37 Commits

Author SHA1 Message Date
braveyao@webrtc.org
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
andrew@webrtc.org
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
kjellander@webrtc.org
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
fischman@webrtc.org
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
pbos@webrtc.org
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
pwestin@webrtc.org
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
kjellander@webrtc.org
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
pwestin@webrtc.org
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
leozwang@webrtc.org
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
fischman@webrtc.org
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
pwestin@webrtc.org
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
pwestin@webrtc.org
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
pwestin@webrtc.org
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
stefan@webrtc.org
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
kjellander@webrtc.org
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
andrew@webrtc.org
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
kjellander@webrtc.org
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
hclam@chromium.org
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
leozwang@webrtc.org
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
fischman@webrtc.org
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
dwkang@webrtc.org
6bd737a714 First pass of MediaCodecDecoder which uses Android MediaCodec API.
Background:
As of now, MediaCodec API is the only public interface which enables us
to access low level HW resource in Android. ViEMediaCodecDecoder will be
used for further experiments/exploration.

TODO:
  To fix known issues. (detaching thread from VM and frequent GC)
Review URL: https://webrtc-codereview.appspot.com/933033

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3233 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-04 06:38:19 +00:00
fischman@webrtc.org
d814d71d92 Delete {start,stop}CPULoad() since they're broken.
- stopCPULoad is incorrect; since mIsBackgroudLoadRunning isn't declared
  volatile, the empty while loop in the background thread isn't required to do a
  memory read (as opposed to reading the value just once and caching it).  The
  result is that stopCPULoad() may never return as the .join() waits forever.
- startCPULoad isn't guaranteed to tax the CPU; the JVM is free to replace the
  while loop in startCPULoad() with a thread pause since it can prove it'll
  never exit the loop once entered (b/c of the previous item).

It's not clear what correct behavior here would be so I'm deleting the code
rather than trying to make it work.  This was responsible for at least most if
not all of the hanginess of start/stop'ing multiple calls in series.

BUG=1162

Review URL: https://webrtc-codereview.appspot.com/972008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3202 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 23:00:41 +00:00
fischman@webrtc.org
be5b5ba490 Enable building WebRTCDemo apk using Release webrtc libs, take 2.
Now passing BUILDTYPE=Release to both the make that builds the libs and the
ndk-build that builds the app makes the app use non-Debug libs.

Review URL: https://webrtc-codereview.appspot.com/966029

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3201 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 18:06:00 +00:00
fischman@webrtc.org
de6f8fbd6d Revert 3190 - Enable building WebRTCDemo apk using Release webrtc libs.
Now passing BUILDTYPE=Release to both the make that builds the libs and the
ndk-build that builds the app makes the app use non-Debug libs.

TBR=leozwang@webrtc.org,

Review URL: https://webrtc-codereview.appspot.com/968010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3191 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 01:18:04 +00:00
fischman@webrtc.org
28afee04ae Enable building WebRTCDemo apk using Release webrtc libs.
Now passing BUILDTYPE=Release to both the make that builds the libs and the
ndk-build that builds the app makes the app use non-Debug libs.

Review URL: https://webrtc-codereview.appspot.com/972007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3190 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 01:09:44 +00:00
leozwang@webrtc.org
aa46ea0b8b Remove ringtone from test app
BUG=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/968009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3184 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-28 19:39:23 +00:00
leozwang@webrtc.org
5835adfef0 Reorganize gyp for Android
BUG=1120
TEST=trybot, local test on xoom and nexus

Message:
It turned out the last CL can only build neon code that
caused problem on Xoom.

Description:
In order to support audo-cpu-detection, I split files into two gypi files, one
contains non-neon code, antoher one ONLY contains neon specific code, so I can
apply different flags to them. Also created two build targets for each of them

We build for linux as before.

Tested on xoom and nexus S.
Review URL: https://webrtc-codereview.appspot.com/930024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3141 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-21 04:10:16 +00:00
leozwang@webrtc.org
6b9543b801 Add libpaced_sender to Android makefile
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/965022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3091 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-13 23:36:41 +00:00
fischman@webrtc.org
f4b26178df Re-initialize enough state on "Stop Call" to be able to stop/start multiple calls in succession.
Review URL: https://webrtc-codereview.appspot.com/965015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3074 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-09 20:57:24 +00:00
leozwang@webrtc.org
06d72d881f Add Android OWNER files
Message:
Add OWNER files so I can review and approve changes for Android.
I also should be owner for all .mk file, but it's OK for now,
please review.

BUG=None
TEST=None
Review URL: https://webrtc-codereview.appspot.com/932016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3069 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-09 17:51:55 +00:00
andrew@webrtc.org
9841d92b8d Reorganize modules/video_render.
The usual elimination of main/source etc.

Review URL: https://webrtc-codereview.appspot.com/929011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3027 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-31 05:22:11 +00:00
andrew@webrtc.org
3c01316d5a Fix Android build after video_capture reorg.
Review URL: https://webrtc-codereview.appspot.com/934011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3026 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-30 22:17:21 +00:00
leozwang@webrtc.org
6ab92ed42d Check if opus exists when build test app on Android
BUG=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/933011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3022 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-30 19:04:58 +00:00
leozwang@webrtc.org
6f19b1b651 Enable Opus
BUG=webrtc issue 992
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/942004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3012 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-27 17:46:55 +00:00
mikhal@webrtc.org
9fedff7c17 Switching to I420VideoFrame
Review URL: https://webrtc-codereview.appspot.com/922004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2983 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-24 18:33:04 +00:00
leozwang@webrtc.org
9eff74a6da Change android NDK library path
TBR=wu
Review URL: https://webrtc-codereview.appspot.com/926004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2968 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 22:05:25 +00:00
andrew@webrtc.org
14b43beb7c Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00