Commit Graph

2239 Commits

Author SHA1 Message Date
kjellander@webrtc.org
5608fe9861 Disabling FileBeforeStreamingTest due to flakiness.
BUG=619
TBR=xians1
TEST=Tested on Linux, Mac and Windows.

Review URL: https://webrtc-codereview.appspot.com/654006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2426 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-20 06:14:31 +00:00
wu@webrtc.org
2259f855ea Remove unused member variables found by clang's -Wunused-private-field.
No intended behavior change.

On behavior of thakis@chromium.org.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/641011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2425 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-19 14:56:50 +00:00
hta@webrtc.org
72e3a89b52 Created a wrapper class for condition_variable that lets me write (hopefully) reliable tests for some of its properties.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/600005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2424 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-19 13:49:48 +00:00
bjornv@webrtc.org
b38fca1ec2 VAD Refactoring: API change of return value from int16_t to int.
This CL change the return int on Process() to meet Google Style. The change affects audio_coding and neteq.

Tests have been changed accordingly and the code has been tested on trybots, vad_unittests, audioproc_unittest, audio_coding_unittests, audio_coding_module_test and neteq_unittests.

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/663005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2423 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-19 11:03:32 +00:00
vspasova@webrtc.org
f477aac844 Removed gflags header from vie_auto_test.
Removed gflags include file from src/video_engine/test/automated/
vie_video_verification.cc as it is no longer needed.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/645005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2422 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-19 09:20:33 +00:00
braveyao@webrtc.org
dfa6b697e2 Refine the error handling made in rev2373
Review URL: https://webrtc-codereview.appspot.com/644005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2421 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-19 06:38:59 +00:00
wu@webrtc.org
67f256fab4 Use 32 as the alignment if possible in VP8 wrapper.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/663004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2420 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-18 21:15:32 +00:00
bjornv@webrtc.org
df596ae444 VAD Refactoring of GMM test section
The CL is organized w.r.t. patch sets as follows:
1) Comments on functionality added.
2) Renamed local variable n to channel for clarity.
3) Dropped the extension _vector of variable |feature_vector| since it doesn't add anything new.
4) Minor comment update w.r.t. |feature|
5) Replaced an else if scheme with two if statements. This way we can use the same calculation for all sub cases which could be a source of error.
6) Moved two code lines to where they are used and rearranged such that avoiding tmp variable.
7) Instead of performing a bit-wise OR operation within an if statement we could perform the bit-wise OR at once.
8) Name change of |shifts0| to |shifts_h0| for clearer reading. Likewise for H1.
9) Renamed |nr| to |gaussian| for clearer reading.
10) Removed multiplication macro.
11) Re-organized local arrays to have the same structure as constants and member arrays used elsewhere in the code.
12) Changed locally declared variable to function declared.
13) Added array initialization at declaration.

Tested with trybots, vad_unittests, audioproc_unittest

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/595006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2417 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-18 18:22:53 +00:00
tina.legrand@webrtc.org
50d5ca5bf2 Refactoring of TestAllCodecs
ACM testing consists of seven individual tests. Up til now we haven't used gtest everywhere, and many of the tests needs some rewriting to follow the style guide.

I've started with this tests, doing formatting, adding the test as a separate test which can now either succeed of fail in a proper way.

Still to do in this test is handling of input file, but that will be changed in a separate CL, because all tests uses the  PCMFile class that will be affected by the change.

BUG=none
TEST=audio_coding_module_test, ACM_AUTO_TEST and ACM_TEST_ALL_CODECS.

Review URL: https://webrtc-codereview.appspot.com/646011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2416 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-18 13:35:52 +00:00
hta@webrtc.org
db2f6cf878 Added missing define guard to sleep.h
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/656006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2415 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-18 13:23:48 +00:00
hta@webrtc.org
86a6aacaee Unittest utilities - starting out with an encapsulated trace-to-screen.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/655005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2414 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-18 13:22:08 +00:00
mflodman@webrtc.org
e3a0712f04 Deregister RTP module before deleting it.
BUG=617
TEST=

Review URL: https://webrtc-codereview.appspot.com/661004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2413 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-18 12:43:41 +00:00
hta@webrtc.org
41adcdbf13 An OS-independent sleep function, and one usage thereof.
BUG=603
TEST=none

Review URL: https://webrtc-codereview.appspot.com/659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2412 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-18 11:24:57 +00:00
henrika@webrtc.org
37198007ea GetRecPayloadType now logs a warning instead of and error when the user asks for the payload type while no packets have been received.
BUG=605
TEST=

Review URL: https://webrtc-codereview.appspot.com/660004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2411 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-18 11:00:12 +00:00
stefan@webrtc.org
190541578a Correct gypi files to match the actual filenames.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/656005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2410 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-18 10:57:05 +00:00
niklas.enbom@webrtc.org
d63d06a289 bump version to 3.8
Review URL: https://webrtc-codereview.appspot.com/657004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2408 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-18 08:36:36 +00:00
braveyao@webrtc.org
4de777ba2b Refine the error processing of StopRecordingMicrophone.
BUG = 
TEST = 
Review URL: https://webrtc-codereview.appspot.com/636007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2406 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-15 02:37:53 +00:00
turaj@webrtc.org
bdf7ee5bab This simple change should adress issue 471.
Previously I uploaded patch 640007 to address issue 471. Today, while discussing that patch with Andrew, we noticed this patch should do the job. Leo is not here to verify it, but Andrew did some test to verify it. I'll ask Leo to do some testing. 

