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Author SHA1 Message Date
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
stefan@webrtc.org
eea2622350 Correctly set SSRCs for extra send RTP modules. 
Fixes a regression introduced in r4096.

BUG=1845
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1585004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4134 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 14:07:54 +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
stefan@webrtc.org
08994cc525 Fix a return value mismatch introduced in r4129. 
TBR=mflodman@webrtc.org
TEST=vie_auto_test, trybots

Review URL: https://webrtc-codereview.appspot.com/1584005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4131 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 13:28:21 +00:00
stefan@webrtc.org
a5cb98cbbd Breaking out RTP header parsing from the RTP module. 
This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video.

Moving bandwidth estimation before the RTP module is also required for RTX.

TEST=vie_auto_test, voe_auto_test, trybots.
BUG=1811
R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1545004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4129 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 12:12:51 +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
elham@webrtc.org
fe6a75e50e Updated WebRTC version to 3.32 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1576004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4122 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-28 17:04:56 +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
pbos@webrtc.org
eceb53241e Default constructors for new VideoEngine structs. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1543004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4115 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-28 08:04:45 +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
solenberg@webrtc.org
a6db54d4c9 - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. 
- Changed implementation of SetReceiveAbsoluteSendTimeStatus API so the RBE instance is changed when at least one channel in a group has the extension enabled.

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

Review URL: https://webrtc-codereview.appspot.com/1553005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4113 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-27 16:02:56 +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
turaj@webrtc.org
e46c8d3875 API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. 
TEST=unit-test, manual, trybots.
R=henrik.lundin@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1384005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4087 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-22 20:39:43 +00:00
solenberg@webrtc.org
561990fd73 - Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp. 
- Changed RemoteBitrateObserver::OnReceivedBitrateChanged() to use a const & instead of non-const *, to avoid unnecessary copying.
- Refactored RemoteBitrateEstimatorTest so it can be instantiated for both single and multi stream BWE (first using a parameterized test, but then as a standard test fixture and a few helper functions).
- Refactored some tests in RemoteBitrateEstimatorTest into a common function CapacityDropTestHelper().

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

Review URL: https://webrtc-codereview.appspot.com/1521004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4086 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-22 19:04:19 +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
hclam@chromium.org
69bb348084 Log error in ViESender::SendRTCPPacket 
Log the packet length and the error of SendRTCPPacket.

R=mikhal@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1512005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4074 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-20 22:39:39 +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
hclam@chromium.org
b3e5acfb66 Cleanup traces in WebRTC 
Remove some unused traces and add a trace counter for encoded video size.

R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1476004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 21:13:02 +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
justinlin@chromium.org
7bfb3a3227 Add more tracing for key frames. 
R=mallinath@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1428004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4015 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-13 22:59:00 +00:00
vikasmarwaha@webrtc.org
941fcc5841 Increased the limit for KViEMaxCaptureDevices from 10 to 256. See issue 1343. 
TBR=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/1463005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4014 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-13 20:28:23 +00:00
elham@webrtc.org
52b3905ec8 Updated WebRTC version to 3.31 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1462004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4011 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-13 17:00:56 +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
andresp@webrtc.org
7707d060bb Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1450008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4007 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-13 10:50:50 +00:00
pbos@webrtc.org
7ee822805d Remove TEXT(x) for BUILDINFO macros. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1453004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4005 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-13 09:29:03 +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