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Author SHA1 Message Date
wu@webrtc.org
d64719d895 Update libjingle to 50191337. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1885005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4461 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 00:00:07 +00:00
fischman@webrtc.org
d3ae3c7b1f Unbreak clang/android build of webrtc. 
TESTED=All target builds once more with clang=1.
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4460 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 23:53:07 +00:00
wu@webrtc.org
7fdbb1c832 We don't need to link with libssl.so when we already depend on openssl. 
This fixes the hidden-symbol linker warnings.

BUG=2149
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1927004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 22:41:36 +00:00
wu@webrtc.org
27c0408a16 Suppressing tsan errors on libjingle_unittest and libjingle_peerconnection_unittest. 
BUG=1205,2080
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1924004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4458 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 22:41:15 +00:00
fischman@webrtc.org
caa7024b86 PeerConnectionTest.java: build on android bots as well as linux ones. 
BUG=1796
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1921005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 21:56:30 +00:00
henrike@webrtc.org
a543114004 Removes no longer needed valgrind-libjingle folder. Was workaround for some bots using wrong valgrind script. 
TBR=wu@webrtc.org

BUG=2146

Review URL: https://webrtc-codereview.appspot.com/1920004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4454 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 17:53:39 +00:00
wu@webrtc.org
d40b4d9685 Fix libjingle memory bots by suppressing some of the errors. 
BUG=1205,2153
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1923004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4453 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 17:32:36 +00:00
mflodman@webrtc.org
d4412feeb0 Adding possibility to use encoding time when trigger underuse for frame based overuse detection. 
BUG=
TEST=Added unittest.
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1885004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4452 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 16:42:21 +00:00
xians@webrtc.org
09e8c47ee5 Merge r4374 from stable to trunk. 
r4374 was mistakenly committed to stable, so this is to re-merge back to trunk.

Store the sequence number in StopSend() and resume it in StartSend().

When restarting the microphone device, we call StopSend() first, then
StartSend() later. Since we reset sequence number in StopSend(), it sometimes
causes libSRTP to complain about packets being replayed. Libjingle work around
it by caching the sequence number in WebRtcVoiceEngine.cc, and call
SetInitSequenceNumber() to resume the sequence number before StartSend().Store the sequence number in StopSend() and resume it in StartSend().

When restarting the microphone device, we call StopSend() first, then
StartSend() later. Since we reset sequence number in StopSend(), it sometimes
causes libSRTP to complain about packets being replayed. Libjingle work around
it by caching the sequence number in WebRtcVoiceEngine.cc, and call
SetInitSequenceNumber() to resume the sequence number before StartSend().

This patch fixes this problem by storing the sequence number in StopSend(), and
resume it in StartSend(). So that we can remove the workaround in libjingle.

BUG=2102
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1922004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4451 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 16:30:19 +00:00
xians@webrtc.org
8fff1f065e Merge r4394 from stable to trunk. 
r4326 was mistakenly committed to stable, so this is to re-merge back to trunk.

Fixed the AGC and interface problems on the new path.

In order to make the AGC work properly, we need to cache the volume value passed
by the callback, compare it with the value returned by
shared->transmit_mixer()->CaptureLevel(). If they are the same, we need to
return 0 to indicate no volume needs changing, otherwise return the new volume.
By doing this, we avoid setting the volume all the same, which allows the users
to change the volume manually.

This patch also fixes some minor issues with the interfaces too: make the int
channel[] const, and correct the order of the input params in
channel::Demultiplex.

R=tommi@webrtc.org

BUG=[2134]
TEST=compile && manual AGC test

Review URL: https://webrtc-codereview.appspot.com/1921004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4450 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 16:27:42 +00:00
xians@webrtc.org
2f84afad30 Merge r4326 from stable to trunk. 
r4326 was mistakenly committed to stable, so this is to re-merge back to trunk.

Add new interface to support multiple sources in webrtc.
CaptureData() will be called by chrome with a flag |need_audio_processing| to
indicate if the data needs to be processed by APM or not. Different from the old
interface that will send the data to all voe channels, the new interface will
specify a list of voe channels that the data is demultiplexing to.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4449 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 16:23:37 +00:00
turaj@webrtc.org
7126b38d8f Handel zero correlation if at the same time distortion is also zero. 
This is the conversation I had with Henrik Lundin regarding this problem.

