| Commit message (Collapse) | Author | Age | Files | Lines |
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Current makefile compiles the resampler only for 64 bit environment.
Allow compilation for 32 bit environment as well.
CRs-Fixed: 735776
Change-Id: I626725551af73fc5ea98a7dbf87cacea9dbbc0ef
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6.0' as a single patch.
Signed-off-by: jinamdar <jaydeep.inamdar@dts.com>
(cherry picked from commit d3668da66643d4cc39058fb65c8db0742748f70f)
Conflicts:
services/audioflinger/AudioFlinger.cpp
services/audioflinger/Threads.cpp
Change-Id: I67e3ba100ff40058919ba827b806aea7bdbaf4bb
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Android 6.0.0 release 26
Change-Id: I8a57007bf6efcd8b95c3cebf5e0444345bdd4cda
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- Move SchedulingPolicyService from audioservice to mediautils
- When starting up a high speed stream config, set request queue thread
to SCHED_FIFO using SchedulingPolicyService
Bug: 24227252
Change-Id: I224b59142bd111caf563779f55cddd62385b9bac
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Add support for QTI audio resampler
audio: Audio resampler support for 192Khz playback
Change-Id: Ia8f24a0874ebf6e16ef7bd1f2759a14f47149875
audio: Add audio mixing support for qti resampler
Change-Id: Ib657aa12b2a72323564148c302ff8891e1bb7433
AudioMixer: Extend use of QTI resampler for 44.1Khz sampling rate
Change-Id: I2a819dbc9f1e3e280cb4fa79328e331883a3e981
AudioMixer: fill 0s at right place when no more buffers available
Change-Id: I50504c5a02eb0c69abfc9b047792b0f6f85b9ce8
audioflinger: add channel count check to use QTI resampler
Change-Id: I8f76dd82b72a0dd8b77343e77e0d0545e1be2114
Change-Id: Ia8f24a0874ebf6e16ef7bd1f2759a14f47149875
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The llvm bug https://llvm.org/bugs/show_bug.cgi?id=21572 was fixed.
BUG: 18373866
Change-Id: Ia529bf53267f636880515ccefb4ca1cf7d731baf
(cherry picked from commit 7630881d853b130e2c1f11cb2dafebe684bcfa91)
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Using Sonic as backbone for time stretching algorithm.
Adding libsonic to needed makefiles.
bug: 19196501
Change-Id: I1ea9221d2f56e4e79fba8746ce0ad350b5079e82
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In preparation for playback rate support and timestretching.
Bug: 19196501
Change-Id: I435accb852d32110dd0b3a9917488522c567ba80
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Create an interface layer between the AudioFlinger and the HAL
that manages the wrapping and format conversion.
Removed unnecessary includes.
Handle rate conversion in getRenderPosition().
Try to open HAL with encoded format before wrapping with SPDIF.
Bug: 17566660
Change-Id: I00ad888ca15ff0f85b85efb8167c7f5ea761a244
Signed-off-by: Phil Burk <philburk@google.com>
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Change-Id: I8e44dbfe02338622eb69193b234743b50f0dd79f
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Change-Id: Ic99890bbba4f856b65535f3df0f928de9e3e9748
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lmp-mr1-dev
* commit '78c02ddb6bf0d676160244a62c11dceb68301b73':
make libserviceutility a shared lib
audio policy: fix remote mic capture
audio policy: suppport for dynamic source
Fix overload of SoftVideoDecoderOMXComponent::updatePortDefinitions
Fix race condition in signaling completion for decode.
AnotherPacketSource: need reset some members before returning from queueDiscontinuity().
Fix looping sound playback
AnotherPacketSource.cpp: Do not queue discontinity signal buffer resulted from seek.
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so that we have only one getpid_cached in mediaserver process
bug: 18919657
Change-Id: Iff3cd932c9110e874b3885f79705f49bf3e3f1fc
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Clang++ chokes on AudioMixer.cpp.
It does not know "how to split this operator."
BUG: 18373866
Change-Id: I3d588d44596c7c3b6f97e5f822545e015b074206
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Change-Id: Ib91ff0696ab2472c62168eb5261bbda5d22f623e
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Use floating point volume in AudioMixer mixing when floating
point input is used with the new mixer engine.
AudioResampler is updated to take floating point volume to match.
Both legacy integer and floating point mixer engines work.
For now, integer volume is used when the new mixer engine
runs in integer input mode, for backward compatibility with
the legacy mixer. The new mixer engine will generally run in
floating point input mode. When the legacy path is removed,
the integer volumes will be removed.
Change-Id: I79e80c292ae7c8b8bdd0aa371a1b2c3a1b618290
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This version supports at most one fast capture client.
