Commit Graph

5429 Commits

Author SHA1 Message Date
asapersson@webrtc.org
8a8c3ef2ae Add ability to configure cpu overuse options via an API.
BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9299006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5736 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-20 13:15:01 +00:00
henrik.lundin@webrtc.org
d66929995f Prevent playout delay wrap-around in VoiceEngine
In the case where a network glitch causes a packet to arrive so late
that the jitter buffer has gone into expand mode, the playout timestamp
could have been increased to a value that is larger than the RTP
timestamp of the late packet when it finally arrives. This causes
the difference to be negative, and would make the value wrap (unsigned).

With this fix, the difference is set to zero when the playout
timestamp is ahead of the incoming RTP timestamp. Further down in the
method, a zero-value will lead to the averaging filter not being updated.

BUG=3080
R=henrika@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5735 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-20 12:04:09 +00:00
henrika@webrtc.org
800b8dbda6 Removes error printout in voe_cmd_test which was caused by attempts to transmit RTCP packets even if a transport object was not registered.
BUG=none
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5734 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-20 08:07:41 +00:00
andresp@webrtc.org
c14807959b Extend perf tests to perform rampup on single stream.
R=kjellander@webrtc.org, stefan@webrtc.org
BUG=3065

Review URL: https://webrtc-codereview.appspot.com/10049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5733 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-20 03:23:55 +00:00
jiayl@webrtc.org
c8ac17ca04 Adjust the captured window rect when the window is maximized.
GetWindowRect includes the window frames for maximized window even they are off screen, causing content outside the window being captured falsely. The fix is to remove the left/right/bottom window frame from the captured rect. Mouse capturing is adjusted accordingly as well.

BUG=3076
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/10149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5732 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-20 00:06:41 +00:00
henrike@webrtc.org
ffe2620c97 (Auto)update libjingle 63352036-> 63363208
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5731 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19 22:20:10 +00:00
stefan@webrtc.org
16395228f5 Properly account for retransmitted packets when not using the pacer.
This regression was introduced in r5728.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5729 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19 19:34:07 +00:00
stefan@webrtc.org
7c6ff2da26 Fixes RTX related bugs.
- An RTX packet with no payload should be dropped prior to parsing RTX header since it doesn't have an RTX header. This can for example happen when sending padding-only packets over the RTX stream.
- The retransmit code path when the pacer is disabled doesn't properly update the abs-send-time and ts-offset header extensions.

TEST=trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5728 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19 18:14:52 +00:00
pbos@webrtc.org
9af85c4ac2 Disabling SendsSetSimulcastSsrcs.
Disabling as bots are turning red. This should be because
VideoSendStream::ReconfigureVideoCodec caps video_codec.startBitrate to
max bitrates and as the start bitrate is just enough to transmit there
might be some rounding errors here causing the top stream not to be
sent. Since no REMB is received (send-side test) this remains as the
transmit bitrate.

I need some more time to figure out if this is the case so I'm disabling
these for now to avoid reverting the big CL. VideoSendStreams aren't
used in production yet.

TBR=mflodman@webrtc.org
BUG=3078

Review URL: https://webrtc-codereview.appspot.com/10229005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5727 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19 15:49:18 +00:00
henrik.lundin@webrtc.org
1e98a15adb Revert "Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets"
Build bots turned red.

TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19 14:16:52 +00:00
henrik.lundin@webrtc.org
e5be877476 Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets
BUG=2935
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5725 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19 13:36:58 +00:00
pbos@webrtc.org
add4073593 Disable flaky CanSwitchToUseAllSsrcs.
Test flakes on bots, disabling while investigating.

R=minyue@webrtc.org
TBR=mflodman@webrtc.org
BUG=3078

Review URL: https://webrtc-codereview.appspot.com/10119006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5724 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19 12:57:35 +00:00
pbos@webrtc.org
709e29742e Simplify pacer interface.
New interface uses two bitrates (max/min). The pace multiplier is also
removed from the interface and instead utilized outside. Min bitrate
will be filled with padding if there's not enough media to transmit.

Also fixes a bug in minimum transmission bitrate that made it ignore
REMBs. A regression test has been added to catch it.

BUG=3014
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5723 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19 10:59:52 +00:00
pbos@webrtc.org
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
wuchengli@chromium.org
ac4b87c258 Fix a deadlock in ViEEncoder::DeliverFrame.
The deadlock can happen when using HW encoder. HW encoder calls
the encode complete callback on libjingle worker thread instead
of ViECaptureThread. The capture thread can hold VieEncoder::|data_cs_|
and wait for ModuleRtpRtcpImpl::|critical_section_module_ptrs_|.
When libjingle worker thread runs encode complete callback, it
can hold ModuleRtpRtcpImpl::|critical_section_module_ptrs_| and
wait for VieEncoder::|data_cs_|.

|default_rtp_rtcp_| is not guarded by |data_cs|. So move it out of
the critical section to avoid the deadlock.

BUG=chromium:352567
TEST=Run apprtc loopback on CrOS.
Run apprtc between CrOS and Linux.
Run vie_auto_test.

