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557 Commits

Author SHA1 Message Date
fischman@webrtc.org
3145a642b7 Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes. 
BUG=1980
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4270 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 20:20:05 +00:00
mflodman@webrtc.org
e6168f5f41 Adding a first simple version of overuse detection, but not hooked up. 
BUG=
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1717004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4268 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 11:23:01 +00:00
mflodman@webrtc.org
1c986e7c89 Removed ViE file API. 
R=asapersson@webrtc.org, niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1723004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4267 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 09:12:49 +00:00
solenberg@webrtc.org
91811e2b04 Remove unused multi stream bandwidth estimator. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1712004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4264 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-25 20:36:14 +00:00
hclam@chromium.org
2e402ce873 Enqueue packet in pacer if sending fails 
If a packet cannot be sent while pacer is in use it should be
queued. This avoid packet loss due to congestion.

BUG=1930
R=pwestin@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1693004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-20 20:18:31 +00:00
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
stefan@webrtc.org
508a84b255 Wire up pacer-based padding. 
This connects the pacer-based padding with the RTP modules, which will
generate padding packets roughly according to what the pacer suggests.
It will only generate padding packets of maximum size to keep the number
off padding packets as small as possible. This also sets a limit of how much
padding + media bitrate which the pacer is allowed to "request" from the
RTP modules.

Padding will for now only be generated by the first sending RTP module.

BUG=1837
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1612005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 12:53:37 +00:00
hclam@chromium.org
7262ad1385 Fix AV sync issue 
r4229 introduced an AV sync issue due to an error.
This is a one linear fix and provides the correct
current video delay for synchronization.

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1675004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-15 06:51:27 +00:00
hclam@chromium.org
9b23ecb939 Log current and target AV delay in ViESyncModule 
R=mikhal@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1668006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4229 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-14 23:30:58 +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
elham@webrtc.org
5137b9752f Updated WebRTC version to 3.33 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1645004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4204 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-10 17:03:51 +00:00
mflodman@webrtc.org
509754c4c9 Making no NACK mode work again in VideoEngine. 
BUG=1910
TEST=ViE autotest loopback with no protection and some percent packet loss
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1631004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4203 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-10 15:50:12 +00:00
pbos@webrtc.org
1819fd711a RW lock access to ssrc maps in VideoCall. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1640004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4202 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-10 13:48:26 +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
7f1b0ae888 Fix init list for VideoSendStream::Config::Rtp. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1616004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4183 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-05 11:39:18 +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
stefan@webrtc.org
de98478965 Update the remote bitrate estimator before passing the packet to the RTP module. 
This solves the problem of reconstructed packets biasing the bandwidth estimate.

TEST=vie_auto_test --automated, trybots
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1594005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4171 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 12:15:40 +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
stefan@webrtc.org
c3cc375499 Add support for padding in pacer. 
This improves pacer-based padding by making sure it limits padding according to:
- Never pad more than 800 kbps.
- Padding + media should not go above a given target bitrate.

Also adds appropriate unittests to make sure we reach the given targets.

BUG=1837
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1582005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4168 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 09:36:56 +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
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