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Author SHA1 Message Date
wuchengli@chromium.org
f425b55eeb Add tests of texture frames in video_send_stream_test. 
Also fix a bug in ViEFrameProviderBase::DeliverFrame that
a texture frame was only delivered to the first callback.

BUG=chromium:362437
TEST=Run video engine test and webrtc call on CrOS.
R=kjellander@webrtc.org, pbos@webrtc.org, stefan@webrtc.org, wuchengli@google.com

Review URL: https://webrtc-codereview.appspot.com/15789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6506 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-20 12:04:05 +00:00
jiayl@webrtc.org
594aefa807 Do not call CaptureCursor in ScreenCapturerWinGdi if no MouseShapeObserver. 
It's wasted work and affects frame rate adaptation in Chrome.

BUG=
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/19789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6499 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-19 22:04:41 +00:00
fgalligan@google.com
304ca76be1 Revert 6481 and 6482 
Revert 6482 "Update webrtc to fix unpack_lib expansion."
Revert 6481 "Update generated asm offsets scripts."

The roll has not been successful. Reverted based on the request of the
committer.


TBR=turaj

Review URL: https://webrtc-codereview.appspot.com/17759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6495 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-19 17:08:46 +00:00
turaj@webrtc.org
8de8c9155e Maintain constantness of the input to iSAC-fix decoder, and prevent heap-buffer overflow. 
To save memory in iSAC-fix, decoder operated directly on the recieved bitstream. However, this breaks constantness of input when decoder performed in-place big to little Endian conversion. Furthermore, for bit-streams with odd lengths, this meant writing outside the memory. That is because the last byte will be shifted to the Most Significat Byte which might be outside the allocated memory.

If we care about memory, the solution is to do a big-to-little Endian conversion everytime we read a Word16 from the bitstream.

BUG=845,chrome:379458
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6494 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-19 15:47:09 +00:00
henrik.lundin@webrtc.org
9158df2aa4 Adding an empty constructor implementation to the AudioSink class 
Turns out it was needed.

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6493 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-19 12:34:31 +00:00
bjornv@webrtc.org
84f8ec1f9c Changes to tests and tools in audio_processing. 
- Disables ApmTest.EchoCancellationReportsCorrectDelays
This test relys completely on the structure of how reported system delays are handled in AEC. In addition it assumes a fix setup of delay logging buffers. This test should be refactored.

- Adds flag to turn off reported_delay in audioproc
Now it is feasible to turn on and off the use of reported system delays.

BUG=N/A
R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6492 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-19 12:14:33 +00:00
stefan@webrtc.org
077593b805 Ensure that the start bitrate can be set multiple times. 
If the start bitrate is set twice, it will be set to the sum of the start
bitrates of the currently registered bitrate observers, or left unchanged
if the current estimate actually is greater than the sum.

BUG=3503
R=henrik.lundin@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6491 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-19 12:13:00 +00:00
henrik.lundin@webrtc.org
496a98463b Adding test::AudioSink interface and derived classes 
The AudioSink interface is supposed to be used by tests that produce
audio output. Two implementation classes are also provided:

OutputAudioFile: Writes the audio to a pcm file.
AudioChecksum: Calculates the MD5 checksum of the audio.

These will both be used in future changes.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6490 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-19 10:02:11 +00:00
bjornv@webrtc.org
5c3f4e3b0f Fixes and re-enables tests disabled on Android 
Several tests were disabled in r6325 and r6326. Also, see issue 3445. This CL fixes the remaining four of the audio_processing related ones. Affects the tests:
- SystemDelayTest.CorrectDelayAfterStableBufferBuildUp
- SystemDelayTest.CorrectDelayDuringDrift
- SystemDelayTest.ShouldRecoverAfterGlitch
- ApmTest.EchoCancellationReportsCorrectDelays

The tests assumes reported delays are used, which now is explicitly set.

BUG=3445
TESTED=trybots
R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-19 09:51:29 +00:00
fgalligan@google.com
a19b930b5b Update webrtc to fix unpack_lib expansion. 
Add on fix for:https://webrtc-codereview.appspot.com/12789004/

*NOTE* This CL will break the Android bots as they are built in a
Chromium checkout, which will pull in old libvpx DEPS. They will
cycle to green when we roll libvpx into Chromium. We must do the
rolls in this order because we have to land webrtc and libvpx at
the same time into Chromium.

