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Author SHA1 Message Date
henrike@webrtc.org
28e2075280 Adds trunk/talk folder of revision 359 from libjingles google code to 
trunk/talk


git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 00:45:36 +00:00
tnakamura@webrtc.org
6aa6229953 Update version number to 3.35 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1778004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4316 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-09 18:43:02 +00:00
tnakamura@webrtc.org
c79b9295cd Update version number to 3.34 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1770006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4315 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-09 18:40:52 +00:00
pbos@webrtc.org
fc496d95df Add root_path_android.cc to webrtc/test/Android.mk. 
Fixes the broken android-platform build (build that uses .mk files).

TBR=andrew@webrtc.org,henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/1777004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4314 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-09 15:24:16 +00:00
pbos@webrtc.org
f3f1358360 Fixed implicit-int-conversion bugs. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1776004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4313 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-09 14:04:46 +00:00
stefan@webrtc.org
cab716cc7d Fix a circular dependency by removing an unnecessary dependency, add a missing include_tests check and missing lib references for android. 
TBR=henrikg@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1776005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4312 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-09 13:43:24 +00:00
stefan@webrtc.org
f56d612c70 Create gyp target for bwe components. 
R=henrikg@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1775004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4311 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-09 12:32:35 +00:00
pbos@webrtc.org
af8d5afec9 Initial port of FullStackTest to new VideoEngine API. 
Deferring network loss, delay and such to a later CL.

BUG=1872
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1756004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4310 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-09 08:02:33 +00:00
henrike@webrtc.org
5fc4d34f54 Arguments need to be separated when implementing gyp-actions. 
TBR=andrew@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1774004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4309 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-09 02:08:25 +00:00
hclam@chromium.org
1a7b9b94be Cleanup WebRTC tracing 
The goal of this change is to:
1. Remove unused tracing events.
2. Organize tracing events to facilitate measurement of end to end latency.

The major change in this CL is to use ASYNC_STEP such that operation
flow can be traced for the same frame.

R=marpan@webrtc.org, pwestin@webrtc.org, turaj@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1761004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4308 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-08 21:31:18 +00:00
henrike@webrtc.org
e80a934b36 Added modules_unittests.isolate for ndk-apk builds. 
TBR=csharp@chromium.org, frankf@chromium.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1750004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4307 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-08 21:19:57 +00:00
henrike@webrtc.org
a950300b0e Disables unit tests that don't work on Android for Android. 
BUG=N/A
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1747004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4306 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-08 18:53:54 +00:00
henrike@webrtc.org
a2073af728 Fixes build breakage when building WebRTC in Chromium and having include_tests=1. 
TBR=fischman@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1770004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4305 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-08 18:14:58 +00:00
henrike@webrtc.org
bd3eee3e24 Fixes broken gyp-condition. 
TBR=andrew@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1771004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4304 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-08 17:34:20 +00:00
henrike@webrtc.org
34773d9b6b Unreverts revert: Makes it possible to find files used by some unit tests when running them as Chrome native tests. 
TBR=andrew@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1754005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4303 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-08 14:55:23 +00:00
pbos@webrtc.org
1932fe1865 Use scoped_ptr<> for loopback.cc 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1764004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4302 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-05 17:02:37 +00:00
stefan@webrtc.org
66b2e5c05a Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the 
rtp_rtcp implementation.

This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.

With this change the dead-or-alive and packet timeout APIs are removed.

TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1745004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-05 14:30:48 +00:00
mcasas@webrtc.org
d4d9480c05 Added gum4.html, a multiple camera opening demo, each opening with a different resolution and/or frame rate. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4300 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-05 09:12:04 +00:00
pbos@webrtc.org
db7d82f26f Revert 4298 "Makes it possible to find files used by some unit t..." 
> Makes it possible to find files used by some unit tests when running them as Chrome native tests.
> 
> BUG=N/A
> R=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1749004

Broke Android NDK/Android.mk builds.

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1752006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4299 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-05 08:49:09 +00:00
henrike@webrtc.org
caf2fcca6a Makes it possible to find files used by some unit tests when running them as Chrome native tests. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4298 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-05 04:15:38 +00:00
mflodman@webrtc.org
21beaf97e7 Adding Stefan as VideoEngine owner, removing Per. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1762004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-04 12:29:08 +00:00
braveyao@webrtc.org
0b8636a783 In AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(), continue enumerating if one individual device failed. 
BUG=
TEST=manual Test
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1753005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4295 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-04 07:24:12 +00:00
henrike@webrtc.org
1303af31d6 Makes it possible to build ndk-apks of native unit tests if the workspace is inside a chromium checkout. 
Alternative solution to http://webrtc-codereview.appspot.com/1748004/.

BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1753006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4294 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 21:50:33 +00:00
pbos@webrtc.org
d900e8bea8 Proper spacing for end-of-namespace comments. 
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 15:12:26 +00:00
tina.legrand@webrtc.org
45426eadf5 In call to Opus decoder: frame length too large 
BUG=https://code.google.com/p/webrtc/issues/detail?id=1201
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1752004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4292 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 13:32:04 +00:00
tina.legrand@webrtc.org
f6f033f8bd Possible divide by 0 in ACM. 
BUG=https://code.google.com/p/webrtc/issues/detail?id=1551
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1757004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4291 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 12:00:14 +00:00
tina.legrand@webrtc.org
b1698ab827 Error in update of read index in ACM 
Fixing a bug where we increase read index with too few samples when the input is stereo.

