Commit Graph

1731 Commits

Author SHA1 Message Date
mflodman@webrtc.org
64f86fba19 Fix test app render bug.
Review URL: https://webrtc-codereview.appspot.com/669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2435 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-21 12:32:39 +00:00
mflodman@webrtc.org
8baed51f6e This CL is part of enabling cpplint check for video_engine uploads.
BUG=627
TEST=vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/653006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2434 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-21 12:11:50 +00:00
mflodman@webrtc.org
9ba151bdf9 Removed cpplint warnings from all impl-files to be able to add this check as presubmit step. I don't want to change the API right now, will come later, so there are several NOLINT comments added to get around this for now.
BUG=627
TESTS=vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/661005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2433 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-21 10:02:13 +00:00
hta@webrtc.org
2bd8d62d3b Sleep using no compile flags
BUG=603
TEST=

Review URL: https://webrtc-codereview.appspot.com/665004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2432 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-21 09:57:24 +00:00
mflodman@webrtc.org
67f98ec63a Removed flaky REMB test. This test is now covered by:
- RemoteBitrateEstimatorTest
- BitrateControllerTest
- RtcpFormatRembTest
- ViERembTest

BUG=477
TEST=See above.

Review URL: https://webrtc-codereview.appspot.com/667004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2431 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-21 09:29:53 +00:00
kma@webrtc.org
173538faa3 Refactored function WebRtcIsacfix_GetLpcCoef() in iSAC-fix.
One reason behind it is for further optimization of it in ARM.
Review URL: https://webrtc-codereview.appspot.com/646012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2429 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-20 17:17:15 +00:00
hta@webrtc.org
3168e5349c Working unit test for critical sections.
This extends unittest coverage, and allows to add more tests if these functions
ever are found to behave strangely.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/632005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2427 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-20 06:45:56 +00:00
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
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
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
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
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
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
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
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
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
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
marpan@webrtc.org
8866bb1132 FEC: Added another set of packet masks for the FEC.
These FEC codes perform better for bursty (consecutive loss) 
than the existing set (which were designed for random loss). 
Updates to the unittests and test_fec accordingly.
Review URL: https://webrtc-codereview.appspot.com/581005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2369 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-05 16:42:20 +00:00
bjornv@webrtc.org
20e13edede New attempt to revert r2362, since drover failed.
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/640005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2368 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-05 13:07:56 +00:00
bjornv@webrtc.org
cb89c6f914 Revert 2363 - 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/605007

TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/634006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2366 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-05 12:25:35 +00:00
stefan@webrtc.org
f72881406f 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/605007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2363 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-05 10:44:00 +00:00
bjornv@webrtc.org
d2acea6b30 Minor style changes
Original CL=577007

Tested on trybots.

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/637007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2362 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-05 08:09:23 +00:00
marpan@webrtc.org
da7fdf4af8 Fix to scaler in libyuv for odd size frames.
Review URL: https://webrtc-codereview.appspot.com/633004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2360 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-04 21:56:13 +00:00
turaj@webrtc.org
ba108aee21 This CL contains some refactoring. Spectrum coding is main place that is affected. Therefore, I have bit-exactness test, test_spectrum_
coding.c, to be sure about the changelist. You can go through the test to be sure the changes are tested. However, I don't intend to commi
t the test, as it would be a source of confusion and requires hack to iSAC to be able to run the test. It is basically a one-time test. 

The part which not covered in this test is where we limit payload for super-wideband bit-stream. I'll add a test for that as well. 

I kept format changes at minimum in all files except isac.c, which was in bad shape, and coding changes were minimum. I'm planning to uplo
ad following patches to this CL where I try to address formatting issues. But I don't intend to change variable names, for the moment. 

The refactoring is not yet finished, so you would find part of the code which could be cleaned up, especially KLT transforms in entropy_co
ding.c
Review URL: https://webrtc-codereview.appspot.com/580004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2359 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-04 20:04:58 +00:00
andrew@webrtc.org
2cc55096d5 Fix syntax error in jpeg.gypi.
TBR=mflodman@webrtc.org
BUG=none
TEST=build on Linux.

Review URL: https://webrtc-codereview.appspot.com/639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2358 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-04 20:01:23 +00:00
mflodman@webrtc.org
ad6083f414 Added condition for which jpeg lib to use.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/638004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2357 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-04 19:10:43 +00:00
tina.legrand@webrtc.org
77fd39aa99 ACM PCM16B, fixing a copy-and-paste error.
Review URL: https://webrtc-codereview.appspot.com/631006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2355 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-04 11:47:49 +00:00
phoglund@webrtc.org
e6f235cfa5 Attempt to fix broken encoding.
TBR=niklase@webrtc.org
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/637004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2353 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-04 11:04:05 +00:00
niklas.enbom@webrtc.org
9cf4d72d5d git-svn-id: http://webrtc.googlecode.com/svn/trunk@2352 4adac7df-926f-26a2-2b94-8c16560cd09d 2012-06-04 10:58:43 +00:00
niklas.enbom@webrtc.org
82bf033380 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2351 4adac7df-926f-26a2-2b94-8c16560cd09d 2012-06-04 10:57:51 +00:00
niklas.enbom@webrtc.org
265e38c336 Fixing test gypi for bit rate controller
Review URL: https://webrtc-codereview.appspot.com/636004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2350 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-04 10:12:44 +00:00
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
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
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
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
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
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
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
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
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
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
andrew@webrtc.org
51b4f3e6a8 Try to fix MixingTest on the Win bots.
- Relax the constraints on recording duration.
- Remove unneeded file deletes. (These files will be properly
  overwritten anyway).

TBR=henrike@webrtc.org
BUG=issue534
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/600006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2295 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-24 17:26:05 +00:00
stefan@webrtc.org
42e78ac087 Disable normal_async_test and rename tests to vp8_integrationtests.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/598004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2294 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-24 14:36:34 +00:00
mflodman@webrtc.org
6af9594d71 Added gyp variable to include/exclude all tests.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/597004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2292 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-24 13:23:35 +00:00