Me:
In Expand::AnalyseSignal() we compute correlation and distortion, then calculate the ratio of correlation to distortion. There if distortion is zero we expect that correlation to be zero. Although in practice this might be true, I suppose we rarely hit into absolutely periodic signal, but in one of the tests the assertion in line 455 of expand.cc was triggered. The distortion is computed over a shorter length of the signal, while correlation is computed over longer segments. Therefore, I guess, if the signal has just enough zeros at the beginning we can end up in situation that distortion is zero but not the correlation. Do you agree? I didn't have time to attempt to solve this, but if my line of thought is correct, we should not have that assert. Perhaps, if correlation is zero we set the ratio to 0, otherwise, ratio would be the largest value of its own type. Any thoughts?

Henrik:
I agree with you. Go ahead with your solution.

R=minyue@google.com

Review URL: https://webrtc-codereview.appspot.com/1888006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4448 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 16:05:09 +00:00
pbos@webrtc.org
2d1a55caed Add some virtual and OVERRIDEs in webrtc/modules/audio_coding/ 
BUG=163
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1900004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4447 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 15:54:00 +00:00
pbos@webrtc.org
e72428442d Fix some chromium-style warnings in webrtc/modules/desktop_capture/ 
BUG=163
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1904004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4446 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 15:32:43 +00:00
pbos@webrtc.org
0193158634 Fix some chromium-style warnings in webrtc/modules/pacing/ 
BUG=163
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1902005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4445 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 15:18:19 +00:00
pbos@webrtc.org
f3e4ceee47 Fix some chromium-style warnings in webrtc/modules/rtp_rtcp/ 
BUG=163
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1904005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4444 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 15:17:19 +00:00
pbos@webrtc.org
8f23df51d4 Fix some chromium-style warnings in webrtc/modules/remote_bitrate_estimator/ 
BUG=163
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1905004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4443 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 15:16:52 +00:00
pbos@webrtc.org
4fac8a4699 Fix some chromium-style warnings in webrtc/modules/bitrate_controller/ 
BUG=163
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1903004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4442 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 15:16:20 +00:00
phoglund@webrtc.org
a96d8771f2 Added libjingle_peerconnection_java_unittest to buildbot_tests.py 
The test apparently needs a custom LD_PRELOAD, so I made the script capable of handling custom environments.

TBR=kjellander@webrtc.org
BUG=1796

Review URL: https://webrtc-codereview.appspot.com/1916004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4441 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 10:50:30 +00:00
andrew@webrtc.org
0a4ca8f0bb Move internal aec_core defines out of header. 
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1915004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4440 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 08:13:08 +00:00
wu@webrtc.org
7446870a0f Suppress failing tests on Linux Memcheck bot. 
BUG=2153
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/1914004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4439 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 23:36:42 +00:00
wu@webrtc.org
9c9fc767b1 Fixing the memory check bots by suppressing some of the tests. 
BUG=1205,2078,2080
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1913004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4438 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 22:54:08 +00:00
wu@webrtc.org
933946ac55 Suppress libjingle_peerconnection_unittest failures on linux memcheck build bot. 
BUG=2153
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1912004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4437 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 22:28:29 +00:00
wu@webrtc.org
0342e65f8d Disable peerconnection tests that are failing on memcheck. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1910006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4436 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 22:28:14 +00:00
wu@webrtc.org
ae7bf1525b Disable p2p tests that are failing on memory test. 
BUG=1972
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1911004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4435 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 21:40:39 +00:00
fischman@webrtc.org
b59c6dd397 Add svn:ignore properties for all spuriously-removed dirs on Linux64 Release (internal). 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4434 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 19:34:07 +00:00
fischman@webrtc.org
85f07f59ee PeerConnectionTest.java: use java_home gyp var instead of hardcoding /usr. 
BUG=1796
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1899005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4433 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 18:11:07 +00:00
turaj@webrtc.org
fd7e3c52d8 Correcting Turaj's email. 
TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1910004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4432 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 17:25:07 +00:00
fischman@webrtc.org
3d496fb046 Roll chromium_revision 205140:214260 to pick up build fixes for ninja iOS device build. 
TESTED=git try
BUG=2106
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1888005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4431 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 17:14:35 +00:00
henrike@webrtc.org
9638564340 Adds no parent to talk folder. 
BUG=1933
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1896004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4430 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 15:51:54 +00:00
pbos@webrtc.org
7f7162a003 Fix some chromium-style warnings in webrtc/modules/video_coding/ 
BUG=163
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1901005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4429 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 15:18:31 +00:00
pbos@webrtc.org
e6c3966530 Fix some chromium-style warnings in webrtc/test/ 
BUG=163
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1907004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4428 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 13:08:38 +00:00
pbos@webrtc.org
a6f56acc53 Fix some chromium-style warnings in webrtc/tools/ 
BUG=163
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1908004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4427 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 12:50:59 +00:00
pbos@webrtc.org
096515b070 Fix some chromium-style warnings in webrtc/modules/audio_device/ 
BUG=163
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1897005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4426 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 12:32:59 +00:00
braveyao@webrtc.org
10bbfeff5b Apprtc: add 'event' parameter to onkeydown event handler. 
BUG=
TEST=Manual test
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/1898005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4425 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 09:27:49 +00:00
agalusza@google.com
d818dcb939 Sets up framework for decoding with errors: collects frame sizes (in number of packets) in JB and passes this information to VCMSessionInfo with rtt_ms as FrameData. 
R=marpan@google.com, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1841004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4424 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 21:48:11 +00:00
henrike@webrtc.org
a0b2f1794b Adds files still expected by the libjingle bots. 
BUG=2146
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1897004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4423 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 21:34:08 +00:00
fischman@webrtc.org
d6134c7cfd PeerConnectionTest.java: make the test work for the bots' v4l2loopback. 
- Make the test agnostic to the actual resolution used, since v4l2_file_player
  is playing a non-640x480 file (go/httfw)
- Teach DeviceInfoLinux::FillCapabilityMap() about I420 since that's what
  v4l2_file_player is feeding.

