27ab19d9b4
Roll chromium_revision 272489:277350 + fix sanitizer options
...
Rolling to this new Chromium revision required us to introduce
a sanitizer_options similar to the one in Chromium's base
(see https://code.google.com/p/chromium/codesearch#chromium/src/base/base.gyp&l=977
and https://codereview.chromium.org/238123003 ) in order
to get the same defaults for ASan and LSan. Without it
compilation will break since LeakSanitizer (LSan) is enabled by
default in Clang r209387 that is pulled with this roll.
I setup so that we pull in the sanitizer_options.cc and
tsan_suppressions.cc files using DEPS, so we don't have to maintain
them separately for now. We can still use our own TSan suppressions.txt
file as we do today with no changes needed.
This roll also brings in http://crrev.com/276676 so we can enable
GN build for WebRTC.
Overview of changes in Chrome DEPS:
$ svn diff http://src.chromium.org/chrome/trunk/src/DEPS -r 272489:277350
which can be compared with the output of:
$ svn cat http://webrtc.googlecode.com/svn/trunk/DEPS | grep chromium_deps | sed 's/^ *//' | sort | uniq
in a WebRTC checkout, gives the following relevant changes:
* third_party/android_tools 6fc0e1:c6e658
* third_party/libjpeg_turbo 263594:272637
* third_party/libyuv 1000:1007
* third_party/nss 271760:277057
* tools/gyp 1921:1927
* tools/swarming_client ae8085:aea506
The following also shows that Clang is upgraded from r206824 to r209387:
$ svn diff http://src.chromium.org/chrome/trunk/src/tools/clang/scripts/update.sh -r 272489:277350
BUG=3441
TEST=Trybots are not passing since after the recipe switch, SVN-based try jobs doesn't seem to support auto-detecting that a sync is needed if there's a DEPS change.
R=andrew@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6516 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-21 19:30:29 +00:00
a24d366e1c
- Exit from a camera thread lopper loop() method only after all camera release calls are completed. This fixes camera exceptions observed from time to time when calling camera functions on a terminated looper.
...
- Allocate real texture for camera preview.
- Add fps and camera frame duration logging.
- Get camera frame timestamp in Java code and pass it to jni code so the frame timestamp is assigned as soon as possible. Jni code will not use these timestamps yet until timestamp ntp correction and zeroing in webrtcvideengine.cc will be addressed.
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6513 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 20:55:54 +00:00
0b893b1e05
Do not hold the critical section in VideoCaptureAndroid::SetCaptureRotation since it would case possible deadlock with OS Camear thread.
...
BUT=3464
TEST=Manual Test with WebRTCDemo
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6509 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 16:00:30 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
620048172c
Create a joint encoder/decoder wrapper for iSAC in ACM
...
This CL extends the ACMISAC wrapper class to inherit from AudioDecoder
as well (the type of object that NetEq uses). The class has it's own
lock protecting the iSAC instance. This way, we can remove the
neteq_decode_lock_ (a.k.a. decoder_lock_) in a later CL.
The old AcmAudioDecoderIsac class is deleted.
R=kwiberg@webrtc.org , tina.legrand@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6377 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09 18:39:00 +00:00