BUG=377062
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6482 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-18 19:20:45 +00:00
fgalligan@google.com
8f06a8aeb0 Update generated asm offsets scripts. 
Libvpx updated the unpack scripts to fix building dependencies.

Roll libvpx 269083:278063
See https://codereview.chromium.org/295313002/
https://codereview.chromium.org/298063002/
https://codereview.chromium.org/305533008/
https://codereview.chromium.org/305703002/
https://codereview.chromium.org/298383003/
https://codereview.chromium.org/302863004/
https://codereview.chromium.org/320923003/
https://codereview.chromium.org/325313007/
https://codereview.chromium.org/346563002/
for the libvpx changes.

See https://codereview.chromium.org/313243004/
for the WebView changes.

*NOTE* This CL will break the Android bots as they are built in a
Chromium checkout, which will pull in old libvpx DEPS. They will
cycle to green when we roll libvpx into Chromium. We must do the
rolls in this order because we have to land webrtc and libvpx at
the same time into Chromium.

BUG=377062
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6481 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-18 17:38:08 +00:00
bjornv@webrtc.org
b947d954a5 Neon version of FilterAdaptation() 
The performance gain on a Nexus 7 reported by audioproc is ~5.2%.

The output is bit exact.

Measured total of 15% speed gain on N7 compared to C.

R=bjornv@webrtc.org, cd@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17699004

Patch from Scott LaVarnway <slavarnw@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6480 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-18 14:55:49 +00:00
henrik.lundin@webrtc.org
12396aba42 Update PacketSource and RtpFileSource 
The NextPacket method should now return NULL when the end of the
source was reached. In the RtpFileSource, this means that when
the end of file is reached, NULL is returned. Also, when an RTCP
packet is encountered, the next packet will be read from file
immediately.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6479 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-18 12:20:31 +00:00
henrik.lundin@webrtc.org
d8de0669c9 Revert "Restore ptypes.txt file" 
This reverts r6460. It turns out the file was no longer needed after
all.

BUG=2996
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6478 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-18 11:09:53 +00:00
turaj@webrtc.org
ec869bf781 Revert 6473 "Update generated asm offsets scripts." 
The roll has not been successful. Reverted based on the request of the committer.

> Update generated asm offsets scripts.
> 
> Libvpx updated the unpack scripts to fix building dependencies.
> 
> Roll libvpx 269083:277778
> See https://codereview.chromium.org/295313002/
> https://codereview.chromium.org/298063002/
> https://codereview.chromium.org/305533008/
> https://codereview.chromium.org/305703002/
> https://codereview.chromium.org/298383003/
> https://codereview.chromium.org/302863004/
> https://codereview.chromium.org/320923003/
> https://codereview.chromium.org/325313007/
> for the libvpx changes.
> 
> See https://codereview.chromium.org/313243004/
> for the WebView changes.
> 
> *NOTE* This CL will break the Android bots as they are built in a
> Chromium checkout, which will pull in old libvpx DEPS. They will
> cycle to green when we roll libvpx into Chromium. We must do the
> rolls in this order because we have to land webrtc and libvpx at
> the same time into Chromium.
> 
> BUG=377062
> TBR=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/15809004

TBR=fgalligan@google.com

Review URL: https://webrtc-codereview.appspot.com/18589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6475 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 19:07:56 +00:00
fgalligan@google.com
32196decd6 Update generated asm offsets scripts. 
Libvpx updated the unpack scripts to fix building dependencies.

Roll libvpx 269083:277778
See https://codereview.chromium.org/295313002/
https://codereview.chromium.org/298063002/
https://codereview.chromium.org/305533008/
https://codereview.chromium.org/305703002/
https://codereview.chromium.org/298383003/
https://codereview.chromium.org/302863004/
https://codereview.chromium.org/320923003/
https://codereview.chromium.org/325313007/
for the libvpx changes.

See https://codereview.chromium.org/313243004/
for the WebView changes.

*NOTE* This CL will break the Android bots as they are built in a
Chromium checkout, which will pull in old libvpx DEPS. They will
cycle to green when we roll libvpx into Chromium. We must do the
rolls in this order because we have to land webrtc and libvpx at
the same time into Chromium.

