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Author SHA1 Message Date
stefan@webrtc.org
8ccb9f9716 Fixes some pacer/padding issues found while testing. 
- A bug was introduced in r4234 causing no paced packets to be sent.
- Only update the sequence number counter if a padding packet is actually going to be sent, to avoid packet loss.
- Have all packets go through the pacer if pacing is enabled to avoid reordering.
- Fix race condition on reading capture_time_ms_/timestamp_ in rtp_sender.cc.

BUG=1837
TEST=trybots and vie_auto_test --automated
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1682004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4246 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-19 14:13:42 +00:00
henrike@webrtc.org
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
fischman@webrtc.org
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
kjellander@webrtc.org
20a993f88a Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. 
Take two of http://review.webrtc.org/1657004/
This time with execution on trybots.

BUG=1925
TEST=win,win_rel,mac,mac_rel,linux,linux_rel trybots passing.
R=mflodman
TBR=mflodman

Review URL: https://webrtc-codereview.appspot.com/1658004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4221 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 14:38:01 +00:00
kjellander@webrtc.org
935d705370 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. 
Disable on Windows due to failures on bots.

BUG=1925
TEST=compile on Linux and Windows.
R=mflodman
TBR=mflodman

Review URL: https://webrtc-codereview.appspot.com/1657004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4220 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 13:59:57 +00:00
kjellander@webrtc.org
7124dd8561 Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test. 
BUG=1790
TEST=Just local compilation.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1654004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4217 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 08:28:09 +00:00
kjellander@webrtc.org
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
fischman@webrtc.org
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
mflodman@webrtc.org
3ba883f0fc Removing functionality for inserting pre-encoded frames instead of raw 
video frames. The functionality hasn't been used for a long time and
should be done properly if used in the future.

This is a pre-step for implementing CPU overload control.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1630004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4194 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-07 13:57:57 +00:00
pbos@webrtc.org
025f4f152b Stats+Config moved into VideoSend/ReceiveStreams. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1561006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4182 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-05 11:33:21 +00:00
pbos@webrtc.org
6998c8ef7a Remove XvRenderer. 
One test renderer per platform is sufficient, multiple code paths are
bad.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1612004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4170 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 11:56:06 +00:00
mikhal@webrtc.org
6eb0f6a4d9 Setting SSRC in vie_loopback_test 
BUG=1822
R=pwestin@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1603004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4159 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-03 22:54:40 +00:00
pbos@webrtc.org
4213633a4d Use int for FPS instead of size_t. 
BUG=
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1578005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4136 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 15:13:12 +00:00
pbos@webrtc.org
7bdfff3503 Remove assert for aborting FrameGeneratorCapturer. 
BUG=
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1586004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4133 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 13:58:11 +00:00
pbos@webrtc.org
26d12105a4 Fake VideoCapturer based on FrameGenerator 
BUG=1793
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4132 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 13:41:03 +00:00
pbos@webrtc.org
1ecee9a15a Break video_engine/new_include/common.h into smaller parts. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1571005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4128 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 11:34:32 +00:00
andrew@webrtc.org
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
mflodman@webrtc.org
a066cbf37c Don't return an estimated receive BW for channels not receiving video. 
BUG=1834
TEST=ViE RTP autotest
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1572004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4121 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-28 15:00:15 +00:00
pbos@webrtc.org
4079c31c0a Include gflags with "gflags/gflags.h" instead of <> 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1551004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4120 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-28 10:38:11 +00:00
stefan@webrtc.org
3496ef1087 Improve vie_autotest_rtp_rtcp by reenabling important tests and reducing flakiness. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1567004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4118 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-28 08:36:02 +00:00
fischman@webrtc.org
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
mflodman@webrtc.org
7f944f3027 Adding Mac test renderer, some test refactoring and made cpplint pass. 
BUG=1667
TEST=Rendered video in Mac loopback test.
R=pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1554004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4112 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-27 15:52:38 +00:00
stefan@webrtc.org
0afd84067a Disabling a flaky expectation in vie_autotest_rtp_rtcp.cc. 
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1566004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4106 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-27 08:58:16 +00:00
pbos@webrtc.org
28556f5658 Make sure GlxRenderer frees its resources. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1544004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4098 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-24 10:54:56 +00:00
stefan@webrtc.org
c74c3c2447 Adds integration test for RTX and fixes bugs found. 
BUG=1811
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4096 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-23 13:48:22 +00:00
stefan@webrtc.org
5c58f63d3f Fix regression where retransmission bitrate is no longer estimated. 
BUG=1813
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1530004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4095 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-23 13:36:55 +00:00
pbos@webrtc.org
d445d2229e CreateEmptyFrame casts from size_t to int. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1540004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4094 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-23 12:59:51 +00:00
pbos@webrtc.org
9b30348cfc FrameGenerator class for future fake capture device. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1511004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4093 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-23 12:37:11 +00:00
pbos@webrtc.org
771cdcbb09 Control new VideoEngine tests with gflags. 
BUG=1703
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1497005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4092 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-23 12:20:16 +00:00
henrike@webrtc.org
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
pbos@webrtc.org
d2541e81c6 Remove <iostream> usage from loopback.cc 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1522004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4077 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-21 11:09:36 +00:00
pbos@webrtc.org
375deb4e19 Suffix VcmCapturer's privates with underscore_ 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1506005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4076 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-21 09:32:22 +00:00
solenberg@webrtc.org
cb9cff0c71 Add functions to ViE API to enable/disable the absolute send time header extension. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1487004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4065 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-20 12:00:23 +00:00
fischman@webrtc.org
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
pbos@webrtc.org
21632124dd Include gflags properly and X11 include order in VideoEngine. 
BUG=

