buildbot@webrtc.org
a663d90ae3
(Auto)update libjingle 79104430-> 79104922
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7602 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 22:29:18 +00:00
glaznev@webrtc.org
5f38c8d1b8
Android AppRTCDemo improvements:
...
- Add a room list to ConnectActivity with buttons to add/remove rooms.
- Add loopback call button.
- Add option to toggle full screen / letterbox video.
- Add camera fps settings.
- Fix device to landscape orientation for HD video until issue 3936
will be fixed.
- Fix a few crashes by avoiding calling peer connection and
GAE signaling function while connection is closing.
- Better handling GAE channel error - catch channel exceptions and
display dialog with error messages.
BUG=3939, 3935
R=kjellander@webrtc.org , pthatcher@webrtc.org , tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7601 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 22:18:52 +00:00
pbos@webrtc.org
96a93259b3
Implement external decoder support in WebRtcVideoEngine2.
...
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/30839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7594 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 14:46:44 +00:00
henrik.lundin@webrtc.org
2236267b5e
Disable PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate under MSan
...
This test is flaky on MSan bots.
BUG=3980
TBR=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7591 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 13:38:50 +00:00
kjellander@webrtc.org
5072e0f6cd
Update Android projects to API level 21.
...
The update in https://webrtc-codereview.appspot.com/23309004
was not enough, so this updates to 21 instead.
This is required in order to roll chromium_revision to
keep up with Chrome, as third_party/android_tools have now
dropped support for API level 20.
Commands used:
third_party/android_tools/sdk/tools/android update project --name OpenSlDemo --target android-21 --path webrtc/examples/android/opensl_loopback
third_party/android_tools/sdk/tools/android update project --name WebRTCDemo --target android-21 --path webrtc/examples/android/media_demo/
third_party/android_tools/sdk/tools/android update project --name AppRTCDemo --target android-21 --path talk/examples/android/
Then I restored the changes of the ANDROID_SDK_ROOT -> ANDROID_HOME since it seems the Chromium build toolchain doesn't set it properly when
build/android/envsetup.sh is sourced.
BUG=
R=glaznev@webrtc.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7587 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 23:26:10 +00:00
kjellander@webrtc.org
c2c94a9a9f
Change default JVM location to /usr/lib/jvm/java-7-openjdk-amd64
...
Given that OpenJDK 1.7 is the recommended Java SDK for
Chromium these days, we should get rid of linking to the old
non-standardized link referring to a Sun Java 1.6 SDK.
Instead of requiring all users to set JAVA_HOME, I prefer
have the most common path as default and and close webrtc:2113
as won't fix after this is submitted.
BUG=2113
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7584 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 19:01:41 +00:00
kjellander@webrtc.org
78c222bfae
Update all .isolate files for the new format.
...
R=kjellander@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27809004
Patch from Marc-Antoine Ruel <maruel@chromium.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7583 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 18:08:09 +00:00
kjellander@webrtc.org
8a130c1084
Update Android projects to API level 20.
...
This is required in order to roll chromium_revision to
keep up with Chrome, as third_party/android_tools have now
dropped support for API level 19.
Commands used:
third_party/android_tools/sdk/tools/android update project --name OpenSlDemo --target android-20 --path webrtc/examples/android/opensl_loopback
third_party/android_tools/sdk/tools/android update project --name WebRTCDemo --target android-20 --path webrtc/examples/android/media_demo/
third_party/android_tools/sdk/tools/android update project --name AppRTCDemo --target android-20 --path talk/examples/android/
Then I restored the changes of the ANDROID_SDK_ROOT -> ANDROID_HOME since it seems the Chromium build toolchain doesn't set it properly when
build/android/envsetup.sh is sourced.
BUG=
R=glaznev@webrtc.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7582 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 17:13:37 +00:00
pbos@webrtc.org
b7ed7799e7
Implement conference-mode temporal-layer screencast.
...
Renames VideoStream::temporal_layers to temporal_layer_thresholds_bps to
convey that it contains thresholds needed to ramp up between them (1
threshold -> 2 temporal layers, etc.).
