70f74f3f7b
Add overshoot of target bitrate for screenshare with temporal layers.
...
Set the codec target bitrate higher than TL0 but lower than TL1, making
sure frame rate is not too low (but still lower than TL1) and that
overshooting for complex scenes don't overly exceed TL1 bitrates.
BUG=4083
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7929 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 10:57:10 +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
f4c19480fc
Remove jitter_estimate_test.h
...
BUG=2156
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7866 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 21:08:39 +00:00
f244760827
Add histograms for receive statistics:
...
- decoded frames per second ("WebRTC.Video.DecodedFramesPerSecond")
- percentage of delayed frames to rendered ("WebRTC.Video.DelayedFramesToRenderer")
- average delay (of delayed frames) to renderer ("WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs")
BUG=crbug/419657
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7847 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 14:13:26 +00:00
9115cde6c9
Merge VP8 changes.
...
R=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/35389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7841 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:36:40 +00:00
86b6d65ef1
Remove no longer used video codec test framework.
...
Moves one test to the vp8 unittests which might still be good to have.
Also does a bit of clean up in vp8 unittests.
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7835 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 00:02:45 +00:00
fb01376eca
Adjust some parameters for VP9 tests.
...
Needed for the next/upcoming libvpx roll.
BUG=
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7821 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 06:25:51 +00:00
001f3b9818
Adjust parameter in videoprocessor_integration_test for vp9.
...
TBR=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/33459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7791 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 02:00:12 +00:00
ceca014b8b
Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeBitRateVP9.
...
BUG=4059
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7789 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 01:05:43 +00:00
84515f841d
Roll chromium_revision 309cf65..24b4c73
...
Two VP9 tests needed to be disabled (see webrtc:4059) to make all tests pass.
Relevant changes:
* src/third_party/android_tools: ea50ccc..4c47ef6
* src/third_party/icu: dd72764..866ff69
* src/third_party/libvpx: 2e5ced5..429874c
* src/third_party/nss: 258342e..bb4e75a
* src/third_party/yasm/source/patched-yasm: c960eb1..4671120
* src/tools/gyp: 0a381c0..fe00999
* src/tools/swarming_client: 5b827c9..1d4965c
Details: 309cf65..24b4c73
/DEPS
Clang version was not updated in this roll.
BUG=4059
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7778 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-02 08:48:08 +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
83b5200f95
Add framerate for complete received frames to histogram stats:
...
"WebRTC.Video.CompleteFramesReceivedPerSecond".
BUG=crbug/419657
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7762 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-28 10:17:13 +00:00
91d928e737
Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader
...
This is in preparation for creating a new class RtpFileWriter which
will use the same RtpPacket struct.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7749 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-26 15:50:30 +00:00
4f16c874c6
Simplifying VideoReceiver and JitterBuffer.
...
Removing frame_buffers_ array and dual-receiver mechanism. Also adding
some thread annotations to VCMJitterBuffer.
R=stefan@webrtc.org
BUG=4014
Review URL: https://webrtc-codereview.appspot.com/27239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7735 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-24 09:06:48 +00:00
ad0e71c9a3
Update mock_frame_dropper.h to use size_t
...
This mock was missed in the work of
https://webrtc-codereview.appspot.com/23129004 since the file
is not currently used by any test in this repo.
BUG=chromium:81439
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7727 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-21 09:40:57 +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
e1745cbb7c
Adjust parameter in vp9 rate control test.
...
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7638 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 02:55:53 +00:00
5f1e2e42a8
Increase speed setting for VP9 (from 5 to 6) and re-enable end_to_end test.
...
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7637 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 02:02:28 +00:00
0bae1fab4a
Wire up bandwidth stats to the new API and webrtcvideoengine2.
...
Adds stats to verify bandwidth and pacer stats.
BUG=1788
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 14:05:29 +00:00
e451b756a8
Update rate control parameter in vp9 test.
...
TBR=hellner@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7599 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 21:26:08 +00:00
4765ca55f9
Roll chromium_revision: 28d1981..d3db2ff
...
Pick up the libvpx roll: https://codereview.chromium.org/674753002
Summary of changes (28d1981..d3db2ff
/DEPS):
* third_party/android_tools 36bf7ac..ea50ccc
* third_party/boringssl 7ea8481..751e889
* third_party/icu 8ac906f..d8b2a9d
* third_party/libvpx efe9712..2e5ced5
* third_party/usrsctp/usrsctplib
* tools/gyp 1990:1991
* tools/swarming_client a57d7db..bcb3bc3
Clang is not updated in this roll.