R=henrik.lundin@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5721 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19 03:44:20 +00:00
henrike@webrtc.org
8b61011b6f (Auto)update libjingle 63293120-> 63352036
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5720 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-18 21:39:10 +00:00
kjellander@webrtc.org
08e2dd80a3 Exclude WebRtcVideoMediaChannelTest.AddRemoveCapturerMultipleSources for Valgrind on Mac
This test seems to crash the bot running memcheck on Mac 10.6 at:
http://build.chromium.org/p/client.webrtc.fyi/builders/Mac%2010.6%20Memcheck
over and over again. I won't file a bug since it's unlikely this
will get fixed as long as Valgrind is not supported on newer OS X
releases.
I mainly wanted to make this bot work better so we can one day start
fixing the leaks it's reporting. One step at the time.

TBR=phoglund
BUG=

Review URL: https://webrtc-codereview.appspot.com/10159005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5719 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-18 21:37:28 +00:00
jiayl@webrtc.org
886c94f07c Adds a method to WindowCapturer to bring a window to the front.
BUG=3071
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/10079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5718 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-18 17:10:36 +00:00
henrike@webrtc.org
e9793ab8b8 (Auto)update libjingle 63111035-> 63293120
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5717 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-18 14:36:23 +00:00
henrik.lundin@webrtc.org
dcc301be07 Adding thread annotations to NetEq4
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5716 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-18 11:49:22 +00:00
pbos@webrtc.org
3f655aa5f7 Add #include <cstdlib> for std::abs.
IWYU violation. Fixes a breakage in the libc++ build of Chromium.

BUG=
R=earthdok@chromium.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5715 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-18 11:10:11 +00:00
henrika@webrtc.org
944cbeb292 Resolves TSan v2 warnings in voe_auto_test.
See bug report for details.

BUG=1590
R=tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5714 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-18 10:32:33 +00:00
henrik.lundin@webrtc.org
ed8b281265 Re-comitting r5711: "Fixing a flaky test in video_engine_tests"
The CL was reverted in r5712, due to bots going red. However, these bots
are unrelated to this CL.

Original description:
VideoSendStreamTest.SuspendBelowMinBitrate was flaky. The problem was
that when the first non-padding packet was sent after the stream was
resumed, the statistics had not always been updated so that
stats.suspended was false. After seeing the first non-padding packet
after suspension, the test will now go into a state where it waits for
the statistics to be changed.

BUG=3068
R=pbos@webrtc.org
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5713 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-18 08:43:29 +00:00
turaj@webrtc.org
12499ff20b Revert 5711 "Fixing a flaky test in video_engine_tests"
> Fixing a flaky test in video_engine_tests
> 
> VideoSendStreamTest.SuspendBelowMinBitrate was flaky. The problem was that when the first non-padding packet was sent after the stream was resumed, the statistics had not always been updated so that stats.suspended was false. After seeing the first non-padding packet after suspension, the test will now go into a state where it waits for the statistics to be changed.
> 
> BUG=3068
> R=pbos@webrtc.org
> TBR=stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/10069004

TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10089005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5712 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-18 00:23:55 +00:00
henrik.lundin@webrtc.org
d0f0c76cd9 Fixing a flaky test in video_engine_tests
VideoSendStreamTest.SuspendBelowMinBitrate was flaky. The problem was that when the first non-padding packet was sent after the stream was resumed, the statistics had not always been updated so that stats.suspended was false. After seeing the first non-padding packet after suspension, the test will now go into a state where it waits for the statistics to be changed.

BUG=3068
R=pbos@webrtc.org
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5711 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-17 21:09:10 +00:00
andresp@webrtc.org
4e69f782b0 Small refactor on send_side_bandwidth_estimation.
R=stefan@webrtc.org
BUG=3065

Review URL: https://webrtc-codereview.appspot.com/10029005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5710 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-17 17:07:48 +00:00
fischman@webrtc.org
ccb33a67b9 turn-prober: enable running headlessly and only emit output on error.
With these changes I have the script running in a 10m cronjob on my desktop and
emailing me on failure.  (extremely poor man's monitoring; still, baby steps)

BUG=2187
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/9659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5709 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-17 16:27:41 +00:00
andresp@webrtc.org
a714eafb83 Refactor rampup tests:
- Cleanup test done condition (should be the same but with less code).
 - Split up functions blocks inside methods that were large.

R=stefan@webrtc.org
BUG=3065

Review URL: https://webrtc-codereview.appspot.com/10029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5708 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-17 15:34:57 +00:00
andresp@webrtc.org
44eb87e6dd Tool to establish a loopback call via apprtc turn server.
For now the test keeps track of video bandwidth estimation and plots it
using google visualization libraries after the test is concluded.
There is also scripts to run a test and record the tcpdump.

