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Author SHA1 Message Date
elham@webrtc.org
c0aa29c98c Updated WebRTC version to 3.37 
TBR=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1894004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4417 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 16:57:21 +00:00
braveyao@webrtc.org
b6433b7a1e Access receiving_ under receive_cs critical section 
Note: InsertRTPPacket/InsertRTCPPacket could be merged into 
ReceivedRTPPacket, as there are no other callers.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1869005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 09:02:46 +00:00
mflodman@webrtc.org
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
yujie.mao@webrtc.org
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
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
elham@webrtc.org
b7eda43810 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on 
several SSRCs"

R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1774006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-15 21:08:27 +00:00
elham@webrtc.org
8543c1c77c Updated WebRTC version to 3.36 
TBR=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1780005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4341 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-15 17:19:45 +00:00
pbos@webrtc.org
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
pbos@webrtc.org
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
stefan@webrtc.org
717d147ebb Support sending multiple report blocks and keeping track of statistics on several SSRCs. 
BUG=1811
TEST=vie_auto_test --automated, voe_auto_test --automated, trybots
R=andresp@webrtc.org, tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1768004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 13:39:27 +00:00
tnakamura@webrtc.org
6aa6229953 Update version number to 3.35 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1778004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4316 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-09 18:43:02 +00:00
tnakamura@webrtc.org
c79b9295cd Update version number to 3.34 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1770006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4315 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-09 18:40:52 +00:00
pbos@webrtc.org
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
stefan@webrtc.org
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
pbos@webrtc.org
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
hclam@chromium.org
1a7b9b94be Cleanup WebRTC tracing 
The goal of this change is to:
1. Remove unused tracing events.
2. Organize tracing events to facilitate measurement of end to end latency.

The major change in this CL is to use ASYNC_STEP such that operation
flow can be traced for the same frame.

R=marpan@webrtc.org, pwestin@webrtc.org, turaj@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1761004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4308 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-08 21:31:18 +00:00
henrike@webrtc.org
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
pbos@webrtc.org
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
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
21beaf97e7 Adding Stefan as VideoEngine owner, removing Per. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1762004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-04 12:29:08 +00:00
pbos@webrtc.org
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
pbos@webrtc.org
65a1f2cb2b Remove log of undefined input values in GetCodec. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1755004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4286 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-02 13:02:14 +00:00
fischman@webrtc.org
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
yujie.mao@webrtc.org
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
fischman@webrtc.org
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
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