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1788,1667
Review URL: https://webrtc-codereview.appspot.com/23269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7578 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 13:08:10 +00:00
pbos@webrtc.org
3bf3d238c8
Configure A/V sync in WebRtcVideoEngine2.
...
Sets up A/V sync for the first video receive channel with the default
voice channel. This is only done when conference mode is disabled to
preserve existing behavior. Ideally we'd know which voice channel to
sync with here.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/23249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7577 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 12:59:34 +00:00
minyue@webrtc.org
2dc6f3154d
Adapting bitrate according to maxplaybackrate for Opus.
...
BUG=
R=mflodman@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7575 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 05:33:10 +00:00
tkchin@webrtc.org
14146e40aa
arm64 iOS build.
...
Allows successful build of arm64 libraries using
GYP_DEFINES="OS=ios target_arch=arm64 target_subarch=arm64".
Note that not all libraries will be NEON optimized (eg common_audio),
however most importantly libvpx will be. WEBRTC_ARCH_ARM needs to be
defined so that libvpx doesn't post-process, which is significantly
detrimental to performance.
BUG=3898
R=kjellander@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7573 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 00:14:39 +00:00
jiayl@webrtc.org
50ca986bc1
Improve the logging when a TCP connection is deleted.
...
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7572 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 23:50:54 +00:00
minyue@webrtc.org
8219529b98
Cleaning up r7562-7567.
...
Wrongly used git svn dcommit for committing a CL.
Then two reverts were applied.
Still something needs to be cleaned.
BUG=
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7568 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 08:23:54 +00:00
buildbot@webrtc.org
879fac81d1
(Auto)update libjingle 78822708-> 78823675
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7567 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 07:50:13 +00:00
minyue@webrtc.org
5f73a37597
Revert 7563 "before rebase" due to wrong submission
...
> before rebase
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7566 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 07:49:58 +00:00
minyue@webrtc.org
c11cc8d947
Revert 7564 "to submit" due to wrong submission
...
> to submit
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7565 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 07:46:47 +00:00
minyue@webrtc.org
de386bf67b
to submit
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7564 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 07:20:09 +00:00
minyue@webrtc.org
c673bb9f29
before rebase
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7563 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 07:19:57 +00:00
minyue@webrtc.org
0b62672576
adding default rates
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7562 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 07:19:49 +00:00
pbos@webrtc.org
776e6f289c
Use external VideoDecoders in VideoReceiveStream.
...
Removes direct VideoCodec use from the new API, exposes VideoDecoders
through webrtc/video_decoder.h similar to VideoEncoders.
Also includes some preparation for wiring up external decoders in
WebRtcVideoEngine2 by adding AllocatedDecoders that specify whether they
were allocated internally or externally.
Additionally addresses a data race in VideoReceiver that was exposed with this change.
R=mflodman@webrtc.org , stefan@webrtc.org
TBR=pthatcher@webrtc.org
BUG=2854,1667
Review URL: https://webrtc-codereview.appspot.com/27829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7560 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 15:28:39 +00:00
buildbot@webrtc.org
1abc146aa5
(Auto)update libjingle 78738075-> 78738103
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7554 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 08:14:14 +00:00
perkj@webrtc.org
7998089789
ApprtDemo Android: Switch between front and back camera.
...
This adds a UI icon for switching between the front and back camera.
This cl adds the possibility to change between the front and back camera while in a call
or before the other end have connected.
BUG=3786
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7553 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 08:10:03 +00:00
minyue@webrtc.org
2623695dfb
Renaming bandwidth to bitrate in webrtcvoiceengine.
...
"bandwidth" is usually a misleading mentioning. It can mean network throughput, audio frequency contents, etc.
This is to remove the confusion inside webrtcvoiceengine
BUG=
R=juberti@webrtc.org , pbos@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7551 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 02:27:08 +00:00
henrike@webrtc.org
269fb4bc90
move xmpp and p2p to webrtc
...
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.
