't put the libvpx waterfall at its own port.
Chrome bots are now using the correct scheduler.
Created separate watchlist e-mail for libvpx builds to avoid spamming WebRTC when these builds fail.

BUG=None
TEST=Tested on local master and slaves.

Review URL: https://webrtc-codereview.appspot.com/620004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2331 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-31 10:42:51 +00:00
wu@webrtc.org
7d3b07a516 Update to chromium r139469.
Review URL: https://webrtc-codereview.appspot.com/615004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2330 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-31 00:31:42 +00:00
andrew@webrtc.org
36ccce4f58 Remove documentation folders.
Review URL: https://webrtc-codereview.appspot.com/606007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2329 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-30 17:28:24 +00:00
andrew@webrtc.org
16fcb247b2 Disable flaky VolumeTests only on Linux.
BUG=issue367
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/611005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2328 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-30 17:26:32 +00:00
leozwang@webrtc.org
e7e64e3468 Fix compilation errors on ChromeOS
Description:
This cl fixes two compilation errors on ChromeOS/ARM, it could
also be reproduced by gcc 4.5+.

I also add comments about error message and how I solve them.

BUG=webrtc issue 554
TEST=try bots and build on chromeos arm
Review URL: https://webrtc-codereview.appspot.com/611006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2327 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-30 16:46:09 +00:00
niklas.enbom@webrtc.org
0cb79cc851 Fixing gyp bug in https://webrtc-codereview.appspot.com/599006
Review URL: https://webrtc-codereview.appspot.com/609006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2325 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-30 14:32:42 +00:00
stefan@webrtc.org
dc257b5781 Add option to configure error concealment and disable by default.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/597005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2324 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-30 11:25:00 +00:00
mflodman@webrtc.org
327ada1cb0 Refactored IncomingVideoStream and VideoRenderFrame, to get code in better shape when hunting BUG=481.
BUG=481
TEST=Compiles on all platformas and autotest passes.

Review URL: https://webrtc-codereview.appspot.com/608005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2323 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-30 10:45:18 +00:00
phoglund@webrtc.org
9259e7bd03 Added a step for restarting pulseaudio.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/611007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2322 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-30 09:07:02 +00:00
bjornv@webrtc.org
281b7983db Resolved Coverity warnings.
This CL includes changes to resolve Coverity warnings 14086, 14110 and 14111.

Tested with trybots and audioproc_unittests.

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/606004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2321 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-30 07:41:57 +00:00
leozwang@webrtc.org
b5ea03adbb Add print out stats summary to integrationtest.cc
Stats summary prints out cpu usage.

BUG=
TEST=test on linux
Review URL: https://webrtc-codereview.appspot.com/602004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2320 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-30 00:34:50 +00:00
andrew@webrtc.org
459955f821 Move audio_frame_operations to the utility module.
TBR=henrika@webrtc.org
BUG=issue551
TEST=voe_auto_test, webrtc_utility_unittest, trybots

Review URL: https://webrtc-codereview.appspot.com/599006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2318 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-29 22:13:14 +00:00
andrew@webrtc.org
aafa49bb85 Disable flaky VolumeTest.DefaultSpeakerVolumeIsAtMost255.
This test failed on six CLs in a row recently.

TBR=xians@webrtc.org
BUG=issue367
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/595007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2317 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-29 22:05:15 +00:00
andrew@webrtc.org
5f23d64cf2 Set the stream delay to zero if too low.
- Return a stream warning instead of an error.
- Add a few delay offset tests.

BUG=
TEST=audioproc_unittest

Review URL: https://webrtc-codereview.appspot.com/607004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2316 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-29 21:14:06 +00:00
leozwang@webrtc.org
2fc6e388c0 Check return value of fwrite. [Video Module]
Description:
On ChromeOS/ARM, compiler enforces to check return result of a function.
Currently, we don't check return result of fwrite, it causes building errors.

The following files need to patch. The patch should be similar, before I patch all
of them, I will start with 3 files, once we agree upon the solution, we will expand
it to all of them.

The question is should we do 
1. if (error) { return -1;} 
or 
2. if (error) { /*ignor the error*/ }

I took "return -1" in this patch, but I'm OK with either. Please let me know your
thoughts and I will upload a new patch.
Review URL: https://webrtc-codereview.appspot.com/583010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2315 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-29 17:33:13 +00:00
kjellander@webrtc.org
8a7a019b55 Syncing tests on try master with build master.
Also adding video_codecs_test_framework_integrationtests since it's executing fast enough and catches a lot.

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/605006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2314 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-29 14:49:39 +00:00
pwestin@webrtc.org
1eef9c16ff Bitrate bugfixes
Review URL: https://webrtc-codereview.appspot.com/609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2313 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-29 09:28:43 +00:00
stefan@webrtc.org
5abab0b1b5 Revert 2311 - Disable error concealment.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/609004

TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/604006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2312 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-29 09:04:40 +00:00
stefan@webrtc.org
3348b2990b Disable error concealment.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2311 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-29 08:44:00 +00:00
kjellander@webrtc.org
fe87f8ce08 Libvpx buildbot waterfall.
I think I found a decent way to avoid duplicating the whole configuration for the libvpx waterfall. It's not perfect but it works. I still haven't figured out what the best way to manage the slaves would be however, since they will need to know which master to connect to, or they'll pick the default they're configured for in slaves.cfg, which is the WebRTC waterfall (can be overridden with the TESTING_MASTER property, but that's only to be used for development and testing.

BUG=None
TEST=Tested on local master and slaves.

