generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
3,681 Commits 1 Branch 0 Tags 108 MiB
6998c8ef7a
Commit Graph

3681 Commits

This Branch
This Branch
All Branches
Author SHA1 Message Date
niklas.enbom@webrtc.org
a5961b855e Update third party license file 
R=jan.linden@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3939 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 18:49:28 +00:00
pwestin@webrtc.org
0d95e06a2f Bugfix custom call stop. 
BUG=1717
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1388004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3938 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 18:25:03 +00:00
andrew@webrtc.org
ea83c6ac9d Allow voe_cmd_test to select Opus mono (now the default). 
* Opus handles stereo and mono on the same payload type, so we need a different mechanism to choose between them.
* Assorted cleanups.

BUG=webrtc:1710
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3937 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 15:57:36 +00:00
andrew@webrtc.org
8c845cb623 Relax VoE's max packet length threshold. 
The earlier threshold would cause packets from a currently available
codec (L16, 32 kHz, stereo) to be discarded.

TESTED=voe_cmd_test using L16, 32 kHz, stereo now works properly.
R=henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1305008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3936 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 15:28:02 +00:00
phoglund@webrtc.org
258f55efc0 Disabled flaky test. 
BUG=1719
TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1387004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3935 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 12:35:00 +00:00
pbos@webrtc.org
68e5a68f07 Revert 3933 "Remove traces of deprecated WebRtc_Word types." 
> Remove traces of deprecated WebRtc_Word types.
> 
> BUG=314
> R=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1385004

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1386004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3934 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 09:30:12 +00:00
pbos@webrtc.org
265a5d298a Remove traces of deprecated WebRtc_Word types. 
BUG=314
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1385004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3933 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 09:11:20 +00:00
braveyao@webrtc.org
3c48f31e5b WebRTCDemo Android app to route audio to headphone when it's plugged in. 
BUG=1654
TEST=WebRTCDemo app

Review URL: https://webrtc-codereview.appspot.com/1348004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3932 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 03:18:00 +00:00
fbarchard@google.com
03d0c66376 Make libyuv fat on linux instead of thin. 
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/1382004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3931 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 01:01:24 +00:00
andrew@webrtc.org
28e82bfec6 Replace Resampler with PushResampler in transmit_mixer. 
* VoE can now exchange 44.1 kHz audio with AudioDevice.
* Changes still required in AudioDevice to remove the 44 kHz workarounds and
enable native 44.1 kHz.

BUG=webrtc:1395
TESTED=voe_cmd_test loopback running through codecs using all combinations of {8, 16, 32} kHz and {1, 2} channels, and Opus (48 kHz, stereo)
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1373004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3930 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 00:30:36 +00:00
andrew@webrtc.org
342353780d Consolidate common_audio into a single target. 
In principle should reduce gyp processing time, but the difference was not measurable. In any case, it's a good simplification that aligns with having a single common_video target.

R=bjornv@webrtc.org, kma@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1375004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3928 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 23:43:26 +00:00
andrew@webrtc.org
dff69c56b0 Add AEC suppression level option to audioproc. 
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/1368007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3927 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 23:01:09 +00:00
sergeyu@chromium.org
23516638fa Move WEBRTC_THREAD_RR and WEBRTC_CLOCK_TYPE_REALTIME to system_wrappers.gypi . 
WEBRTC_THREAD_RR and WEBRTC_CLOCK_TYPE_REALTIME are used only in code compiled
in system_wrappers, so they don't need to be in common.gypi.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1368005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3926 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 22:53:51 +00:00
andresp@webrtc.org
72d0b0cf1f Add self to video_engine watchlist. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1305009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3925 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 22:20:53 +00:00
stefan@webrtc.org
4980679d35 Fixes two bugs in receive statistics. 
- Reported bitrate wasn't reset correctly when no frames had been received.
- Internal framerate estimate wasn't reset when no frames had been received.

BUG=1713

Review URL: https://webrtc-codereview.appspot.com/1377004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3924 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 22:05:07 +00:00
pwestin@webrtc.org
d35964a1ce Fixing AV sync. 
Increased 2 const to allow for a bigger difference in AV sync.

BUG=1711

Re-wrote the ComputeDelays to be readable and remove the possibilities of returning values lower than base_target_delay_ms

R=mflodman@webrtc.org, mikhal@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1367004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3922 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 16:06:10 +00:00
mikhal@webrtc.org
6faba6edc9 VCM: Setting buffering delay in timing 
Review URL: https://webrtc-codereview.appspot.com/1338004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3921 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 15:39:34 +00:00
mikhal@webrtc.org
dd807ac474 Adding buffered mode to loopback test 
Review URL: https://webrtc-codereview.appspot.com/1371004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3920 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 15:19:47 +00:00
solenberg@webrtc.org
8efc623fc2 Apply Chromium C++ style to RemoteRateControl. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3919 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 08:33:46 +00:00
sergeyu@chromium.org
15e32ccd30 Add DesktopCapturer interface for desktop capturers. 
The new DesktopCapturer interface will be used for screen and window
captures. Beside DesktopCapturer itself also added classes/interfaces
that it depends on.

