dd8f6f3d48
Rename rtpDumpPktHdr_t to RtpDumpPacketHeader.
...
_t names are reserved in POSIX.
BUG=162
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7945 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 09:18:42 +00:00
0b1534c52e
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
...
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.
This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".
BUG=chromium:81439
TEST=none
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 22:09:40 +00:00
273a414b0e
Report encoded frame size in VideoSendStream.
...
Implements reporting transmitted frame size in WebRtcVideoEngine2.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=4033
Review URL: https://webrtc-codereview.appspot.com/33399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 15:23:21 +00:00
4591fbd09f
Use size_t more consistently for packet/payload lengths.
...
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
cc774a69cb
Mark all virtual overrides in the hierarchies of RtpDump and
...
VCMPacketizationCallback as such.
This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.
This also marks all other such overrides in the affected files.
BUG=none
TEST=none
R=henrike@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7161 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 22:45:54 +00:00
47658f1269
Mark all virtual overrides in the hierarchy of AudioPacketizationCallback,
...
RTPStream, and NetEq as such. Also mark all other virtual overrides in the same
files.
This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.
This also deletes ACMTest.cc, which existed solely to define ~ACMTest(), which
was marked pure virtual in the header. (Pure virtual destructors still need a
definition.) Because there is another pure virtual method in this class, the
class is already abstract, so there's no benefit to making the desturctor pure.
Making it non-pure allows removing the separate source file.
BUG=none
TEST=none
R=henrik.lundin@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7144 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 22:14:59 +00:00
047a46f8b4
Remove Android.mk build files.
...
These files are generally not maintained and break, some contain files
that don't exist anymore and do not build anymore. If we need to add
some of these back we should really set up a bot for them.
R=andrew@webrtc.org , glaznev@webrtc.org , henrike@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/15249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6974 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 08:48:51 +00:00
62bafae661
Some refactoring inside rtp_rtcp/.
...
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.
BUG=
R=stefan@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 12:10:51 +00:00
637c55f45b
Add support of texture frames for video capturer.
...
This is a reland of r6252. The video_capture_tests failure on
builder Android Chromium-APK Tests should be flaky.
- Add ViECapturer unittest.
- Add CloneFrame function in I420VideoFrame.
- Encoders do not support texture yet and texture frames
are dropped in ViEEncoder for now.
Corresponding CLs:
https://codereview.chromium.org/277943002
http://cl/66620352
BUG=chromium:362437
TEST=WebRTC video stream forwarding, video_engine_core_unittests,
common_video_unittests and video_capture_tests_apk.
TBR=fischman@webrtc.org , perkj@webrtc.org , stefan@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6258 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 07:00:51 +00:00
89e8ffb395
Revert "Add support of texture frames for video capturer."
...
This reverts commit 83c89cd003be75d7d06ef9a2b139588f08d280ca.
Reason: The Buildbot has detected a new failure on builder
Android Chromium-APK Tests.
BUG=chromium:362437
TBR=fischman@webrtc.org , perkj@webrtc.org , stefan@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6253 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-27 14:12:58 +00:00
efe15355ee
Add support of texture frames for video capturer.
...
- Add ViECapturer unittest.
- Add CloneFrame function in I420VideoFrame.
- Encoders do not support texture yet and texture frames
are dropped in ViEEncoder for now.
Corresponding CLs:
https://codereview.chromium.org/277943002
http://cl/66620352
BUG=chromium:362437
TEST=WebRTC video stream forwarding. Run video_engine_core_unittests and common_video_unittests.
R=fischman@webrtc.org , perkj@webrtc.org , stefan@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6252 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-27 12:40:27 +00:00
74767401f2
Fix a bug preventing FilePlayer from playing encoded wav files
...
A bug in ACM2 prevented decoding and playout of wav files where the
audio data was encoded (i.e., not just linear PCM 16 bit data).
This CL fixes the issue, and adds a unit test for the FilePlayer.
BUG=3386
R=henrike@webrtc.org , tina.legrand@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21499005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6248 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-26 13:37:45 +00:00
21299d4e00
Remove the use of AudioFrame::energy_ from AudioProcessing and VoE.
...
