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Author SHA1 Message Date
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
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
phoglund@webrtc.org
df373984e2 Renamed test_bwe. 
BUG=
TEST=
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/642004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2364 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-05 11:00:24 +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
andrew@webrtc.org
3007b26591 Roll Chromium 134666:140240. 
This brings in a common.gypi change which enables warnings-as-errors on
Mac. Necessitated some modifications to third-party gyps.

BUG=none
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/618004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2361 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-04 23:14:37 +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
phoglund@webrtc.org
f1d6e0a65b Removed the obsolete sanity check and added new test HTML files. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/630004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-04 10:06:52 +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
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