We don't want to abandon patch 640007 as it will save some complexity. 
Review URL: https://webrtc-codereview.appspot.com/648004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2405 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-14 23:46:35 +00:00
marpan@webrtc.org
352d09ab28 Updates to videoprocessor_integration test:
-added metric for expected key frame size mismatch
   -fix to start bitrate
   -updates to some expected values in tests
Review URL: https://webrtc-codereview.appspot.com/641007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2404 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-14 18:35:00 +00:00
marpan@webrtc.org
f088448c37 Libyuv Scalerunittest: Added PSNR check to some tests in scaler unittest:
-for downsampling to 1/2x1/2
    -for the odd frame sizes cases
Review URL: https://webrtc-codereview.appspot.com/642009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2403 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-14 17:00:45 +00:00
mflodman@webrtc.org
139c4678c1 Fixed a/v sync issue.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2402 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-14 11:08:51 +00:00
leozwang@webrtc.org
46d83fa26c Use digital mode on mobile
Use fixed digital mode in test app on android

BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/636010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2401 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-12 23:47:20 +00:00
marpan@webrtc.org
c35f1d26c5 FEC: Fix to coverity issue 14448: unintended sign extension.
Review URL: https://webrtc-codereview.appspot.com/647004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2400 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-12 20:12:13 +00:00
stefan@webrtc.org
f0d4696ab3 Add support for SSE intrinsics on gcc in libvpx.
BUG=none
TEST=build on Linux with -Dtarget_arch=ia32 -Ddisable_sse2=1, trybots

Review URL: https://webrtc-codereview.appspot.com/646009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2398 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-12 08:33:53 +00:00
mflodman@webrtc.org
d41851480c Bumped version number to 3.7.
Review URL: https://webrtc-codereview.appspot.com/642007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2397 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-12 08:31:36 +00:00
bjornv@webrtc.org
b1c3276f5a VAD Refactoring: WebRtcVad_Process()
Code style: Indentation, braces

Tested with trybot, vad_unittests, audioproc_unittest

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/579012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2396 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-12 08:19:24 +00:00
kjellander@webrtc.org
5f9f1db12a This change make PulseAudio only start for the tests on the LinuxLargeTests bot.
I put back the --daemonize flag too, since the audio_e2e_test didn't find the pulseaudio daemon otherwise.

TBR=phoglund@webrtc.org
BUG=None
TEST=Tested locally on LinuxLargeTests bot.

Review URL: https://webrtc-codereview.appspot.com/635009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2395 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-12 08:07:30 +00:00
tina.legrand@webrtc.org
5e7ca608d5 Use new fileutil functions for trace in ACM
I this CL I have changed to use filutil functions in the ACM tests. I have also reformated the file Tester.cc, and fixe one minor bug in TestAllCodecs.cc.

BUG=issue195
TEST=audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/636006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2394 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-12 07:16:24 +00:00
andrew@webrtc.org
1c284734cd Fix master's "Start PulseAudio" step.
TBR=kjellander@webrtc.org
BUG=none
TEST=manually on the LinuxLargeTests bot.

Review URL: https://webrtc-codereview.appspot.com/632010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2393 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-12 06:15:45 +00:00
andrew@webrtc.org
0594916a21 Add audio_e2e_test to LinuxLargeTests.
Ensure pulseaudio is running on the bot.

BUG=none
TEST=audio_e2e_test
Review URL: https://webrtc-codereview.appspot.com/646004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2392 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-12 03:32:18 +00:00
andrew@webrtc.org
9f6577b304 Restore default source in e2e test.
Also wait for VoE to start up.

TBR=kjellander@webrtc.org
BUG=issue502
TEST=audio_e2e_test

Review URL: https://webrtc-codereview.appspot.com/643006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2391 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-12 03:25:03 +00:00
leozwang@webrtc.org
6724c4239b Add VoiceEngine apm settings to test application
Implement apm settings and add a small bug fix

BUG=
TEST=build and test on android
Review URL: https://webrtc-codereview.appspot.com/632008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2390 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-11 21:23:16 +00:00
andrew@webrtc.org
be581640c1 Add a variable for the libjpeg include directory.
- Also clean up the use of libjpeg_gyp_path. Both the Chromium and
  standalone builds can use it.