Review URL: https://webrtc-codereview.appspot.com/595005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2310 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-29 06:23:42 +00:00
mflodman@webrtc.org
ca8d788362 Fix a bug where a RAII object was created for just one line instead of a block.
Found by clang:

../../third_party/webrtc/video_engine/vie_render_manager.cc:157:3: error: expression result unused [-Werror,-Wunused-value]
  ViEManagerWriteScoped(*this);
  ^~~~~~~~~~~~~~~~~~~~~~~~~~~~
1 error generated.

Review URL: https://webrtc-codereview.appspot.com/599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2309 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-28 18:56:20 +00:00
phoglund@webrtc.org
dbaa893525 Completed rewrite of APM extended test.
Removed NS tests since they are already covered by audio_processing_test.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/603004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2308 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-28 14:36:59 +00:00
bjornv@webrtc.org
1747427861 VAD Refactoring: Replaced pointer operation with array index
This CL contains a change of pointer nomenclature to array index. In addition, one place with two hard coded Gaussians has been generalized with a for loop.

Tested with trybots, vad_unittests, audioproc_unittest

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/592004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2307 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-28 12:50:05 +00:00
bjornv@webrtc.org
4e12d3065e VAD Refactoring: Removed assign calls
These calls are not used anywhere in WebRTC and there is no plan on using them.
Removed them and updated corresponding unit tests.

Tested on trybots, vad_unittests, audioproc_unittest

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/608004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2306 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-28 12:25:07 +00:00
tina.legrand@webrtc.org
0de1ee3830 NetEQ: Remove an unnecessary condition, to fix a clang warning
This is a duplicate of issue 606005: https://webrtc-codereview.appspot.com/606005/

BUG=
TEST=neteq_unittests

Review URL: https://webrtc-codereview.appspot.com/605005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2305 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-28 11:37:50 +00:00
kjellander@webrtc.org
b6e4cc776e Valgrind memcheck and TSAN script now uses Chrome+WebRTC suppression files.
Skeleton suppression files for future WebRTC suppressions are added and are included in addition to the ones Chrome are using and maintaining when our wrapper script executes.

Also added tweaked PRESUBMIT checks based on the Chrome code, that verifies
that suppressions are added correctly. I tested that they work by adding an invalid
suppression.

BUG=544
TEST=Tested running tools/valgrind-webrtc/webrtc_tests.sh --tool=tsan -t out/Debug/system_wrappers_unittests and it reports far less errors. Tested adding bad suppression and it was caught by the PRESUBMIT check.

Review URL: https://webrtc-codereview.appspot.com/601004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2304 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-27 20:59:35 +00:00
kma@webrtc.org
0d321da7e1 Refactored ARM specific code in Noise Suppression. Bit exact.
Review URL: https://webrtc-codereview.appspot.com/459006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2303 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-26 01:05:27 +00:00
leozwang@webrtc.org
1755a57cbc Check return result of fwrite, [APM]
Description:
This cl added checking return result of fwrite which makes it compile
on ChromeOS/ARM.

BUG=issue:541
TEST=Build on all platforms
Review URL: https://webrtc-codereview.appspot.com/583009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2302 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-25 19:20:35 +00:00
leozwang@webrtc.org
f14575fd8e Dynamically load codec list
Description:
This cl adds a feature that can query video engine and voice engine and load code list in
gui settings. Currently, codec lists are fixed in resource file, it caused confusion and
problems.

TBR=ronghua
BUG=
TEST=test on android
Review URL: https://webrtc-codereview.appspot.com/583006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2301 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-25 16:47:56 +00:00
leozwang@webrtc.org
351fb6d3b4 Exclude code that don't work on android in voe_cmd_test
Description:
Ths cl makes voe_cmd_test work on android by excluding some code
that are availabel on android today, some highlights
1. change maxnumofchannles
2. disable audio device selection
3. disable set/get volume

BUG=
TEST=test on try bots
Review URL: https://webrtc-codereview.appspot.com/584009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2300 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-25 16:47:35 +00:00
kjellander@webrtc.org
15cd04a314 Enabling vp8_integrationtests on all platforms
BUG=None
TEST=Tested on Linux, Mac and Windows in Debug+Release

Review URL: https://webrtc-codereview.appspot.com/599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2299 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-25 09:46:19 +00:00
kjellander@webrtc.org
21c5bda17c Enabling udp_transport_unittests for TSAN
BUG=536
TEST=Tested running the test within TSAN on Linux.

Review URL: https://webrtc-codereview.appspot.com/591004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2298 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-25 09:45:33 +00:00
turaj@webrtc.org
10d3b5239b I haven't done any refactoring here.
Resolve coverity warnings.

14305.

The warning is not really valid. The 'decode' function should be called with a 'mode' variable, where inside the function it is assumed that mode is either zero or one. If mode is taking other values some varibles are used uninitialized. However, this is an internal function and it is always called with either ZERO or ONE. Therefore, the code operates correctly. I made small changes as I beleive it is a bit nicer way. 

In ACM:
- Conditions on 'mode' is changed.


Tested with trybots.

BUG=None
TEST=None
Review URL: https://webrtc-codereview.appspot.com/564014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2297 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-24 21:20:25 +00:00
andrew@webrtc.org
f45d47ad7d Remove mixing tests from voe_extended_test.cc
These have been moved to:
src/voice_engine/main/test/auto_test/standard/mixing_test.cc

BUG=
TEST=build voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/588005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2296 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-24 17:27:59 +00:00