55 Commits

Author SHA1 Message Date
kjellander
17f008bf33 GYP: Remove targets inside include_tests==1 that are converted to GN.
Remove a large number of targets that are no longer built, to reduce maintenance.
Only targets that have a GN version were removed.

BUG=webrtc:6323
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2340773003
Cr-Commit-Position: refs/heads/master@{#14231}
2016-09-15 11:57:39 +00:00
Alejandro Luebs
a181c9ad17 Keep track of the user-facing number of channels in a ChannelBuffer
Before this change the ChannelBuffer had a fixed number of channels. This meant for example that when the Beamformer would reduce the number of channels to one, the merging filter bank was still merging all the channels, which was unnecessary since they were not processed and just discarded later. This change doesn't change the signal at all. It just reflects the number of channels in the ChannelBuffer, reducing the complexity.

R=henrik.lundin@webrtc.org, peah@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/2053773002 .

Cr-Commit-Position: refs/heads/master@{#13352}
2016-06-30 22:33:47 +00:00
kwiberg
729b21f97f Add clz functions (Count number of Leading Zero bits), 32-and 64-bit variants
Using __builtin_clz on gcc/clang, and a fallback implementation
otherwise.

Also redefine WebRtcSpl_GetSizeInBits(x) as simply 32 - clz32(x).

BUG=chromium:601787

Review-Url: https://codereview.webrtc.org/2014023002
Cr-Commit-Position: refs/heads/master@{#13014}
2016-06-02 11:02:22 +00:00
kjellander
208d19845d Rename APK tests workaround to make it more generic.
We plan to add junit tests running with Robolectric
so naming these files "apk" is slightly confusing.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2020213002
Cr-Commit-Position: refs/heads/master@{#12971}
2016-05-31 11:01:47 +00:00
kjellander
e532aec252 Add isolate files for Android tests
BUG=chromium:583318
TESTED=Passing runs with:
GYP_DEFINES='test_isolation_mode=prepare OS=android' webrtc/build/gyp_webrtc
ninja -C out/Release
NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1882963003

Cr-Commit-Position: refs/heads/master@{#12397}
2016-04-18 03:08:28 +00:00
peah
faed4ab24b Revert of Moved ring-buffer related files from common_audio to audio_processing" (patchset #2 id:20001 of https://codereview.webrtc.org/1858123003/ )
Reason for revert:
Because of down-stream dependencies, this CL needs to be reverted.

The dependencies will be resolved and then the CL will be relanded.

Original issue's description:
> Revert "Revert of Moved ring-buffer related files from common_audio to audio_processing (patchset #8 id:150001 of https://codereview.webrtc.org/1846903004/ )"
>
> This reverts commit c54aad6ae07fe2a44a65be403386bd7d7d865e5b.
>
> BUG=webrtc:5724
> NOPRESUBMIT=true
>
> Committed: https://crrev.com/8864fe5e08f8d8711612526dee9a812adfcd3be1
> Cr-Commit-Position: refs/heads/master@{#12247}

TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5724

Review URL: https://codereview.webrtc.org/1855393004

Cr-Commit-Position: refs/heads/master@{#12248}
2016-04-05 21:57:55 +00:00
peah
8864fe5e08 Revert "Revert of Moved ring-buffer related files from common_audio to audio_processing (patchset #8 id:150001 of https://codereview.webrtc.org/1846903004/ )"
This reverts commit c54aad6ae07fe2a44a65be403386bd7d7d865e5b.

BUG=webrtc:5724
NOPRESUBMIT=true

Review URL: https://codereview.webrtc.org/1858123003

Cr-Commit-Position: refs/heads/master@{#12247}
2016-04-05 21:42:51 +00:00
peah
c54aad6ae0 Revert of Moved ring-buffer related files from common_audio to audio_processing (patchset #8 id:150001 of https://codereview.webrtc.org/1846903004/ )
Reason for revert:
This CL caused a google3 breakage due to dependencies in Google3.

I will fix that, and reland.

