| 48407f71221635aec2d2a382ee24dea28e4f69cf |
|
09-Nov-2015 |
peah <peah@webrtc.org> |
Changed queue implementation to the proposed vector-based solution.
Added unit tests.
BUG=webrtc:5099
TBR=hlundin-webrtc
Review URL: https://codereview.webrtc.org/1398473004
Cr-Commit-Position: refs/heads/master@{#10562}
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| ac4234ccfc964208163d6152977cc1d015448bbb |
|
25-Jun-2015 |
Andrew MacDonald <andrew@webrtc.org> |
Add a [rtc_]build_with_neon variable to unify conditions.
Also consolidate ARM options for gn in an arm_neon_config.
R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org
Review URL: https://codereview.webrtc.org/1181373004.
Cr-Commit-Position: refs/heads/master@{#9501}
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| 57e5fd2e604ff7e60425c3f7654b40da03fc763c |
|
25-May-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
PRESUBMIT: Improve PyLint check and add GN format check.
Add pylintrc file based on
https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc
bit tightened up quite a bit (the one in depot_tools is far
more relaxed).
Remove a few excluded directories from pylint check and fixed/
suppressed all warnings generated.
Add GN format check + formatted all GN files using 'gn format'.
Cleanup redundant rules in tools/PRESUBMIT.py
TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations.
Ran it again with a modification in webrtc/build/webrtc.gni, formatted
all the GN files and ran it again.
R=henrika@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50069004
Cr-Commit-Position: refs/heads/master@{#9274}
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| 23dc68e515acf6b4227adddac36c1f8b16aade9f |
|
24-Apr-2015 |
Andrew MacDonald <andrew@webrtc.org> |
Add the rtc_build_openmax_dl variable to the GN build.
For symmetry with the gyp build.
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49109005
Cr-Commit-Position: refs/heads/master@{#9083}
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| f2497cf5175c9325d3c3a5727f36e31b7f2f4f1c |
|
16-Apr-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Fix unknown option '-msse2' warning
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43169004
Cr-Commit-Position: refs/heads/master@{#9016}
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| a9c0ae284c464e962492c53f0d11bb2fc99da910 |
|
15-Apr-2015 |
Alejandro Luebs <aluebs@webrtc.org> |
Add a sparse FIR filter implementation
A Finite Impulse Response filter implementation which takes advantage of sparse coefficients.
The coefficients are assumed to be uniformly distributed and have an initial offset.
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49659004
Cr-Commit-Position: refs/heads/master@{#9002}
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| 04c50981f8574b4ef3f8751ac37d0eb50a569a2b |
|
19-Mar-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
Add the Ooura FFT to RealFourier.
We are using the Ooura FFT in a few places:
- AGC
- Transient suppression
- Noise suppression
The optimized OpenMAX DL FFT is considerably faster, but currently does
not compile everywhere, notably on iOS. This change will allow us to use
Openmax when possible and otherwise fall back to Ooura.
(Unfortunately, noise suppression won't be able to take advantage of it
since it's not C++. Upgrade time?)
R=aluebs@webrtc.org, mgraczyk@chromium.org
Review URL: https://webrtc-codereview.appspot.com/45789004
Cr-Commit-Position: refs/heads/master@{#8798}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8798 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| 0933d01d094648f6c133e9d5d71e27da4e013e54 |
|
05-Mar-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
Enabling common_audio building with NEON on ARM64
Passed building common_audio_neon and common_audio_unittests both on
Android ARMv7 and Android ARM64. Pass common_audio_unittests tests both
on Android ARMv7 and Android ARM64.
BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org
Change-Id: I8e0722f356db8cca6fc8232f00ae1e898a086f5a
Review URL: https://webrtc-codereview.appspot.com/40629004
Patch from Zhongwei Yao <zhongwei.yao@arm.com>.
Cr-Commit-Position: refs/heads/master@{#8620}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8620 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| ac2d27d9ae74eb8d28ec0d5f12f70fa64461ab90 |
|
26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Fix style violations in common_types.h and config.h
Mostly, it's about moving constructors and descructors to the .cc
files, so that they won't be inlined everywhere.
The reason this CL is so big is that a lot of code was using
common_types.h without declaring a dependency on webrtc_common, which
broke the build once common_types.h started to depend on
common_types.cc.