BUG=none
TEST=build with all combinations of use_libjpeg_turbo and build_libjpeg

Review URL: https://webrtc-codereview.appspot.com/640004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2389 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-11 20:38:48 +00:00
kjellander@webrtc.org
f08f52f136 Fixing issues with slaves.cfg on Windows.
This fix is needed for our own build slaves to work properly on Windows and is caused by the hacky way we created the Libvpx waterfall to avoid duplicating unnecessary Python code.

TBR=phoglund
BUG=None
TEST=Tested on Windows build slave.

Review URL: https://webrtc-codereview.appspot.com/639009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2387 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-11 08:06:40 +00:00
bjornv@webrtc.org
eec739f846 VAD Refactoring: Changed pointer structure in WebRtcVad_FindMinimum().
For easier code reading, a couple of structural changes together with name changes have been performed in the function WebRtcVad_FindMinimum():
- Removed temporary pointers
- Updated comments
- Pointer name changes
- Changed pointer nomenclature to array index
- Made local variable const

Tested with trybots, vad_unittests, audioproc_unittest

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/594005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2386 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-11 07:57:57 +00:00
marpan@webrtc.org
78a3110602 Disable multi_res_encoding in libvpx.
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/639008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2385 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-08 22:53:55 +00:00
tina.legrand@webrtc.org
fa7138f889 Change CriticalSectionScoped to use pointer constructor
BUG=issue183
TEST=audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/638005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2384 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-08 10:51:28 +00:00
leozwang@webrtc.org
276dc1872a Add libremote_bitrate_estimator to android makefile
The order of libraries is bit messy, will clean up later.

BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/646007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2383 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-07 18:58:12 +00:00
kma@webrtc.org
f85b35a2f4 Refactored Neon code for AECM module, by using pure assembly code.
Bit exact.
Review URL: https://webrtc-codereview.appspot.com/447008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2382 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-07 16:17:17 +00:00
leozwang@webrtc.org
38506ef4d3 Disable cpu detection on arm-neon
Descritpion
Probably it's a bug in vpx script, disable it for now.
Review URL: https://webrtc-codereview.appspot.com/640006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2381 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-07 15:26:02 +00:00
stefan@webrtc.org
d81ab1397b abs() was used instead of fabsf(), which returns int and not float and therefore truncated the return value.
Also fixes problems with the remote_bitrate_estimator_unittest.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/641006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2380 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-07 13:48:04 +00:00
phoglund@webrtc.org
f7d0c77b48 Added the bitrate estimator test to the trybots.
TBR=kjellander@webrtc.org
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/642005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2378 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-07 09:28:12 +00:00
tina.legrand@webrtc.org
90af7f841c Changing Celt to run on 20 msec frames
BUG=none
TEST=-

Review URL: https://webrtc-codereview.appspot.com/641004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2377 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-07 08:57:27 +00:00
phoglund@webrtc.org
d2956d8cda Renamed test_bwe.
TBR=kjellander@webrtc.org
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/635007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2376 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-07 08:36:35 +00:00
stefan@webrtc.org
9354cc965c Refactoring the receive-side bandwidth estimation into its own module.
Each REMB group has one remote_bitrate_estimator object. For now the
estimator keeps one estimate for every SSRC. In a later commit this will
be unified and one estimate will be used for all SSRC in one group.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/637009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2375 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-07 08:10:14 +00:00
andrew@webrtc.org
f4c6aa2e81 Improve the reliablity of the audio e2e test.
- Use higher quality resampling.
- Add a longer delay before pacat recording.

TBR=kjellander@webrtc.org
BUG=issue502
TEST=manually

Review URL: https://webrtc-codereview.appspot.com/646005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2374 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-07 04:00:15 +00:00
braveyao@webrtc.org
b0bcf13dd4 Trival fix to relative paths of audio files in voe_ui_win_test
BUG  = 
TEST = voe_ui_win_test
Review URL: https://webrtc-codereview.appspot.com/635005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2373 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-07 02:21:44 +00:00
marpan@webrtc.org
5f97232cac Removing a TODO in the FEC: renaming the exisiting packets mask to indicate random mode,
and refactored and renamed corresponding table file.
Review URL: https://webrtc-codereview.appspot.com/632007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2372 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-06 22:34:38 +00:00
wu@webrtc.org
cac603f390 Fix for the alignment problems/mismatch in ViECapture and VP8Encoder.
BUG=576
TEST=unittest

Review URL: https://webrtc-codereview.appspot.com/637010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2371 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-05 23:52:59 +00:00
marpan@webrtc.org
f4c2de9e2f Added some tests to videoprocessor_integrationtest, for testing:
-encooder response to changing bit rate and frame rate
   -frame dropper and spatial resize
   -temporal layers
Review URL: https://webrtc-codereview.appspot.com/613006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2370 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-05 21:07:28 +00:00