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Author SHA1 Message Date
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
pbos@webrtc.org
b5bf54c4e7 Permit arbitrary payload names for kVideoCodecGeneric. 
BUG=1575

Review URL: https://webrtc-codereview.appspot.com/1282005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-05 13:27:38 +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
solenberg@webrtc.org
a442d4d983 Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. 
Today I had to figure out this code was legacy. Now next person doesn't have to.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1247004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 09:14:36 +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
stefan@webrtc.org
e1a7193869 Fix flakiness in network up/down event tests when running under memcheck. 
TBR=pwestin@webrtc.org

BUG=1524

Review URL: https://webrtc-codereview.appspot.com/1261005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 17:01:48 +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
stefan@webrtc.org
bfacda60be Add interface to signal a network down event. 
- In real-time mode encoding will be paused until the network is back up.
- In buffering mode the encoder will keep encoding, and packets will be
  buffered at the sender. When the buffer grows above the target delay
  encoding will be paused.
- Fixes a couple of issues related to pacing which was found with the new test.
- Introduces different max bitrates for pacing and for encoding. This allows
  the pacer to faster get rid of the queue after a network down event.

(Work based on issue 1237004)

BUG=1524
TESTS=trybots,vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/1258004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 16:36:01 +00:00
pwestin@webrtc.org
a078d5cc38 Bugfix for extended RTP/RTCP test 
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1234004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3713 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 20:03:03 +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
marpan@webrtc.org
94bc4cf905 Add min and target bitrate to VideoCodec. 
Review URL: https://webrtc-codereview.appspot.com/1214004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3710 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 17:13:08 +00:00
pbos@webrtc.org
8911ce46a4 Generic video-codec support. 
Labels frames as key/delta, also marks the first RTP packet of a frame as such,
to allow proper reconstruction even if packets are received out of order.

BUG=1442
TBR=ajm@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1207004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3680 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 16:39:03 +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
2dc0367406 Added destructors for tests to control destruct order 
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1197005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3667 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 21:36:10 +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
mikhal@webrtc.org
efe4edb6da Enabling bufffering mode with no sync module or VoE 
BUG= 1454

Review URL: https://webrtc-codereview.appspot.com/1149006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3625 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-06 23:29:33 +00:00
bemasc@google.com
ea386147f1 Update integration tests for idempotent RTP header settings. 
Review URL: https://webrtc-codereview.appspot.com/1152004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3593 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 23:43:14 +00:00
pbos@webrtc.org
0b6293aaaa Fixed typo in vie_autotest_loopback.cc. 
Review URL: https://webrtc-codereview.appspot.com/1114004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3542 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 12:13:10 +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
mikhal@webrtc.org
3897255b63 Add VoE interface to VieRTP test 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1097015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3527 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-16 01:35:59 +00:00
mikhal@webrtc.org
ef9f76a59d Adding a receive side API for buffering mode. 
At the same time, renaming the send side API.

Review URL: https://webrtc-codereview.appspot.com/1104004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-15 23:22:18 +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
stefan@webrtc.org
07b667db5e Remove MultiStreamMode from test. 
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1101010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3512 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-14 11:35:20 +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
stefan@webrtc.org
0cb48a0a18 Set SingleStream BWE in unittests. 
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/1094004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3494 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-11 08:30:23 +00:00
phoglund@webrtc.org
147c73ea60 Made it possible to render custom call output to file. 
This is to enable quality tests using the custom call.

BUG=
TESTED=locally

Review URL: https://webrtc-codereview.appspot.com/1093005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3483 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-07 08:52:08 +00:00
kjellander@webrtc.org
fe3d606f15 Enable indefinitely running vie_auto_test option 
When doing test automation, the prompt in vie_auto_test is not working as expected on Windows when the test is run from a Buildbot. As soon a prompt is presented to the test runner, vie_auto_test exits, assuming the user pressed Ctrl-D.

By adding a third option for the Stop/Modify call prompt that allows running the call indefinitely (and making that the default), no prompt is displayed when the --auto_custom_call flag is used.

BUG=none
TEST=Execution with vie_auto_test.exe --auto_custom_call --override "Enter destination IP.=192.168.3.11" and by running vie_auto_test in interactive mode.
+ Trybots passing.

Review URL: https://webrtc-codereview.appspot.com/1099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3478 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-06 09:36:37 +00:00
kjellander@webrtc.org
fa53d8717c Fixing/disabling Windows x64 warnings 
Disabled MSVC #4267 warnings in common.gypi to enable x64 builds
for Windows.
Fixed MSVC #4267 warnings in test/testsupport.
Added third_party/directxsdk to .gitignore.

With http://review.webrtc.org/1070008 landed, this should make it possible
to build for x64 on Windows.

BUG=1348
TEST=Compiling with http://review.webrtc.org/1070008 applied:
set GYP_DEFINES="target_arch=x64"
set GYP_GENERATORS=ninja
gclient sync
ninja -C out\Debug_x64

Review URL: https://webrtc-codereview.appspot.com/1060008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3464 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-04 10:07:17 +00:00
mikhal@webrtc.org
dbe97d2550 Adding a send side API for streaming 
Review URL: https://webrtc-codereview.appspot.com/1070009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3457 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-01 19:33:21 +00:00
mikhal@webrtc.org
e07c661a29 VP8: Making key frame interval a tunnable parameter 
Review URL: https://webrtc-codereview.appspot.com/1070006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3444 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-31 16:37:13 +00:00
mflodman@webrtc.org
59d209562f Moving ViE test files and deleting files no longer used. 
BUG=977
TEST=Try bots.