Requires https://gist.github.com/fischman/2e9a9b2efd2ad363ef82 be applied to the
v4l2loopback driver code.

BUG=1796
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1891004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 20:43:15 +00:00
fischman@webrtc.org
147d44a450 AppRTCDemo: replace the use of query-string parameters for pre-JB devices. 
Replaces the use of a query-string parameter with a (once-per-session)
JS-to-Java function call, because query-string parameters on file:// URLs are
busted on ICS and earlier Android releases
(https://code.google.com/p/android/issues/detail?id=17535).

Also added channel.html to the list of inputs to cause edits to it to cause a
rebuild of the .apk.

BUG=1949
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1890004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 19:07:33 +00:00
niklas.enbom@webrtc.org
7694562805 Land http://webrtc-codereview.appspot.com/1632005/ 
TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1895004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4420 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 18:37:32 +00:00
henrike@webrtc.org
ea40bd0cc8 Presubmit script for preventing changes to protected files and add the full list of those files. 
BUG=2090
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1855004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4419 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 18:20:07 +00:00
elham@webrtc.org
c0aa29c98c Updated WebRTC version to 3.37 
TBR=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1894004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4417 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 16:57:21 +00:00
phoglund@webrtc.org
8400246fce Improved error messages when binaries are missing. Also stderr = stdout now. 
Now that these scripts are called from browser tests, we need to print everything on stdout since the tests will throw away stderr when invoking programs. I chose to assign sys.stderr to sys.stdout. Otherwise I would have missed stuff like parser.error, which print to stderr.

The error message will get improved because the old code did not catch the case when the binary was missing, which lead to a very confusing error when that was the case. This gets fixed now.

BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1886004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4416 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 11:01:03 +00:00
kma@webrtc.org
f87177a757 To fix a bug in InverseFFTAndWindow() function in AECM. 
It's a bufer overwritting issue, and thus Android AppRTCDemo app was broken (reported by Ami).
Tested with audioproc offline test. Bit-exact.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4415 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 23:43:33 +00:00
henrike@webrtc.org
1e09a71126 Update talk folder to revision=49952949 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4413 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 19:17:59 +00:00
fischman@webrtc.org
367f640bea webrtc/.gitignore: add parts of talk/examples/android and third_party/llvm to the list. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1887004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4412 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 17:50:10 +00:00
kma@webrtc.org
b6a6a24fda Updated WebRtcNsx_PrepareSpectrumNeon() in accordance with the new real FFT interface in APM. For reference, you can check https://webrtc-codereview.appspot.com/1830004/diff/92001/webrtc/modules/audio_processing/ns/nsx_core.c, line 594 "static void PrepareSpectrumC()". 
Tested with audioproc. Bit exact.

R=andrew@webrtc.org, johannkoenig@google.com

Review URL: https://webrtc-codereview.appspot.com/1859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4411 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 16:24:34 +00:00
braveyao@webrtc.org
b6433b7a1e Access receiving_ under receive_cs critical section 
Note: InsertRTPPacket/InsertRTCPPacket could be merged into 
ReceivedRTPPacket, as there are no other callers.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1869005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 09:02:46 +00:00
sergeyu@chromium.org
abab1d8456 Don't set clang_use_chrome_plugins in common.gypi 
This caused a failure on chrome os ASAN bots (where that flag is disabled):
http://build.chromium.org/p/chromium.memory/builders/Chromium%20OS%20%28x86%29%20ASAN/builds/5491

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1882004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 00:55:46 +00:00
henrike@webrtc.org
14c966c706 Fixes resources and data path in modules_unittests.isolate. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1859005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4407 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 22:44:04 +00:00