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4673 Commits

Author SHA1 Message Date
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
stefan@webrtc.org
717d147ebb Support sending multiple report blocks and keeping track of statistics on several SSRCs. 
BUG=1811
TEST=vie_auto_test --automated, voe_auto_test --automated, trybots
R=andresp@webrtc.org, tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1768004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 13:39:27 +00:00
stefan@webrtc.org
9de89a6f6b Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered. 
R=pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1782004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4321 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 12:42:15 +00:00
stefan@webrtc.org
452d853c43 Fix three uninitialized members in rtp_receiver_impl.cc. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1781004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4320 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 10:54:56 +00:00
pbos@webrtc.org
08933a5dfb Initialize payload-type frequency in channel.cc. 
Uninitialized values triggered divide-by-zero crashes in voe_auto_test.

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

Review URL: https://webrtc-codereview.appspot.com/1780004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4319 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 10:06:29 +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
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
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
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
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
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
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
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
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
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
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
sergeyu@chromium.org
3348ae2b97 mac: Mark kCGLPFAFullScreen as allowed in a 10.6-only function. 
kCGLPFAFullScreen is marked deprecated starting with 10.6 in the 10.9 SDK,
but it's functional on 10.6 and this code only runs on 10.6 and will go away
when support for 10.6 is dropped.

BUG=webrtc:1958
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1710004

Patch from Nico Weber <thakis@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4255 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-21 23:33:10 +00:00
hclam@chromium.org
6eb53f71d6 Fix memory bot failure 
Exit the method with critical setting held. This should make
the memory bot happy.

TBR=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1704005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-20 23:01:39 +00:00
hclam@chromium.org
2e402ce873 Enqueue packet in pacer if sending fails 
If a packet cannot be sent while pacer is in use it should be
queued. This avoid packet loss due to congestion.

BUG=1930
R=pwestin@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1693004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-20 20:18:31 +00:00
mikhal@webrtc.org
9ca7360b97 VCM: removing max jitter estimate 
BUG= 1921
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1690004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4249 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-20 20:13:07 +00:00
stefan@webrtc.org
8ccb9f9716 Fixes some pacer/padding issues found while testing. 
- A bug was introduced in r4234 causing no paced packets to be sent.
- Only update the sequence number counter if a padding packet is actually going to be sent, to avoid packet loss.
- Have all packets go through the pacer if pacing is enabled to avoid reordering.
- Fix race condition on reading capture_time_ms_/timestamp_ in rtp_sender.cc.

BUG=1837
TEST=trybots and vie_auto_test --automated
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1682004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4246 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-19 14:13:42 +00:00
fbarchard@google.com
d7148c86c5 Use 3 threads for higher than 720p resolutions 
BUG=1893
TEST=untested
R=ajm@google.com, andrew@webrtc.org, dingkai@google.com, marpan@google.com, marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1684004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4243 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-18 22:06:42 +00:00
hclam@chromium.org
30fb7b83d5 Add a log message to see video delay break down 
Shows video delay in terms of:
1. Min playout delay
2. Jitter delay
3. Max decode time
4. Render delay

R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1674004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4242 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-18 21:37:09 +00:00
sergeyu@chromium.org
a20eb91154 Make ScreenCapturerMac work in versions of OSX before Lion. 
The screen capturer was broken when moving code to webrtc: width
and height parameters for glReadPixels were swapped by mistkake.

BUG=crbug.com/244102
R=alexeypa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1678005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4239 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 22:22:40 +00:00