Review URL: https://webrtc-codereview.appspot.com/1046004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3414 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-25 12:45:39 +00:00
wjia@webrtc.org
a3c82bf667 Remove '<(library)' in gyp files. 
This will remove all usage of '<(library)' in all webrtc gyp files. 
Review URL: https://webrtc-codereview.appspot.com/1049005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3392 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-18 23:42:21 +00:00
stefan@webrtc.org
3b7feb2a5d Convert psnr and ssim to strings before printing them. 
Review URL: https://webrtc-codereview.appspot.com/1042004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3380 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-17 13:35:01 +00:00
stefan@webrtc.org
77a584be1d Made ViEToFileRenderer use a separate thread for rendering frames to file. 
Review URL: https://webrtc-codereview.appspot.com/1021011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-15 16:34:34 +00:00
stefan@webrtc.org
e7dc7f8553 Disable full stack PSNR/SSIM triggers on Mac and Win for now due to flakiness. Adding plots of PSNR and SSIM. 
TBR=mflodman

BUG=1271

Review URL: https://webrtc-codereview.appspot.com/1032005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3360 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-11 12:55:19 +00:00
stefan@webrtc.org
e468f08078 Disable PSNR/SSIM thresholds for the Gilber-Elliot test. 
This is to avoid flakiness as the GE model can cause quite big freezes
from time to time. Will keep the test running to get the plots.

TBR=phoglund

BUG=1271

Review URL: https://webrtc-codereview.appspot.com/1030004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-10 15:17:36 +00:00
phoglund@webrtc.org
d005468e9b Generalized mechanism for excluding gtests on platforms, disabled broken tests on mac. 
BUG=1268
TEST=vie_auto_test on mac and linux
TBR=mflodman, kjellander

Review URL: https://webrtc-codereview.appspot.com/1027006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3347 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-09 16:53:42 +00:00
henrika@webrtc.org
d66eb8c4eb Disabled GQoS since it breaks ViE auto test. 
BUG=1266
TEST=vie_auto_test.exe --automated --gtest_filter=-ViERtpFuzzTest* --capture_test_ensure_resolution_alignment_in_capture_device=false

Review URL: https://webrtc-codereview.appspot.com/1025005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3345 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-09 09:13:00 +00:00
stefan@webrtc.org
fcd8585874 Enable external encoders with internal picture source. 
CL enables registering of external encoder with internal picture
source on API by adding simple passthrough parameter that is already
supported within video engine.

Review URL: https://webrtc-codereview.appspot.com/1006006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-09 08:35:40 +00:00
mikhal@webrtc.org
658d423e81 Using Convert in lieu of ExtractBuffer: Less error prone (as we don't need to compute buffer sizes etc.). This cl is first in a series (doing all of WebRtc would make it quite a big cl). While at it, fixing a few headers. 
BUG=988

Review URL: https://webrtc-codereview.appspot.com/995014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3343 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-08 19:19:59 +00:00
phoglund@webrtc.org
df3a15f63b Removed spaces from full stack test labels, consolidated graphs 
NOTE TO SELF: save history on master when deploying!

BUG=
TEST=Ran locally

Review URL: https://webrtc-codereview.appspot.com/1021007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3337 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-07 10:06:52 +00:00
stefan@webrtc.org
1960219530 Make protection method, filename and resolution configurable for FullStackTest. 
Review URL: https://webrtc-codereview.appspot.com/991007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3315 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 12:45:16 +00:00
mikhal@webrtc.org
0b18fb38e6 vie auto test: Adding a constructor for NetworkParameters 
BUG=

Review URL: https://webrtc-codereview.appspot.com/995013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3310 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-18 19:47:52 +00:00
mikhal@webrtc.org
622c8bd0cc ViE autotest: Adding loss models to the external transport 
Review URL: https://webrtc-codereview.appspot.com/1000004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3309 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-18 17:21:51 +00:00
mflodman@webrtc.org
4aee6b637d Added API to get receive side video delay. 
BUG=1222

Review URL: https://webrtc-codereview.appspot.com/997004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3294 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 14:02:10 +00:00
stefan@webrtc.org
32519398b6 Remove latency excl network and add render time diff stats. 
Review URL: https://webrtc-codereview.appspot.com/996004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 09:03:27 +00:00
elham@webrtc.org
ddebc17bee Fix for buffer overflow, WebRTC issue 1196 
Review URL: https://webrtc-codereview.appspot.com/998004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3286 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 21:55:47 +00:00
mflodman@webrtc.org
7acb65a870 Added jitter to fake network pipe. 
Review URL: https://webrtc-codereview.appspot.com/988004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 15:53:11 +00:00
stefan@webrtc.org
91c91df35a Track the actual render time rather than the decode time. 
Review URL: https://webrtc-codereview.appspot.com/993004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3282 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 15:26:01 +00:00