37bb4974e7
Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals.
...
R=juberti@google.com , mikhal@webrtc.org , stefan@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5027 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-23 23:59:45 +00:00
901ae77618
Android: Fixes WebRTCDemo build (missing Java code).
...
TBR=ajm@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/2395005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4961 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-14 21:46:53 +00:00
f53622d42e
WebRTCDemo: Fixes warning for devices with pre-17 API level. Also fixes broken build build.xml and project.properties.
...
BUG=2083
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2375004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4951 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-11 21:28:26 +00:00
6c82e04cee
Android standalone: remove some usages of deprecated APIs and prevent further regressions.
...
Also:
- Fixed WebRTCDemo UI to say "SwitchToBack" at startup since default camera is front
- Rebuild WebRTCDemo APK when resources/layout/strings change
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2337004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4916 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03 18:57:48 +00:00
4e65e07e41
VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android.
...
Besides being ~40% the size of the previous implementation, this makes it so
that VideoCaptureAndroid can stop and restart capture, which is necessary to
support onPause/onResume reasonably on Android.
BUG=1407
R=henrike@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2334004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4915 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03 18:23:13 +00:00
82f014aa0b
OpenSL (not default): Enables low latency audio on Android.
...
BUG=1669
R=andrew@webrtc.org , fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2032004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4719 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-10 18:24:07 +00:00
c7f708679d
Clamp camera id to legal values.
...
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2184004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4694 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 18:17:45 +00:00
dde7d4c6ed
Roll chromium_revision 214260:217707 and gflags 45:84
...
gflags roll is needed mostly to pick up fixes for warnings triggered by newer
compiler/settings pulled in by the chromium roll. Had to switch from the old
google-gflags project the current gflags project to pick up this fix (see
https://code.google.com/p/gflags/source/detail?r=74 for details).
Update android build.xml file to reflect tools moves in new SDK pulled in by the chromium_revision roll.
R=niklas.enbom@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2043004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4555 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 23:31:30 +00:00
d3ae3c7b1f
Unbreak clang/android build of webrtc.
...
TESTED=All target builds once more with clang=1.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4460 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-31 23:53:07 +00:00
129afc29fb
Correctly rebuild WebRTCDemo after jni/ source file changes
...
BUG=1980
TEST=Modify source file under jni/ and WebRTCDemo will rebuild
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1831004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4377 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-20 04:43:08 +00:00
cab716cc7d
Fix a circular dependency by removing an unnecessary dependency, add a missing include_tests check and missing lib references for android.
...
TBR=henrikg@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1776005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4312 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-09 13:43:24 +00:00
a2073af728
Fixes build breakage when building WebRTC in Chromium and having include_tests=1.
...
TBR=fischman@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/1770004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4305 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-08 18:14:58 +00:00
546c91dc2e
Build all java files into jar for each module on Android
...
BUG=None
TEST=All java files in each module are built into jar and used by WebRTCDemo app
R=fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1696004
Patch from Jeremy Mao <yujie.mao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4284 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-01 17:52:39 +00:00
d4803ced60
WebRTCViEDemo: Use global reference when passing variables across different threads
...
There are JNI local reference changes in ICS when Android SDK
target level API >= 14.
http://android-developers.blogspot.com/2011/11/jni-local-reference-changes-in-ics.html
BUG=NONE
TEST=WebRTCViEDemo works well using MediaCodec Decoder/Renderer
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1744004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4283 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-01 14:55:37 +00:00
0021632f40
Re-add WebRTCDemo dependencies as dependencies (not just inputs) because they also need to be built for this target!
...
BUG=1980
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1734004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4275 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-27 17:35:32 +00:00
3145a642b7
Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes.
...
BUG=1980
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4270 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-26 20:20:05 +00:00
f27389ca9f
WebRTCDemo: ensures that using front and back camera work as expected.
...
I.e. egress: Real world up is stream up.
Ingress: stream up is app up.
Local (preview): Real world up is app up.