BUG=377062
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6473 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 17:55:23 +00:00
stefan@webrtc.org
a15fbfdcde Add round-robin selection of send stream to pad on. 
BUG=1812
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 17:32:05 +00:00
niklas.enbom@webrtc.org
9c09e6ee2b Add high perf mode to VP8 
R=marpan@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6470 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 16:32:08 +00:00
henrike@webrtc.org
a685c9df62 base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/ 
BUG=N/A
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6467 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 14:48:44 +00:00
henrike@webrtc.org
5654b305e5 Rebase webrtc/base with r6464 version of talk/base: 
cd webrtc/base
svn diff -r 6463:6464 http://webrtc.googlecode.com/svn/trunk/talk/base >
6464.diff
patch -p0 -i 6464.diff

BUG=3379
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12749005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6466 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 14:37:05 +00:00
minyue@webrtc.org
d42da54768 Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..." 
> Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling.
> 
> TEST=passed_all_trybots
> R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/16619005

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6462 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 09:50:12 +00:00
kjellander@webrtc.org
851a09e71a Initial GN work for WebRTC 
This CL makes it possible to build the 'webrtc_base'
target using GN.
The majority of our GYP stuff in webrtc/build/common.gypi has been
translated into the configs of webrtc/BUILD.gn.
The webrtc/base/base.gyp file is translated into webrtc/base/BUILD.gn.

This CL depends on https://codereview.chromium.org/322373002/ for the
jsoncpp BUILD.gn file and the ssl config.
To build inside Chromium, https://codereview.chromium.org/321313006/
needs to be landed first.

BUG=webrtc:3441
TEST=
Successful compilation of WebRTC as standalone:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_clang=true" && ninja -C out/Default
I also ran:
gn gen out/Default --args="build_with_chromium=false have_dbus_glib=true"
but it fails to compile: something is probably wrong with with pkg-config for that.

For Chromium, I symlinked src/third_party/webrtc to the webrtc subfolder of the
WebRTC checkout and applied the following patches:
https://codereview.chromium.org/322373002 (for jsoncpp and ssl config)
https://codereview.chromium.org/321313006 (enable building WebRTC)
Then I built successfully using:
gn gen out/Default && ninja -C out/Default webrtc_base

R=brettw@chromium.org
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6461 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 08:54:03 +00:00
henrik.lundin@webrtc.org
2ca2188906 Restore ptypes.txt file 
The file was lost when the neteq folders where moved and renamed.

BUG=2996
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6460 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 08:51:01 +00:00
phoglund@webrtc.org
6b061425c2 Updated W3C getusermedia tests to the latest version of the spec. 
BUG=webrtc:3455
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 08:46:58 +00:00
minyue@webrtc.org
8f8503d947 Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling. 
TEST=passed_all_trybots
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6458 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 08:02:05 +00:00
henrike@webrtc.org
9f36c087f1 Makes it possible to prevent some third party libraries (jsoncpp and openssl) from being linked. This makes it possible to link webrtc with external implementations of those libraries in case the project depending on webrtc requires another version of those libraries. 
BUG=3379
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17699005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 21:35:20 +00:00
asapersson@webrtc.org
d980307197 Add max limit of number for overuses. When limit is reached always apply the rampup delay. 
BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6451 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 14:27:19 +00:00
asapersson@webrtc.org
4b12d40008 Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class. 
BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6449 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 14:09:28 +00:00
kjellander@webrtc.org
d6e2213edd Remove ivinnichenko from webrtc/test/OWNERS 
Apparently, We're doing a poor job of cleaning out
really old OWNERS.

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6447 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 11:42:27 +00:00
henrik.lundin@webrtc.org
1e3c5c248a Importing ThreadChecker class from Chromium 
The ThreadChecker class is imported/re-implemented from Chromium.
The implementation is changed to depend on WebRTC primitives.

R=andrew@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6446 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 11:34:44 +00:00
bjornv@webrtc.org
b099a6f9ab Adds aluebs@webrtc.org as owner to audio_processing 
BUG=N/A
TESTED=trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6445 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 10:30:58 +00:00
bjornv@webrtc.org
721f970cba common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16 
We should avoid macros in general (see style guide) and the shift ones are particular dangerous since they assume that the user apply a non-negative shift.

Related CL: https://webrtc-codereview.appspot.com/16669004

BUG=3348,3353
TESTED=trybots and manually on linux
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6444 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 10:30:14 +00:00
mflodman@webrtc.org
eb16b811fb Implements start bitrate for new video API. 
Added  a new rampup test.

