3b84b3a58c
Add RTCP packet types to packet builder:
...
REMB, TMMBR, TMMBN and
extended reports: RRTR, DLRR, VoIP metric.
BUG=2450
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9299005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6537 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-25 12:22:17 +00:00
6b061425c2
Updated W3C getusermedia tests to the latest version of the spec.
...
BUG=webrtc:3455
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6459 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 08:46:58 +00:00
4b12d40008
Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class.
...
BUG=2450
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6449 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 14:09:28 +00:00
d6e2213edd
Remove ivinnichenko from webrtc/test/OWNERS
...
Apparently, We're doing a poor job of cleaning out
really old OWNERS.
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6447 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 11:42:27 +00:00
a1bfc50a72
Pass GYP DEPTH variable to isolate.
...
Similar change to https://codereview.chromium.org/322403003/
This will make it possible to handle different
directory levels for special builds of WebRTC, without
breaking GYP when the .isolate files are processed and
their contents is verified.
Also update all our .isolate files to use the <(DEPTH)
variable.
BUG=343106
TEST=Successful compile+test on Linux using:
ninja -C out/Release
tools/swarming_client/isolate.py run -s out/Release/tools_unittests.isolated
Also trybots passing all tests.
R=pbos@webrtc.org
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6427 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-13 09:02:15 +00:00
7b82c18979
Add kjellander@webrtc.org as OWNER for *.isolate
...
This should make project-wide changes for isolate files
easier and make it more obvious who's a suitable reviewer
for them.
BUG=
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6379 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10 05:42:53 +00:00
582367f251
Updated conformance tests and w3c-ified them.
...
I intend here to put these up for review on W3C. This moves the tests
to use the W3C-style vendor prefix handling and updates the tests to
the latest drafts.
This yields 44 Pass 24 Fail and 13 pass 54 fail 1 timeout on Firefox.
As far I can tell all failures are correct; in particular FF media
media stream tracks do not adhere to the standard.
Also I can't get FF to get a remote video up in the peerconnection
test, just the local one.
BUG=webrtc:3455
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6370 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09 09:47:44 +00:00
6ae48c6609
Make VideoSendStream/VideoReceiveStream configs const.
...
Benefits of this is that the send config previously had unclear locking
requirements, a lock was used to lock parts parts of it while
reconfiguring the VideoEncoder. Primary work was splitting out video
streams from config as well as encoder_settings as these change on
ReconfigureVideoEncoder. Now threading requirements for both member
configs are clear (as they are read-only), and encoder_settings doesn't
stay in the config as a stale pointer.
CreateVideoSendStream now takes video streams separately as well as the
encoder_settings pointer, analogous to ReconfigureVideoEncoder.
This change required changing so that pacing is silently enabled when
using suspend_below_min_bitrate rather than silently setting it.
R=henrik.lundin@webrtc.org , mflodman@webrtc.org , pthatcher@webrtc.org , stefan@webrtc.org
BUG=3260
Review URL: https://webrtc-codereview.appspot.com/20409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6349 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06 10:49:19 +00:00
94454b71ad
Fix the chain that propagates the audio frame's rtp and ntp timestamp including:
...
* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.
Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.
Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.
BUG=3111
R=henrik.lundin@webrtc.org , turaj@webrtc.org , xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc
Review URL: https://webrtc-codereview.appspot.com/14559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 20:34:08 +00:00
eae7924836
Adding back platform specific renderer to video loopback test.
...
BUG=3039
TEST=locally on Mac and Win, video_loopback test
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6339 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 09:32:51 +00:00
e6e139159f
Android: cleanup gtest_target_type conditions.
...
Ever since crrev.com/133053 OS==android implies:
gtest_target_type=shared_library
Similar to Chromium's crrev.com/271222 where base.gyp's conditions are changed
(which the affected conditions in this cl comes from).
R=henrike@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6332 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 20:46:50 +00:00
88fbb2d86b
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
...
Same as https://webrtc-codereview.appspot.com/19519004 . The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux ...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing ...
(tested locally).
BUG=3380
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17619005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 21:18:46 +00:00
2fa7f79094
Revert 6202 "Switch to using base/constructormagic.h and remove ..."
...
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
>
> BUG=N/A
> R=andrew@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/19519004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14579007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 11:07:29 +00:00
125ffd709d
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
...
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 15:20:44 +00:00
a826006132
Add NACK and RPSI packet types to RTCP packet builder.
...
Fixes bug found when parsing received RPSI packet.
BUG=2450
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6194 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 09:53:51 +00:00
cb711f77d2
Add interface to propagate audio capture timestamp to the renderer.
...
BUG=3111
R=andrew@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 17:39:11 +00:00
60015d27ae
Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main}
...
