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961 Commits

Author SHA1 Message Date
magjed@webrtc.org
e575e9c40f Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h 
The purpose of this CL is to be able to reuse the class WebRtcVideoRenderFrame in webrtcvideoengine.cc.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7888 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-14 11:09:23 +00:00
pthatcher@webrtc.org
dee76f3b89 Move the obvious/easy Jingle-specific code into webrtc/libjingle. 
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7886 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-12 21:04:42 +00:00
guoweis@webrtc.org
8c9d79a29d Add adapter_type into Candidate object. 
Expose adapter_type from Candidate such that we could add jmidata on top of this.

Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.

This is migrated from issue 32599004

BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7885 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-12 19:21:14 +00:00
tommi@webrtc.org
c57310b982 Switch kStatsValueName* constants to be enums instead of char*. 
This is to guard against potentially assigning a value name to an incorrect value, non-static string or otherwise assume they can be treated as strings.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7884 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-12 17:41:28 +00:00
pthatcher@webrtc.org
40b276ea7b Cleanup little things found when refactoring. 
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/33519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7880 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-12 02:44:30 +00:00
pbos@webrtc.org
2b19f06312 Wire up RTT statistics to webrtc::Call. 
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788

Review URL: https://webrtc-codereview.appspot.com/32249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7876 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-11 13:26:09 +00:00
pbos@webrtc.org
13518951e3 Remove old_factory from WebRtcVideoEngine. 
Minor pending cleanup.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/28239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7875 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-11 13:14:30 +00:00
perkj@webrtc.org
128fabaf7b Revert "Revert 7826 "Change Android PeerConnectionUnittest to build usin..."" 
Original cl description:

Change Android PeerConnectionUnittest to build using Chrome macros.
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest

This also add a new build target to build java PeerConnection using Chromes build macros.

BUG=4031
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7874 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-11 12:25:57 +00:00
buildbot@webrtc.org
a85307737c (Auto)update libjingle 81702493-> 81755413 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7860 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 09:01:18 +00:00
tommi@webrtc.org
aa2c342c10 Add back a constructor to fix FYI build. 
TBR=perkj

Review URL: https://webrtc-codereview.appspot.com/24349005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7854 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 20:23:06 +00:00
tkchin@webrtc.org
87776a8935 iAppRTCDemo: WebSocket based signaling. 
Updates the iOS code to use the new signaling model. Removes old Channel API
code. Note that this no longer logs messages to UI. UI update forthcoming.

BUG=
R=glaznev@webrtc.org, jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7852 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 19:32:35 +00:00
pthatcher@webrtc.org
0babb4a4e6 Fix a comment. 
R=juberti@webrtc.org, pbos@webrtc.org, sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7851 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 19:01:45 +00:00
tommi@webrtc.org
c9d155faeb Move implementation of types in statstypes. to its cc file. 
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7850 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 18:18:06 +00:00
henrika@webrtc.org
a954c07ee1 AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer 
BUG=4034
R=andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 16:22:09 +00:00
tommi@webrtc.org
5c3ee4bce6 Add empty implementation file that will hold statstypes.h implementation. 
The implementation for the types currently in statstypes.h is split between statstypes.h and statscollector.cc.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7844 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 10:47:01 +00:00
glaznev@webrtc.org
eef85387ec Fix AppRTCDemo closing error for KK and JB Android devices. 
- Do not allow connection output when sending http delete
request to ws server - this causes IOException for KK and JB devices.
- Avoid creating dialog box with error message when activity
has been already closed / paused -
this causes resource leak error message for KK devices.
- Plus some code clean up to support async http messages in
websocket channel wrapper and use Handler for running
peerconnection client funcitons on UI thread.

R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 01:29:17 +00:00
andrew@webrtc.org
3b3c406908 Revert 7826 "Change Android PeerConnectionUnittest to build usin..." 
Broke gclient runhooks on internal bots. e.g.
http://chromegw/i/internal.client.webrtc/builders/Linux64%20Debug/builds/3575

> Change Android PeerConnectionUnittest to build using Chrome macros.
> The purpose is to be able to run the tests using Chromes buildbots. To run:
> CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest
> 
> This also add a new build target to build java PeerConnection using Chromes build macros.
> 
> BUG=4031
> R=kjellander@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/28189004

TBR=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7829 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 17:21:50 +00:00
perkj@webrtc.org
ed7824b1c0 Change Android PeerConnectionUnittest to build using Chrome macros. 
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest

This also add a new build target to build java PeerConnection using Chromes build macros.

BUG=4031
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7826 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 15:41:01 +00:00
glaznev@webrtc.org
e2a9261f3e Improve AppRTCDemo connection speed by sending all 
http POST requests asynchronously.

R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7820 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-05 20:11:06 +00:00
kjellander@webrtc.org
bd8cc0b914 Add codereview.settings to the /talk subdirectory 
With this, it will be possible to create CLs from
Git repos created using
https://chromium.googlesource.com/external/webrtc/trunk/talk
(which is what you get when working with the repo currently
put in Chrome's src/third_party/libjingle/source/talk).