BUG=1763
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1642004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4227 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-14 05:37:13 +00:00
dd97ef4e28
Revert 4211 "Build all java files into jar for each module on An..."
...
Reason for revert: behold the meltdown of the "trunk" bots on http://build.chromium.org/p/chromium.webrtc.fyi/waterfall
Turns out that include in gyp is fraught with peril: https://code.google.com/p/gyp/wiki/InputFormatReference#Including_Other_Files
> Build all java files into jar for each module on Android
>
> BUG=
> R=fischman@webrtc.org , niklas.enbom@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1636004
>
> Patch from Jeremy Mao <yujie.mao@intel.com >.
TBR=fischman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/1660005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4222 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12 17:39:29 +00:00
6c35e0b0f7
Reorganize test targets in WebRTC
...
This CL will lower the number of test targets in WebRTC by:
Add common_audio_unittests and merge the following targets into it (copied from http://review.webrtc.org/1584006 ):
* resampler_unittests
* signal_processing_unittests
* vad_unittests
Merge into modules_unittests:
* bitrate_controller_unittests
* desktop_capture_unittests
* media_file_unittests
* remote_bitrate_estimator_unittests
* rtp_rtcp_unittests
* paced_sender_unittests
Merge into test_support_unittests:
* channel_transport_unittests
channel_transport.gyp was also removed in favor for test.gyp.
I had to remove a main method from rtcp_format_remb_unittest.cc
since it caused the fileutils.h code to not be able to find the
right project root path in ordrer to provide correct paths
to test files.
Buildbot configuration update will be synced with the commit of this CL.
TEST=trybots
BUG=1843
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4213 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-11 08:29:17 +00:00
1374965680
Build all java files into jar for each module on Android
...
BUG=
R=fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1636004
Patch from Jeremy Mao <yujie.mao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4211 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10 23:34:27 +00:00
f791b1cebf
Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp.
...
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1574004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4126 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29 00:38:02 +00:00
68c05f498c
Remove libvpx_intrinsics_sse4_1.a in Android.mk since this target is no longer generated in libvpx
...
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1569004
Patch from Jeremy Mao <yujie.mao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4114 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-28 05:49:43 +00:00
191c596912
Adds print out of incoming resolution.
...
BUG=N/A
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1532004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4091 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-23 11:57:25 +00:00
8d6eb56085
Avoid NPE crash on Android platforms that don't support getting preview framerate.
...
- catch Camera.setParameters() signaling errors through RuntimeException (!)
- make video_demo_apk rebuild when .java sources change
BUG=1778
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1493004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4059 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-17 17:33:31 +00:00
f5d4cb1958
Include files from webrtc/.. paths in video_engine/
...
BUG=1662
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1492004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4056 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-17 13:44:48 +00:00
e874a8f24b
Enable WebRTC demo application on x86 Android
...
Steps to build the demo application for x86 Android:
source build/android/envsetup.sh --target-arch=x86
gclient runhooks
ninja -C out/Debug
cd webrtc/video_engine/test/android
ndk-build APP_ABI=x86
ant debug
R=fischman@webrtc.org , leozwang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1478004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4053 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-17 05:41:07 +00:00
d6ed000585
This is the first step to convert building the Android WebRTC demo to a proper GYP target, android ndk toolchains is being used to build the jni cpp files instead of using ndk-build.
...
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1444005
Patch from Jeremy Mao <yujie.mao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3999 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-10 16:34:01 +00:00
6a36f0e46f
Since the layout of the Android WebRTC demo application is fixed, if we start the demo application in portrait postion, the activity will be destroyed and then created again, force the demo application to start in landscape position to avoid activity re-creation.
...
BUG=webrtc:1741
TEST=Build and run the Android WebRTC demo application
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1439006
Patch from Jeremy Mao <yujie.mao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3994 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-09 17:40:33 +00:00
e525309004
WebRTCDemo Android doesn't hangle activity recreation correctly.
...
Also optimize Statsview a little bit.
