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1309 Commits

Author SHA1 Message Date
henrike@webrtc.org
89c674053e Adds all unittests to android NDK-APK framework. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1872004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4474 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-02 16:53:47 +00:00
pbos@webrtc.org
9162080527 Fix some chromium-style warnings in webrtc/modules/audio_processing/ 
BUG=163
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1902004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-02 11:44:11 +00:00
sergeyu@chromium.org
17758e96c5 Fix crash in DesktopRegion::Intersect(). 
BUG=crbug.com/266933
R=alexeypa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1938004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4468 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 19:51:04 +00:00
agalusza@google.com
a7e360e89b Removed lines preventing simultaneous kHardNack and decoding with errors. Also made changes recommended by gcl lint (with the exception of changing non-const references to pointers). 
Propagated orthogonal API for decoding with errors from VideoCodingModule to VCMJitterBuffer.
Modified VCMJitterBuffer to allow three error modes: kNoErrors, kSelectiveErrors, kWithErrors.

R=marpan@google.com, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1846004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4463 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 03:15:08 +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
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
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
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
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
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
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
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
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
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
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
sergeyu@chromium.org
099b8c9e8e Update include paths in device_info_external.cc 
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1875004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 18:41:43 +00:00
andrew@webrtc.org
61e596fc49 Add a Config class interface to AudioProcessing for passing options. 
Pass the Config down to all AudioProcessing components.

Also add an EchoCancellationImplWrapper to optionally create different
EchoCancellationImpls.

BUG=2117
TBR=turaj@webrtc.org
TESTED=git try

Review URL: https://webrtc-codereview.appspot.com/1843004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4400 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 18:28:29 +00:00
niklas.enbom@webrtc.org
8e3bbedacd Fix include path in video_capture_external.cc 
Fix build error introduced in r4337

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1873004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4397 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 16:55:58 +00:00
kma@webrtc.org
fc8aaf02e1 Formalized Real 16-bit FFT for APM. 
It also prepares for introducing Real 16-bit FFT Neon code from Openmax to SPL. CL https://webrtc-codereview.appspot.com/1819004/ takes care of that, but this CL is a prerequisite of that one.
Tested audioproc with an offline file. Bit exact.

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

Review URL: https://webrtc-codereview.appspot.com/1830004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-24 17:38:23 +00:00
sergeyu@chromium.org
d102e66ef9 Fix ScreenCapturerLinux not to use XDamage when requested. 
When moving this code to webrtc I added line "use_x_damage=true" for
debugging and forgot to remove it when landing this code, so the
capturer always tries to use XDamage.

BUG=crbug.com/263003
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1854004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-23 20:05:42 +00:00
fischman@webrtc.org
c6d5b50b41 AppRTCDemo: build fixes for iOS build in webrtc 
BUG=1421,1450,1451
TESTED=git try, also the same patch (along with a bunch of other, non-webrtc changes) in a libjingle checkout allows building iOS AppRTCDemo
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4371 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-18 02:02:07 +00:00
tnakamura@webrtc.org
d2102afa2a Undo libvpx include changes in r4348 to fix build. 
A longer term fix is needed, but this at least quickly unblocks the build.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1816005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-17 18:48:24 +00:00
tnakamura@webrtc.org
64e2cbf184 clean up incomplete revert in r4357 
Also revert r4319, will follow up with pbos

Reason for recent series of reverts: video freezes when testing with packet loss

R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1817004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4359 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 21:52:59 +00:00
tnakamura@webrtc.org
aa4d96a134 Revert r4301 
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 19:25:04 +00:00
pbos@webrtc.org
0c4e05afbb Include files from webrtc/.. paths in media_file/. 
BUG=1662
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1784005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4351 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 13:05:40 +00:00
pbos@webrtc.org
9b82dced8d Make sure first RTP packet counts as in-order. 
BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1811004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4350 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 13:03:35 +00:00
pbos@webrtc.org
2e10b8e4a0 Include files from webrtc/.. paths in bitrate_controller/. 
BUG=1662
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1787004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 12:54:53 +00:00
pbos@webrtc.org
a4407329d4 Include files from webrtc/.. paths in video_coding/. 
BUG=1662
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1783006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4348 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 12:32:05 +00:00
elham@webrtc.org
4a44ea21d7 Revert r4320 "Fix three uninitialized members in rtp_receiver_impl.cc" 
TBR=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1803004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-15 21:46:06 +00:00
elham@webrtc.org
4888fd4827 Revert r4321 "Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered" 
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1790006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4345 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-15 21:21:48 +00:00
elham@webrtc.org
b7eda43810 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on 
several SSRCs"

R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1774006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-15 21:08:27 +00:00
elham@webrtc.org
6f5707e184 Revert r4328 
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1774005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4343 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-15 20:59:52 +00:00
pbos@webrtc.org
df119c9a45 Remove dead video_capture for QuickTime. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4339 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-12 18:08:13 +00:00
pbos@webrtc.org
a9b74ad716 Include files from webrtc/.. paths in video_capture/. 
BUG=1662
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1788004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4337 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-12 10:03:52 +00:00
pbos@webrtc.org
8b06200802 Include files from webrtc/.. paths in utility/. 
BUG=1662
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1786004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4336 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-12 08:28:10 +00:00
pbos@webrtc.org
0ed57c51a3 Remove dead code testAPI.cc. 
BUG=
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1783005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4335 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-12 08:23:05 +00:00
pbos@webrtc.org
5aa3f1b4c0 Include files from webrtc/.. paths in video_render/. 
BUG=1662
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1782006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4334 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-12 08:12:08 +00:00
pbos@webrtc.org
811269df40 Include files from webrtc/.. paths in audio_device/. 
BUG=1662
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1785005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4330 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-11 13:24:38 +00:00
pbos@webrtc.org
db6e3f8bc5 Fix root-relative includes for pacing/. 
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4329 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-11 09:50:05 +00:00
stefan@webrtc.org
e4736eee20 Fixes a crash when sending SR reports from a sender only module. 
BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1790004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-11 08:28:35 +00:00
braveyao@webrtc.org
aeba6e8740 ModuleRTPRTCP call rtcp_sender_.TMMBR() directly instead of calling its own API. 
BUG=2051
TEST=autotest
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1790005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4327 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-11 08:06:37 +00:00
pbos@webrtc.org
96edd56170 Sorted headers under rtp_rtcp/. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1781005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4325 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 15:40:42 +00:00