BUG=3379
Review URL: https://webrtc-codereview.appspot.com/26999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 22:20:11 +00:00
buildbot@webrtc.org
ae694effd8
(Auto)update libjingle 78642371-> 78680406
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7545 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 17:37:17 +00:00
buildbot@webrtc.org
fbd55cb27d
(Auto)update libjingle 78616359-> 78642371
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7540 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 05:35:35 +00:00
tommi@webrtc.org
f15dee6980
Check if a datachannel in the current local description is an sctp channel before assuming rtp.
...
When generating an offer from a local description when 'sctp' is not explicitly set in the
media session options, we were generating an offer with an RTP datachannel even though the
channel in the local description was already sctp.
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7539 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 22:15:04 +00:00
glaznev@webrtc.org
243eb8e9af
Adding setting screen to AppRTCDemo.
...
- Move server URL from connection screen
to the setting screen.
- Add setting for local video resolution.
- Auto save last entered room number.
- Use full screen mode in video renderer and fix
texture offsets recalculation when rendering type is
dynamically changed.
BUG=3935,3953
R=kjellander@webrtc.org , pbos@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7534 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 17:22:15 +00:00
buildbot@webrtc.org
068b529f46
(Auto)update libjingle 78583324-> 78583691
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7532 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 16:20:42 +00:00
pthatcher@webrtc.org
2e7ee4b28b
Fix the SrtpFilter crash caused by two local offers.
...
BUG=http://crbug.com/421774
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7530 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 16:10:29 +00:00
pbos@webrtc.org
efc82c2c73
Implement screencast settings for WebRtcVideoEngine2.
...
Adds support for screencast_min_bitrate and sets content type
corresponding to the capture type.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/29959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7529 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 13:58:00 +00:00
braveyao@webrtc.org
1732df6129
Use flags set by the port allocator.
...
Currently, port allocator flags are ignored. This is inconvenient if you
want to have your own PortAllocatorFactory subclass.
BUG=webrtc:3958
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7524 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 03:01:37 +00:00
buildbot@webrtc.org
3f7bcc126d
(Auto)update libjingle 78430441-> 78445452
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7522 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-24 17:26:28 +00:00
buildbot@webrtc.org
c7ed8db7fd
(Auto)update libjingle 78427027-> 78430441
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7521 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-24 12:59:08 +00:00
perkj@webrtc.org
470988742a
Add HD support to Android if we detect a hardware video encoder that can be used. This Change the internal class MediaCodecVideoEncoder to have a one public method for checking if the platform is supported. It also adds &hd=true to the reqest url a hardware encoder is detected.
...
BUG=3934
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7520 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-24 11:38:19 +00:00
pthatcher@webrtc.org
c9d6d14020
patch from issue 25469004
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7517 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 23:37:22 +00:00
buildbot@webrtc.org
8fe75ee234
(Auto)update libjingle 78381351-> 78389679
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7516 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 23:07:23 +00:00
buildbot@webrtc.org
fb5e9fc44e
(Auto)update libjingle 78344087-> 78381351
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7515 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 21:36:17 +00:00
asapersson@webrtc.org
580d367b14
Add macros and APIs for webrtc histograms.
...
BUG=crbug/419657
Code that links system_wrappers.gyp:system_wrappers should either:
- provide implementations for the APIs, or
- link with default implementations in system_wrappers.gyp:system_wrappers_default.
R=andresp@webrtc.org , kjellander@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7508 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 12:57:56 +00:00
buildbot@webrtc.org
9d446f2e16
(Auto)update libjingle 78296920-> 78342456
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7507 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 12:22:06 +00:00
buildbot@webrtc.org
a9f0898e7d
(Auto)update libjingle 78273470-> 78296920
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7501 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 22:02:00 +00:00
glaznev@webrtc.org
7bb4a9881d
Merging Henrik's and Peter's changes for AppRTCDemo
...
from https://github.com/hkjellander/AppRTCDemo .
Description of changes:
- Add connect screen with an option to enter room number or select loopback mode.
- Add 'hangup' and 'WebRTC statistics' buttons to AppRTCDemo activity.
