tkchin@webrtc.org
acca675bcf
Implement mac version of AppRTCDemo.
...
- Refactored and moved AppRTCDemo to support sharing AppRTC connection code between iOS and mac counterparts.
- Refactored OpenGL rendering code to be shared between iOS and mac counterparts.
- iOS AppRTCDemo now respects video aspect ratio.
BUG=2168
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6291 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 22:26:06 +00:00
fischman@webrtc.org
abe01dd634
AppRTCDemo(android): run in full-screen & immersive mode.
...
Also:
- Only show stats HUD on demand
- Only collect stats when HUD is showing
- Don't render solid green frame when video is not present in either direction
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6275 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-29 21:46:52 +00:00
fischman@webrtc.org
43a1395370
AppRTCDemo(android): README updates for a shrinking envsetup.sh world.
...
There was duplicated (and out of date!) information in README relative to
getting-started so de-duped to point to getting-started as the canonical
reference.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15589006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6265 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 17:29:09 +00:00
buildbot@webrtc.org
727ff69829
(Auto)update libjingle 67872893-> 67873348
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6244 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 23:20:53 +00:00
buildbot@webrtc.org
75cb3dc5f2
(Auto)update libjingle 67869540-> 67872893
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6243 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 23:13:35 +00:00
tkchin@webrtc.org
1732a591e7
Add a UIView for rendering a video track.
...
RTCEAGLVideoView provides functionality to render a supplied RTCVideoTrack using OpenGLES2.
R=fischman@webrtc.org
BUG=3188
Review URL: https://webrtc-codereview.appspot.com/12489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6192 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 23:26:01 +00:00
fischman@webrtc.org
a150bc9bbf
PeerConnection(android): allow initializing either (or neither) of {Voice,Video}Engine.
...
Enables applications that don't want to pay the init/startup cost or request
extra permissions (e.g. audio-only app, or DataChannel-only app).
BUG=3234
Review URL: https://webrtc-codereview.appspot.com/15489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6164 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 22:00:50 +00:00
fischman@webrtc.org
14ea7e8922
AppRTCDemo(android): added a Heads-Up Display for bandwidth estimation.
...
- tap display to toggle visibility
- increased getStats frequency to 1hz.
R=glaznev@google.com
Review URL: https://webrtc-codereview.appspot.com/19419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6039 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-01 20:57:55 +00:00
fischman@webrtc.org
dd92feb6dd
AppRTCDemo(android): send the created SDP, not the local description after setting it
...
This is required to allow explicit filtering of ICE candidates.
BUG=3288
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6038 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-01 19:06:18 +00:00
tkchin@webrtc.org
ff2733204d
Implement ObjC DataChannel wrapper
...
R=fischman@webrtc.org
BUG=3112
Review URL: https://webrtc-codereview.appspot.com/16369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6031 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-30 18:32:33 +00:00
fischman@webrtc.org
7c82adae61
AppRTCDemo was blocking the main thread for network requests. This fixes it by making the background queue serial instead of using @synchronize to make the background operations serial.
...
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16379004
Patch from Bridger Maxwell <bridgeyman@gmail.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6028 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-30 00:17:47 +00:00
fischman@webrtc.org
f27fdeb9c9
AppRTCDemo(android): don't initialize process-globals more than once.
...
BUG=3257
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6001 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 16:32:38 +00:00
mallinath@webrtc.org
a0d3067575
Use CreatePeerConnection method which accepts port_allocator.
...
Other method will be removed, in a different CL.
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20369006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5987 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-26 00:00:15 +00:00
tkchin@webrtc.org
19b1be159e
Provide GetStats method in RTCPeerConnection
...
BUG=3144
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12069006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5960 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 21:05:38 +00:00
tkchin@webrtc.org
ec3d8ecdcc
Fix typo by renaming RTCSessionDescriptonDelegate -> RTCSessionsDescriptionDelegate
...
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5946 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-21 18:47:24 +00:00
fischman@webrtc.org
d1fe6b728e
AppRTCDemo(android): fix a couple of SDP-related regressions.
...
- r5834 made it so that empty fields are a fatal SDP parsing error, exposing
opportunities for improvement in the preferISAC; changed split/join to use
\r\n instead of \n and now omitting the trailing space on the m=audio line
that triggered the new failure.
- DTLS requires a different role for each endpoint so conflicts with loopback
calling. apprtc.py suppresses DTLS for that reason in loopback calls, so the
android demo app now only enables DTLS by default if it is not suppressed by a
constraint (matching Chrome).
BUG=3164,3165,2507
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5847 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-04 21:40:46 +00:00
fischman@webrtc.org
49c5ba32bb
AppRTCDemo(iOS): now works in the iOS Simulator!
...
...which has no camera device emulation or pass-through, so no local video
view.