This allows use of webrtc field trials and opens up the possibility to try the different code paths when running the unit tests by wiring them up to a --force_fieldtrials.
Tested: running a test target that links with the above with a flag --force_fieldtrials=invalid leads the test to crash.
BUG=crbug/367114
R=mflodman@webrtc.org , phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6181 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 09:39:51 +00:00
caba2d2a37
Add DeliveryStatus enum to DeliverPacket().
...
Allows signalling why packet delivery failed. Especially enables
signaling that delivery fails because the incoming packet had an unknown
SSRC. This allows an application to react and create receivers for the
new streams.
R=mflodman@webrtc.org
BUG=3228
Review URL: https://webrtc-codereview.appspot.com/12289005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6150 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 13:57:12 +00:00
a36ad6929d
Add webrtc field trials API.
...
From now on it is expected that code linking system_wrappers.gyp:system_wrappers
provides an implementation for field_trial API or links with the default one in
system_wrappers.gyp:field_trial_default.
Note: Since there is no use of webrtc::field_trial API inside webrtc this CL on
itself does not forces the clients to actually define it. It however lays the
API and updates the gyp rules to link with so that it is ready to use.
Tested: Introduced a use of field trial in system wrappers and make sure all
bots were building successfully.
BUG=crbug/367114
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6147 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 12:24:04 +00:00
023b101f4e
Move gflags usage to video_loopback.
...
gflags aren't used by the test environment and is an unnecessary
dependency. They're only used by the video_loopback target, so moving
them there.
R=mflodman@webrtc.org
BUG=3113
Review URL: https://webrtc-codereview.appspot.com/12379006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6120 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 11:26:40 +00:00
f2aafe4355
Added include of assert.h for files calling assert but missing the include.
...
BUG=N/A
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19409005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6022 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 17:54:17 +00:00
de1429e9ad
Add thread annotations to Call API.
...
Also constified a lot of pointers and reordered members to make
protected members more grouped together.
R=kjellander@webrtc.org , stefan@webrtc.org
BUG=2770
Review URL: https://webrtc-codereview.appspot.com/15399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5998 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 13:00:21 +00:00
8f69330310
Replace scoped_array<T> with scoped_ptr<T[]>.
...
scoped_array is deprecated. This was done using a Chromium clang tool:
http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar ...
except for the few not-built-on-Linux files which were updated manually.
TESTED=trybots
BUG=2515
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 23:10:28 +00:00
cd70119a10
Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase.
...
BUG=3111
TEST=new performance tests
R=niklas.enbom@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5976 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 22:10:24 +00:00
7de47bce12
Remove use of tmpnam.
...
This solves compilation with the Mac SDK 10.9.
BUG=3120, 3151
TEST=git try -t modules_tests:VideoProcessorIntegrationTest*
R=fischman@webrtc.org , henrike@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10739005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5917 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 08:04:26 +00:00
2c89b5cb27
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
...
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done
(and then removed the talk/ impact)
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 20:08:03 +00:00
b1f5010075
VoE changes to allow forwarding of packets from VoE to ViE BWE.
...
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5757 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-24 10:38:25 +00:00
f577ae9eac
Remove internal codecs from VideoSendStream.
...
Replaces VideoCodec in VideoSendStream::Config with an EncoderSettings
struct. The EncoderSettings struct uses an external encoder for all
codecs. This means that external users, such as libjingle, will provide
the encoders themselves, removing the previous distinction of internal
and external codecs.
For now VideoSendStream translates to VideoCodec internally. In the
interrim (before the corresponding change is implemented in
VideoReceiveStream) tests convert EncoderSettings to VideoCodecs.
Removes Call::GetVideoCodecs().
Disables RampUpTest.WithPacingAndRtx as its further exposed with changes
to bitrates used in tests.
BUG=2854,2992
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5722 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19 08:43:57 +00:00
3349ae0cdc
Implement minimum transmit bitrate.
...
Utilizing minimum transmission bitrate prevents low remote bitrate
estimates (bitrate estimation dips) when encoding non-complex content
such as screenshare of a static image even though there's nothing wrong
with the link.
Requires pacing to be enabled for now, pending issue 3036.
BUG=3014
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5694 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13 12:52:27 +00:00
95153cc4cd
Remove platform-specific code from new-API tests.
...
We've had problems that seem to manifest in run_tests.mm getting stuck
on exit. For our automated test targets only full_stack.cc was making
use of the platform-specific renderers provided by webrtc_test_common
and since no one currently monitors these the use case is hypothetical.
Readding platform-specific renderers to video_loopback is tracked with
issue 3039, though as far as I'm aware no one's currently using the
video_loopback target.
BUG=2987
R=kjellander@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5686 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-12 13:22:00 +00:00
2bd5944144
Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed.
...
This was disabled in r5598.
