generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
5,800 Commits 1 Branch 0 Tags 108 MiB
9158df2aa4
Commit Graph

5800 Commits

This Branch
This Branch
All Branches
Author SHA1 Message Date
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
buildbot@webrtc.org
117afeec91 (Auto)update libjingle 69188577-> 69260070 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6437 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 07:11:01 +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
glaznev@webrtc.org
c6c1dfd7ea Add extra logging and latency restriction to VP8 HW encoder. 
- Do not allow encoder to accumulate more than 100 ms of
data in input buffers.
- Add optional extra logging (disabled by default) to track
encoder buffers timing.

BUG=
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6435 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 22:59:08 +00:00
buildbot@webrtc.org
a6764ab869 (Auto)update libjingle 69144530-> 69164179 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6434 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 18:24:39 +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
buildbot@webrtc.org
db56390f7e (Auto)update libjingle 69143161-> 69144530 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6432 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 13:05:48 +00:00
pbos@webrtc.org
f99c2f2dbc Add NACK feedback parameter to WebRtcVideoEngine2. 
Also fixing enabling/disabling of NACK. Previous implementation was made
under the assumption that NACK should always be enabled which caused
both missing NACK settings in SDP as well as broken interop between old
and new WebRtcVideoEngines.

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6431 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 12:27:38 +00:00
pbos@webrtc.org
e322a175f6 Implement RTX tests+fixes in WebRtcVideoEngine2. 
BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6430 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 11:47:28 +00:00
pbos@webrtc.org
9fbb717aca Remove engine_codecs_ cache from unittests. 
Used interchangably with engine_.codecs() becomes confusing and it's not
really used that much.

BUG=1788
R=pthatcher@google.com, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17689005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6429 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 09:34:13 +00:00
kjellander@webrtc.org
d54ec1256c Fix GYP DEPTH for libjingle isolate files 
In https://review.webrtc.org/13679004/ the libjingle isolate
files in patch set #2 were not tested, which caused a failure when
6427 was committed. This fixes the talk/build/isolate.gypi with a
similar change.

BUG=343106
TEST=Successful local compile on Linux
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6428 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 09:16:23 +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
buildbot@webrtc.org
c800c1cc40 (Auto)update libjingle 69131548-> 69132244 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6426 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 07:56:17 +00:00
pbos@webrtc.org
1c8223c590 Initial owners file for talk/media/webrtc/. 
Including pthatcher@webrtc.org (already root owner) and
mflodman@webrtc.org.

BUG=
R=juberti@google.com, juberti@webrtc.org
TBR=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15679008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6425 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 07:29:26 +00:00
buildbot@webrtc.org
7e71b77f8a (Auto)update libjingle 69102234-> 69116997 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6424 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 01:14:01 +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
jiayl@webrtc.org
1a6c6281ca Revert r6420 'Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck' 
Failing tests are disabled for memcheck.

TBR=wu@webrtc.org
BUG=2626

Review URL: https://webrtc-codereview.appspot.com/13699004

Review URL: https://webrtc-codereview.appspot.com/13699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 21:59:29 +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
jiayl@webrtc.org
ddeec048c0 Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck 
This reverts commit c3272a942f04f9dd0db3f6bf0d201bcf47c3fa08.

TBR=wu@webrtc.org
BUG=2626

Review URL: https://webrtc-codereview.appspot.com/13689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6420 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 21:42:46 +00:00
buildbot@webrtc.org
3f3f428d2b (Auto)update libjingle 69097619-> 69099564 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6419 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 21:36:26 +00:00
jiayl@webrtc.org
6c6f33b5bb Fix the flaky RTP DataChannel test. 
BUG=2891
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6418 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 21:05:19 +00:00
buildbot@webrtc.org
18dfa8d574 (Auto)update libjingle 69069003-> 69082899 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6417 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 18:11:02 +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
xians@webrtc.org
4cb012858f Fixed GetStats when local and remote track are using the same ssrc. 
R=hta@chromium.org, kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6414 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 14:57:05 +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
buildbot@webrtc.org
b90619c07f (Auto)update libjingle 69049090-> 69054765 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6412 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 09:19:08 +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
pbos@webrtc.org
276637b107 Disable flaky test on DrMemory Full. 
VideoSendStreamTest.RetransmitsNackOverRtxWithPacing fails
often on DrMemory Full.

