generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
3,652 Commits 1 Branch 0 Tags 108 MiB
4213633a4d
Commit Graph

3652 Commits

This Branch
This Branch
All Branches
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
pbos@webrtc.org
a048d7cb0a Include files from webrtc/.. paths in rtp_rtcp/ 
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1557004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 14:27:38 +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
pbos@webrtc.org
9aca5b34e1 Remove #pragma once 
BUG=1830
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1568004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4130 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 13:19:09 +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
stefan@webrtc.org
ace7ad2302 Switch frame list implementation to std::map. 
This reduces the complexity of insert and find (by timestamp) from linear to logarithmic, which has a big impact on large frame lists.

BUG=1726
TEST=trybots, vie_auto_test --automated
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1561005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4127 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 07:41:48 +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
marpan@webrtc.org
a6ae644e52 Add comment about test_packet_masks_metrics. 
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1577004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4124 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-28 17:42:12 +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
pbos@webrtc.org
8c34ceeef1 Include "gtest/gtest.h", not by full path, on WEBRTC_ANDROID_PLATFORM_BUILD 
BUG=
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1571004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4119 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-28 09:24:03 +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
15c1c61e2c Include files from webrtc/.. paths in audio_conference_mixer/ 
BUG=1662
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1565004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4117 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-28 08:13:20 +00:00
pbos@webrtc.org
7fad4b8c9f Include files from webrtc/.. paths in audio_processing/ 
BUG=1662
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4116 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-28 08:11:59 +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
pbos@webrtc.org
acaf3a1b13 Include files from webrtc/.. paths in system_wrappers/ 
BUG=1662
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1550004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4111 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-27 15:07:45 +00:00
pbos@webrtc.org
1e50231ff8 Include files from webrtc/.. paths in test/channel_transport/ 
BUG=1662
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1548004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4110 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-27 15:02:23 +00:00
pbos@webrtc.org
6f3d8fcfc0 Include files from webrtc/.. paths in video_processing/ 
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1558004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4109 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-27 14:12:16 +00:00
pbos@webrtc.org
47ce120efb Include files from webrtc/.. paths in remote_bitrate_estimator/ 
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1552004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4108 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-27 12:41:33 +00:00
pbos@webrtc.org
aa30bb7ef5 Include files from webrtc/.. paths in common_audio/ 
BUG=1662
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1535005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-27 09:49:58 +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
34741c8b0e Include files from webrtc/.. paths in test/ 
BUG=1662
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4105 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-27 08:02:22 +00:00
stefan@webrtc.org
7f3f8bc5a6 Refactor jitter buffer to use separate lists for decodable and incomplete frames. 
This changes the design of the jitter buffer to keeping track of decodable frames from the point when packets are inserted in the buffer, instead of searching for decodable frames when they are needed.

To accomplish this the frame_list_, which previously contained all frames (incomplete or complete, continuous or not), is split into a list of decodable_frames_ (complete, continuous) and a list of incomplete_frames_ (either incomplete or non-continuous). These frame lists are updated every time a packet is inserted.

This is another step in the direction of doing most of the work in the jitter buffer only once, when packets are inserted, instead of doing it every time we look for a frame or try to get a nack list.

BUG=1798
TEST=vie_auto_test, trybots
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1522005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4104 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-27 07:02:45 +00:00
sergeyu@chromium.org
ead3c6d508 Fix bugs in DesktopRegion::IntersectWith() and DesktopRect::IntersectWith(). 
IntersectWith() didn't work correctly which breaks screen capturers in chromium.

BUG=crbug.com/243160
R=alexeypa@chromium.org, wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1560004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-24 21:07:20 +00:00
pbos@webrtc.org
8665da8926 Remove dead testRateControl.cc 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1556004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4101 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-24 13:29:29 +00:00
pbos@webrtc.org
a01f7f6509 Removed dead testH263Parser.cc 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1555004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4100 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-24 13:01:57 +00:00
pbos@webrtc.org
c1f0eb2c03 Remove dead bitstreamTest.cc. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1553004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4099 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-24 12:46:08 +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
stefan@webrtc.org
a7dc37d568 Log the type of recycled frames. 
Also correct the logging of incoming key frame packets.

BUG=1814
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1537004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-23 07:21:05 +00:00
hclam@chromium.org
8c49c1eab3 Log a message when a key frame packet is received 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1518004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-22 21:18:59 +00:00
solenberg@webrtc.org
46db413e22 Fix failing tests on 32 bit Linux. 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1534004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4088 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-22 20:53:42 +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
sergeyu@chromium.org
6ec25073e3 Disable WindowCapturer tests on OSX and Linux 
R=alexeypa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1533004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-22 18:47:07 +00:00
sergeyu@chromium.org
6ebfd346ae Add direct_dependent_settings in common.gypi. 
When building chromium targets that depend on webrtc, compiler settings must
have the include path to webrtc and webrtc-specific defines that the headers
may depend on. Added direct_dependent_settings in common.gyp, so that all
webrtc target propagate these settings to dependencies.

R=andrew@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1371005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-22 18:22:21 +00:00
braveyao@webrtc.org
5f8f112a7b Not to request to TURN server for local tests. Follow-up work to issue1197. 
BUG=1197
TEST=Manual test
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/1340004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4083 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-22 07:27:05 +00:00