5a63655ab0
Rename Call::Create{Receive,Send}Stream().
...
Renaming the methods to include Video. Long-term there will hopefully be
AudioSendStream/AudioReceiveStreams as well.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5135 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 10:40:25 +00:00
0b72f5863b
Add experimental noise suppression dummy API.
...
Add this flag to the voe_cmd_test.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5134 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-19 15:17:51 +00:00
5d85819dd2
Fix DesktopAndCursorComposer to restore frames to the original state.
...
Screen capturers may reuse frame buffers and they expect that the
frame content isn't changed by the frame consumer.
DesktopAndCursorComposer draws mouse cursor on generated frames and
it was releasing the frames with the mouse cursor on them. Fixed
it to restore frame content erasing mouse cursor before returning
desktop frames.
BUG=crbug.com/316297
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/3899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5133 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-19 02:15:47 +00:00
7a05ae5c69
Adding back main() to the test. Now it is possible to choose between ACM1 and ACM2, furthermore, the test can simulate a channel with packet loss and FEC can be activated. Packet loss pattern is based on channel implementation in Channel{.cc,.h}, which currently is a determenistic pattern with 1 every 3rd packet is discarded.
...
The main() was deleted in r4731.
BUG=
R=andrew@webrtc.org , minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2370004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5132 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18 18:16:53 +00:00
ce8e0936d9
Rename AutoMute to SuspendBelowMinBitrate
...
Changes all instances throughout the WebRTC stack.
BUG=2436
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18 12:18:43 +00:00
28bf50f0ec
Fix test broken with r5128.
...
TBR=pbos@webrtc.org
BUG=2530
Review URL: https://webrtc-codereview.appspot.com/3979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5129 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18 11:58:24 +00:00
b082ade3db
Hook up audio/video sync to Call.
...
Adds an end-to-end audio/video sync test.
BUG=2530, 2608
TEST=trybots
R=henrika@webrtc.org , mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5128 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18 11:45:11 +00:00
4cfa6050f6
Fix breakage after introducing new test.
...
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3899005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5127 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-15 13:15:56 +00:00
69969e2e2f
Improve Call tests for RTX.
...
Also does some refactoring to reuse RtpRtcpObserver.
BUG=1811
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5126 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-15 12:32:15 +00:00
6e95d7afab
Increment RTP timestamps for padding packets
...
This CL makes the padding packets get their own RTP timestamps,
rather than having the same timestamp as the last sent video
packet. The purpose is to solve Issue 2611, where the overuse-
detector does not react to padding packets.
A test was implemented to verify that the padding packets do
get their own timestamps.
BUG=2611
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5125 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-15 08:59:19 +00:00
6488761f2e
Implement VideoSendStream::SetCodec().
...
Removing assertion that SSRC count should be the same as the number of
streams in the codec. It makes sense that you don't always use the same
number of streams under one call. Dropping resolution due to CPU overuse
for instance can require less streams, but the SSRCs should stay
allocated so that operations can resume when not overusing any more.
This change also means we can get rid of the ugly SendStreamState whose
content wasn't defined. Instead we use SetCodec to change resolution
etc. on the fly. Should something else have to be replaced on the fly
then that functionality simply has to be implemented.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3499005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5123 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-14 08:58:14 +00:00
e8722856f9
Disable all vie_auto_tests on Linux for now (take 2)
...
Turns out OS_LINUX is not working in this context
(see http://review.webrtc.org/3539005/ )
WEBRTC_LINUX is the right define to use.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5119 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 15:51:49 +00:00
c8489852ec
Disable all automated vie_auto_tests on Linux for now
...
Since the switch from icewm to openbox window manager on
Linux in Chrome infra, causes the test to hang when
creating Windows.
TEST=trybots compile step
BUG=chromium:318760
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3539005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5118 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 15:44:54 +00:00
9b82f5a6ed
Fix for RTX in combination with pacing.
...
Retransmissions didn't get sent over RTX when pacing was enabled since
the pacer didn't keep track of whether a packet was a retransmit or not.
BUG=1811
TEST=trybots
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5117 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 15:29:21 +00:00
03f33709f8
Inject config when creating channels to override the existing one.
...
BUG=
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5116 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 00:02:48 +00:00
e8433eb115
Reimplementing NetEq4's AudioVector
...
The current implementation using std::vector is too slow.
This CL introduces a new implementation, using a regular
array as data container.
In AudioMultiVector::ReadInterleavedFromIndex, a special case for
1 channel was implemented, to further reduce runtime. Finally,
AudioMultiVector::Channels was reimplemented.
The changes in this CL reduces the runtime of neteq4_speed_test
by 33%.