Made the change getchar() --> getc(stdin) as seems like getchar() isn't supported on android anymore.
(getchar() was causing the error: undefined reference to '__srget')
Update rate control parameter in vp9 test.
R=andrew@webrtc.org
TBR=ajm@google.com
Review URL: https://webrtc-codereview.appspot.com/23229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7598 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 20:10:26 +00:00
96dc685143
Add stats for video:
...
- number of sent/received RTCP NACK/FIR/PLI per minute
- percentage of unique sent/received NACK requests
- percentage of discarded/duplicated packets by the jitter buffer
- permille of sent/received key frames
BUG=crbug/419657
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7592 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 14:40:38 +00:00
ed45896759
Adjust/increase rate control thresold for a vp9 test.
...
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7589 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 07:08:52 +00:00
5b88317820
Add VP9 codec to VCM and vie_auto_test.
...
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.
This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422 , which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in:
see https://code.google.com/p/webrtc/issues/detail?id=3932
R=kjellander@webrtc.org , mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
776e6f289c
Use external VideoDecoders in VideoReceiveStream.
...
Removes direct VideoCodec use from the new API, exposes VideoDecoders
through webrtc/video_decoder.h similar to VideoEncoders.
Also includes some preparation for wiring up external decoders in
WebRtcVideoEngine2 by adding AllocatedDecoders that specify whether they
were allocated internally or externally.
Additionally addresses a data race in VideoReceiver that was exposed with this change.
R=mflodman@webrtc.org , stefan@webrtc.org
TBR=pthatcher@webrtc.org
BUG=2854,1667
Review URL: https://webrtc-codereview.appspot.com/27829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7560 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 15:28:39 +00:00
580d367b14
Add macros and APIs for webrtc histograms.
...
BUG=crbug/419657
Code that links system_wrappers.gyp:system_wrappers should either:
- provide implementations for the APIs, or
- link with default implementations in system_wrappers.gyp:system_wrappers_default.
R=andresp@webrtc.org , kjellander@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7508 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 12:57:56 +00:00
82462aade0
Adds support for sending first set of packets at increasingly higher bitrates to probe the link and faster ramp up to a high bitrate.
...
Also wires up a finch experiment to control this.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7505 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 11:57:05 +00:00
3f8f5554a0
Disable TestVp8Impl.BaseUnitTest on MSan.
...
MemorySanitizer uses generic (non-optimized) libvpx which is not bit
exact. This may be a bug in upstream libvpx, but we're forced to disable
it now as it blocks launching the MSan bot.
R=stefan@webrtc.org
TBR=marpan@webrtc.org
BUG=3904
Review URL: https://webrtc-codereview.appspot.com/24089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7491 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 10:30:30 +00:00
b1dac33cac
Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
...
BUG=3932
R=marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/27779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7470 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 18:54:46 +00:00
c87b74717b
Adjust/increase rate control thresold for a vp9 test.
...
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7423 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 17:55:57 +00:00
573c78e31c
Add VP9 codec to VCM and vie_auto_test.
...
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.
Passes trybots.
R=kjellander@webrtc.org , mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7422 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 16:44:47 +00:00
3cefbc99f4
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
...
This also marks all virtual overrides of other classes in the same files.
This will make a subsequent change I intend to do safer, where I'll change the
argument types of the base Transport functions, by breaking the compile if I
miss any overrides.
This also highlighted a number of unused functions. I've removed some of these.
TBR=mflodman@webrtc.org , pkasting@chromium.org
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/28709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 09:42:53 +00:00
b8caf6a504
GN: Enable libvpx, add link target and convert some test targets
...
Libvpx now supports GN and this CL turns on compiling it.
I also introduced an executable target 'webrtc_tests'
that depends on all in WeBRTC + tests in order to get a full
linking step executed (since we've seen link problems for GN
when rolling WebRTC into Chromium).
I also converted a few test targets and made a GN file for
third_party/gflags.
BUG=3441
TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc
dir to a workspace wit this CL applied.
R=brettw@chromium.org
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 18:05:02 +00:00
f21ea918ad
GN: Add common configs to all targets.
...
This is needed to ensure we have the same build with GN
as with GYP, since GYP includes the common.gypi on a global level.
Several fixes has been needed in the past because some code have
been built without the right defines.
