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Author SHA1 Message Date
hclam@chromium.org
ad7efa6944 Port Chromium's trace_event.h to WebKit and add 
trace_event.h is ported from Chromium code.

These files are defined new for WebRTC:
* event_tracer.h
* event_tracer.cc
* event_tracer_unittest.cc
Review URL: https://webrtc-codereview.appspot.com/933034

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3262 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 21:19:08 +00:00
kma@webrtc.org
02d9df4544 Updated webrtc_resources_revision to 11, for adding two test files for APM and iSAC. 
Review URL: https://webrtc-codereview.appspot.com/973014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3261 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 19:33:56 +00:00
stefan@webrtc.org
71258c594b Add a third full stack test and support for random jitter in ext transport. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/975005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 15:14:56 +00:00
mflodman@webrtc.org
eaf7cf26fe Adding a simple fake network pipe to use for testing. Next CL will contain an external transport implementation using this link and I'll follow up later making this more advanced. 
Review URL: https://webrtc-codereview.appspot.com/935032

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3259 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 11:47:22 +00:00
kjellander@webrtc.org
f98ffc6db3 Removing default trybot names 
This is removing the default try bot names added in r3031. It doesn't seem like we need to avoid sending all jobs to these bots, even if they're much slower.

BUG=none
TEST=none

Review URL: https://webrtc-codereview.appspot.com/978004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3258 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 10:00:41 +00:00
turaj@webrtc.org
42259e7ebc VoE Changes to enable dual_streaming. 
TEST=added new unit-test

This CL depends on issue 933015 http://webrtc-codereview.appspot.com/933015/
which is under review. Should be committed after issue 933015 is committed.
Committed: https://code.google.com/p/webrtc/source/detail?r=3231
Review URL: https://webrtc-codereview.appspot.com/970005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3257 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 02:15:12 +00:00
turaj@webrtc.org
36965b1803 Bug fix for iSAC fixed-point. The bug was the result of changes in iSAC floating-point to add 48 kHz extension. 
TBR=tlegrand@google.com

TEST=voe_cmd_test, ACM unittest.
Review URL: https://webrtc-codereview.appspot.com/974011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3256 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-10 23:52:43 +00:00
marpan@webrtc.org
55edaecc93 Revert r3254 due to bot failure on android. 
TBR=andrew@webrtc.org, leozwang@google.com
Review URL: https://webrtc-codereview.appspot.com/971018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3255 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-10 19:59:03 +00:00
marpan@webrtc.org
1f3476dd83 Roll libvpx to 000c8414b510. 
Relevant updates/fixes:

000c8414b510: Moved denoiser frame copy/updates out of loopfilter thread.
Multi-threading bug fix: http://code.google.com/p/webm/issues/detail?id=497

ef2248a2a376: Added work buffer for denoiser.
Denoiser bug fix: http://code.google.com/p/webm/issues/detail?id=485

464b1df6d45b: Updates to qp-regulate and rate correction factor.
Rate control improvement: http://code.google.com/p/webrtc/issues/detail?id=1153
Review URL: https://webrtc-codereview.appspot.com/971017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3254 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-10 18:11:30 +00:00
phoglund@webrtc.org
5bbe069f28 Reformatted event* classes. 
TEST=Ran trybots.

Review URL: https://webrtc-codereview.appspot.com/972012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3253 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-10 10:44:37 +00:00
phoglund@webrtc.org
3bb42ef0d6 Made e2e audio quality test write its results to perf. 
The https://chromereviews.googleplex.com/5573026/ patch will mark the test step as perf-printing - this cl will make the test actually print perf lines.

TEST=Ran locally.

