sergeyu@chromium.org
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
turaj@webrtc.org
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
pbos@webrtc.org
9c5fb76662
Exclude AV-sync test from Valgrind platforms.
...
Test is performance-dependent and was observed to never sync on the
linux_memcheck bot.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5131 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18 16:22:50 +00:00
henrik.lundin@webrtc.org
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
stefan@webrtc.org
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
stefan@webrtc.org
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
stefan@webrtc.org
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
stefan@webrtc.org
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
henrik.lundin@webrtc.org
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
pbos@webrtc.org
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
sergeyu@chromium.org
183c727bca
Disable datachannel_unittest.cc
...
the test fails to compile because it uses incorrect gmock path (as
some other tests).
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5121 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 22:59:20 +00:00
sergeyu@chromium.org
a23f0ca4ba
Update talk to 56619788
...
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3839005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5120 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 22:48:52 +00:00
kjellander@webrtc.org
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
kjellander@webrtc.org
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
stefan@webrtc.org
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
turaj@webrtc.org
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
henrik.lundin@webrtc.org
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
asapersson@webrtc.org
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
minyue@webrtc.org
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
andrew@webrtc.org
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
andrew@webrtc.org
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
pbos@webrtc.org
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
andrew@webrtc.org
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
marpan@webrtc.org
00ed170795
Roll libvpx 225010:232686.
...
R=andrew@webrtc.org
TBR=ajm@google.com
Review URL: https://webrtc-codereview.appspot.com/3649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5105 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-08 18:37:18 +00:00
andrew@webrtc.org
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
stefan@webrtc.org
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
henrikg@webrtc.org
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
kjellander@webrtc.org
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
andrew@webrtc.org
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
elham@webrtc.org
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
fbarchard@google.com
a7855a88b3
Fix for xgetbv on Visual Studio 2010.
...
BUG=none
TEST=local build of webrtc with 2010. python build\gyp_chromium --depth=. -G msvs_version=2010 -fninja all.gyp & ninja -C out\Debug
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5092 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-06 21:18:48 +00:00
marpan@webrtc.org
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
kjellander@webrtc.org
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
asapersson@webrtc.org
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
wu@webrtc.org
16d6254e8c
Update talk to 56183333.
...
TEST=try bots
R=sheu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/3469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5087 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-05 23:45:14 +00:00
kjellander@webrtc.org
d16d307218
Fix bad Google Storage uploads of resource files.
...
The files in this CL seem to have hit some kind of bug
during upload, causing the downloaded files to get another
SHA-1 hash than the .sha1 file. This makes them become
redownloaded every time runhooks execute.
Re-uploading them one by one seems to have resolved this.
TEST=trybots passing
BUG=2294
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5086 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-05 21:03:04 +00:00
kjellander@webrtc.org
0e03360591
Add OWNERS for resources/
...
Make it possible for all our committers to
upload resource .sha1 files in here.
TEST=none
BUG=2294
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5085 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-05 21:02:49 +00:00
kjellander@webrtc.org
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
kjellander@webrtc.org
1e8b671d05
Roll chromium_revision 231713:232627
...
This will pick up r232015 which fixes an error during
GYP execution when GYP_DEFINES is not set.
TEST=trybots passing compile step.
BUG=none
R=henrika@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5083 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-05 12:32:05 +00:00
kjellander@webrtc.org
da7f6589aa
Add svn:ignore to avoid re-download of resources
...
Without this, the bots will download all resources for
every build. This consumes a lot of unnecessary traffic.
I tried experimenting with patterns ignoring everything
except the .sha1 files but wasn't able to get it working,
so this will have to do for now.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5082 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-05 09:27:51 +00:00
fischman@webrtc.org
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
fischman@webrtc.org
7b273a545d
PeerConnection iOS: update README instructions
...
This is needed to account for https://codereview.chromium.org/25535004/
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5079 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04 18:48:12 +00:00
wu@webrtc.org
07a6fbe83d
Update talk to 56092586.
...
R=jiayl@webrtc.org , mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5078 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04 18:41:34 +00:00
kjellander@webrtc.org
3779c1cb0a
Fix invalid .sha1 files for audio_coding
...
It seems like multiple runs of the upload_to_google_storage.py
script created .sha1.sha1 files that sneaked in with
https://code.google.com/p/webrtc/source/detail?r=5076
This caused the wrong files getting downloaded during sync.
This affected the modules_unittests and the neteq_unittests
which started failing (due to wrong version of the resource files).
TEST=trybots passing
BUG=2294
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5077 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04 14:54:47 +00:00
kjellander@webrtc.org
80174583bd
Replace old resources download script with depot_tools
...
With help from hinoka@, we're now using a more efficient approach
to download only the files that have changed from Google Storge.
When uploading new resource files, use
upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename
which of course requires gsutil authentication setup.
NOTICE: Before deploying this, svn:ignore should be removed for
the resources folder, or the bots will run into problems with a
non-versioned file being found in the checkout during sync (as
this CL adds resources to version control).
All developers will also need to be informed to wipe their local
resources dir to avoid getting an error during checkout due to the
already existing non-versioned resources directory.
BUG=2294
TEST=locally running gclient runhooks
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2095004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04 12:07:57 +00:00
kjellander@webrtc.org
a452fc29e6
Remove resources/ svn:ignore to prepare for updated resource handling
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5075 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04 12:07:09 +00:00
kjellander@webrtc.org
58bcdeee2c
Roll chromium_revision 229708:231713
...
Recent changes in how the build dir is used for bots
(see https://codereview.chromium.org/38873003 for details)
requires us to roll to a more recent version
of Chromium to get our android_apk trybot back into
a working state.
This roll needs to be landed at the same time as the
client.webrtc and tryserver.webrtc masters are updated
with the changes in https://codereview.chromium.org/53283002
TEST=trybots passing (except the iOS ones since they require
the above change to be applied to be able to compile)
BUG=2560
R=fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3159004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5074 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04 09:40:03 +00:00
asapersson@webrtc.org
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
kjellander@webrtc.org
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
sheu@chromium.org
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