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7819 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-05 13:47:37 +00:00
kjellander@webrtc.org
599e299b9d cricket::VideoFrame int64 to int64_t. 
Needed for successful compile of ios arm64.

BUG=3898
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30359004

Patch from Zeke Chin <tkchin@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7817 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-05 09:42:57 +00:00
bemasc@webrtc.org
9b5467e88d Fix assertion failure when closing data channel, and add a unit test. 
BUG=4066
R=jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7816 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 23:16:52 +00:00
glaznev@webrtc.org
4b407aa985 Update AppRTCDemo README with information on 3-dot-apprtc server 
and new command line arguments.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7815 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 22:42:59 +00:00
guoweis@webrtc.org
7169afd9d5 With IPv6 enabled, it's important to know whether IPv6 is really used or not. BestConnection is tracked for this purpose. Also added a test case to verify the end to end behavior. 
BUG=411086
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30919005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 17:59:29 +00:00
glaznev@webrtc.org
369746bcb8 Support new WebSocket signaling format. 
- Support new GAE message format and new signaling
sequence, which allows connection to 3-dot-apprtc server.
- Add UI setting to switch between GAE / WebSockets signaling.
- Some clean ups to better support command line application
execution.

BUG=3937,3995,4041
R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 17:28:52 +00:00
pbos@webrtc.org
0fb6ad2004 Check if cpu_monitor_ exists before Stop(). 
R=asapersson@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/25279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 13:44:29 +00:00
asapersson@webrtc.org
d8aed6b321 Verify that cpu_monitor exists before calling Stop(). 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 12:37:47 +00:00
pthatcher@webrtc.org
eb0954248d Don't reset sequence number for a stream on deactivate/reactivate. 
BUG=chromium:431908
R=pbos@webrtc.org, sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 00:34:10 +00:00
glaznev@webrtc.org
d01955179a Change minimum video encoder initialization resolution to 
176x144 to ensure HW encoder can be initialized.

R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7787 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-02 23:41:18 +00:00
perkj@webrtc.org
beee9cec22 Change back so that Android ApprtcDemo only use one MediaStream containing both audio and video. 
The reason is that the desktop apprtcdemo only handle one MediaStream and this doesn't play audio if it receive two streams.

TEST=Test that a call with audio and video can be setup between an Android device and a desktop client using apprtc.appspot.com.
BUG=4051,3786
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7781 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-02 14:38:18 +00:00
pthatcher@webrtc.org
146e0fd30f Fix the build by putting in a typecast to avoid a comparison between 
signed and unsigned ints introduced in cl/81073932.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7776 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 20:07:52 +00:00
glaznev@webrtc.org
dea5173edf Add start bitrate and vp8 hw acceleration option to 
Android AppRTCDemo.

- Add an option to set VP8 encoder start bitrate
usig x-google-start-bitrate line in remote SDP.
- Allow to enabled/disable VP8 hw decoder and
encoder acceleration using appRTC settings.

BUG=4046
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7775 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 20:02:13 +00:00
buildbot@webrtc.org
32ec0dd032 (Auto)update libjingle 81063831-> 81073932 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7774 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 17:57:36 +00:00
pbos@webrtc.org
273a414b0e Report encoded frame size in VideoSendStream. 
Implements reporting transmitted frame size in WebRtcVideoEngine2.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=4033

Review URL: https://webrtc-codereview.appspot.com/33399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 15:23:21 +00:00
tommi@webrtc.org
2c13f659c7 Add a platform specific typedef for SOCKET in the peerconnection_server example since it's not universally 'int'. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7763 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-28 10:37:31 +00:00
tkchin@webrtc.org
3e9ad26112 Refactor iOS AppRTC parsing code. 
Moved parsing code to JSON categories for the relevant objects.
No longer prefer ISAC as audio codec.

BUG=
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31989005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7755 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-27 00:52:38 +00:00
sprang@webrtc.org
a71bb6033b Revert 7750 "Don't reset sequence number for a stream on deactiv..." 
> Don't reset sequence number for a stream on deactivate/reactivate.
>
> BUG=chromium:431908
> R=pbos@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/32199004

TBR=sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7752 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 19:33:15 +00:00
sprang@webrtc.org
31f7a0e710 Don't reset sequence number for a stream on deactivate/reactivate. 
BUG=chromium:431908
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 16:55:52 +00:00
perkj@webrtc.org
2faf7eea6f Revert "Revert "This adds an Android apk for running tests on the Java layer of PeerConnection."" 
This reverts commit 308e7ff613.

Original cl description:

This adds an Android apk for running tests on the Java layer of PeerConnection.