We want to remove energy_ entirely as we've seen that carrying around
this potentially invalid value is dangerous.
Results in the removal of AudioBuffer::is_muted(). This wasn't used in
practice any longer, after the level calculation moved directly to
channel.cc
Instead, now use ProcessMuted() in channel.cc, to shortcut the level
computation when the signal is muted.
BUG=3315
TESTED=Muting the channel in voe_cmd_test results in rms=127.
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6159 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 19:00:59 +00:00
ceffdbc371
Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord()
...
R=henrike@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6018 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 17:11:07 +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
34c5da6b5e
Cleaned up logging in video_coding.
...
Converted all calls to WEBRTC_TRACE to LOG(). Also removed a large number of less useful logs.
BUG=3153
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5887 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-11 14:08:35 +00:00
8b2ec15d1e
Convert WEBRTC_TRACE to LOG in utility.
...
BUG=3153
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5886 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-11 07:59:43 +00:00
4f0801bd39
AviRecorder is missing a critical section.
...
BUG=2885
TEST=AUTOTEST
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5600 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-24 09:19:36 +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
de7c9e8884
Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0.
...
Move the logic to common.gypi to reduce the chance of the define being
unprotected in the future.
BUG=b/12018143
TESTED=git try, and local Linux build with -Denable_video=0
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5244 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-09 16:23:00 +00:00
621df678c8
WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN.
...
Mostly to remove a long-standing TODO...
TESTED=trybots
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5013 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-22 10:27:23 +00:00
4e65e07e41
VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android.
...
Besides being ~40% the size of the previous implementation, this makes it so
that VideoCaptureAndroid can stop and restart capture, which is necessary to
support onPause/onResume reasonably on Android.
BUG=1407
R=henrike@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2334004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4915 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03 18:23:13 +00:00
e546f02c84
Remove include_dirs from utility.
...
BUG=1662
TEST=compile on trybots
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4890 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-01 09:29:09 +00:00
eb524d997b
Remove deprecated AudioCodingModule::Destroy.
...
Have Channel hold a pointer rather than reference, and shorten the name.
TESTED=trybots
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2256004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4820 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-23 23:02:24 +00:00
8fa436bd65
Remove use of vcm->ResetDecoder from modules/utility.
...
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2203006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4750 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-16 11:26:35 +00:00
eda189be14
Remove redundant STR_CASE_CMP macro definitions.
...
R=minyue@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2187005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4711 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 17:50:10 +00:00
f1e807c0e5
Removing FrameForStorage
...
R=pwestin@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2142004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4688 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 22:34:41 +00:00
9dba525627
* Update libjingle to 50389769.
...
* Together with "Add texture support for i420 video frame." from
wuchengli@chromium.org .
https://webrtc-codereview.appspot.com/1413004
RISK=P1
TESTED=try bots
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1967004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4489 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 20:36:57 +00:00
f696f253b2
Invert dependency between webrtc_utility and media_file targets to reflect reality.
...
BUG=2166
R=henrike@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1953004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4488 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 18:45:19 +00:00
12dc1a38ca
Switch C++-style C headers with their C equivalents.
...
The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.
BUG=1833
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1917004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 16:22:53 +00:00
8b06200802
Include files from webrtc/.. paths in utility/.
...
BUG=1662
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1786004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4336 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 08:28:10 +00:00
d900e8bea8
Proper spacing for end-of-namespace comments.
...
BUG=
R=mflodman@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1760006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-03 15:12:26 +00:00
fec34d7afa
Merge webrtc_utility_unittests into modules_unittests.
...
This CL eliminates the webrtc_utility_unittests test target.
NOTICE: Upon committing, this test must be removed from the
Buildbot configuration.
BUG=1843
TEST=trybots passing. Compiled and ran modules_unittests, verified the
AudioFrameOperationsTest test executes and passes.
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1584004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4181 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-05 08:58:46 +00:00
342353780d
Consolidate common_audio into a single target.
...
In principle should reduce gyp processing time, but the difference was not measurable. In any case, it's a good simplification that aligns with having a single common_video target.