Original issue's description:
> Moved ring-buffer related files from common_audio to audio_processing
>
> BUG=webrtc:5724
> NOPRESUBMIT=true
>
> Committed: https://crrev.com/711ccc8d96490f58cc3d7fd9207c19d4d881d4dc
> Cr-Commit-Position: refs/heads/master@{#12227}

TBR=ivoc@webrtc.org,henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5724

Review URL: https://codereview.webrtc.org/1856323002

Cr-Commit-Position: refs/heads/master@{#12232}
2016-04-05 07:02:35 +00:00
peah
711ccc8d96 Moved ring-buffer related files from common_audio to audio_processing
BUG=webrtc:5724
NOPRESUBMIT=true

Review URL: https://codereview.webrtc.org/1846903004

Cr-Commit-Position: refs/heads/master@{#12227}
2016-04-05 05:57:48 +00:00
terelius
85fa7d5311 Move swap_queue.h to base/
This will let us use the SwapQueue as a message queue for the event log's output thread. See https://codereview.webrtc.org/1687703002/

Review URL: https://codereview.webrtc.org/1812823007

Cr-Commit-Position: refs/heads/master@{#12113}
2016-03-24 08:51:59 +00:00
kjellander
9ac4df1ba6 iOS: Enable modules_unittests and common_audio_unittests
Instead of excluding the whole test binaries, only exclude the parts that cause the
compilation to fail for modules_unittests and common_audio_unittests.

BUG=webrtc:4752, webrtc:4755, webrtc:5544
TESTED=Successful build with:
GYP_DEFINES='OS=ios target_arch=x64' webrtc/build/gyp_webrtc
ninja -C out/Debug-iphonesimulator modules_unittests common_audio_unittests
NOTRY=True

Review URL: https://codereview.webrtc.org/1698033002

Cr-Commit-Position: refs/heads/master@{#11675}
2016-02-18 21:15:17 +00:00
kjellander@webrtc.org
f22695c3d8 Remove build_with_libjingle and exclude failing iOS tests from 'All' target.
This will make it possible to remove the build_with_libjingle=1 and key=''
GYP_DEFINES the bots are using (https://codereview.chromium.org/1450313002/).
It will also pave the road for enabling more WebRTC native tests on iOS.

BUG=webrtc:4755,webrtc:3185,webrtc:5165
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
Local compilation with:
GYP_DEFINES='OS=ios target_arch=arm' webrtc/build/gyp_webrtc
ninja -C out/Release-iphoneos
GYP_DEFINES='OS=ios target_arch=arm chromium_ios_signing=0' webrtc/build/gyp_webrtc
ninja -C out/Release-iphoneos
GYP_DEFINES='OS=ios target_arch=arm64' webrtc/build/gyp_webrtc
ninja -C out/Release-iphoneos
GYP_DEFINES='OS=ios target_arch=ia32' webrtc/build/gyp_webrtc
ninja -C out/Release-iphonesimulator
GYP_DEFINES='OS=ios target_arch=x64' webrtc/build/gyp_webrtc
ninja -C out/Release-iphonesimulator

R=henrika@webrtc.org

Review URL: https://codereview.webrtc.org/1457053003 .

Cr-Commit-Position: refs/heads/master@{#10711}
2015-11-19 14:39:54 +00:00
peah
48407f7122 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}
2015-11-09 13:24:56 +00:00
Andrew MacDonald
ac4234ccfc 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}
2015-06-25 01:25:59 +00:00
Henrik Kjellander
f2497cf517 Fix unknown option '-msse2' warning
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43169004

Cr-Commit-Position: refs/heads/master@{#9016}
2015-04-16 06:57:12 +00:00
Alejandro Luebs
a9c0ae284c 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}
2015-04-14 22:51:22 +00:00
Richard Coles
d417c93c10 Remove android_webview_build conditions.
Now that android_webview_build is no longer supported, remove build
conditionals referencing it and also remove the extra level of
indirection used to reference the cpufeatures target.

BUG=chromium:440793
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44119005

Patch from Richard Coles <torne@chromium.org>.

Cr-Commit-Position: refs/heads/master@{#8963}
2015-04-09 15:36:13 +00:00
andrew@webrtc.org
04c50981f8 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
2015-03-19 20:07:43 +00:00
andrew@webrtc.org
d2c09dd339 Make building openmax_dl conditional in gyp.
Intentionally not modifying the GN build.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48479004

Cr-Commit-Position: refs/heads/master@{#8688}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8688 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-11 22:07:18 +00:00
andrew@webrtc.org
0933d01d09 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
2015-03-05 19:14:21 +00:00
kjellander@webrtc.org
035e9123e9 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
2015-01-28 19:57:44 +00:00
andrew@webrtc.org
041035b390 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
2015-01-26 21:23:53 +00:00
kjellander@webrtc.org
d7e34e1086 Make it easier to use external libyuv + cleanup GYP files.
It is now easier to use an external libyuv library.
Fix some GYP errors.
Remove the temporary webrtc_base target (depends on
https://codereview.chromium.org/865603002/ being landed
first).