BUG=2879
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6443 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 08:57:39 +00:00
henrik.lundin@webrtc.org
63e46077a3 Add thread annotations to parts of ACMGenericCodec 
This change adds annotations to all member variables that could be
annotated without acquiring any new locks, or changing the lock
structure in any other way.

BUG=3041
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6441 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 08:02:25 +00:00
kjellander@webrtc.org
2bae3211b1 Add missing sources to webrtc/base/base.gyp 
During my work on setting up a GN build for
WebRTC, I discovered that the base.gyp is trying
to remove source files (for the Chromium build)
that are not added in the initial set.
I assume these files should be listed and that
GYP just doesn't complain when it's trying to
remove a file that is not present in the sources
list.

natserver_main.cc is also removed, since it's not used anywhere.

There are also a couple of other header files that are
used in other code that probably also should be listed in
base.gyp (please do this in another CL):
* compile_assert.h
* dscp.h
* move.h
* template_util.h

BUG=None
TEST=Trybots passing clobber compile step.
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6438 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 07:11:19 +00:00
glaznev@webrtc.org
ab23d493e0 Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc. 
Review URL: https://webrtc-codereview.appspot.com/20659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6436 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 23:31:35 +00:00
bjornv@webrtc.org
af6f02f7bd Neon version of OverdriveAndSuppress() 
audioproc reports the average frame time going from 279us to 255us with the test data used.

the output does not match the c version, but the difference seen is +-1.

Performance gain on Nexus7: 8.8%

BUG=3131
TESTED=trybots and manually
R=bjornv@webrtc.org, cd@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19539004

Patch from Scott LaVarnway <slavarnw@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6433 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 14:50:23 +00:00
kjellander@webrtc.org
a1bfc50a72 Pass GYP DEPTH variable to isolate. 
Similar change to https://codereview.chromium.org/322403003/
This will make it possible to handle different
directory levels for special builds of WebRTC, without
breaking GYP when the .isolate files are processed and
their contents is verified.

Also update all our .isolate files to use the <(DEPTH)
variable.

BUG=343106
TEST=Successful compile+test on Linux using:
ninja -C out/Release
tools/swarming_client/isolate.py run -s out/Release/tools_unittests.isolated
Also trybots passing all tests.

R=pbos@webrtc.org
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6427 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 09:02:15 +00:00
wu@webrtc.org
8e256eec4f Revert 6415 "Update generated asm offsets scripts." 
> Update generated asm offsets scripts.
> 
> This is the same CL as https://webrtc-codereview.appspot.com/16629004/
> Relanding and TBR from previous lgtm.
> 
> Libvpx updated the unpack scripts to fix building dependencies.
> 
> Roll libvpx 269083:275816
> See https://codereview.chromium.org/295313002/
> https://codereview.chromium.org/298063002/
> https://codereview.chromium.org/305533008/
> https://codereview.chromium.org/305703002/
> https://codereview.chromium.org/298383003/
> https://codereview.chromium.org/302863004/
> https://codereview.chromium.org/320923003/
> for the libvpx changes.
> 
> See https://codereview.chromium.org/313243004/
> for the WebView changes.
> 
> *NOTE* This CL will break the Android bots as they are built in a
> Chromium checkout, which will pull in old libvpx DEPS. They will
> cycle to green when we roll libvpx into Chromium. We must do the
> rolls in this order becuase we have to land webrtc and libvpx at
> the same time into Chromium.
> 
> BUG=377062
> TBR=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/17689004

TBR=fgalligan@google.com

Review URL: https://webrtc-codereview.appspot.com/13709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6423 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 23:03:17 +00:00
henrike@webrtc.org
3c13ed3b93 json.h include different header files depending on WEBRTC_CHROMIUM_BUILD being defined or not. Since json.h/cc is not even used in chromium it is the wrong flag to use. Instead add WEBRTC_EXTERNAL define. Also added OWNERS for base which is a copy of system_wrappers owners as the two folders are being merged. 
BUG=3379
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 21:43:04 +00:00
stefan@webrtc.org
cb254aac3b Enable pacing by default and remove the option to disable it from the new API. 
BUG=1672
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6416 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 15:12:25 +00:00
fgalligan@google.com
27f062ae6f Update generated asm offsets scripts. 
This is the same CL as https://webrtc-codereview.appspot.com/16629004/
Relanding and TBR from previous lgtm.

