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73 Commits

Author SHA1 Message Date
wu@webrtc.org
9dba525627 * Update libjingle to 50389769. 
* Together with "Add texture support for i420 video frame." from
wuchengli@chromium.org.
https://webrtc-codereview.appspot.com/1413004

RISK=P1
TESTED=try bots
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1967004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 20:36:57 +00:00
pbos@webrtc.org
12dc1a38ca Switch C++-style C headers with their C equivalents. 
The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.

BUG=1833
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1917004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 16:22:53 +00:00
tnakamura@webrtc.org
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
stefan@webrtc.org
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
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
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
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
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
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
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
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
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
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
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
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
mikhal@webrtc.org
dd807ac474 Adding buffered mode to loopback test 
Review URL: https://webrtc-codereview.appspot.com/1371004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3920 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 15:19:47 +00:00
mikhal@webrtc.org
47128ab5ab Removing vie file related code from vie_custom_call 
Follow up on https://code.google.com/p/webrtc/source/detail?r=3900

Review URL: https://webrtc-codereview.appspot.com/1361004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3911 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 20:09:54 +00:00
pbos@webrtc.org
b238d1210b WebRtc_Word32 -> int32_t in video_engine/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1302005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3801 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:41:51 +00:00
mflodman@webrtc.org
367804cce2 Clean packets on the network when closing + made loopback test actually run again. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1290006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3776 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 10:42:50 +00:00
pbos@webrtc.org
b5bf54c4e7 Permit arbitrary payload names for kVideoCodecGeneric. 
BUG=1575

Review URL: https://webrtc-codereview.appspot.com/1282005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-05 13:27:38 +00:00
solenberg@webrtc.org
a442d4d983 Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. 
Today I had to figure out this code was legacy. Now next person doesn't have to.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1247004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 09:14:36 +00:00
stefan@webrtc.org
e1a7193869 Fix flakiness in network up/down event tests when running under memcheck. 
TBR=pwestin@webrtc.org

BUG=1524

Review URL: https://webrtc-codereview.appspot.com/1261005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 17:01:48 +00:00
stefan@webrtc.org
bfacda60be Add interface to signal a network down event. 
- In real-time mode encoding will be paused until the network is back up.
- In buffering mode the encoder will keep encoding, and packets will be
  buffered at the sender. When the buffer grows above the target delay
  encoding will be paused.
- Fixes a couple of issues related to pacing which was found with the new test.
- Introduces different max bitrates for pacing and for encoding. This allows
  the pacer to faster get rid of the queue after a network down event.

(Work based on issue 1237004)

BUG=1524
TESTS=trybots,vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/1258004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 16:36:01 +00:00
pwestin@webrtc.org
a078d5cc38 Bugfix for extended RTP/RTCP test 
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1234004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3713 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 20:03:03 +00:00
pwestin@webrtc.org
26e35e1d06 Move the VIE tests to use external transport instead of the built in udp transport 
Review URL: https://webrtc-codereview.appspot.com/1216010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3712 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 19:21:27 +00:00
marpan@webrtc.org
94bc4cf905 Add min and target bitrate to VideoCodec. 
Review URL: https://webrtc-codereview.appspot.com/1214004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3710 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 17:13:08 +00:00
pbos@webrtc.org
8911ce46a4 Generic video-codec support. 
Labels frames as key/delta, also marks the first RTP packet of a frame as such,
to allow proper reconstruction even if packets are received out of order.

BUG=1442
TBR=ajm@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1207004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3680 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 16:39:03 +00:00
pwestin@webrtc.org
684f0577fb Revert r3667 and r3665 
Review URL: https://webrtc-codereview.appspot.com/1199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 23:20:57 +00:00
pwestin@webrtc.org
2dc0367406 Added destructors for tests to control destruct order 
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1197005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3667 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 21:36:10 +00:00
pwestin@webrtc.org
361bac7a4f Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. 
Review URL: https://webrtc-codereview.appspot.com/1029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 17:52:42 +00:00
mikhal@webrtc.org
efe4edb6da Enabling bufffering mode with no sync module or VoE 
BUG= 1454

Review URL: https://webrtc-codereview.appspot.com/1149006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3625 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-06 23:29:33 +00:00
bemasc@google.com
ea386147f1 Update integration tests for idempotent RTP header settings. 
Review URL: https://webrtc-codereview.appspot.com/1152004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3593 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 23:43:14 +00:00
pbos@webrtc.org
0b6293aaaa Fixed typo in vie_autotest_loopback.cc. 
Review URL: https://webrtc-codereview.appspot.com/1114004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3542 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 12:13:10 +00:00
mikhal@webrtc.org
3897255b63 Add VoE interface to VieRTP test 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1097015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3527 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-16 01:35:59 +00:00
mikhal@webrtc.org
ef9f76a59d Adding a receive side API for buffering mode. 
At the same time, renaming the send side API.

Review URL: https://webrtc-codereview.appspot.com/1104004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-15 23:22:18 +00:00
stefan@webrtc.org
07b667db5e Remove MultiStreamMode from test. 
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1101010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3512 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-14 11:35:20 +00:00
stefan@webrtc.org
0cb48a0a18 Set SingleStream BWE in unittests. 
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/1094004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3494 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-11 08:30:23 +00:00
phoglund@webrtc.org
147c73ea60 Made it possible to render custom call output to file. 
This is to enable quality tests using the custom call.

BUG=
TESTED=locally

Review URL: https://webrtc-codereview.appspot.com/1093005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3483 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-07 08:52:08 +00:00
kjellander@webrtc.org
fe3d606f15 Enable indefinitely running vie_auto_test option 
When doing test automation, the prompt in vie_auto_test is not working as expected on Windows when the test is run from a Buildbot. As soon a prompt is presented to the test runner, vie_auto_test exits, assuming the user pressed Ctrl-D.

By adding a third option for the Stop/Modify call prompt that allows running the call indefinitely (and making that the default), no prompt is displayed when the --auto_custom_call flag is used.

BUG=none
TEST=Execution with vie_auto_test.exe --auto_custom_call --override "Enter destination IP.=192.168.3.11" and by running vie_auto_test in interactive mode.
+ Trybots passing.

Review URL: https://webrtc-codereview.appspot.com/1099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3478 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-06 09:36:37 +00:00
mikhal@webrtc.org
dbe97d2550 Adding a send side API for streaming 
Review URL: https://webrtc-codereview.appspot.com/1070009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3457 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-01 19:33:21 +00:00
mikhal@webrtc.org
e07c661a29 VP8: Making key frame interval a tunnable parameter 
Review URL: https://webrtc-codereview.appspot.com/1070006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3444 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-31 16:37:13 +00:00
mflodman@webrtc.org
59d209562f Moving ViE test files and deleting files no longer used. 
BUG=977
TEST=Try bots.

Review URL: https://webrtc-codereview.appspot.com/1046004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3414 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-25 12:45:39 +00:00
henrika@webrtc.org
d66eb8c4eb Disabled GQoS since it breaks ViE auto test. 
BUG=1266
TEST=vie_auto_test.exe --automated --gtest_filter=-ViERtpFuzzTest* --capture_test_ensure_resolution_alignment_in_capture_device=false

Review URL: https://webrtc-codereview.appspot.com/1025005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3345 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-09 09:13:00 +00:00