Review URL: https://webrtc-codereview.appspot.com/933036

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3252 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-10 10:08:00 +00:00
braveyao@webrtc.org
72feb0b2e2 Not to enum NOTPRESENT audio devices with CoreAudio on Win 
BUG = 
TEST = Manual test
Review URL: https://webrtc-codereview.appspot.com/939037

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-07 22:36:07 +00:00
leozwang@webrtc.org
8e49b02f3d Add more audio codec information into codec list 
BUG=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/974009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-07 22:26:57 +00:00
mikhal@webrtc.org
451aa5dd9d Adding vp8 sequence coder: simple command line encode and decode. 
Goal is to replace existing normal test and affiliates (will be done in follow up cl's)
BUG =1070

Review URL: https://webrtc-codereview.appspot.com/935029

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3249 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-07 21:23:02 +00:00
andrew@webrtc.org
3a5a8a8bcc Properly zero out unmixed frames. 
BUG=6770157

Review URL: https://webrtc-codereview.appspot.com/933037

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3248 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-07 19:37:16 +00:00
kma@webrtc.org
0e739508e0 Added buildbot benchmarking in iSAC and APM into Android platform build. 
Review URL: https://webrtc-codereview.appspot.com/964022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3247 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-07 15:26:28 +00:00
mikhal@webrtc.org
b968213f3c vp8 test: Updating creation of enc/dec 
Review URL: https://webrtc-codereview.appspot.com/937036

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3246 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-06 19:14:26 +00:00
mikhal@webrtc.org
251f64e9e8 Updating vp8 test structure 
Review URL: https://webrtc-codereview.appspot.com/935031

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3245 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-06 17:56:20 +00:00
mikhal@webrtc.org
60d25f90ff Updating Vp8 unit tests - Initiating the switch to gtest-based tests, and adding a stride test. 
This is a follow up on r3227.

Review URL: https://webrtc-codereview.appspot.com/929038

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3244 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-06 16:48:18 +00:00
henrik.lundin@webrtc.org
75f8c78d08 Fixing path to ptypes.txt in NetEqRTPplay 
The default path to the file ptypes.txt needed by NetEqRTPplay
had gone old. Updating to new repo layout.

Also purging old payload types from the file itself.

Review URL: https://webrtc-codereview.appspot.com/966035

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3243 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-06 15:22:00 +00:00
wjia@webrtc.org
df94329a4b Use different cpufeatures library when building with chrome. 
There are more than one target when building with chrome. They have different build setup.
This patch just puts content of build/android/cpufeatures.gypi inside system_wrappers.gyp.
In the future, if more modules will use cpufeatures lib, we can move the code into a gypi file.
Review URL: https://webrtc-codereview.appspot.com/939030

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3242 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-05 22:54:52 +00:00
hclam@chromium.org
81cffd1f2c Port Chromium's atomicops to WebRTC 
Porting Chromium's base/atomicops.h and friends into WebRTC.

Included the original unit test in Chromium.
Review URL: https://webrtc-codereview.appspot.com/964026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3241 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-05 22:51:01 +00:00
leozwang@webrtc.org
63a243a031 Replace the last occurrence of .s with .h 
BUG=None
TEST=trybot
Review URL: https://webrtc-codereview.appspot.com/935027

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3240 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-05 07:12:15 +00:00
leozwang@webrtc.org
96bcac8fbb Expose Set and Get Recording/Playout sample rate apis 
Message:
This is the first cl to add Set/Get Recording and Playout sample rate apis.
In this cl, apis are enabled but returns -1, will add android
implementation in next cl, it's easy for review and coding.

Description:
This CL expose fours voice engine apis,
SetRecordingSampleRate,
RecordingSampleRate, 
SetPlayoutSampleRate,
PlayoutSampleRate. 

BUG=none
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/626004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3239 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-04 19:11:55 +00:00
fischman@webrtc.org
f4e070eca5 Added auto-call feature to WebRTCDemo. 
This (compile-time switchable) option automatically starts & stops calls in
series to stress-test the setup/teardown codepaths.  When startCPULoad() is
removed (https://webrtc-codereview.appspot.com/972008/) this showed no
hangs/crashes after completing 200 start/stop pairs.

Also fixed a tiny shutdown-order bug (onDestroy() calling super.onDestroy()
before performing self-shutdown) and changed default video frame resolution to
640x480 to more effectively stress the device (and be a more compelling demo).

