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189 Commits

Author SHA1 Message Date
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
elham@webrtc.org
14c9909ef6 Updated WebRTC version to 3.27 
Review URL: https://webrtc-codereview.appspot.com/1235004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3714 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 21:59:19 +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
stefan@webrtc.org
3d0b0d6902 Follow-up fix for r3681. 
TESTS=trybots and vie_auto_test
BUG=1514

Review URL: https://webrtc-codereview.appspot.com/1216006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3689 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-19 10:04:57 +00:00
elham@webrtc.org
f1ea0df728 Updated WebRTC version number to 3.26 
Review URL: https://webrtc-codereview.appspot.com/1219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3683 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 17:45:04 +00:00
stefan@webrtc.org
abc9d5b6aa Change VCM interface to take target bitrate in bits per second. 
This also solves issue 1469.

TESTS=trybots
BUG=1469

Review URL: https://webrtc-codereview.appspot.com/1215004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3681 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 17:00:51 +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
mikhal@webrtc.org
bda7f305c5 Adding RTX on source 
Review URL: https://webrtc-codereview.appspot.com/1190004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-15 23:21:52 +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
mikhal@webrtc.org
15960c2b67 Increasing size of nack list in buffered mode. 
Review URL: https://webrtc-codereview.appspot.com/1187007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3666 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 20:52:49 +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
pbos@webrtc.org
927296fd1b Lazy capture_device_info acquisition. 
BUG=1484

Review URL: https://webrtc-codereview.appspot.com/1169005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3641 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-08 13:12:29 +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
elham@webrtc.org
ec6226eedc Updated version number to 3.25 
Review URL: https://webrtc-codereview.appspot.com/1149005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3600 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-04 18:05:56 +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
mflodman@webrtc.org
6648093911 Destroy VCM and VPM instead of delete. 
Review URL: https://webrtc-codereview.appspot.com/1149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3586 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 14:51:23 +00:00
mikhal@webrtc.org
0d8d010017 Handle multiple calls to set initial delay 
BUG= 1419

Review URL: https://webrtc-codereview.appspot.com/1113005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3562 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-22 19:30:44 +00:00
mflodman@webrtc.org
59b2d5fbce Stop and restart fix. 
BUG=1398
TEST=Local stop and start test.

Review URL: https://webrtc-codereview.appspot.com/1115004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3545 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 16:00:27 +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
fischman@webrtc.org
aea96d36e3 Rename webrtc::StatsObserver to webrtc::CallStatsObserver 
to avoid ODR violations with peerconnectioninterface.h in libjingle.

Conflict introduced in
https://webrtc-codereview.appspot.com/1060005/diff/14010/webrtc/modules/interface/module_common_types.h#newcode326

TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1105011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3540 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-19 22:09:36 +00:00
mikhal@webrtc.org
77fced32e2 fixing nack list size calculation 
Review URL: https://webrtc-codereview.appspot.com/1093012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3535 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-19 19:42:35 +00:00
elham@webrtc.org
10741b32b8 Updated version number to 3.24 
Review URL: https://webrtc-codereview.appspot.com/1110004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3533 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-19 19:05:14 +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
mflodman@webrtc.org
294e5b0b82 Reset ssrc when calling SetSendCodec. 
BUG=1398
TEST=Tested locally.

Review URL: https://webrtc-codereview.appspot.com/1107004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3511 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-14 11:25:26 +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
fbarchard@google.com
0ee57c2436 Increase maximum resolution to 4k x 3k. 
BUG=1375
TESTED=try bots
Review URL: https://webrtc-codereview.appspot.com/1097008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3503 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-12 04:57:56 +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
mikhal@webrtc.org
3d305c64b4 Updates to send side streaming mode: 
1. Disabling frame-droppers from the vie encoder and not the channel.
2. Accounting for qpMax in the VP8 wrapper.

Review URL: https://webrtc-codereview.appspot.com/1101007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3492 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-10 18:42:55 +00:00
tnakamura@webrtc.org
79481474ad Update version number to 3.23 
TBR=niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/1105004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-08 19:56:09 +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
mflodman@webrtc.org
4fd5527ab1 Don't report an error for GetEstimatedReceiveBandwidth if there is no valid 
estimate.

BUG=1377

Review URL: https://webrtc-codereview.appspot.com/1095005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3479 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-06 17:46:39 +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
elham@webrtc.org
c4e45f67c0 Updated version number to 3.22 
Review URL: https://webrtc-codereview.appspot.com/1096004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3469 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-04 21:34:05 +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
stefan@webrtc.org
becf9c897c Fix mismatch between different NACK list lengths and packet buffers. 
This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors.

BUG=1289

Review URL: https://webrtc-codereview.appspot.com/1065007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-01 15:09:57 +00:00
stefan@webrtc.org
b586507986 Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. 
Also make sure RTT is computed independently of whether it's time to send RTCP messages or not.

BUG=1298

Review URL: https://webrtc-codereview.appspot.com/1060005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-01 14:33:42 +00:00
braveyao@webrtc.org
7050f96bff Set frame length for frame converting in external renderer 
ISSUE = Issue 1342
TEST  = Manual Test
Review URL: https://webrtc-codereview.appspot.com/1083005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3451 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-01 02:49:29 +00:00