BUG=1363
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5115 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-12 13:15:02 +00:00
38599510df
Parse next RTCP XR report block after an unsupported block type.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5114 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-12 08:08:26 +00:00
3e427263ee
Reducing opus_test runtime to pass Android test
...
BUG=2609
R=solenberg@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5111 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-11 22:03:52 +00:00
e03cafaebc
MIPS optimizations for AECM audio processing module
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2279005
Patch from Ljubomir Papuga <lpapuga@mips.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5110 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-11 20:10:01 +00:00
b0730108a2
Move audio_processing dependencies to a variable.
...
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5108 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-11 17:20:27 +00:00
57eb858698
Remove ".." from include_dirs in build/common.
...
BUG=1662
TEST=compile on trybots
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2332004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5107 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-11 10:20:27 +00:00
6e908b3adf
Remove unnecessary include_dirs from audio_processing.
...
TBR=aluebs
TESTED=trybots
Review URL: https://webrtc-codereview.appspot.com/3659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5106 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-08 19:52:05 +00:00
5973f3a24a
Remove unneeded includes from trace_posix.cc.
...
TESTED=trybots
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5103 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-08 17:30:07 +00:00
48df38114d
Fix for making sure that the packet in order checks are done prior to updating the last received packet state.
...
Without this fix all packets are considered out-of-order by the rtp receiver, causing the last received state
in the rtp receiver to never get valid.
Also makes sure that only valid timestamps and receive times are used for audio/video sync.
BUG=2608
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5102 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-08 15:18:52 +00:00
bff9620116
Fix log build error for Chromium builds.
...
This only happens when building in Chromium. Can't roll due to this.
../../third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc: In function 'Window {anonymous}::GetTopLevelWindow(Display*, Window)':
../../third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc:39:7: error: 'LS_INFO' was not declared in this scope
../../third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc:39:7: note: suggested alternative:
../../third_party/webrtc/system_wrappers/interface/logging.h:71:29: note: 'webrtc::LS_INFO'
See for example http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20%5Blatest%20WebRTC%2Blibjingle%5D/builds/3039/steps/compile/logs/stdio
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5100 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-08 10:37:27 +00:00
4c828e145e
Remove update_resources.py as it's no longer used.
...
After http://review.webrtc.org/2095004/ has been landed
for normal WebRTC builds, and https://codereview.chromium.org/62273004/
and https://codereview.chromium.org/60513012/ for our Android
APK builds with a Chromium checkout, we should be fine to remove
this script.
I have verified that the runhooks step on the Android testers
is using the download_from_google_storage.py script to pull
the resources from Google Storage.
BUG=webrtc:2294
TEST=a few trybots passing compile step.
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5099 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-08 09:08:36 +00:00
f1a48174d4
Replace disabled logging with a restricted logging mode.
...
This will enable some low-level webrtc logging in a Chromium build,
while limiting the binary size impact.
For a Mac Release build, it results in an increase to Chrome.app of 37k
and libpeerconnection.so of 25k. For comparison, enabling full logs
costs 230k and 218k respectively.
BUG=b/11470432
TESTED=voe_cmd_test produces logs of the appropriate severity.
R=fischman@webrtc.org , henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5097 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-07 23:47:26 +00:00
5adc89747a
Updated WebRTC version to 3.46
...
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5093 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-06 22:27:51 +00:00
bde3056567
Fix for video_processor_intergration_tests to run in parallel.
...
BUG=2601.
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5091 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-06 20:59:29 +00:00
c4225b63bb
Update getUserMedia W3C conformance tests.
...
This CL updates these tests to the spec as of
http://dev.w3.org/2011/webrtc/editor/archives/20130824/getusermedia.html
There are still a lot of functionality that lacks testing. I've put a bunch of TODOs in there but I'm unlikely to get time to implement them all any time soon...
TEST=local testing with Chrome Canary.
BUG=
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5090 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-06 13:26:34 +00:00
8bad50e845
Sending status fix for module.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5089 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-06 10:45:58 +00:00
7a36cb408b
Add missing dependencies to .isolate files
...
Also fix invalid paths in video_engine_tests.isolate.
TEST=trybots passing compile step (no .isolate use is deployed on them yet)
BUG=chromium:300017
R=pbos@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3399005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5084 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-05 14:28:57 +00:00
b8cb85b348
Fix broken build on x86 Android
...
BUG=2545
R=fischman@webrtc.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3019004
Patch from Lu Quiang <qiang.lu@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5081 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04 19:06:08 +00:00
766154aa1d
Removed unused code.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5073 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04 08:35:50 +00:00
e2df8b7f01
Make video quality analysis unittests print to log instead of stdout.
...
I think it's best to avoid printing these perf numbers since
when we turn on perf measurements for Android, it will be for
all tests as far as I understand it works today.