BUG=1162

Review URL: https://webrtc-codereview.appspot.com/939032

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3238 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-04 16:53:43 +00:00
perkj@webrtc.org
2cf22a6abc Revert 3231 - VoE Changes to enable dual_streaming. 
TEST=added new unit-test

This CL depends on issue 933015 http://webrtc-codereview.appspot.com/933015/
which is under review. Should be committed after issue 933015 is committed.
Review URL: https://webrtc-codereview.appspot.com/970005

TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/929040

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3236 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-04 10:02:02 +00:00
stefan@webrtc.org
e861359925 Adds two full stack performance metrics for end-to-end delay. 
Review URL: https://webrtc-codereview.appspot.com/937034

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3235 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-04 09:40:46 +00:00
dwkang@webrtc.org
6bd737a714 First pass of MediaCodecDecoder which uses Android MediaCodec API. 
Background:
As of now, MediaCodec API is the only public interface which enables us
to access low level HW resource in Android. ViEMediaCodecDecoder will be
used for further experiments/exploration.

TODO:
  To fix known issues. (detaching thread from VM and frequent GC)
Review URL: https://webrtc-codereview.appspot.com/933033

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3233 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-04 06:38:19 +00:00
fbarchard@google.com
781cf06124 libyuv r508 with scaler fix for overread horizontally that was caught by valgrind. 
BUG=none
TEST=valgrind bots
Review URL: https://webrtc-codereview.appspot.com/968013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3232 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-04 03:07:29 +00:00
turaj@webrtc.org
767d87cf24 VoE Changes to enable dual_streaming. 
TEST=added new unit-test

This CL depends on issue 933015 http://webrtc-codereview.appspot.com/933015/
which is under review. Should be committed after issue 933015 is committed.
Review URL: https://webrtc-codereview.appspot.com/970005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 22:51:37 +00:00
turaj@webrtc.org
226db898f7 Dual-stream implementation, not including VoE APIs. 
Review URL: https://webrtc-codereview.appspot.com/933015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3230 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 22:13:31 +00:00
turaj@webrtc.org
277ec8e3f5 Fix a bug when iSAC-48kHz was added. 
I discovered this by running extended VoE test on "Codecs."

TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/973010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3229 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 21:16:23 +00:00
mikhal@webrtc.org
f18de86db1 Revert 3227 
> vp8 unittest: Adding qcif stride test
> 
> Review URL: https://webrtc-codereview.appspot.com/930030

TBR=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/929037

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3228 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 20:08:57 +00:00
mikhal@webrtc.org
ab83bb39ad vp8 unittest: Adding qcif stride test 
Review URL: https://webrtc-codereview.appspot.com/930030

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3227 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 19:12:29 +00:00
turaj@webrtc.org
b0dff12d2b 48 kHz extension to iSAC. 
Test:
-manual test with voe_cmd_test.
-manual test with RTPEncode & NetEqRTPPlay.
-manual test with simpleKenny.
-Bit-exact test of iSAC-swb and iSAC-wb with head revision of trunk. The bit-exactness is confirmed on all files generated by running webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACLongtest.txt
Review URL: https://webrtc-codereview.appspot.com/937025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 17:43:52 +00:00
phoglund@webrtc.org
0bacb635cb Removed stale version of fuzzer; it's now internal. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/971015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3225 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 14:28:40 +00:00
stefan@webrtc.org
8d0cd07d0c Add test to verify that padding only frames are passing through the RTP module. 
Review URL: https://webrtc-codereview.appspot.com/934023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3224 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 14:01:46 +00:00
tina.legrand@webrtc.org
5b4fe494e7 Changing default bitrate to 64000 bps for Opus. 
Default settings for Opus in WebRtc is stereo, but we had default rate to 32 kbps. This is too low for stereo, where we need to encode using 64 kbps to get the quality we like.