BUG=3938
R=kjellander@webrtc.org , pbos@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7500 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 17:43:37 +00:00
buildbot@webrtc.org
fb5410a8b7
(Auto)update libjingle 78262388-> 78262615
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7496 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 15:45:17 +00:00
pbos@webrtc.org
eacc6e4657
Remove some disabled tests in WebRtcVideoEngine2.
...
Removes some tests that shouldn't have to be implemented or have already
been through other tests.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/25929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7495 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 15:36:54 +00:00
buildbot@webrtc.org
a5c36b397a
(Auto)update libjingle 78193292-> 78199328
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7485 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 20:44:16 +00:00
guoweis@webrtc.org
b6173abe59
Fix local address leakage when IceTransportsType is relay
...
As part of implementing IceTransportsType constraint, we should hide the raddr which is the mapped address to prevent local address leakage.
BUG=1179
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7484 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 20:40:21 +00:00
buildbot@webrtc.org
1288cbb704
(Auto)update libjingle 78106439-> 78193292
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7482 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 19:29:16 +00:00
glaznev@webrtc.org
a8c0edd29f
Avoid using EGLContext class for Android 4.1 and below.
...
Support for this class was added in Android 4.2, so
disable surface decoding for lower Android versions.
BUG=3901
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7478 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-20 19:08:05 +00:00
pbos@webrtc.org
fa553ef605
Set up start bitrate in WebRtcVideoEngine2.
...
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/27789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7476 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-20 11:07:07 +00:00
henrike@webrtc.org
28100cb388
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
...
BUG=N/A
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 22:03:39 +00:00
buildbot@webrtc.org
7992b40994
(Auto)update libjingle 77953038-> 77970462
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7471 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 21:20:28 +00:00
glaznev@webrtc.org
58202946a7
Cleaning up Android AppRTCDemo.
...
- Move signaling code from Activity to a separate class
and add interface for AppRTC signaling. For now
only pure GAE signaling implements this interface.
- Move peer connection, video source and peer connection
and SDP observer code from Activity to a separate class.
- Main Activity class will do only high level calls and
event handling for peer connection and signaling classes.
- Also add video renderer position update and use full
screen for local preview until the connection is established.
BUG=
R=braveyao@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7469 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 17:42:38 +00:00
henrike@webrtc.org
d1ba6d9cbf
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.
...
BUG=3379
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27709005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 17:30:28 +00:00
buildbot@webrtc.org
81ddc78536
(Auto)update libjingle 77701902-> 77709729
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7450 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 22:39:24 +00:00
buildbot@webrtc.org
1ecbe45c7e
(Auto)update libjingle 77689511-> 77696841
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7449 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 20:29:28 +00:00
pbos@webrtc.org
43336b6b9f
Remove unused (no-op) VideoOptions.
...
Removing VideoOptions: adapt_input_to_encoder, adapt_view_switch,
video_one_layer_screencast and video_high_bitrate.
R=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/23079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7448 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 19:12:06 +00:00
henrike@webrtc.org
a4351a045d
libjingle: use _stricmp instead of deprecated stricmp.
...
BUG=N/A
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7447 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 17:07:41 +00:00
pbos@webrtc.org
7fe1e03dd6
Wire up external encoders.
...
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/30649005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7440 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 04:25:33 +00:00
buildbot@webrtc.org
f68cc0b0c3
(Auto)update libjingle 77554188-> 77629208
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7439 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 01:17:42 +00:00
henrike@webrtc.org
1e6a5dd14e
Removes xmllite from talk/xmllite since webrtc/xmllite is used instead.
...
BUG=3379
R=marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/23039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7436 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 18:27:11 +00:00
buildbot@webrtc.org
3c16d8bd1c
(Auto)update libjingle 77414393-> 77554188
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7428 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 06:35:10 +00:00
xians@webrtc.org
3cefbc99f4
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
...
This also marks all virtual overrides of other classes in the same files.
This will make a subsequent change I intend to do safer, where I'll change the
argument types of the base Transport functions, by breaking the compile if I
miss any overrides.
