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Author SHA1 Message Date
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
henrik.lundin@webrtc.org
fd59c39caa Delete last file in neteq4 folder 
The .isolate file can now be safely removed, since issue 3462 is
resolved.

BUG=2996
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6388 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 20:26:27 +00:00
andrew@webrtc.org
919914d71b MIPS optimizations for ISAC (patch #1) 
Implemented functions:
    - WebRtcIsacfix_AutocorrMIPS
    - WebRtcIsacfix_FilterArLoop
    - WebRtcIsacfix_FilterMaLoopMIPS
    - WebRtcIsacfix_AllpassFilter2FixDec16MIPS (only MIPS DSP)
    - WebRtcIsacfix_PitchFilterCore (only MIPS DSPR2)

Gain achieved: from aprox. 15% (MIPS32) up to aprox. 40% (MIPS DSPR2)

R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17559005

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 18:13:15 +00:00
mflodman@webrtc.org
0d7ab0a634 Adding the new video folder and pacer to the wathclist. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6386 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 13:59:37 +00:00
kwiberg@webrtc.org
12cd443752 Noise suppression: Change signature to work on floats instead of ints 
Internally, it already worked on floats. This patch just changes the
signature of a bunch of functions so that floats can be passed
directly from the new and improved AudioBuffer without converting the
data to int and back again first.

(The reference data to the ApmTest.Process test had to be modified
slightly; this is because the noise suppressor comes immediately after
the echo canceller, which also works on floats. If I truncate to
integers between the two steps, ApmTest.Process doesn't complain, but
of course that's exactly the sort of thing the float conversion is
supposed to let us avoid...)

BUG=
R=aluebs@webrtc.org, bjornv@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 11:13:09 +00:00
kjellander@webrtc.org
1014101470 Revert 6380 "Replace libjingle_root with talk_root variable." 
It turns out this doesn't fix the problem we're trying to solve...

> Replace libjingle_root with talk_root variable.
> 
> This CL is similar to https://review.webrtc.org/9019004/
> It is needed in order to be able to build with different
> copies of libjingle. Having the libjingle_root variable didn't
> make this possible, since relative paths in the .isolate files
> ended up at the wrong directory level and .isolate files doesn't
> support all the normal GYP variables like <(DEPTH).
> 
> BUG=chromium:343106
> TEST=trybots passing compile step with clobber.
> R=tommi@webrtc.org, wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/15709004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6384 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 10:13:38 +00:00
buildbot@webrtc.org
3eb2c2f4c3 (Auto)update libjingle 68891947-> 68893961 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 09:39:06 +00:00
pbos@webrtc.org
86f613d6b8 Move WebRtcVideoEngine2 fakes to unittest header. 
BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6382 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 08:53:05 +00:00
asapersson@webrtc.org
734a532723 Add additional metric (relative standard deviation of encode time) for overuse detection. 
This code is currently only for testing.

BUG=1577
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6381 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 06:35:22 +00:00
kjellander@webrtc.org
0238682984 Replace libjingle_root with talk_root variable. 
This CL is similar to https://review.webrtc.org/9019004/
It is needed in order to be able to build with different
copies of libjingle. Having the libjingle_root variable didn't
make this possible, since relative paths in the .isolate files
ended up at the wrong directory level and .isolate files doesn't
support all the normal GYP variables like <(DEPTH).

BUG=chromium:343106
TEST=trybots passing compile step with clobber.
R=tommi@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6380 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 05:46:31 +00:00
kjellander@webrtc.org
7b82c18979 Add kjellander@webrtc.org as OWNER for *.isolate 
This should make project-wide changes for isolate files
easier and make it more obvious who's a suitable reviewer
for them.

BUG=
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6379 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 05:42:53 +00:00