BUG=1740
TEST=Manual test with WebRTCDemo Android
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1439005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3993 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-09 08:52:50 +00:00
ebdfa8dcba
Add fischman into OWNERS of WebRTCDemo Android.
...
BUG=
TBR=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1450005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3991 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-09 07:30:38 +00:00
3c48f31e5b
WebRTCDemo Android app to route audio to headphone when it's plugged in.
...
BUG=1654
TEST=WebRTCDemo app
Review URL: https://webrtc-codereview.appspot.com/1348004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3932 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 03:18:00 +00:00
342353780d
Consolidate common_audio into a single target.
...
In principle should reduce gyp processing time, but the difference was not measurable. In any case, it's a good simplification that aligns with having a single common_video target.
R=bjornv@webrtc.org , kma@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1375004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3928 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-30 23:43:26 +00:00
c41478f7eb
Ensure build_demo.py run subprocesses with bash shell.
...
It turns out the default shell becomes /bin/sh on Lucid. By specifying the shell for subprocess.check_call we ensure bash is used.
Thanks to yujie.mao@intel.com for pointing this out.
BUG=1659
TEST=Successful build with build_demo.py both on Ubuntu Lucid and Precise.
TBR=leozwang
Review URL: https://webrtc-codereview.appspot.com/1343004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3875 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-19 11:50:42 +00:00
6f41ca9fd2
WebRTCDemo: Enable making multiple calls.
...
Previously after the first call subsequent attempts to bind the RTP/RTCP ports would fail, since r3754.
BUG=1618
Review URL: https://webrtc-codereview.appspot.com/1302007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3817 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-10 20:33:27 +00:00
b238d1210b
WebRtc_Word32 -> int32_t in video_engine/
...
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1302005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3801 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-09 13:41:51 +00:00
6faf71d27b
Remove the old unused udp_transport
...
Review URL: https://webrtc-codereview.appspot.com/1272009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3788 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-08 23:25:25 +00:00
10eb92039b
Add GYP target for WebRTC Video demo for Android.
...
Add a build target for the Video demo app for Android that only
exists when OS=='android' during build.
Note that this doesn't solve webrtc:1029, it's more like a workaround
waiting for the complete solution, which is to great a proper GYP target
that doesn't involve an action and an external script.
BUG=1029
TEST=Built successfully with:
source build/android/envsetup.sh
gclient runhooks
ninja -C out/Debug
Also verified the target is not present when OS is not 'android'.
Review URL: https://webrtc-codereview.appspot.com/1286004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3769 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-05 13:36:32 +00:00
82dcc9ff11
Remove UDP transport API from ViE
...
Review URL: https://webrtc-codereview.appspot.com/1232004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3754 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-02 20:37:14 +00:00
458194ba65
Fix broken audio.
...
The problem was introduced in 3712, no need to external transport in
real test app, revert the change.
TBR=pwestin@webrtc.org
BUG=1539
Review URL: https://webrtc-codereview.appspot.com/1266005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3735 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-27 20:55:54 +00:00
add50b94a5
WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14.
...
(required bumping minSdkVersion to 14)
This fixes a RuntimeException thrown on GalaxyNexus (but not N7, N4, or NS)
during startPreview() after the sequence of Start(), Stop(), Start(); seemingly
GN's OMX stack can't deal with parallel startPreview() & setPreviewDisplay() in
this situation.
Also:
- Only set the surface in the camera when valid
- Remove duplicate assignment
- Fix error check on voiceChannel allocation to account for multiple channel creation due to orientation change causing onDestroy()/onCreate() on the app, and rampant use of process-static holders for VoE data.
BUG=1537
Review URL: https://webrtc-codereview.appspot.com/1259005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3731 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-27 16:48:34 +00:00
26e35e1d06
Move the VIE tests to use external transport instead of the built in udp transport
...
Review URL: https://webrtc-codereview.appspot.com/1216010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3712 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-22 19:21:27 +00:00
684f0577fb
Revert r3667 and r3665
...