Libvpx updated the unpack scripts to fix building dependencies.

Roll libvpx 269083:275816
See https://codereview.chromium.org/295313002/
https://codereview.chromium.org/298063002/
https://codereview.chromium.org/305533008/
https://codereview.chromium.org/305703002/
https://codereview.chromium.org/298383003/
https://codereview.chromium.org/302863004/
https://codereview.chromium.org/320923003/
for the libvpx changes.

See https://codereview.chromium.org/313243004/
for the WebView changes.

*NOTE* This CL will break the Android bots as they are built in a
Chromium checkout, which will pull in old libvpx DEPS. They will
cycle to green when we roll libvpx into Chromium. We must do the
rolls in this order becuase we have to land webrtc and libvpx at
the same time into Chromium.

BUG=377062
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6415 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 14:59:28 +00:00
kjellander@webrtc.org
7e3d62d709 Revert 6411 "Revert 6407 "Revert 6405 "Update generated asm offs..." 
Turns out the previous revert was based on invalid assumptions.
The libvpx in Chromium was reverted in 
http://chromegw.corp.google.com/viewvc/chrome?view=rev&revision=271259
which ends up with libvpx r269083. Therefore we should restore
that same libvpx revision for WebRTC, which this revert will do.

> Revert 6407 "Revert 6405 "Update generated asm offsets scripts.""
> 
> > Revert 6405 "Update generated asm offsets scripts."
> > 
> > TBR=fgalligan@google.com
> > BUG=N/A
> > 
> > Review URL: https://webrtc-codereview.appspot.com/20639004
> 
> TBR=henrike@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/15739004

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6413 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 11:07:07 +00:00
minyue@webrtc.org
c01cc3d3a8 Revert 6407 "Revert 6405 "Update generated asm offsets scripts."" 
> Revert 6405 "Update generated asm offsets scripts."
> 
> TBR=fgalligan@google.com
> BUG=N/A
> 
> Review URL: https://webrtc-codereview.appspot.com/20639004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6411 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 08:48:34 +00:00
asapersson@webrtc.org
2881ab1e36 Increased kMaxRampUpDelayMs (120 to 240s). 
Add support for triggering on encode rsd metric if its thresholds are configured. Added unit tests.

BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 08:46:46 +00:00
henrike@webrtc.org
286cd7683c Revert 6405 "Update generated asm offsets scripts." 
TBR=fgalligan@google.com
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/20639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6407 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 00:38:32 +00:00
fgalligan@google.com
4aeb94186a Update generated asm offsets scripts. 
Libvpx updated the unpack scripts to fix building dependencies.

Roll libvpx 269083:275816
See https://codereview.chromium.org/295313002/
https://codereview.chromium.org/298063002/
https://codereview.chromium.org/305533008/
https://codereview.chromium.org/305703002/
https://codereview.chromium.org/298383003/
https://codereview.chromium.org/302863004/
https://codereview.chromium.org/320923003/
for the libvpx changes.

See https://codereview.chromium.org/313243004/
for the WebView changes.

BUG=377062
R=andrew@webrtc.org, michaelbai@chromium.org

Review URL: https://webrtc-codereview.appspot.com/16629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6405 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 17:12:51 +00:00
henrik.lundin@webrtc.org
5b111b06fa Re-land "Create a joint encoder/decoder wrapper for iSAC in ACM" 
The change was reverted since it was thought to cause a flaky test.
But the test kept flaking after the change was reverted.

This effectively reverts r6394, relanding r6377.

BUG=3496
TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6404 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 14:37:21 +00:00
phoglund@webrtc.org
8454ad1b3e Reland: Making WebRTC able to play and record audio to files for tests. 
By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to
WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to
play out audio to a file and feed audio in from a file. We want to do
so we can better test WebRTC-using applications by recording what the
audio stack outputs and feeding known audio in for quality tests.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6403 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 14:12:04 +00:00
stefan@webrtc.org
fbb567dacd Add APIs to enable padding with redundant payloads. 
Also makes a small change to the tests to remove flakiness. We can't do
BWE only based on rtp timestamps if we preemptively resend packets instead
of sending padding packets.

BUG=1812,2992
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6400 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 13:41:36 +00:00