The only testcase is currently the same test we run on Java standalone.
To run the test adb shell am instrument -w org.webrtc.test/android.test.InstrumentationTestRunner

BUG=4031
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7748 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 07:35:37 +00:00
glaznev@webrtc.org
58edb83fd4 Add video encoder fps and bitrate statistics to 
Android AppRTCDemo UI.

BUG=4045
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7747 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 00:39:42 +00:00
pbos@webrtc.org
008731868a Implement settable min/start/max bitrates in Call. 
These parameters are set by the x-google-*-bitrate SDP parameters. This
is implemented on a Call level instead of per-stream like the currently
underlying VideoEngine implementation to allow this refactoring to not
reconfigure the VideoCodec at all but rather adjust bandwidth-estimator
parameters.
Also implements SetMaxSendBandwidth in WebRtcVideoEngine2 as it's a SDP
parameter and allowing it to be dynamically readjusted in Call.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/26199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7746 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-25 14:03:34 +00:00
glaznev@webrtc.org
dab5d92df6 Use mirror image for Android AppRTCDemo local preview. 
Similar to Chrome apprtc using mirror image for camera
local preview provides better experience when device
is rotated.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7741 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 17:31:01 +00:00
kjellander@webrtc.org
8562f23acb OWNERS: Remove tomasl@ and mallinath@ 
mallinath@ has left the team and tomasl@ says he doesn't
know why he's owner in webrtc/test/channel_transport

R=henrika@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 10:05:05 +00:00
kjellander@webrtc.org
308e7ff613 Revert "This adds an Android apk for running tests on the Java layer of PeerConnection." 
This reverts r7732

Reason: Breaks compilation on Linux:
[813/818] LINK libjingle_media_unittest
FAILED: cd ../../talk; build/build_jar.sh /usr/lib/jvm/java-7-openjdk-amd64 ../out/Debug/libjingle_peerconnection_test.jar ../out/Debug/obj/talk/libjingle_peerconnection_test_jar.gen app/webrtc/javatests/src:../out/Debug/libjingle_peerconnection.jar:../third_party/junit/junit-4.11.jar app/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java app/webrtc/javatests/src/org/webrtc/PeerConnectionTestJava.java
build/build_jar.sh: Entering directory `/mnt/data/b/build/slave/linux64/build/src/talk'
app/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java:46:warning: [deprecation] Assert in junit.framework has been deprecated
import static junit.framework.Assert.*;
                             ^
app/webrtc/javatests/src/org/webrtc/PeerConnectionTestJava.java:36:error: cannot find symbol
  @Test
   ^
  symbol:   class Test
  location: class PeerConnectionTestJava
app/webrtc/javatests/src/org/webrtc/PeerConnectionTestJava.java:43:error: cannot find symbol
  @Test
   ^
  symbol:   class Test
  location: class PeerConnectionTestJava
2 errors
1 warning
ninja: build stopped: subcommand failed.

TBR=perkj@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/32169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-23 21:23:00 +00:00
perkj@webrtc.org
2751f2ab4c This adds an Android apk for running tests on the Java layer of PeerConnection. 
The only testcase is currently the same test we run on Java standalone.
To run the test adb shell am instrument -w org.webrtc.test/android.test.InstrumentationTestRunner

R=kjellander@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-23 16:00:57 +00:00
thorcarpenter@google.com
88d14f483b Remove expensive and unnecessary memory alloc for sending black frames on video 
mute.

Remove old crusty is_black_ member var in webrtcvideoengine which was not adding value.

R=henrike@webrtc.org, tpsiaki@google.com

Review URL: https://webrtc-codereview.appspot.com/26229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-22 01:04:26 +00:00
magjed@webrtc.org
bdcf38c894 cricket::VideoFrame: Refactor ConvertToRgbBuffer into base class 
There is also an implementation in Chromium that can be removed if/when this lands:
content/renderer/media/webrtc/webrtc_video_capturer_adapter.cc

R=fbarchard@google.com, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7728 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-21 10:53:00 +00:00
pkasting@chromium.org
4591fbd09f Use size_t more consistently for packet/payload lengths. 
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-20 22:28:14 +00:00
glaznev@webrtc.org
edc6e57a92 Support loopback mode and command line execution 
for Android AppRTCDemo when using WebSocket signaling.

- Add loopback support for new signaling. In loopback mode
only room connection is established, WebSocket connection is
not opened and all candidate/sdp messages are automatically
routed back.
- Fix command line support both for channek and new signaling.
Exit from application when room connection is closed and add
an option to run application for certain time period and exit.
- Plus some fixes for WebSocket signaling - support
POST (not used for now) and DELETE request to WebSocket server
and making sure that all available TURN server are used by
peer connection client.

BUG=3995,3937
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7725 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-20 21:16:12 +00:00
magjed@webrtc.org
f58b455cf7 cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. 
In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.

This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.

R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7702

Committed: https://code.google.com/p/webrtc/source/detail?r=7707

Review URL: https://webrtc-codereview.appspot.com/29949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7721 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-19 18:09:14 +00:00