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1500004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4057 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-17 14:25:02 +00:00
pbos@webrtc.org
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
fischman@webrtc.org
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
pbos@webrtc.org
29d5839233 New VideoEngine API implementation on top of old one, first steps. 
BUG=1668
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1360004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4044 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 12:08:03 +00:00
mflodman@webrtc.org
4dee30927a Remove SetOverUseDetectorOptions and cleaned ViESharedData. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1486004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 11:13:18 +00:00
andresp@webrtc.org
29b2219914 Adding a factory to remote bitrate estimator and allow it to be set via config. 
Additionally:
 - clean api to set remote bitrate estimator mode.
 - clean api to set over use detector options.

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

Review URL: https://webrtc-codereview.appspot.com/1448006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4027 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 12:10:58 +00:00
fbarchard@google.com
c9cb4fffac Fix typo in log statement. witdh should be width. 
BUG=none
TESTED=try bots
Review URL: https://webrtc-codereview.appspot.com/1466004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4016 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 05:02:08 +00:00
phoglund@webrtc.org
c53480fbcf Disabled flaky codec test (RunsCodecTestWithoutErrors) 
BUG=1734
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1460004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4009 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-13 15:10:02 +00:00
fischman@webrtc.org
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
fischman@webrtc.org
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
braveyao@webrtc.org
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
braveyao@webrtc.org
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
andrew@webrtc.org
d72262dc01 Fix compile errors in ViE with latest clang. 
Rolling to the latest Chromium picks up a new clang, which catches a fresh error:

error: 'reinterpret_cast' to class 'webrtc::VideoEngineImpl *' from its base at non-zero offset 'webrtc::VideoEngine *' behaves differently from 'static_cast' [-Werror,-Wreinterpret-base-class]
 VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
                              ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
../../webrtc/video_engine/vie_codec_impl.cc:36:31: note: use 'static_cast' to adjust the pointer correctly while downcasting
  VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
                              ^~~~~~~~~~~~~~~~
                              static_cast

This was triggered by André's change here:
https://code.google.com/p/webrtc/source/detail?r=3986
which made VideoEngineImpl a derived class of VideoEngine (good).

Picked up one other error as well:
error: implicit conversion from 'long' to 'int' changes value from 9223372036854775807 to -1 [-Werror,-Wconstant-conversion]
        AutoTestSleep(std::numeric_limits<long>::max());
        ~~~~~~~~~~~~~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

This fixes the errors and is required before stable can be rolled in Chromium.

TBR=mflodman,andresp

Review URL: https://webrtc-codereview.appspot.com/1450004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3989 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-09 02:12:07 +00:00
pbos@webrtc.org
3004c79c6a Fix clang errors in non-GYP_DEFINES=clang=1 build 
BUG=1623
R=stefan@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1368004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 12:36:21 +00:00
pwestin@webrtc.org
42636e82d0 Removing bad code resulting in flaky test. 
BUG=1723
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1390004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3941 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 21:02:04 +00:00
pwestin@webrtc.org
0d95e06a2f Bugfix custom call stop. 
BUG=1717
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1388004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3938 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 18:25:03 +00:00