BUG=3441
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/28589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7317 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-28 17:37:22 +00:00
70e2d11ea8
Reduce jitter delay for low fps streams.
...
Enabled by finch flag.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31389005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7288 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 14:06:56 +00:00
38344ed280
Move thread_annotations.h to webrtc/base/.
...
R=andresp@webrtc.org , mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 06:05:00 +00:00
94ff92ceec
Use VPX_IMG_FMT_*/VPX_PLANE_* defines
...
The compatibility layer has been removed upstream:
https://gerrit.chromium.org/gerrit/gitweb?p=webm%2Flibvpx.git;a=commit;h=9cdaa3d72eade9ad162ef8f78a93bd8f85c6de10
BUG=webrtc:3839
R=marpan@google.com , marpan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7279 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 17:31:47 +00:00
6cd6ba8ae0
Expose VP8/H264 defaults through video_encoder.h.
...
Reduces code duplication quite a bit, these identical defaults were set
in quite a few different places.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=3070
Review URL: https://webrtc-codereview.appspot.com/19299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7220 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18 12:42:28 +00:00
ab990ae43a
Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""
...
Re-lands r7114 after landing r7204 to adress the compile error causing
the rollback in r7151.
BUG=3070
TBR=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7207 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 09:02:25 +00:00
a0d7827b16
Add ability to downscale content to improve quality.
...
BUG=3712
R=marpan@google.com , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7164 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-12 11:51:47 +00:00
6071b0636d
Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver 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 highlighted a number of unused functions which I've removed.
-- This is was reviewed in https://webrtc-codereview.appspot.com/19309004/ , but
-- a new cl was needed to resolve a small conflict before committing.
BUG=none
TEST=none
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7162 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-12 07:42:33 +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
307d3dbdee
Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."
...
Speculative revert, seems to be reason for flaky Win FYI bot compile break.
> Expose VideoEncoders with webrtc/video_encoder.h.
>
> Exposes VideoEncoders as part of the public API and provides a factory
> method for creating them.
>
> BUG=3070
> R=mflodman@webrtc.org , stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/21929004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7151 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 09:48:30 +00:00
1972ff8a6e
Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.
...
This will make a subsequent change I intend to do safer, where I'll change the
return type of one of the base Module functions, by breaking the compile if I
miss any overrides.
This also highlighted a number of unused functions (in many cases apparently
virtual "overrides" of no-longer-existent base functions). I've removed some of
these.
This also highlighted several cases where "virtual" was used unnecessarily to
mark a function that was only defined in one class. Removed "virtual" in those
cases.
BUG=none
TEST=none
R=andrew@webrtc.org , henrik.lundin@webrtc.org , mallinath@webrtc.org , mflodman@webrtc.org , stefan@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7146 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 06:20:28 +00:00
b420191743
Expose VideoEncoders with webrtc/video_encoder.h.
...
Exposes VideoEncoders as part of the public API and provides a factory
method for creating them.
BUG=3070
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7114 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09 10:40:56 +00:00
6d08ca6379
GN: Prefix WebRTC specific variables with "rtc_"
...
BUG=3441
TESTED=Trybots + Running GN in a Chromium checkout with
src/third_party/webrtc symlinked to the WebRTC checkout
with this CL applied, both with the default GN settings
and using: --args="os=\"android\" cpu_arch=\"arm\""
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/27379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 17:36:10 +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
b96ea2aab5
Remove former team members from OWNERS and WATCHLISTS
...
Remove the following (CCed) former team members from all
OWNERS files and the WATCHLISTS file:
* fischman@
* leozwang@
* mikhal@
* pwestin@
* wu@
BUG=
R=henrike@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6973 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 06:12:08 +00:00
42ee5b54b5
GN: Disable Chromium clang plugins for standalone build.
...
Now that WebRTC has rolled the chromium_revision past
http://crrev.com/284372 in r6784, clang has become the
default compiler. Since WebRTC standalone code doesn't
yet compile the Chromium Clang plugins enabled, this CL
disables them for the parts of the code that doesn't yet pass
compilation with them enabled.
The buildbots are using Goma which is not yet switched
over to Clang by default. That's why they're not red yet.
BUG=163
TEST=Passing compile locally on Linux using:
gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja
-C out/Debug
gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja
-C out/Release
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/16279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 14:15:35 +00:00
a84b0a6dab
Small refactor on ViE to remove redudant conditions and long ifdefs.
...
BUG=3694
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-14 16:46:46 +00:00