R=alexeypa@chromium.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1322007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3917 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-29 20:10:57 +00:00
mikhal@webrtc.org
865ada3a52 Don't reset the last je value and mode 
Review URL: https://webrtc-codereview.appspot.com/1369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3916 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-29 19:09:41 +00:00
andrew@webrtc.org
50b2efef6e Add a wrapper around PushSincResampler and the old Resampler. 
The old resampler is used whenever it supports the requested rates. Otherwise
the sinc resampler is enabled.

Integrated with output_mixer in order to test the change through
output_mixer_unittest. The sinc resampler will not yet be used, since we don't
feed VoE with any rates that trigger it.

BUG=webrtc:1395
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1355004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3915 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-29 17:27:29 +00:00
stefan@webrtc.org
5b7120c81b Fix two issues where we might end up busy looping in decoder_render mode. 
This happens if
- Next frame is far into the future (> 200 ms).
- Next frame is ready for decode/render but incomplete.

BUG=1696
TESTS=trybots

Review URL: https://webrtc-codereview.appspot.com/1354005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3914 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-29 16:41:30 +00:00
pwestin@webrtc.org
b0061f94b2 Enable Nack pacing. 
Review URL: https://webrtc-codereview.appspot.com/1357004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3912 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-27 00:41:08 +00:00
mikhal@webrtc.org
47128ab5ab Removing vie file related code from vie_custom_call 
Follow up on https://code.google.com/p/webrtc/source/detail?r=3900

Review URL: https://webrtc-codereview.appspot.com/1361004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3911 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 20:09:54 +00:00
pwestin@webrtc.org
4e545b33b3 Fixed remaining nits from Stefan 
Review URL: https://webrtc-codereview.appspot.com/1323007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3910 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 15:23:34 +00:00
andrew@webrtc.org
8fc05feed4 Add a push-based wrapper around SincResampler. 
Includes a unittest to ensure we meet the same quality thresholds as SincResampler (modulo quantization error).

BUG=webrtc:1395

Review URL: https://webrtc-codereview.appspot.com/1323011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3909 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 14:56:51 +00:00
fbarchard@google.com
42b0b84367 libyuv r680 fixes arm version of I444ToARGB and some lint changes 
BUG=none
TEST=libyuv unittests pass on arm with Neon disabled.
Review URL: https://webrtc-codereview.appspot.com/1356005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3908 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 02:23:32 +00:00
andrew@webrtc.org
1acb3b33bc Add comfort noise disabling and routing mode selection to audioproc. 
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/1358004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3907 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 00:39:27 +00:00
mikhal@webrtc.org
4cea79b830 Removing another instance of file api 
Review URL: https://webrtc-codereview.appspot.com/1356004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3906 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 23:31:27 +00:00
vikasmarwaha@webrtc.org
77ac84814d Added new demo states.html & updated existing demos to work on firefox. 
Review URL: https://webrtc-codereview.appspot.com/1327007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3905 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 23:22:03 +00:00
pwestin@webrtc.org
91563e42da Fix the encoder pause logic. 
BUG=1691

Review URL: https://webrtc-codereview.appspot.com/1352004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3904 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 22:20:08 +00:00
mikhal@webrtc.org
381da4be9c VCM: Adding API for the size(duration) of the jitter buffer. 
Refers to the duration in time of the frames which are ready to be sent to the decoder.

Review URL: https://webrtc-codereview.appspot.com/1319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3903 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 21:45:29 +00:00
mikhal@webrtc.org
8392cd9edd VCM/JB: Using last decoded state for waiting for key 
relanding 1323006

BUG=

Review URL: https://webrtc-codereview.appspot.com/1354004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3902 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 21:30:50 +00:00
mikhal@webrtc.org
dc3cd217b2 VCM/JB: FrameForDecoding->IncompleteFrameForDecoding 
- Update complete frame for decoding
- Remove FrameForDecodingNack

This CL should only be committed after issue http://webrtc-codereview.appspot.com/1313007/

Review URL: https://webrtc-codereview.appspot.com/1316007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3901 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 20:27:04 +00:00
mikhal@webrtc.org
b84f13f185 Disabling avi file interface 
Review URL: https://webrtc-codereview.appspot.com/1351004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3900 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 18:07:32 +00:00
pwestin@webrtc.org
52aa019e98 Avoid adding duplicates in pacer lists. 
Review URL: https://webrtc-codereview.appspot.com/1329007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3899 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 17:35:56 +00:00
stefan@webrtc.org
cb60fb2e6c Make sure timestamps are monotonically increasing. 
BUG=1563