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/10919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5815 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-31 20:22:19 +00:00
fischman@webrtc.org
61e78fca6c
AppRTCDemo(iOS): remote-video reliability fixes
...
Previously GAE Channel callbacks would be handled by JS string-encoding the
payload into a URL. Unfortunately this is limited to the (undocumented,
silently problematic) maximum URL length UIWebView supports. Replaced this
scheme by a notification from JS to ObjC and a getter from ObjC to JS (which
happens out-of-line to avoid worrying about UIWebView's re-entrancy, or lack
thereof). Part of this change also moved from a combination of: JSON,
URL-escaping, and ad-hoc :-separated values to simply JSON.
Also incidentally:
- Removed outdated TODO about onRenegotiationNeeded, which is unneeded
- Move handling of PeerConnection callbacks to the main queue to avoid having
to think about concurrency too hard.
- Replaced a bunch of NSOrderedSame with isEqualToString for clearer code and
not having to worry about the fact that [nil compare:@"foo"]==NSOrderedSame
is always true (yay ObjC!).
- Auto-scroll messages view.
BUG=3117
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/10899006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5814 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-31 20:16:49 +00:00
fischman@webrtc.org
fe16488184
AppRTCDemo(android): specify DtlsSrtpKeyAgreement:true in CreatePeerConnection's constraints.
...
This is required to interop with Chrome now that SDES is disabled in Chrome (as of r5640).
BUG=2774
R=jiayl@chromium.org
Review URL: https://webrtc-codereview.appspot.com/10749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5809 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-28 19:58:03 +00:00
henrike@webrtc.org
1ca08f65e3
Fix after auto update in r5787. APPRTCVideoView.h/m was removed incorrectly.
...
BUG=3121
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5793 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-26 16:42:14 +00:00
henrike@webrtc.org
dce3feb0b0
(Auto)update libjingle 63738002-> 63773382
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5787 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-26 01:17:30 +00:00
henrike@webrtc.org
ae3347a546
Fix after auto update: removed files were brought back.
...
BUG=N/A
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5782 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-25 18:17:02 +00:00
fischman@webrtc.org
76d4f389bb
AppRTCDemo(iOS): allow rooms with no incoming audio.
...
Also fix a compile-time warning for a leftover unimplemented method
(RTCVideoRenderer:setTransform).
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/10629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5780 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-25 17:40:38 +00:00
henrike@webrtc.org
6e3dbc2a77
(Auto)update libjingle 63648983-> 63738002
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5779 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-25 17:09:47 +00:00
fischman@webrtc.org
385a722646
PeerConnection(iOS): make ARC-clean talk/.../objc* and talk/examples/ios
...
- Removes a strong-reference cycle between RTCPeerConnection and
RTCPeerConnectionObserver
- Gives RTCPeerConnectionObserver a virtual dtor
- Ensures RTCPeerConnectionTest tears down correctly
- Ensures AppRTCDemo tears down correctly
This is the talk/ half; the webrtc/ half is in https://webrtc-codereview.appspot.com/10539005
BUG=3054,3055,3100
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/10499005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5771 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-25 05:16:29 +00:00
fischman@webrtc.org
7fa1fcb72c
AppRTCDemo(ios): style/cleanup fixes following cr/62871616-p10
...
BUG=2168
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/9709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5768 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-25 00:11:56 +00:00
henrike@webrtc.org
d3d6bce9ed
(Auto)update libjingle 62865357-> 62871616
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5674 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-10 20:41:22 +00:00
pbos@webrtc.org
371243dfa3
Remove std:: prefixes from C functions in talk/.
...
std::memcpy -> memcpy for instance. This change was motivated by a
compile report complaining that std::rand() was used instead of rand(),
probably with a stdlib.h include instead of cstdlib. Use of C functions
without the std:: prefix is a lot more common, so removing std:: to
address this.
BUG=
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5657 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-07 15:22:04 +00:00
henrike@webrtc.org
40b3b68cdf
Update libjingle 62364298->62472237
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5632 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-03 18:30:11 +00:00
henrike@webrtc.org
1bbfb57d71
Rollback of r5629 "(Auto)update libjingle 62364298-> 62368661".
...
BUG=N/A
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5631 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-03 17:37:52 +00:00
henrike@webrtc.org
31413dc635
(Auto)update libjingle 62364298-> 62368661
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5629 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-03 16:47:01 +00:00
fischman@webrtc.org
bcfc1670d6
AppRTCDemo(android): don't send local SDP until it's set.
...
This fixes a race condition where the remote participant could receive the
offer, create & set its answer locally, send it back, and then try to set the
answer before the local set completed. Observed intermittently in loopback
calls when setLocalDescription is intentionally delayed (debugging something
else).