This also highlighted a number of unused functions. I've removed some of these.
TBR=mflodman@webrtc.org , pkasting@chromium.org
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/28709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 09:42:53 +00:00
glaznev@webrtc.org
dae40dcde9
Change setting VP8 codec specific info values by HW VP8 encoder
...
to follow SW implementation.
This fixes video freezing observed when connecting Android AppRtcDemo
on devices with hW encoder support with Chrome apprtc.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7414 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 17:53:09 +00:00
glaznev@webrtc.org
95bacfed08
Remove bad waiting code from video decoder release function.
...
Instead keep surface texture object alive while video codec
is re-initialized with a different resolution.
BUG=
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7401 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 00:00:11 +00:00
buildbot@webrtc.org
97abeee282
(Auto)update libjingle 77263371-> 77296420
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7400 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-08 22:24:30 +00:00
pbos@webrtc.org
575d126a3d
Protect send_/recv_streams_ in WebRtcVideoEngine2.
...
Important as OnLoadUpdate() won't be called on the worker thread and the
list of streams can't be concurrently modified while delivering this
callback to all send streams.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/22959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7395 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-08 14:48:08 +00:00
jiayl@webrtc.org
742922b313
Make the media content send only if offerToReceive is false while local streams exist.
...
We previously do not add the media content if offerToReceive is false.
BUG=3833
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7390 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 21:32:43 +00:00
pbos@webrtc.org
d6bda09503
Initialize sctp_paddrparams in OpenSctpSocket().
...
Addresses 'use-of-uninitialized-value' detected with MemorySanitizer.
params.spp_address.sa_family was used without being initialized before
when calling usrsctp_setsockopt with SCTP_PEER_ADDR_PARAMS.
R=jiayl@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/23909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7389 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 19:23:43 +00:00
glaznev@webrtc.org
46ffc70878
Temporary fix to allow Invoke() calls for VP8 HW encoder and decoder.
...
BUG=
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7387 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 17:11:36 +00:00
pbos@webrtc.org
963b979510
Remove potential deadlock in WebRtcVideoEngine2.
...
Fixes lock-order inversions between capturer's SignalVideoFrame and
WebRtcVideoSendStream. Additionally also removes all deadlock
suppressions for WebRtcVideoEngine2.
R=stefan@webrtc.org
TBR=kjellander@webrtc.org
BUG=1788,2999
Review URL: https://webrtc-codereview.appspot.com/26729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7386 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 14:27:27 +00:00
kjellander@webrtc.org
6ed1cf49f0
Isolate: Remove use of --ignore_broken_items
...
BUG=chromium:395700
R=jam@chromium.org
Review URL: https://webrtc-codereview.appspot.com/30659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7383 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 09:17:35 +00:00
henrike@webrtc.org
528fc650d8
Fixing build issue with L-sdk
...
Based on https://codereview.appspot.com/153000043/
BUG=https://code.google.com/p/chromium/issues/detail?id=420293
R=niklas.enbom@webrtc.org , serya@chromium.org , yfriedman@chromium.org
Review URL: https://webrtc-codereview.appspot.com/29659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7374 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-06 17:56:43 +00:00
pbos@webrtc.org
42684be21b
Wire up CPU adaptation in WebRtcVideoEngine2.
...
Includes clean-up work to be able to use the webrtc::Call::Config that's
set up. This introduced a CallFactory to spawn a FakeCall with the
config used and allowed removal of FakeWebRtcVideoChannel2.
BUG=1788
R=mflodman@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7370 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-03 11:25:45 +00:00
glaznev@webrtc.org
25cc745d6b
Switch to SW video decoder on Android after getting 2 or more
...
critical errors from HW decoder.
BUG=410730
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7368 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-02 16:58:05 +00:00
henrike@webrtc.org
4530b2ca48
Revert 7355 "Fix parallelization in libjingle_p2p_unittest."
...
Breaks waterfall.
TBR=pbos@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/22909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7357 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-01 15:43:55 +00:00
pbos@webrtc.org
fd29205e6e
Fix parallelization in libjingle_p2p_unittest.