Review URL: https://webrtc-codereview.appspot.com/1325008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3898 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-24 23:54:31 +00:00
fbarchard@google.com
c63772eb39 libyuv license file updates for Android WebView license check. 
BUG=none
TEST=try bots
Review URL: https://webrtc-codereview.appspot.com/1313013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3897 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-24 02:12:49 +00:00
andrew@webrtc.org
df9c0e5ec9 Revert 3892 "VCM/JB: Using last decoded state for waiting for key" 
> VCM/JB: Using last decoded state for waiting for key
> 
> Review URL: https://webrtc-codereview.appspot.com/1323006

Although I have no idea why, it appears this might be causing failures in ViEStandardIntegrationTest.RunsFileTestWithoutErrors. I was unable to reproduce locally. This is a trial revert to verify. If the errors continue to happen, I will restore this change.

TBR=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1321010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3896 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-24 02:13:18 +00:00
andrew@webrtc.org
2e65346e98 Add a root codereview.settings file. 
Chromium has one of these as well:
http://src.chromium.org/viewvc/chrome/codereview.settings

It's needed for drover reverts to work properly.

TBR=kjellander

Review URL: https://webrtc-codereview.appspot.com/1319014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3895 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-24 02:08:50 +00:00
fbarchard@google.com
6fec15941a Update template to follow chromium copyright style 
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/1322008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3894 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-24 01:01:28 +00:00
andresp@webrtc.org
b5eeaa92ba Adding extra options to interact with external encoder/decoder. 
Review URL: https://webrtc-codereview.appspot.com/1327006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3893 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 22:50:53 +00:00
mikhal@webrtc.org
1248d4effc VCM/JB: Using last decoded state for waiting for key 
Review URL: https://webrtc-codereview.appspot.com/1323006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3892 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 20:57:06 +00:00
fbarchard@google.com
3a9a3cdaa2 Roll libyuv to r676 for improved llvm compatibility 
BUG=none
TESTED=try bots
Review URL: https://webrtc-codereview.appspot.com/1313010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3891 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 18:05:46 +00:00
stefan@webrtc.org
8ca8a71de2 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." 
This reverts commit aae26db1da5803482b094357c546b8454ab1c26d.

BUG=1613
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1327008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3890 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 16:48:32 +00:00
turaj@webrtc.org
a942692725 Buf fix for r3883. 
Review URL: https://webrtc-codereview.appspot.com/1319012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3889 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 16:08:29 +00:00
stefan@webrtc.org
ccd4b2aec8 Add a default RTT to CallStats and use different values for buffered/real-time mode. 
BUG=1613

Review URL: https://webrtc-codereview.appspot.com/1326007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3888 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 15:58:23 +00:00
pbos@webrtc.org
d25b602dc0 VP8: Avoid copying the codec struct on Reset(). 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1319013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3887 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 13:08:04 +00:00
mflodman@webrtc.org
efdf778d3f BUG=1351 
Propose of this CL: Close the camera properly on MacOS in order to allow other apps to use it.

Changes in this CL:
1. video_capture_qtkit_info_objc.mm _captureDevicesInfo is never released. I have found this memory leak using Instruments from XCode. The patch is releasing it in dealloc.

2. In video_capture_qtkit_objc.h:
a) _captureDeviceName is not needed. Is allocated in the class but never used.
b) I don't see the role of the  NSAutoreleasePool. also if you use it you have to release it when the class is destroyed. Otherwise you will leak memory. Libjingle has for each thread a pool on mac os.

3. In video_capture_qtkit_objc.mm
a) the camera is not stopped properly . See the changes from dealloc. NOTE : If you don't call [[_captureVideoDeviceInput device] close] other apps will not be able to use the camera since you are not closing your app

b) Removed QTCaptureDevice* videoDevice = (QTCaptureDevice*)[_captureDevices objectAtIndex:0]; I don't know why this because the desired camera is opened in setCaptureDeviceById and can be different than position 0 in the camera array. At this moment if you have two cameras and user want to pick the one on index 1 the app also locks the one on 0 .

Other changes I have done to improve (and are not in this CL):
a) I have set the FPS properly to the desired. I have succeeded to reduce the CPU with 3 % doing this. The current code for setting FPS is commented in webrtc
b) I have removed _rLock from the equation. I don't know if it's good or not but I hadn't understood what exactly we are trying to protect with this. Anyway in the current implementation is never released.

Review URL: https://webrtc-codereview.appspot.com/1097014

Patch from Silviu Caragea <silviu.cpp@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3886 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 11:57:56 +00:00