BUG=2960
TESTED=test passes locally and runs & passes on git try --bot=linux_baremetal
R=henrike@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5627 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-01 00:07:08 +00:00
c0e9aebe8f
Add SetConfig method to FakeNetworkPipe and to DirectTransport
...
This method allow the user to change the network configuration
during run-time. This is useful when testing how components react
to changing bandwidth.
BUG=2636
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5612 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-26 13:34:52 +00:00
55fcd716f3
Disable libjingle_peerconnection_java_unittest
...
Broken by libjingle roll in r5590.
TBR=henrike@webrtc.org
BUG=2960
TEST=git try --bot=linux_baremetal --revision=5597
Review URL: https://webrtc-codereview.appspot.com/9029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5598 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-23 18:47:27 +00:00
0f2809a5ac
Add RTCP packet class.
...
Adds packet types: sr, rr, bye, fir.
BUG=2450
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5592 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-21 08:14:45 +00:00
a07923339b
Remove external encryption API for VoE.
...
BUG=
R=henrika@webrtc.org , henrikg@webrtc.org , phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5564 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 11:27:22 +00:00
346094cb01
Incorrect overhead calculation when using FEC + RTP extension headers.
...
When frames are fragmented inte multiple RTP packets in order to not
exceed a maximum packet size, the header overhead calculation must
take into account that FEC redundancy packets may use more than the
12 bytes of the basic RTP header. For example, a csrc list or extension
headers may be present.
BUG=2899
R=phoglund@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8769005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5562 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 08:40:33 +00:00
c279a5d72c
Wire up RTX in VideoReceiveStream.
...
Also adds a test to make sure that a retransmitted frame is actually
received and decoded on the remote side. The previous NACK test checked
retransmission, but not that the receiver actually takes care of the
retransmitted packet.
BUG=2399
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5422 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-24 09:30:53 +00:00
79cf3acc79
Removes usage of ListWrapper from several files.
...
BUG=2164
R=andrew@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5373 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-13 15:21:30 +00:00
faada6e604
Integrate fake_network_pipe into direct_transport.
...
TEST=trybots
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5321 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 20:28:25 +00:00
5fe2d65c43
Remove metrics_unittests
...
This target has been merged into video_engine_tests in r5284.
BUG=webrtc:1843
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5319 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 13:27:37 +00:00
a9890800e0
Update talk to 58127566 together with
...
https://webrtc-codereview.appspot.com/5309005/ .
R=mallinath@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5277 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 00:21:03 +00:00
2018269dc3
Revert 5274 "Update talk to 58113193 together with https://webrt ..."
...
> Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/ .
>
> R=mallinath@webrtc.org , niklas.enbom@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5719004
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5275 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 22:54:25 +00:00
a129b6cd13
Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/ .
...
R=mallinath@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 22:40:39 +00:00
724947b8ef
Add SwapFrame() to VideoSendStreamInput.
...
Optionally prevents doing a frame copy when putting frames into a
VideoSendStream. PutFrame() is still there, which copies the frame.
Also removes time_since_capture_ms as a parameter, since
I420VideoFrame::render_time_ms() denotes when the frame was captured.
BUG=2657
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5265 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 16:26:16 +00:00
f9bdbe3619
Roll chromium_revision 232627:238260
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This brings us the updated swarming_client
that has moved out from Chromium into a standalone
project.
Because of this, all .isolate files needed to be
updated as well, similar to the changes in
https://codereview.chromium.org/29993003
TEST=trybots passing
BUG=none
R=andrew@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5260 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 13:37:12 +00:00
8b8819262f
Improve VideoSendStreamTest::MaxPacketSize
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This CL was submitted as issue https://webrtc-codereview.appspot.com/4849004/ , but was reverted because of flakiness. This new issue will correct that.
Patch Set 1 contains the code that was submitted in 4849004.
BUG=2428
R=pbos@webrtc.org , phoglund@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5251 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-10 10:05:17 +00:00
797522f9f2
Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..."
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> Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered.
>
> BUG=2428
> R=pbos@webrtc.org , phoglund@webrtc.org , stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/4849004
It caused a failure in video_engine_tests on the Linux Tsan bot.
TBR=sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5240 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-06 17:42:32 +00:00
7104fc1906
Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered.
...
BUG=2428
R=pbos@webrtc.org , phoglund@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5229 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 16:15:11 +00:00
c49d5b7df8
Move implementation files out of the webrtc/ root.
...
Leaves the root for public headers. Also fixes the issue of requiring
root OWNERS approval for changes in the Call implementation and adding
end-to-end tests.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5049005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5223 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 12:11:47 +00:00
7e9315b42e
Adds support for sending redundant payloads over RTX.
...
TEST=trybots
BUG=1812
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5209 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04 10:24:26 +00:00