R=bjornv@webrtc.org , kma@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1375004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3928 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-30 23:43:26 +00:00
6e788df19e
Remove vim/emacs modelines from .gypi files
...
BUG=1655
Review URL: https://webrtc-codereview.appspot.com/1326005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3857 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-16 12:40:34 +00:00
c75102eba7
WebRtc_Word32 -> int32_t in utility/
...
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1307005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3797 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-09 13:32:55 +00:00
4ff956f428
Fixes data race in WebRTCAudioDeviceTest.Construct reported by ThreadSanitizer
...
BUG=159112
Review URL: https://webrtc-codereview.appspot.com/1201007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3750 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-02 11:59:11 +00:00
93bea51517
Removed CPU APIs from VoEHardware. Code is now only used by test applications.
...
Recommitting https://code.google.com/p/webrtc/source/detail?r=3736 after fixing build break.
BUG=8404677
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3739 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-28 15:58:49 +00:00
7a7a008031
Changing non-const reference arguments to pointers, ACM
...
Part of refactoring of ACM, and recent lint-warnings.
This CL changes non-const references in the ACM API to pointers.
BUG=issue1372
Committed: https://code.google.com/p/webrtc/source/detail?r=3543
Review URL: https://webrtc-codereview.appspot.com/1103012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3555 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-21 10:27:48 +00:00
eb7ebf20ed
Revert 3543
...
> Changing non-const reference arguments to pointers, ACM
>
> Part of refactoring of ACM, and recent lint-warnings.
> This CL changes non-const references in the ACM API to pointers.
>
> BUG=issue1372
>
> Review URL: https://webrtc-codereview.appspot.com/1103012
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1116004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3544 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-20 15:57:31 +00:00
374aa49e1a
Changing non-const reference arguments to pointers, ACM
...
Part of refactoring of ACM, and recent lint-warnings.
This CL changes non-const references in the ACM API to pointers.
BUG=issue1372
Review URL: https://webrtc-codereview.appspot.com/1103012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3543 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-20 15:22:23 +00:00
ae1a58bba4
Replace AudioFrame's operator= with CopyFrom().
...
Enforce DISALLOW_COPY_AND_ASSIGN to catch offenders.
Review URL: https://webrtc-codereview.appspot.com/1031007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3395 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-22 04:44:30 +00:00
a3c82bf667
Remove '<(library)' in gyp files.
...
This will remove all usage of '<(library)' in all webrtc gyp files.
Review URL: https://webrtc-codereview.appspot.com/1049005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3392 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-18 23:42:21 +00:00
ad0f3baf90
Removing redundant codec unittest targets.
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The following targets have been merged into audio_coding_unittests:
* cng_unittests
* g711_unittests
* g722_unittests
* isacfix_unittests
* pcm16b_unittests
Some of them were empty and were created with the assumption they were
needed in order to get code coverage (which was actually not needed).
The following test has been removed since it was empty:
* audio_conference_mixer_unittests
BUG=none
TEST=trybots passing (well, except for the unittests that are removed, they're failing until the try server configuration is updated)
Review URL: https://webrtc-codereview.appspot.com/971008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3222 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03 10:52:29 +00:00
418443c531
Remove operator overloading from RTPFragmentationHeader.
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Instead supply a CopyFrom() method.
TEST=vie_auto_test
Review URL: https://webrtc-codereview.appspot.com/972004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3158 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-23 19:17:23 +00:00
f3adba499e
Add Android include path so that header files can follow google style
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BUG=1011
TEST=bot
Review URL: https://webrtc-codereview.appspot.com/930018
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3107 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-15 18:17:40 +00:00
c381a8487a
Fix valgrind issue.
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Code was removed by mistake in r2983.
BUG=1020
Review URL: https://webrtc-codereview.appspot.com/938006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3021 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-30 18:22:02 +00:00
a5d4c31735
Fixing IsZeroSize call
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Review URL: https://webrtc-codereview.appspot.com/940004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3001 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-25 21:31:31 +00:00
9fedff7c17
Switching to I420VideoFrame
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Review URL: https://webrtc-codereview.appspot.com/922004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2983 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-24 18:33:04 +00:00
14b43beb7c
Move src/ -> webrtc/
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TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00