BUG=4185
R=andresp@webrtc.org, andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8154 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 19:17:26 +00:00
andrew@webrtc.org
6b6301588e 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
2015-01-15 00:09:53 +00:00
andresp@webrtc.org
86e1e487e7 Move system_wrappers.gyp files to the proper directory.
Build targets should not refer to non-subpaths as was happening before when
 source/system_wrappers.gyp refers to ../interface/ files.

R=kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 09:30:52 +00:00
andrew@webrtc.org
d730b288c8 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
2015-01-07 21:34:23 +00:00
andrew@webrtc.org
1090a6eccf 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
2014-12-18 21:36:18 +00:00
andrew@webrtc.org
1153322cf8 Build fix for MIPS Android Webview build.
Excluding optimizations to support MIPS32R6 platform for Android Webview build (see also https://code.google.com/p/webrtc/source/detail?r=7580).

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7729 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-21 16:28:32 +00:00
andrew@webrtc.org
a3ed713dad 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
2014-10-31 21:51:03 +00:00
kjellander@webrtc.org
78c222bfae Update all .isolate files for the new format.
R=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27809004

Patch from Marc-Antoine Ruel <maruel@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7583 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 18:08:09 +00:00
andrew@webrtc.org
508c91683c Build fix for MIPS32R6.
Exclude MIPS optimizations for MIPS32R6 build since some of the instructions
are not supported. This is temporary fix, until the MIPS32R6 code is added.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25989004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7580 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 16:26:17 +00:00
henrik.lundin@webrtc.org
8aa4d2d2cd 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
2014-10-30 13:23:25 +00:00
andrew@webrtc.org
aada86b261 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
2014-10-27 18:18:17 +00:00
bjornv@webrtc.org
fab5439112 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
2014-10-15 04:38:42 +00:00
andrew@webrtc.org
4165f7aa22 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
2014-10-08 18:01:27 +00:00
andrew@webrtc.org
325cff01b4 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
2014-10-01 17:42:18 +00:00
kjellander@webrtc.org
0372b93118 Partial revert of r7014 (Android APK refactor)
This reverts selected parts of r7014 to enable
rolling WebRTC in Chromium DEPS.

This works around the problem with GYP includes
being processed in the first pass (i.e. variables
cannot be used for paths). Using a dependency with
a path using a variable that is conditioned for
build_with_chromium being 0 or 1 solves the Chromium
build.

These changes will be restored once I've finished
a major GYP refactoring that will break out all
test related code (at least the parts that includes
the Android APK targets) into a separate chain
of GYP targets that are not processed when generating
projects for Chromium (which is why r7014 is breaking
the Chromium build).

BUG=3741
TESTED=Passing compilation of standalone using:
GYP_DEFINES="OS=android component=static_library fastbuild=1 target_arch=arm" webrtc/build/gyp_webrtc
ninja -C out/Debug
Then verified the *_apk  targets are generated and compiled.

Passing compilation from a Chromium checkout with third_party/webrtc
directory removed and a new empty third_party/webrtc mapped to the
standalone checkout using:
sudo mount --bind /path/to/trunk/webrtc third_party/webrtc
Then running build/gyp_chromium
I also verified WebRTC GYP targets exist and are able to compile.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7040 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 14:34:46 +00:00
kjellander@webrtc.org
3bd4156d75 Android APK tests built from a normal WebRTC checkout.
Restructure how the Android APK tests are compiled now
that we have a Chromium checkout available (since r6938).

This removes the need of several hacks that were needed when
building these targets from inside a Chromium checkout.
By creating a symlink to Chromium's base we can compile the required
targets. This also removes the need of the previously precompiled
binaries we keep in /deps/tools/android at Google code.

All the user needs to do is to add the target_os = ["android"]
entry to his .gclient as described at
https://code.google.com/p/chromium/wiki/AndroidBuildInstructions

Before committing this CL, the Android APK buildbots will need
to be updated.
This also solves http://crbug.com/402594 since the apply_svn_patch.py
usage will be similar to the other standalone bots.
It also solves http://crbug.com/399297

BUG=chromium:399297, chromium:402594
TESTED=Locally compiled all APK targets by running:
GYP_DEFINES="OS=android include_tests=1 enable_tracing=1" gclient runhooks
ninja -C out/Release

checkdeps

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

Review URL: https://webrtc-codereview.appspot.com/22149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7014 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 11:06:37 +00:00
andrew@webrtc.org
af7fdfcde8 Add LTO support for Android Chromium.
This is to add support for a Link-Time Optimizations experiment in Android Chromium. As it is disabled by default, it won't change anything for most configurations.
BUG=chromium:407544
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7009 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-29 17:41:13 +00:00
kwiberg@webrtc.org
877083c4d4 New utility class for easy debug dumping to WAV files
There are currently a number of places in the code where we dump audio
data in various stages of processing for debug purposes. Currently
these all write raw, uncompressed PCM files, which isn't supported by
the most common audio players, and requires the user to supply
metadata such as sample rate, sample size and endianness, etc.