Review URL: https://webrtc-codereview.appspot.com/1199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13 23:20:57 +00:00
361bac7a4f
Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work.
...
Review URL: https://webrtc-codereview.appspot.com/1029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13 17:52:42 +00:00
eb91792cfd
Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings.
...
Review URL: https://webrtc-codereview.appspot.com/1105007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3528 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-18 14:40:18 +00:00
4013ac478e
Roll Chromium revision 176094:182149
...
This gets us (for build/):
* GYP updates for Mac 64-bit builds (r178644)
* Lots of updates to Android scripts
* Support Visual Studio Express 2012.
* asan=1 now enables line numbers in symbolized ASan reports (r179326)
See
http://build.chromium.org/f/chromium/perf/dashboard/ui/changelog.html?url=trunk%2Fsrc%2Fbuild%2F&range=176094%3A182149&mode=html
for more info
In addition to this all our DEPS references to Chromium's DEPS file are
updated.
BUG=none
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/1106004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3516 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-14 19:13:30 +00:00
e6e344a7dc
Sync libvpx and its gyp wrapper from Chromium.
...
TBR=kjellander
BUG=webrtc:1213
Review URL: https://webrtc-codereview.appspot.com/1096007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3505 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-12 19:35:18 +00:00
18a21a03c6
Android NDK build tools
...
This CL enables building with Android NDK in the way that Chromium buildbots do it.
== Overview ==
* Add Android dependencies to DEPS (SDK, NDK, Android test runner). This also makes it possible to use Android's build/android/run_tests.py script to execute tests on Android devices.
* Add a Python script to build the WebRTC Video demo for Android using ndk-build and Ant. This is designed as an annotation script for Buildbots but is also fine to run locally.
* Update Android.mk so it works with the compiler output from a build performed by build/android/buildbot/bb_run_bot.py (which is how Chrome buildbots build).
== Syncing Android dependencies ==
To get the dependencies added in DEPS synced out, you must change the last line
of your .gclient file to look like this:
];target_os = ["android"]
That will append another variable to the .gclient file that causes these
dependencies to be synced during gclient sync.
If you want to get additional platform-specific dependencies in the same
checkout, add them to the list too, e.g. target_os = ["android", "unix"].
== Android.mk ==
The fix in Android.mk is needed since Chrome is building using build/android/buildbot/bb_run_bot.py, which only output the libraries into out/Debug. With the change it works for both that and a normal build (which copies the library files from out/Debug/obj.target/subpath to out/Debug anyway as a part of the build).
== svn:ignore ==
NOTICE: Before submitting, the following directories should be added to svn:ignore in third_party to avoid them from being removed and re-synced for every build:
* android_testrunner
* android_tools
* WebKit
This has to be done in a manual SVN commit since it's not possible to include in a git-svn CL (and I don't want to migrate this to a SVN CL).
BUG=none
TEST=local builds
Review URL: https://webrtc-codereview.appspot.com/1024009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3497 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-11 17:43:19 +00:00
f222a00881
Use TRACE_EVENT to track time spent in VP8 encoding
...
Using the TRACE_EVENT macro to log VP8 encoding events.
Review URL: https://webrtc-codereview.appspot.com/968011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3264 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-11 22:27:55 +00:00
8e49b02f3d
Add more audio codec information into codec list
...
BUG=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/974009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3250 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-07 22:26:57 +00:00
f4e070eca5
Added auto-call feature to WebRTCDemo.
...
This (compile-time switchable) option automatically starts & stops calls in
series to stress-test the setup/teardown codepaths. When startCPULoad() is
removed (https://webrtc-codereview.appspot.com/972008/ ) this showed no
hangs/crashes after completing 200 start/stop pairs.
Also fixed a tiny shutdown-order bug (onDestroy() calling super.onDestroy()
before performing self-shutdown) and changed default video frame resolution to
640x480 to more effectively stress the device (and be a more compelling demo).
BUG=1162
Review URL: https://webrtc-codereview.appspot.com/939032
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3238 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-04 16:53:43 +00:00