BUG=163
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26089004
Cr-Commit-Position: refs/heads/master@{#8516}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| 722739108a9a1b30cbcb8285ce0b76762b356fb3 |
|
23-Feb-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530)
Includes GN changes from
https://webrtc-codereview.appspot.com/39249004/
Android changes for JNI were required due to
https://codereview.chromium.org/843103003
Other relevant changes:
* src/buildtools: 5c5e924..93b3d0a
* src/third_party/boringssl/src: d306f16..b180ee9
* src/third_party/icu: 4e3266f..2081ee6
* src/third_party/libvpx: 5cdd302..33bbffe
* src/third_party/usrsctp/usrsctplib: 190c8cb..13718c7
* src/tools/gyp: 4d7c139..3464008
* src/tools/swarming_client: bdad118..1b7bfec
Details: https://chromium.googlesource.com/chromium/src/+/b0c3ed3..2c3ffb2/DEPS
Clang version was not updated in this roll.
R=dpranke@chromium.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40079004
Cr-Commit-Position: refs/heads/master@{#8466}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8466 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| 035e9123e9c7de3b09b37bc8e57907e4af7ce219 |
|
28-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Move channel_buffer.{h,cc} to common_audio.
In https://code.google.com/p/webrtc/source/detail?r=8166
I added a check preventing GYP files from referencing
sources above their directory level.
This CL fixes the disallowed reference added in
https://code.google.com/p/webrtc/source/detail?r=8157
by moving channel_buffer.{h,cc} to common_audio for real.
BUG=4185
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35939004
Cr-Commit-Position: refs/heads/master@{#8190}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8190 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| 041035b390e25b1d388d73ef52c9e68a550a982f |
|
26-Jan-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
Add an AudioRingBuffer class wrapper for the ring_buffer.h C interface.
Integrate it in Blocker to demonstrate use.
TEST=beamforming sounds good.
R=aluebs@webrtc.org, mgraczyk@chromium.org, sahark@google.com
Review URL: https://webrtc-codereview.appspot.com/36799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8157 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| 6b6301588ef2d7b5f5d442aa95bef442a43ead53 |
|
15-Jan-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
Move ring_buffer to common_audio.
In preparation for adding a C++ wrapper in common_audio. Also, change
the return type of Init to void and call it from Create.
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8068 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| d730b288c8873d6459250327be02c1895c86e652 |
|
07-Jan-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
Remove WebRtcSpl_ScaleAndAddVectorsWithRoundNeon
This function isn't used anymore. The file and header are also removed.
BUG=4002,3273
R=andrew@webrtc.org
Change-Id: I4b65dec57e6adc2ac2253031501f3b6de6937fac
Review URL: https://webrtc-codereview.appspot.com/35519004
Patch from Yang Zhang <yang.zhang@arm.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8019 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| 1090a6eccf9f998caa30ca64d19a804f3fd7449f |
|
18-Dec-2014 |
andrew@webrtc.org <andrew@webrtc.org> |
Remove obsolete target_arch == armv7.
Also, use arm_version >= 7 so things will continue to work when building
for ARMv8 and higher targets.
BUG=3906
R=kjellander@webrtc.org, tkchin@webrtc.org, zhongwei.yao@arm.com
Review URL: https://webrtc-codereview.appspot.com/38379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7957 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| 1751ee7d326677cb5843898228e9288c35f76682 |
|
02-Dec-2014 |
andrew@webrtc.org <andrew@webrtc.org> |
Remove -flax-vector-conversions flag for ARM NEON building.
Pass compilation on both ARMv7 and ARM64. The generated
binary (audioproc) is byte to byte (with symbol striped) same as
before. The output of audioproc -aecm is also byte to byte same between
C and NEON version on ARMv7 and ARM64.
Change-Id: Ibdf40fe085f6bad1311f59bf9318bbcf37dd7ce5
BUG=3850
R=andrew@webrtc.org, jridges@masque.com
Review URL: https://webrtc-codereview.appspot.com/30239004
Patch from Zhongwei Yao <zhongwei.yao@arm.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7783 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| a3ed713dad5ccad03e2f5d775081143babd19097 |
|
31-Oct-2014 |
andrew@webrtc.org <andrew@webrtc.org> |
Add a WavReader counterpart to WavWriter.
Don't bother with a C interface as we currently have no need to call
this from C code. The first use will be in the audioproc tool.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7585 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| 8aa4d2d2cd46bca6da7e071583482dd7ed0e2d0c |
|
30-Oct-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Creating a C++ wrapper class for VAD
Also adding a mock. This work is part of an ongoing effort to
encapsulate encoders in AudioEncoder classes. The CNG encoder will also
be implemented as an AudioEncoder class, and will also contain a VAD
C++ wrapper.
BUG=3926
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7570 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| aada86b261146336d74ea09acedaf40e5c2f4618 |
|
27-Oct-2014 |
andrew@webrtc.org <andrew@webrtc.org> |
Add a simple AudioConverter class.
This will be used to refactor AudioProcessing/AudioBuffer. We can
enable alternate downmixing schemes in AudioProcessing by pulling
the conversion logic out of AudioBuffer.