TEST=trybots passing tools_unittests
BUG=none
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3109005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5072 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-03 18:34:51 +00:00
5dd2ecb32d
Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
...
This reverts commit f4ca3808bd9ec2293ec205f2f4a7d9739ce1f2df.
TBR=niklas.emblom@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/3269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5071 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 23:41:04 +00:00
74e6e8458e
Remove extra copy in VideoCaptureImpl::IncomingFrameI420
...
Add support for aliasing a I420VideoFrame (and internally, a Plane) to an
existing memory buffer without taking ownership. Use this to remove an extra
copy in VideoCaptureImpl::IncomingFrameI420.
BUG=1128
BUG=chromium:310271
TEST=local build, run Chromium on ARM, build, run Chromium/unittests on Linux
TBR=fischman@webrtc.org , mflodman@webrtc.org , mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3239005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5070 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 21:48:16 +00:00
d705649edf
Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
...
This reverts commit 99f9743fe39066ba93b41f2b0a417696cbbd06fb.
Revert while build breakage is fixed.
BUG=None
TBR=niklas.emblom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5069 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 21:20:15 +00:00
1a4ed0d70c
Remove extra copy in VideoCaptureImpl::IncomingFrameI420
...
Add support for aliasing a I420VideoFrame (and internally, a Plane) to an
existing memory buffer without taking ownership. Use this to remove an extra
copy in VideoCaptureImpl::IncomingFrameI420.
BUG=1128
TEST=local build, run Chromium on ARM, build, run Chromium/unittests on Linux
R=fischman@webrtc.org , mflodman@webrtc.org , mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5068 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 20:32:28 +00:00
58cd31665c
Address Clag Analyzer issues.
...
Following are the issues related to NetEq 4, discovered by Clang Analyzer.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b44b95.html#EndPath
Valid; perhaps unlikely, addressed.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-6beef6.html#EndPath
Valid, addressed.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2e3883.html#EndPath
Valid; Addressed
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-293659.html#EndPath
Valid; Addressed.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b875cd.html#EndPath
Valid; Addressed.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/index.html
Not valid;
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-86f2ed.html#EndPath
Not Valid; the assert statement will be short-circuited, however I also added a check of nullity of |packet|.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-3a5669.html#EndPath
Not Valid: |energy_input| and |energy_expand| are both non-negative, therefore if-statement condition on line 226 is not satisfied unless |energy_input| >= 1. Therefore |energy_input| cannot be zero after normalization to 14-bits, i.e. operations on lines 228 & 229.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2f914f.html#EndPath
Valid; addressed.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2332b1.html#EndPath
Valid; addressed.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-de8dea.html#EndPath
Not valid; |out_len| is set when Process() is called, however, it makes sense to initialize to zero when declaring |out_len|.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b671a3.html#EndPath
Valid; addressed.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2729005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5064 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 15:15:55 +00:00
7d6bd22019
Propagate estimated RTT from receivers to rtt observer.
...
BUG=1613
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5063 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 12:14:34 +00:00
da2c37b759
Video bandwidth not reported correctly
...
ViEChannel::GetBandwidthUsage fails to aggregate video_bitrate_sent in
the same way as the total, fec and nack.
BUG=2579
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5062 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 09:49:03 +00:00
773e72797f
Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.cc
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Chromium issue:
https://code.google.com/p/chromium/issues/detail?id=310146
BUG=2551
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2759004
Patch from Daniel Nicoara <dnicoara@chromium.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5061 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 01:51:21 +00:00
de748c806c
Remove unused make_scoped_ptr which causes an "ambiguous" error with chromium build.
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TEST=build
R=andrew@webrtc.org , fischman@webrtc.org
TBR=andrew
Review URL: https://webrtc-codereview.appspot.com/3149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5059 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-30 20:43:27 +00:00
dce70ccb0b
Add delay limit to ChokeFilter.
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BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3079005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5058 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-30 19:18:07 +00:00
d6e46638ec
Logging for BWE test framework.
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BUG=
R=stefan@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5055 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-30 16:06:26 +00:00
47ebbaddbb
Make video/ only depend on video_engine_core.
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Fixes Android/Chromium build error. Previous dependencies included
VideoEngine tests that couldn't build on this configuration.
BUG=2535
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5050 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-29 13:11:56 +00:00
def22b455b
Stop DirectTransports in VideoSendStreamTests.
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Prevents racy packet delivery during or after Call destruction.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3099005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5049 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-29 10:12:10 +00:00
55e1723713
Avoid a leak in AudioCodingModuleTest.TestIsac. The leak was caught by LSAN.
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BUG=2515
TEST=reproduced locally on linux and verified the fix resolves the issue.
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5048 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-29 04:40:09 +00:00
0aeb22e32c
Adding tl0idx consideration for continuity
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R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5046 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28 22:26:14 +00:00