BUG=3471
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6409 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 08:46:21 +00:00
buildbot@webrtc.org
d41eaeb7cd (Auto)update libjingle 69005149-> 69049090 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 07:13:26 +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
buildbot@webrtc.org
e9e8007ab4 (Auto)update libjingle 68985065-> 69005149 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6406 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 18:41:17 +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
henrik.lundin@webrtc.org
ab85187e63 Remove unused resource 
The file resources/audio_coding/neteq_universal.rtp is no longer
used in any test. Removing the hash file neteq_universal.rtp.sha1.

BUG=2996
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6402 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 13:59:44 +00:00
pbos@webrtc.org
9e65a3b013 Re-land webrtcmediaengine.cc part of r6397. 
webrtcvideoengine.cc un-reverted by a bot roll in r6399 so half of r6397
is still applied. The applied fix (diff between r6397) is to put
WebRtcVideoEngine2 in ifdefs and only build for WEBRTC_CHROMIUM_BUILDs
corresponding to webrtcmediaengine.h.

BUG=
R=minyue@webrtc.org
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19719005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 13:42:37 +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
buildbot@webrtc.org
5d223a7d2d (Auto)update libjingle 68982444-> 68983526 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6399 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 13:05:05 +00:00
minyue@webrtc.org
6604c6df26 Revert 6397 "(Auto)update libjingle 68949184-> 68982444" 
> (Auto)update libjingle 68949184-> 68982444

TBR=buildbot@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6398 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 13:02:36 +00:00
buildbot@webrtc.org
af214d804f (Auto)update libjingle 68949184-> 68982444 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6397 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 12:46:49 +00:00
minyue@webrtc.org
e08a11c4a1 Revert 6395 "Making WebRTC able to play and record audio to file..." 
> 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/20609004

TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6396 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 10:40:30 +00:00
phoglund@webrtc.org
fa042ca15d 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/20609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 09:57:23 +00:00
henrik.lundin@webrtc.org
c726b1fc33 Revert r6377 "Create a joint encoder/decoder wrapper for iSAC in ACM" 
BUG=3469
TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6394 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 08:35:53 +00:00
bjornv@webrtc.org
18026abd82 common_audio/signal_processing: Removes macro WEBRTC_SPL_RSHIFT_U16 
This macro is only used at a few places and implies a cast to uint16_t before right shifting. All places already use uint16_t. Further, the amount of shifts applied in the macro has no sanity check for negativity, makes the macro dangerous to use.

BUG=3348,3353
TESTED=trybots and manually
R=kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6393 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 06:53:20 +00:00
bjornv@webrtc.org
782978cfcb common_audio/signal_processing: Moves WEBRTC_SPL_UMUL_16_16_RSFT16 to iSAC fix 
This macro is only used by the fixed point version of iSAC. Replacing the (five) locations in arith_routines_logist.c, where it is used, with the actual operation.

BUG=3348,3353
TESTED=trybots and manually
R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6392 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 06:39:03 +00:00
bjornv@webrtc.org
3f83072c26 modules/audio_processing: Adds a config for reported delays 
There are platforms and devices where the reported delays are untrusted and we currently solve that with an extended filter length and a slightly more conservative delay handling.
With this change we give the user the possibility to turn off reported system delay values completely.

- Includes new unit tests.

TESTED=trybots and manual testing
R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6391 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 04:48:11 +00:00
jiayl@webrtc.org
e61b8e32d8 Adds end to end DataChannel tests. 
BUG=2626
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 23:54:13 +00:00
glaznev@webrtc.org
a40210aee2 Add support for NVidia VP8 HW encoder. 
- Some changes in HW VP8 encoder search logic to detect HW codec
with supported color space format.
- Support yuv420 and nv12 formants for encoder input.
- Add some extra logging and encoder frame drop statistics.

BUG=3176
R=fischman@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6389 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 23:48:29 +00:00