...
Adding VirtualSocketServers to SessionTest and RelayServerTest to avoid
contention on real ports.
R=juberti@webrtc.org
BUG=2597
TEST=third_party/gtest-parallel/gtest-parallel -w 64 out/Debug/libjingle_p2p_unittest
Review URL: https://webrtc-codereview.appspot.com/26679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7355 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-01 12:31:31 +00:00
henrik.lundin@webrtc.org
4cebd84c79
Reland "Remove DTMF status methods from Voice Engine" r7276
...
This reverts r7277.
TBR=henrika@webrtc.org ,pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7353 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-01 08:23:21 +00:00
xians@webrtc.org
7aad5e5cce
Revert 7338 "Fixed the android build by making the interface pur..."
...
> Fixed the android build by making the interface pure virtual.
>
> TBR=asapersson@webrtc.org , bjornv@webrtc.org ,
>
> Review URL: https://webrtc-codereview.appspot.com/24789004
TBR=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7341 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 15:26:15 +00:00
xians@webrtc.org
90d1979d77
Fixed the android build by making the interface pure virtual.
...
TBR=asapersson@webrtc.org , bjornv@webrtc.org ,
Review URL: https://webrtc-codereview.appspot.com/24789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7338 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 15:15:22 +00:00
pbos@webrtc.org
1795c358fc
Add default implementation of Add/RemoveObserver.
...
Needed to remove Add/RemoveObserver from RTCVideoEncoderFactory in
Chromium before removing these completely. This is done to keep the
chromium.webrtc.fyi bots happy and to make rolling webrtc revisions
easier.
R=stefan@webrtc.org
BUG=3876
Review URL: https://webrtc-codereview.appspot.com/23839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7332 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 09:45:25 +00:00
kjellander@webrtc.org
8cad9432d5
Revert 7327 "Update isolate.gypi files + link to isolate_driver.py"
...
Breaks debug compilation (didn't run all trybots when testing this).
> Update isolate.gypi files + link to isolate_driver.py
>
> This updates the isolate.gypi copies we're forced to
> maintain in our code repo to Chromium revision c264a05.
>
> Since isolated testing is now using a new launch script
> in tools: isolate_driver.py, that is added to our links
> script.
>
> BUG=395700
> TESTED=Ran one of our tests with:
> ninja -C out/Release tools_unittests_run
> tools/isolate_driver.py run --isolated out/Release/tools_unittests.isolated --isolate webrtc/tools/tools_unittests.isolate
>
> R=henrika@webrtc.org , jam@chromium.org
>
> Review URL: https://webrtc-codereview.appspot.com/26649004
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7328 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 08:44:00 +00:00
kjellander@webrtc.org
02cd3067d2
Update isolate.gypi files + link to isolate_driver.py
...
This updates the isolate.gypi copies we're forced to
maintain in our code repo to Chromium revision c264a05.
Since isolated testing is now using a new launch script
in tools: isolate_driver.py, that is added to our links
script.
BUG=395700
TESTED=Ran one of our tests with:
ninja -C out/Release tools_unittests_run
tools/isolate_driver.py run --isolated out/Release/tools_unittests.isolated --isolate webrtc/tools/tools_unittests.isolate
R=henrika@webrtc.org , jam@chromium.org
Review URL: https://webrtc-codereview.appspot.com/26649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7327 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 08:34:57 +00:00
glaznev@webrtc.org
359d720004
Allow Android apps to set video renderer scaling type.
...
Also add type check for EGL context object received from apps and
switch to byte buffer video decoding if EGL context is incorrect
BUG=3851
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7326 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-29 23:07:08 +00:00
jiayl@webrtc.org
7dfb7fa189
Reland disallowing blocking calls on the worker thread.
...
This fixed the issue that invoking the call when the thread is not started.
BUG=3559
R=juberti@webrtc.org , thorcarpenter@google.com
Review URL: https://webrtc-codereview.appspot.com/24769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7325 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-29 22:45:55 +00:00
asapersson@webrtc.org
626624061e
Disable flaky tests:
...