The unit test is largely stolen from voice_engine/utility_unittest.cc.
As commented, the voice_engine routines should be replaced with
AudioConverter.
BUG=chromium:405270
R=aluebs@webrtc.org, mgraczyk@chromium.org
TBR=kwiberg
Review URL: https://webrtc-codereview.appspot.com/30779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7538 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| fab543911287c699fce2deef4aae691e982ae8f0 |
|
15-Oct-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
common_audio: Removed version API from signal_processing
The Signal Processing version API is not used anymore.
BUG=3353
R=kwiberg@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7451 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| 4165f7aa226584f6865df5ce37b0cb0bae4150d8 |
|
08-Oct-2014 |
andrew@webrtc.org <andrew@webrtc.org> |
Add a variable for deciding when to use openmax_dl.
Modifies the previous condition to additionally not use openmax_dl on
iOS. Remove the All target's direct dependency on it, as it is now
pulled in by the targets that need it.
Add gn support since an openmax_dl gn target is available.
BUG=chromium:415393, webrtc:3906
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7397 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| 325cff01b42064313b221f2c8819ce01218f5ca4 |
|
01-Oct-2014 |
andrew@webrtc.org <andrew@webrtc.org> |
Import LappedTransform and friends.
Add code for doing block-based frequency domain processing. Developed
and reviewed in isolation. Corresponding export CL:
https://chromereviews.googleplex.com/95187013/
R=bercic@google.com, kjellander@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7359 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| b8caf6a5042e4a8bf0230b8a202a0fbfc414fb59 |
|
30-Sep-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
GN: Enable libvpx, add link target and convert some test targets
Libvpx now supports GN and this CL turns on compiling it.
I also introduced an executable target 'webrtc_tests'
that depends on all in WeBRTC + tests in order to get a full
linking step executed (since we've seen link problems for GN
when rolling WebRTC into Chromium).
I also converted a few test targets and made a GN file for
third_party/gflags.
BUG=3441
TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc
dir to a workspace wit this CL applied.
R=brettw@chromium.org
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| f21ea918ad9e4dcbe7f372fd32d130c082641e36 |
|
28-Sep-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
GN: Add common configs to all targets.
This is needed to ensure we have the same build with GN
as with GYP, since GYP includes the common.gypi on a global level.
Several fixes has been needed in the past because some code have
been built without the right defines.
BUG=3441
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/28589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7317 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| a6cefcaceb49608f57d6f12ae18a6956e95f5351 |
|
16-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
gn: Fix cflags usage
R=brettw@chromium.org
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29519004
Patch from Cem Kocagil <ckocagil@chromium.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7198 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| 6d08ca6379a3200dcfb8df79fe420637e69b1a10 |
|
07-Sep-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
GN: Prefix WebRTC specific variables with "rtc_"
BUG=3441
TESTED=Trybots + Running GN in a Chromium checkout with
src/third_party/webrtc symlinked to the WebRTC checkout
with this CL applied, both with the default GN settings
and using: --args="os=\"android\" cpu_arch=\"arm\""
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/27379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| 524b8f7304090eec09e4251ca234e0ea083ce858 |
|
31-Aug-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
GN: Implement voice engine, common audio, audio coding and audio processing
NOTICE: Assembly offsets generation for audio processing will
not be ported to GN and the process of removing them is tracked
in https://code.google.com/p/webrtc/issues/detail?id=3580.
The GN files are based upon the GYP files as of r7009.
BUG=3441
TESTED=Passing builds with:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default
I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now:
gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default
Compilation of Chromium's 'all' target with src/third_party/webrtc
symlinked to the WebRTC checkout with this CL applied, both
with the default GN settings and using
--args="is_debug=false os=\"android\" cpu_arch=\"arm\""
R=andrew@webrtc.org, brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/15999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|
| 1227ab89a7c08e4e5af051a63daba889ea0d2da7 |
|
23-Jun-2014 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
GN: Add BUILD.gn files + kjellander to OWNERS
This should work as a foundation for all the work that is
left to do to make the parts of WebRTC that Chromium uses
to build with GN.
I implemented some the smaller modules myself in this CL.
The remaining work (TODO's in the .gn files) will be distributed
to various team members.
I'm adding myself to OWNERS files for BUILD.gn files in all the
directories where I'm adding a BUILD.gn file.
BUG=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 clang_use_chrome_plugins=false" && ninja -C out/Default
I built successfully from a Chromium checkout (with
https://codereview.chromium.org/321313006/ applied) using:
gn gen out/Default && ninja -C out/Default webrtc
R=brettw@chromium.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/BUILD.gn
|