perkj@webrtc.org
b5d045e94d
ReAdd PeerConnectionInterface::AddStream to fix Chrome build.
...
AddStream(MediaStreamInterface* stream, const MediaConstraintsInterface* constraints);
This will be removed once Chrome has been updated.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7608 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 13:01:33 +00:00
tommi@webrtc.org
18de6f9622
Change the PeerConnection proxy templates to use blocking method calls instead of using Thread::Send.
...
The problem with Thread::Send is that it processes incoming pending messages and for the proxies,
this can mean that multiple incoming calls can concurrently run on the same thread, resulting in unexpected behavior.
See e.g. crbug.com/429740 (and more)
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7607 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 12:08:48 +00:00
perkj@webrtc.org
c2dd5ee2c0
Prepare for removal of PeerConnectionObserver::OnError.
...
Prepare for removal of constraints to PeerConnection::AddStream.
OnError has never been implemented and has been removed from the spec.
Also, constraints to PeerConnection::AddStream has also been removed from the spec and have never been implemented.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7605 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 11:31:29 +00:00
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
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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
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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