BUG=3037
R=hta@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5707 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-17 14:23:22 +00:00
henrike@webrtc.org
26caf0eb53 Suppresses/disables tsan/memcheck issues due to sync of 63111035.
BUG=3063
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5706 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-14 22:47:07 +00:00
henrike@webrtc.org
18e5911d92 (Auto)update libjingle 63089643-> 63111035
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5705 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-14 17:19:38 +00:00
sprang@webrtc.org
cf6f46d716 References to includes in third_party should be relative, not absolute.
BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5704 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-14 16:22:18 +00:00
mcasas@webrtc.org
4375e1acbb Add support for YUV4MPEG file reading to tools files. (Minor fix).
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5703 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-14 12:51:02 +00:00
mcasas@webrtc.org
6e2d012b58 Add support for YUV4MPEG file reading to tools files.
This CL adds support for reading .y4m files to the infra in
video_quality_analysis.cc, adding new functions 
ExtractFrameFromYuvFile() and ExtractFrameFromY4mFile(),
instad of the previous ExtractFrameFromI420(). The decision
as to which one to use is taken from the file extension,
if it is .y4m then is considered a YUV4MPEG file, otherwise
is taken as a raw .yuv file.

It also removes the pseudo duplicated function 
GetNextI420Frame(), that is used from psnr_ssim_analyzer.c,
and adds support for y4m files there.

Tested/validated via local compile-run.

YUV4MPEG is a trivial container with a file header
and a per-frame header, see [1]

[1]
http://wiki.multimedia.cx/index.php?title=YUV4MPEG2

BUG=https://code.google.com/p/chromium/issues/detail?id=343504



git-svn-id: http://webrtc.googlecode.com/svn/trunk@5702 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-14 12:45:45 +00:00
henrik.lundin@webrtc.org
24779fe7cc Fix a bug where network freeze during CNG causes delay
Wrote a new NetEq unit test to test a network freeze during comfort
noise playout. The network freezes and resumes during the silence
period, and then resumes speech. It was verified that the delay
increased due to the freeze, and this CL contains a fix for that
problem.

BUG=2995
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5701 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-14 12:40:05 +00:00
henrik.lundin@webrtc.org
367000fa8d Remove legacy weirdness in Merge::Downsample
In practice, this will have only marginal effect. The length_limit
was increased from 6.7 ms to 10 ms. This is compared with the
input_length, which is equal to the decoded frame size. Thus,
this change will only affect encoded frame sizes in this range
(including 10 ms).

BUG=2696
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5700 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-14 12:28:39 +00:00
henrike@webrtc.org
f45a55083f (Auto)update libjingle 63019975-> 63089643
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5699 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13 18:51:34 +00:00
henrik.lundin@webrtc.org
54464e6f49 Stopping network threads before tearing down test
Also initializing suspended_in_stats_ to false.

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9959005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5698 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13 15:39:27 +00:00
sprang@webrtc.org
5a320fb06f Race condition in RTPSender
RTPSender::sending_media_ should be guarded by send_critsect_. Fix this
in GetSendSideDelay, SendPadData and TimeToSendPadding.
Also add appropriate thread annotations.

BUG=3029
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5697 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13 15:12:37 +00:00
stefan@webrtc.org
41689018a0 Add max delay to trace based filters and enhances drop tail queues with delay statistics.
R=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5696 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13 15:01:32 +00:00
henrik.lundin@webrtc.org
b10363f3b6 Re-landing "Routing SuspendChange to VideoSendStream::Stats"
This was originally committed as r5687, but reverted due to a flaky
test.

BUG=3040
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5695 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13 13:31:21 +00:00
pbos@webrtc.org
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
henrik.lundin@webrtc.org
6ea4f6397e Enable all RampUpTest.UpDownUp* tests
With issue 2987 fixed, all these tests can be enabled without problems.

BUG=3010
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5693 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13 09:21:26 +00:00
pbos@webrtc.org
b5f3029302 Replace labs with std::abs.
Resolves clang 3.5 warnings on OS X for -Wabsolute-value.

BUG=chromium:351479
R=andrew@webrtc.org, thakis@chromium.org

Review URL: https://webrtc-codereview.appspot.com/9869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5692 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13 08:53:39 +00:00
mallinath@webrtc.org
827faae0ec Fixing incorrect memset.
Found when ENABLE_EXTERNAL_AUTH is enabled in chrome.

TBR=ronghuawu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5691 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13 02:15:12 +00:00
andrew@webrtc.org
dd5d804efb Disable all protobuf dependent targets when enable_protobuf=0.
BUG=3045
TESTED=builds now when enable_protobuf=0 and modules_unittests still
includes ApmTest.* when enable_protobuf=1.

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5690 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13 00:57:52 +00:00
henrike@webrtc.org
c7bec8484b (Auto)update libjingle 62948689-> 63019975
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5689 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-12 19:53:43 +00:00
henrike@webrtc.org
9269ba14f0 (Git)ignore all of /net. Works around issue: gclient sync, git clean -df, gclient runhooks -> failure (regression in r4466).
BUG=N/A
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5688 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-12 15:50:36 +00:00
kjellander@webrtc.org
c2313fb73c Enable VS2013 for Windows compilation by default.
BUG=chromium:340973
TEST=All trybots passing runhooks and compile step (needs clobber flag -c to ensure old compile output doesn't cause it to fail). I also ran all the tests for the Windows trybots, which passed.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5687 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-12 14:41:58 +00:00
pbos@webrtc.org
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