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BUG=chromium:436400
TBR=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27259004
git-svn-id: http://webrtc.googlecode.com/svn/branches/40/webrtc@7759 4adac7df-926f-26a2-2b94-8c16560cd09d
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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/webrtc@7585 4adac7df-926f-26a2-2b94-8c16560cd09d
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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/webrtc@7583 4adac7df-926f-26a2-2b94-8c16560cd09d
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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/webrtc@7580 4adac7df-926f-26a2-2b94-8c16560cd09d
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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/webrtc@7570 4adac7df-926f-26a2-2b94-8c16560cd09d
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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/webrtc@7538 4adac7df-926f-26a2-2b94-8c16560cd09d
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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/webrtc@7451 4adac7df-926f-26a2-2b94-8c16560cd09d
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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/webrtc@7397 4adac7df-926f-26a2-2b94-8c16560cd09d
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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/webrtc@7359 4adac7df-926f-26a2-2b94-8c16560cd09d
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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/webrtc@7040 4adac7df-926f-26a2-2b94-8c16560cd09d
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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/webrtc@7014 4adac7df-926f-26a2-2b94-8c16560cd09d
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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/webrtc@7009 4adac7df-926f-26a2-2b94-8c16560cd09d
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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/webrtc@6932 4adac7df-926f-26a2-2b94-8c16560cd09d
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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/webrtc@6332 4adac7df-926f-26a2-2b94-8c16560cd09d
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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/webrtc@5756 4adac7df-926f-26a2-2b94-8c16560cd09d
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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/webrtc@5250 4adac7df-926f-26a2-2b94-8c16560cd09d
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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="$GYP_DEFINES include_tests=1 enable_tracing=1" gclient runhooks
ninja -C out/Release android_builder_webrtc
BUG=1916
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2338004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4907 4adac7df-926f-26a2-2b94-8c16560cd09d
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WebRtcSpl_CrossCorrelation, WebRtcSpl_ScaleAndAddVectorsWithRound and the inline functions from signal_processing spl_inl.h file.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1791004
Patch from Ljubomir Papuga <lpapuga@mips.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4779 4adac7df-926f-26a2-2b94-8c16560cd09d
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BUG=2343
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2176004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4684 4adac7df-926f-26a2-2b94-8c16560cd09d
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This is a re-land attempt of http://review.webrtc.org/1673004/
It now includes a build/isolate.gypi in WebRTC that includes the same
file as the one that would be included when WebRTC is used in a Chromium
checkout. It is needed since it is not possible to use variables in GYP's
includes sections.
Implemented according to the instructions at
http://www.chromium.org/developers/testing/isolated-testing
Workflow has been like this:
1. create _run GYP target
2. create a stripped down .isolate file
3. export GYP_DEFINES="$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/webrtc@4590 4adac7df-926f-26a2-2b94-8c16560cd09d
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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="$GYP_DEFINES test_isolation_mode=check"
> 4. runhooks
> 5. compile
> 6. test if the test would run (i.e. find it's dependencies) without
> actually executing it:
> tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
> 7. If failing, run the fix_test_cases.py script like this:
> tools/swarm_client/fix_test_cases.py --isolated out/Release/testname.isolated
>
> All tests that run on the bots for WebRTC has got _run target
> and .isolate file created.
>
> "Normal tests" that run fine on any machine:
> * audio_decoder_unittests
> * common_audio_unittests
> * common_video_unittests
> * metrics_unittests
> * modules_integrationtests
> * modules_unittests
> * neteq_unittests
> * system_wrappers_unittests
> * test_support_unittests
> * tools_unittests
> * video_engine_core_unittests
> * voice_engine_unittests
>
> Tests that requires bare-metal and audio/video devices:
> * audio_device_integrationtests
> * video_capture_integrationtests
>
> I also added the isolate boilerplate code for the following
> tests that are not yet pure gtest binaries (which means they
> cannot run isolated yet):
> * video_render_integrationtests
> * vie_auto_test
> * voe_auto_test
>
> TEST=running isolate.py as described above.
> BUG=1916
> R=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1673004
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2040004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4548 4adac7df-926f-26a2-2b94-8c16560cd09d
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Implemented according to the instructions at
http://www.chromium.org/developers/testing/isolated-testing
Workflow has been like this:
1. create _run GYP target
2. create a stripped down .isolate file
3. export GYP_DEFINES="$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/webrtc@4547 4adac7df-926f-26a2-2b94-8c16560cd09d
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BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1872004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4474 4adac7df-926f-26a2-2b94-8c16560cd09d
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This CL will lower the number of test targets in WebRTC by:
Add common_audio_unittests and merge the following targets into it (copied from http://review.webrtc.org/1584006):
* resampler_unittests
* signal_processing_unittests
* vad_unittests
Merge into modules_unittests:
* bitrate_controller_unittests
* desktop_capture_unittests
* media_file_unittests
* remote_bitrate_estimator_unittests
* rtp_rtcp_unittests
* paced_sender_unittests
Merge into test_support_unittests:
* channel_transport_unittests
channel_transport.gyp was also removed in favor for test.gyp.
I had to remove a main method from rtcp_format_remb_unittest.cc
since it caused the fileutils.h code to not be able to find the
right project root path in ordrer to provide correct paths
to test files.
Buildbot configuration update will be synced with the commit of this CL.
TEST=trybots
BUG=1843
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4213 4adac7df-926f-26a2-2b94-8c16560cd09d
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WebRtcSpl_ComplexBitReverse, WebRtcSpl_ComplexFFT, WebRtcSpl_ComplexIFFT, WebRtcSpl_DownsampleFast and WebRtcSpl_FilterARFastQ12.
Also, moved the common table used in complex_fft functions to a separate header file (webrtc/common_audio/signal_processing/include/complex_fft_tables.h).
R=andrew@webrtc.org, kma@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1126004
Patch from Ljubomir Papuga <lpapuga@mips.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4141 4adac7df-926f-26a2-2b94-8c16560cd09d
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* Brings in on-the-fly sample ratio updates (or varispeed) with minor modifications to build in webrtc.
* Moved SSE and NEON optimized functions into their own files to handle run-time detection properly. NEON optimizations now enabled.
TESTED=unit tests and ran voe_cmd_test loopback with both devices using 44.1 kHz to exercise SincResampler in real-time.
R=dalecurtis@chromium.org, kma@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1438004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3987 4adac7df-926f-26a2-2b94-8c16560cd09d
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In principle should reduce gyp processing time, but the difference was not measurable. In any case, it's a good simplification that aligns with having a single common_video target.
R=bjornv@webrtc.org, kma@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1375004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3928 4adac7df-926f-26a2-2b94-8c16560cd09d
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TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
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