This patch adds a simple class that makes it easy to write WAV files
instead. WAV files still contain the same uncompressed PCM data, but
they have a small header that contains all the requisite metadata, and
are supported by virtually all audio players.

Since some of the debug code that will be writing WAV files is written
in plain C, a C API is included as well.

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

Review URL: https://webrtc-codereview.appspot.com/16809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6932 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 07:42:46 +00:00
henrike@webrtc.org
e6e139159f Android: cleanup gtest_target_type conditions.
Ever since crrev.com/133053 OS==android implies:
gtest_target_type=shared_library

Similar to Chromium's crrev.com/271222 where base.gyp's conditions are changed
(which the affected conditions in this cl comes from).

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

Review URL: https://webrtc-codereview.appspot.com/13539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6332 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 20:46:50 +00:00
aluebs@webrtc.org
37ca765650 Add fir_filter to common_audio
It has 3 implementation:
* fir_filter_c with no optimization
* fir_filter_sse which outperforms the C version by a factor of 3x
* fir_filter_neon which outperforms the C version by a factor of 2x

R=andrew@webrtc.org, bjornv@webrtc.org, johannkoenig@google.com

Review URL: https://webrtc-codereview.appspot.com/9759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5756 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-24 10:16:11 +00:00
kjellander@webrtc.org
917306d3fd Change uses of the obsolete armv7 setting to arm_version==7.
BUG=http://crbug.com/234135
R=andrew@webrtc.org, fischman@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5369004

Patch from Mostyn Bramley-Moore <mostynb@opera.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5250 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-10 09:26:07 +00:00
kjellander@webrtc.org
2a97317953 Fix include of isolate.gypi
Recent changes in GYP seem to have broken our previous
"hack" for getting the GYP rule for .isolate files
imported from the Chromium build/isolate.gypi.

The best solution for now is to remove the hack
and check in a copy of Chromium's src/build/isolate.gypi
in WebRTC's build/ dir instead. A similar approach is
used for our build/protoc.gypi file.

TEST=On Linux, I successfully ran:
gclient runhooks
ninja -C out/Release
and verified a bunch of .isolated files were created in
out/Release (which didn't happen before this patch).

I also renamed the build/isolate.gypi from Chromium to
ensure that our own is used and not that one (in case any
paths would be incorrect).

I also ran build/gyp_chromium in a Chromium checkout
with WebRTC in third_party/webrtc having this patch applied
to ensure GYP processing was still working.

Finally, I verified that the same project generation and
compilation from a Chromium checkout worked the way we build
our Android native tests, using:
. build/android/envsetup.sh
GYP_DEFINES="$GYP_DEFINES include_tests=1 enable_tracing=1" gclient runhooks
ninja -C out/Release android_builder_webrtc

BUG=1916
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2338004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4907 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-02 19:31:16 +00:00
andrew@webrtc.org
8bf755d5c5 MIPS optimizations for the functions WebRtcSpl_SqrtFloor, WebRtcSpl_CrossCorrelation, WebRtcSpl_ScaleAndAddVectorsWithRound and the inline functions from signal_processing spl_inl.h file.
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1791004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4779 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-18 17:40:46 +00:00
fischman@webrtc.org
31b4a5ac82 Recognize armv7 target_arch for ios support in webrtc common.gyp
BUG=2343
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2176004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4684 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 16:46:36 +00:00
kjellander@webrtc.org
3365422c41 Isolate GYP target and .isolate files for tests
This is a re-land attempt of http://review.webrtc.org/1673004/
It now includes a build/isolate.gypi in WebRTC that includes the same
file as the one that would be included when WebRTC is used in a Chromium
checkout. It is needed since it is not possible to use variables in GYP's
includes sections.

Implemented according to the instructions at
http://www.chromium.org/developers/testing/isolated-testing

Workflow has been like this:
1. create _run GYP target
2. create a stripped down .isolate file
3. export GYP_DEFINES="$GYP_DEFINES test_isolation_mode=check"
4. runhooks
5. compile
6. test if the test would run (i.e. find it's dependencies) without
   actually executing it:
   tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
7. If failing, run the fix_test_cases.py script like this:
   tools/swarm_client/googletest/fix_test_cases.py --isolated out/Release/testname.isolated

All tests that run on the bots for WebRTC has got _run target
and .isolate file created.