BUG=

Review URL: https://webrtc-codereview.appspot.com/974008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3223 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 12:08:53 +00:00
kjellander@webrtc.org
ad0f3baf90 Removing redundant codec unittest targets. 
The following targets have been merged into audio_coding_unittests:
* cng_unittests
* g711_unittests
* g722_unittests
* isacfix_unittests
* pcm16b_unittests

Some of them were empty and were created with the assumption they were
needed in order to get code coverage (which was actually not needed).

The following test has been removed since it was empty:
* audio_conference_mixer_unittests

BUG=none
TEST=trybots passing (well, except for the unittests that are removed, they're failing until the try server configuration is updated)

Review URL: https://webrtc-codereview.appspot.com/971008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3222 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 10:52:29 +00:00
phoglund@webrtc.org
ba21c95e15 Reformatted data_log. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/974007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3221 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 09:01:21 +00:00
stefan@webrtc.org
c94f8d4e8f Fix OOB read in padding tests. 
BUG=1177

Review URL: https://webrtc-codereview.appspot.com/973009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3220 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 08:57:54 +00:00
phoglund@webrtc.org
78bec2dcbe Fixed bug where we would rewrite *deref_ptr = ...; to // deref_ptr = ...; 
BUG=

Review URL: https://webrtc-codereview.appspot.com/929036

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3219 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 08:48:07 +00:00
henrike@webrtc.org
fc4a7ee807 Fixes chromium build bots. 
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/971014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3213 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-30 16:17:44 +00:00
brykt@google.com
c7896df420 Fixed bug that caused frame_cutter_unittest to fail when built with MVS2008. 
This was caused by not supplying a correct pointer to where fread should read. The files are now opened in binary mode (which I have under stood can cause problems between different OS if it is not done). I also check for EOF when I compare data from fread. Previously the checking for correct amount of bytes read failed when the end of the file had been reached.

BUG=

Review URL: https://webrtc-codereview.appspot.com/937032

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3212 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-30 12:37:14 +00:00
phoglund@webrtc.org
53034fb247 Improved the conformance test: it will now show video tags and better verify that we set up a call. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/930031

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-30 11:59:20 +00:00
phoglund@webrtc.org
99f7c917d2 Reformatted critical_section wrappers. 
BUG=
TEST=ran trybots

Review URL: https://webrtc-codereview.appspot.com/971012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3210 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-30 10:44:49 +00:00
andrew@webrtc.org
219df91095 Delete bad mergeinfo from webrtc/modules/video_capture/windows 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3208 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-30 02:46:24 +00:00
andrew@webrtc.org
dddc02b9dc Use <(webrtc_root) to point to webrtc files in tools.gyp. 
TBR=brykt@google.com

Review URL: https://webrtc-codereview.appspot.com/939034

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3206 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-30 02:28:27 +00:00
fischman@webrtc.org
d814d71d92 Delete {start,stop}CPULoad() since they're broken. 
- stopCPULoad is incorrect; since mIsBackgroudLoadRunning isn't declared
  volatile, the empty while loop in the background thread isn't required to do a
  memory read (as opposed to reading the value just once and caching it).  The
  result is that stopCPULoad() may never return as the .join() waits forever.
- startCPULoad isn't guaranteed to tax the CPU; the JVM is free to replace the
  while loop in startCPULoad() with a thread pause since it can prove it'll
  never exit the loop once entered (b/c of the previous item).

It's not clear what correct behavior here would be so I'm deleting the code
rather than trying to make it work.  This was responsible for at least most if
not all of the hanginess of start/stop'ing multiple calls in series.

BUG=1162

Review URL: https://webrtc-codereview.appspot.com/972008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3202 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-29 23:00:41 +00:00
fischman@webrtc.org
be5b5ba490 Enable building WebRTCDemo apk using Release webrtc libs, take 2. 
Now passing BUILDTYPE=Release to both the make that builds the libs and the
ndk-build that builds the app makes the app use non-Debug libs.

Review URL: https://webrtc-codereview.appspot.com/966029

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3201 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-29 18:06:00 +00:00