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5625 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-01 00:02:27 +00:00
henrike@webrtc.org
b8395ebe14
(Auto)update libjingle 62293974-> 62364298
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-28 21:57:22 +00:00
braveyao@webrtc.org
eaadecaf98
iOS, AppRTCDemo: Fixes exception due to JSON for ice using "urls" instead of "url", which is introduced by r5599.
...
BUG=2962
TEST=
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9109005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5610 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-26 04:16:02 +00:00
henrike@webrtc.org
79a1cff65a
Android, AppRTCDemo: Fixes java exception due to JSON for ice using "urls" instead of "url".
...
BUG=2952
TEST=Manual
TBR=braveyao
Review URL: https://webrtc-codereview.appspot.com/9099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5605 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-24 23:22:18 +00:00
fischman@webrtc.org
c5d506a106
AppRTCDemo(android): clarified README on how to launch app using adb.
...
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8689005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5553 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 17:55:13 +00:00
fischman@webrtc.org
3eda643a91
PeerConnection(java): added MediaConstraints support to AudioSource, now fed to AudioTrack.
...
BUG=2912
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5540 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 04:01:04 +00:00
fischman@webrtc.org
540acde5b3
PeerConnection(java): use MediaCodec for HW-accelerated video encode where available.
...
Still disabled by default until https://code.google.com/p/webrtc/issues/detail?id=2899 is resolved.
Also (because I needed them during development):
- make AppRTCDemo "debuggable" for extra JNI checks
- honor audio constraints served by apprtc.appspot.com
- don't "restart" video when it hasn't been stopped (affects running with the
screen off)
BUG=2575
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/8269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5539 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 03:56:14 +00:00
jiayl@webrtc.org
14d80793a8
PeerConnectionClient needs to initialize SSL.
...
BUG=2911
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5531 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 00:41:59 +00:00
fischman@webrtc.org
82387e4608
Add ability to receive calls for iOS
...
BUG=2701
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7989005
Patch from Sajid Hussain <shussain@temasys.com.sg>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5518 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 18:47:11 +00:00
henrike@webrtc.org
2ce9a64b75
Talk: Removes deprecated example apps and moves the server apps to trunk/talk/examples.
...
BUG=12545067
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7159004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5397 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 16:49:53 +00:00
sergeyu@chromium.org
4b26e2eee3
Update libjingle to 59676287
...
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5390 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-15 23:15:54 +00:00
fischman@webrtc.org
d7568a08c3
PeerConnection(java): Add OnRenegotiationNeeded support
...
Also:
- Make PeerConnectionObserver::OnRenegotiationNeeded() pure virtual to avoid
this sort of mistake in the future.
- Sprinkle @Override annotations on some callback definitions that were missing
them.
- Fix a JNI method-signature-lookup typo (s/(V)V/()V/) in PCOJava::OnError()
- Add an explicit ScopedLocalFrameRef to PCOJava::OnError() to match all other
C++-fired callbacks, for consistency.
BUG=2771
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5376 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-13 22:04:12 +00:00
fischman@webrtc.org
1794693ec8
AppRTCDemo(android): close() the throw-away DataChannel.
...
Otherwise, the PeerConnection remembers the channel enough to include an
m=application line in its offer SDP, causing connection to chrome to fail, since
apprtc.appspot.com doesn't include an RtpDataChannels:true constraint in its
RTCPeerConnection constructor call.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6729005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5354 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-08 18:29:34 +00:00
wu@webrtc.org
2018269dc3
Revert 5274 "Update talk to 58113193 together with https://webrt ..."
...
> Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/ .
>
> R=mallinath@webrtc.org , niklas.enbom@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5719004
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5275 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 22:54:25 +00:00
wu@webrtc.org
a129b6cd13
Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/ .
...
R=mallinath@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 22:40:39 +00:00
fischman@webrtc.org
f41f06b916
PeerConnection(java): rationalize pointer-to-jlong conversion.
...
In r4665 I went a bit crazy with the manual reinterpretation of a pointer to a
jlong (to avoid undefined behavior) but that's what reinterpret_cast<> is for.
So use it directly now.
Added a do-nothing DataChannel to AppRTCDemo to regression test this, since the
only repro I've found of the original bug requires ARM ABI (PeerConnectionTest
on ia32 fails to repro).
BUG=2302
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5269 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 21:07:18 +00:00
kjellander@webrtc.org
f9bdbe3619
Roll chromium_revision 232627:238260
...
This brings us the updated swarming_client
that has moved out from Chromium into a standalone
project.
Because of this, all .isolate files needed to be
updated as well, similar to the changes in
https://codereview.chromium.org/29993003
TEST=trybots passing
BUG=none
R=andrew@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5260 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 13:37:12 +00:00
sergeyu@chromium.org
5bc25c41fc
Update libjingle to 57692857
...
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5217 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 00:24:06 +00:00
sergeyu@chromium.org
a23f0ca4ba
Update talk to 56619788
...
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3839005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5120 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 22:48:52 +00:00