JsepPeerConnectionP2PTestClient.ReceivedBweStatsCombined
JsepPeerConnectionP2PTestClient.ReceivedBweStatsNotCombined
BUG=3871
R=henrike@webrtc.org , kjellander@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7323 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-29 14:30:07 +00:00
pbos@webrtc.org
34f2a9ea72
Initialize SSL in unittest_main.cc.
...
Instead of having each test individually initialize and tear down SSL
move this to unittest_main.cc so that all tests are properly
initialized and new tests "don't have to think about it".
R=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/30549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7316 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-28 11:36:45 +00:00
jiayl@webrtc.org
bebc75e8bd
Fix the duplicated candidate problem when using multiple STUN servers.
...
BUG=3723
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7312 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-26 23:01:11 +00:00
thorcarpenter@google.com
a21d071607
Reverting part of
...
https://webrtc-codereview.appspot.com/15089004/diff/140001/talk/session/media/channelmanager.cc?context=10&column_width=80
because of a major regression hanging the executable on start.
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7309 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-26 17:19:14 +00:00
pbos@webrtc.org
05305116d6
Explicitly initialize SSL for tests.
...
Adding missing SSL initialization/cleanups in
TransportDescriptionFactoryTest and MediaSessionTest.
These being missing prevent these tests from being run individually
without other tests preceding them that initialize SSL.
BUG=3860
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7300 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25 15:50:26 +00:00
jiayl@webrtc.org
3987b6de50
Fix a problem in Thread::Send.
...
Previously if thread A->Send is called on thread B, B->ReceiveSends will be called, which enables an arbitrary thread to invoke calls on B while B is wait for A->Send to return. This caused mutliple problems like issue 3559, 3579.
The fix is to limit B->ReceiveSends to only process requests from A.
Also disallow the worker thread invoking other threads.
BUG=3559
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7290 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 17:14:05 +00:00
pbos@webrtc.org
d60d79a145
Thread annotation of rtc::CriticalSection.
...
Effectively re-lands r5516 which was reverted because talk/-only
checkouts existed. This now resides in webrtc/base/, so no talk/-only
checkouts should be possible.
This change also enables -Wthread-safety for talk/ and fixes a bug in
talk/media/webrtc/webrtcvideoengine2.cc where a guarded variable was
read without taking the corresponding lock.
R=andresp@webrtc.org , mflodman@webrtc.org , pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7284 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 07:10:57 +00:00
pbos@webrtc.org
38344ed280
Move thread_annotations.h to webrtc/base/.
...
R=andresp@webrtc.org , mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 06:05:00 +00:00
glaznev@webrtc.org
8166faeff3
Change Android video renderer to maintain video aspect
...
ratio when displaying camera or decoded video frames.
-
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7282 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 23:58:52 +00:00
glaznev@webrtc.org
90668b1633
Switch HW video decoder to output byte buffers if video
...
renderer EGL context is not provided by app.
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7281 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 21:42:15 +00:00
buildbot@webrtc.org
1b7dcc1647
(Auto)update libjingle 76169599-> 76176062
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7280 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 17:41:48 +00:00
guoweis@webrtc.org
2c1bcea1bc
Enable ipv6 by default for webrtc under a Finch experiment.
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Reapply 23529005 after fixing the build break issue (Chromium:582133002)
Committed: https://code.google.com/p/webrtc/source/detail?r=7253
Review URL: https://webrtc-codereview.appspot.com/23529005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7278 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 16:23:02 +00:00
henrik.lundin@webrtc.org
3987f10c11
Revert "Remove DTMF status methods from Voice Engine" r7276
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This change caused some trouble.
TBR=henrika@webrtc.org ,pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7277 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 13:15:14 +00:00
henrik.lundin@webrtc.org
bf7b9e0081
Remove DTMF status methods from Voice Engine
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These methods are not used.
R=henrika@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7276 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 12:54:04 +00:00
pbos@webrtc.org
0a2087a711
Skeleton for registering external encoders/decoders.
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R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/31429005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7270 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 09:40:22 +00:00