"Normal tests" that run fine on any machine:
* audio_decoder_unittests
* common_audio_unittests
* common_video_unittests
* metrics_unittests
* modules_tests
* modules_unittests
* neteq_unittests
* system_wrappers_unittests
* test_support_unittests
* tools_unittests
* video_engine_core_unittests
* voice_engine_unittests

Tests that requires bare-metal and audio/video devices:
* audio_device_tests
* video_capture_tests

I also added the isolate boilerplate code for the following
tests that are not yet pure gtest binaries (which means they
cannot run isolated yet):
* video_render_tests
* vie_auto_test
* voe_auto_test

TEST=running isolate.py as described above. WebRTC trybots passing. Created a Chromium checkout with third_party/webrtc ToT and this patch applied, passing the runhooks step.
BUG=1916
R=henrike@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2056004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4590 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-22 07:57:00 +00:00
kjellander@webrtc.org
4298f73031 Revert 4547 "Isolate GYP target and .isolate files for tests"
As this breaks the FYI bots in 
http://build.chromium.org/p/chromium.webrtc.fyi/waterfall
due to different path to isolate.gypi (which cannot easily
be resolved due to limitations in GYP)

> Isolate GYP target and .isolate files for tests
> 
> Implemented according to the instructions at
> http://www.chromium.org/developers/testing/isolated-testing
> 
> Workflow has been like this:
> 1. create _run GYP target
> 2. create a stripped down .isolate file
> 3. export GYP_DEFINES="$GYP_DEFINES test_isolation_mode=check"
> 4. runhooks
> 5. compile
> 6. test if the test would run (i.e. find it's dependencies) without
>    actually executing it:
>    tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
> 7. If failing, run the fix_test_cases.py script like this:
>    tools/swarm_client/fix_test_cases.py --isolated out/Release/testname.isolated
> 
> All tests that run on the bots for WebRTC has got _run target
> and .isolate file created.
> 
> "Normal tests" that run fine on any machine:
> * audio_decoder_unittests
> * common_audio_unittests
> * common_video_unittests
> * metrics_unittests
> * modules_integrationtests
> * modules_unittests
> * neteq_unittests
> * system_wrappers_unittests
> * test_support_unittests
> * tools_unittests
> * video_engine_core_unittests
> * voice_engine_unittests
> 
> Tests that requires bare-metal and audio/video devices:
> * audio_device_integrationtests
> * video_capture_integrationtests
> 
> I also added the isolate boilerplate code for the following
> tests that are not yet pure gtest binaries (which means they
> cannot run isolated yet):
> * video_render_integrationtests
> * vie_auto_test
> * voe_auto_test
> 
> TEST=running isolate.py as described above.
> BUG=1916
> R=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1673004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2040004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4548 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 11:29:58 +00:00
kjellander@webrtc.org
d7a4d235d2 Isolate GYP target and .isolate files for tests
Implemented according to the instructions at
http://www.chromium.org/developers/testing/isolated-testing

Workflow has been like this:
1. create _run GYP target
2. create a stripped down .isolate file
3. export GYP_DEFINES="$GYP_DEFINES test_isolation_mode=check"
4. runhooks
5. compile
6. test if the test would run (i.e. find it's dependencies) without
   actually executing it:
   tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
7. If failing, run the fix_test_cases.py script like this:
   tools/swarm_client/fix_test_cases.py --isolated out/Release/testname.isolated

All tests that run on the bots for WebRTC has got _run target
and .isolate file created.

"Normal tests" that run fine on any machine:
* audio_decoder_unittests
* common_audio_unittests
* common_video_unittests
* metrics_unittests
* modules_integrationtests
* modules_unittests
* neteq_unittests
* system_wrappers_unittests
* test_support_unittests
* tools_unittests
* video_engine_core_unittests
* voice_engine_unittests

Tests that requires bare-metal and audio/video devices:
* audio_device_integrationtests
* video_capture_integrationtests

I also added the isolate boilerplate code for the following
tests that are not yet pure gtest binaries (which means they
cannot run isolated yet):
* video_render_integrationtests
* vie_auto_test
* voe_auto_test

TEST=running isolate.py as described above.
BUG=1916
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1673004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4547 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 10:02:06 +00:00
henrike@webrtc.org
89c674053e Adds all unittests to android NDK-APK framework.
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1872004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4474 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-02 16:53:47 +00:00