Commit Graph

273 Commits

Author SHA1 Message Date
bbb07e69e5 Glue code and tests for NACK in new VideoEngine API.
The test works by randomly dropping small bursts of packets until enough
NACKs have been sent back by the receiver. Retransmitted packets are
never dropped in order to assure that all packets are eventually
delivered. When enough NACK packets have been received and all dropped
packets retransmitted, the test waits for the receiving side to send a
number of RTCP packets without NACK lists to assure that the receiving
side stops sending NACKs once packets have been retransmitted.

BUG=2043
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1934004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4482 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 12:01:36 +00:00
7fb9ce0cf5 Fix send times in video_full_stack.
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1947004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4481 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 09:29:50 +00:00
d3ae3c7b1f Unbreak clang/android build of webrtc.
TESTED=All target builds once more with clang=1.
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4460 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-31 23:53:07 +00:00
6879c8adad Hooking up first simple CPU adaptation version.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1767004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4384 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-23 11:35:00 +00:00
129afc29fb Correctly rebuild WebRTCDemo after jni/ source file changes
BUG=1980
TEST=Modify source file under jni/ and WebRTCDemo will rebuild
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1831004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4377 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-20 04:43:08 +00:00
aa4d96a134 Revert r4301
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 19:25:04 +00:00
69215d8432 Include files from webrtc/.. paths in video_engine/.
BUG=1662
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1759005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4324 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 15:02:02 +00:00
adf23a55f8 Direct3D renderer for new VideoEngine API tests.
TEST=Rendered video in video_loopback test.
BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1573004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4323 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 14:07:56 +00:00
f3f1358360 Fixed implicit-int-conversion bugs.
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1776004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4313 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-09 14:04:46 +00:00
cab716cc7d Fix a circular dependency by removing an unnecessary dependency, add a missing include_tests check and missing lib references for android.
TBR=henrikg@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1776005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4312 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-09 13:43:24 +00:00
af8d5afec9 Initial port of FullStackTest to new VideoEngine API.
Deferring network loss, delay and such to a later CL.

BUG=1872
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1756004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4310 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-09 08:02:33 +00:00
a2073af728 Fixes build breakage when building WebRTC in Chromium and having include_tests=1.
TBR=fischman@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1770004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4305 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-08 18:14:58 +00:00
1932fe1865 Use scoped_ptr<> for loopback.cc
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1764004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4302 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-05 17:02:37 +00:00
66b2e5c05a Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
rtp_rtcp implementation.

This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.

With this change the dead-or-alive and packet timeout APIs are removed.

TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1745004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-05 14:30:48 +00:00
d900e8bea8 Proper spacing for end-of-namespace comments.
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-03 15:12:26 +00:00
546c91dc2e Build all java files into jar for each module on Android
BUG=None
TEST=All java files in each module are built into jar and used by WebRTCDemo app
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1696004

Patch from Jeremy Mao <yujie.mao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4284 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-01 17:52:39 +00:00
d4803ced60 WebRTCViEDemo: Use global reference when passing variables across different threads
There are JNI local reference changes in ICS when Android SDK
target level API >= 14.
http://android-developers.blogspot.com/2011/11/jni-local-reference-changes-in-ics.html

BUG=NONE
TEST=WebRTCViEDemo works well using MediaCodec Decoder/Renderer
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1744004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4283 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-01 14:55:37 +00:00
0021632f40 Re-add WebRTCDemo dependencies as dependencies (not just inputs) because they also need to be built for this target!
BUG=1980
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1734004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4275 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-27 17:35:32 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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