BUG=https://code.google.com/p/webrtc/issues/detail?id=714
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1753004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 09:25:34 +00:00
tommi@webrtc.org
ecd3c800c4 Add Magnus to root owners. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1752005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4289 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 08:21:41 +00:00
pbos@webrtc.org
c66aaaf921 Rename unit_test.{cc,h} under module_unittest. 
Squelches the following Windows trybot warning:
warning LNK4042: object specified more than once; extras ignored

BUG=
R=andrew@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1758004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4288 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 07:56:33 +00:00
yujie.mao@webrtc.org
510dfad636 Update myself in webrtc watchlist 
BUG=NONE
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4287 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 01:13:18 +00:00
pbos@webrtc.org
65a1f2cb2b Remove log of undefined input values in GetCodec. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1755004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4286 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-02 13:02:14 +00:00
pbos@webrtc.org
504af45a6f Diff NTP and internal once in VideoCaptureImpl. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1754004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4285 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-02 10:15:43 +00:00
fischman@webrtc.org
546c91dc2e Build all java files into jar for each module on Android 
BUG=None
TEST=All java files in each module are built into jar and used by WebRTCDemo app
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1696004

Patch from Jeremy Mao <yujie.mao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-01 17:52:39 +00:00
yujie.mao@webrtc.org
d4803ced60 WebRTCViEDemo: Use global reference when passing variables across different threads 
There are JNI local reference changes in ICS when Android SDK
target level API >= 14.
http://android-developers.blogspot.com/2011/11/jni-local-reference-changes-in-ics.html

BUG=NONE
TEST=WebRTCViEDemo works well using MediaCodec Decoder/Renderer
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1744004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-01 14:55:37 +00:00
braveyao@webrtc.org
90cc3b95b7 Android opengles renderer: add thread sync to swap frame and draw native. 
BUG=1616
TEST=Manual Test
R=fischman@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1738005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-28 23:53:11 +00:00
hclam@chromium.org
5616abadf5 Suppress excessive logging in video_coding 
Only prints the warning message if a frame was dropped.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1735004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4278 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 19:47:40 +00:00
henrike@webrtc.org
2a7fd5355d Moves tools/update.py to trunk/webrtc/tools and updates it so that it no longer pulls any information from the DEPS file. 
BUG=N/A
R=andrew@webrtc.org, kjellander@google.com, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1730004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 18:36:28 +00:00
henrike@webrtc.org
83cebb25d7 Removes unused main function that is poluting the build. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1728005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4276 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 18:31:13 +00:00
fischman@webrtc.org
0021632f40 Re-add WebRTCDemo dependencies as dependencies (not just inputs) because they also need to be built for this target! 
BUG=1980
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1734004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4275 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 17:35:32 +00:00
fischman@webrtc.org
1d4a2d5daf Move TickTime::QueryOsForTicks out-of-line 
This inline function is no longer expanded on arm Android, but on x86 Android it
will still be expanded. Move it out-of-line to make things consistent.

This change list will also fix a potential bug on webrtc for Android:
Since the inline function won't be expanded on arm Android,
TickTime::MillisecondTimestamp and Clock::GetRealTimeClock()->TimeInMilliseconds
will be treated as function call, due to macro WEBRTC_CLOCK_TYPE_REALTIME's
guard defined in system_wrappers module they will get current time using
CLOCK_REALTIME.

But on x86 Android, the inline function will be expanded to where it's been
called, if the call happens in other compilation units which don't have
WEBRTC_CLOCK_TYPE_REALTIME definition, it will get current time using
CLOCK_MONOTONIC, while Clock::GetRealTimeClock()->TimeInMilliseconds will always
use CLOCK_REALTIME, then there will be two types of time in x86 Android which
will cause some weird issues like all received remote streams will be dropped
due to future render timestamp.

BUG=None
TEST=WebRTCViEDemo application works well on both arm and x86 Android
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1688004

Patch from Jeremy Mao <yujie.mao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4274 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 17:15:20 +00:00
stefan@webrtc.org
4cf1a8af69 Removes kStateFree and kStateDecoding, added a free_frames_ list which simplifies finding a free frame. 
The idea is to have all frames not in use be stored in free_frames_, and whenever a packet from a new frame arrives we can just pop a frame from free_frames_. When a frame is grabbed for decoding it will be removed from all lists, and will be added to free_frames_ when it's returned to the jitter buffer.

We should be able to remove the state enum completely later, as their state is defined by the list they are in. But I'll keep it around for now to simplify the cl.

TEST=try bots and vie_auto_test --automated
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1721004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4273 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 15:20:14 +00:00
phoglund@webrtc.org
7bcc7e3b43 Fixed bad parameter passing in compare_videos.py 
BUG=http://crbug.com/254932
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1733004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4272 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 14:05:26 +00:00
pbos@webrtc.org
2de80ddc72 Fix unnamed-type-template-args warnings on clang. 
BUG=
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1732004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4271 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 10:18:09 +00:00
fischman@webrtc.org
3145a642b7 Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes. 
BUG=1980
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4270 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 20:20:05 +00:00
mflodman@webrtc.org
e6168f5f41 Adding a first simple version of overuse detection, but not hooked up. 
BUG=
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1717004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4268 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 11:23:01 +00:00
mflodman@webrtc.org
1c986e7c89 Removed ViE file API. 
R=asapersson@webrtc.org, niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1723004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4267 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 09:12:49 +00:00
solenberg@webrtc.org
a5fd2f1348 Do basic parsing of RTCP headers in PcapFileReader to enable log filtering. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1697004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4266 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 08:36:07 +00:00
solenberg@webrtc.org
892d750ba6 Add *.DS_Store to .gitignore so that ._.DS_Store is ignored too. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1698004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4265 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 08:22:53 +00:00
solenberg@webrtc.org
91811e2b04 Remove unused multi stream bandwidth estimator. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1712004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4264 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-25 20:36:14 +00:00
stefan@webrtc.org
a4c5abb52a Make sure padding packets are sent. 
BUG=1837
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1717006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-25 15:46:16 +00:00