Change-Id: Idf609bfc80ae22433433d66a5232c043c65506df
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Added a new PatchPanel subclass to AudioFlinger
to handle audio ports and audio patches configuration
and connection.
The first implementation does not add new functionnality.
AudioPolicyManager uses patch panel interface to control
device routing.
AudioFlinger:
- Added PatchPanel class. The first implementation does not
add new functionnality. PatchPanel handles routing commands
for audio HAL after 3.0 or converts to setParameters for audio
HALs before 3.0.
- Added config events to ThreadBase to control synchronized
audio patch connection.
AudioPolicyManager:
- Use PatchPanel API to control device selection isntead of setParameters.
- New base class AudioPort common to audio device descriptors
and input output stream profiles. This class is RefBase and groups
attributes common to audio ports.
- Use same device selection flow for input as for outputs:
getNewInputDevice -> getDeviceForInptusiource -> setInputDevice
Change-Id: Idaa5a883b19a45816651c58cac697640dc717cd9
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Change-Id: I4c6f7b8f88fcf107bb29ee6432feecd4ab6554d2
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Change-Id: I3f07493375ace6e5cfdcd02ad90c4b6fad543b0c
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Change-Id: Ifd5385ad42a81e02e6a6afc6281f09fbff361671
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libaudioresampler is available in both 32-bit and 64-bit,
unlike libaudioflinger which is currently 32-bit only.
Bug: 8141282
Change-Id: I839f7b4e6aaed6984012ca6d514323f927669df6
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Change-Id: Ibc3ef07aa9860b7fd4f9aaff27b0dbe0dcbf1cbf
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Change-Id: Ic579d346c27ff05ea6444faaa60fa6caaec86fbf
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also 32-bit only command-line apps
Change-Id: I9ac557a8d02bbf6986a9b5c3cdce23d400b306a3
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This does not affect the existing resamplers.
New resampler accessed through additional quality settings:
DYN_LOW_QUALITY = 5
DYN_MED_QUALITY = 6
DYN_HIGH_QUALITY = 7
Change-Id: Iebbd31871e808a4a6dee3f3abfd7e9dcf77c48e1
Signed-off-by: Andy Hung <hunga@google.com>
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This will reduce code duplication, and allow us take advantage of more
advanced capabilities of libsndfile in the future.
Change-Id: I25fa2b6d0c21e325aeaf05bda62cf7aab0c5deb4
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Bug: 8834855
Change-Id: I4cd842cdfb09d2aaaaab9df9ac3bec6179709bd3
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we export only symbols needed by clients of this library.
this saves about 130KB (1/3rd of the lib size)
Change-Id: Id81f3ecb299ee3abc0811915cf6efe87180bf15c
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Bug: 8580410
Change-Id: If493d87d60d71be664ad75b140c62acadb75b0d0
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Change-Id: If7e2bc9b2a216524ee9cbb68682e2634933b4973
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Bug: 8223560
Change-Id: Iddbfb06c45d43d9f20bb428215dd4094931e19a7
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Implement rotation to reduce long-term storage use.
Implement optional per-track tee.
Dynamically enable at runtime based on property, instead of at compile-time.
Dynamic frame count not yet implemented.
Bug: 8223560
Change-Id: I3706443c6ec0cb0c6656dc288715a02ad5fea63a
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With ANDROID_SMP set, tracing functionality is completely inline, and
without the performance hits of external library calls, tracing does
not need to be conditionally compiled.
Change-Id: I4b29a9a52c403f0d2ea137c5b7bc05a518a7ca4b
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Tracing functions are meant to be dynamically controlled via sysprops.
Conditional compilation removes this functionality.
Change-Id: I26bc473d104d0b3c50a228dddfda3fa2428d157a
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Audioflinger.cpp and Audioflinger.h files must be split to
improve readability and maintainability.
This CL splits the files as follows:
AudioFlinger.cpp split into:
- AudioFlinger.cpp: implementation of IAudioflinger interface and global methods
- AFThreads.cpp: implementation of ThreadBase, PlaybackThread, MixerThread,
DuplicatingThread, DirectOutputThread and RecordThread.
- AFTracks.cpp: implementation of TrackBase, Track, TimedTrack, OutputTrack,
RecordTrack, TrackHandle and RecordHandle.
- AFEffects.cpp: implementation of EffectModule, EffectChain and EffectHandle.
AudioFlinger.h is modified by inline inclusion of header files containing
the declaration of complex inner classes:
- AFThreads.h: ThreadBase, PlaybackThread, MixerThread, DuplicatingThread,
DirectOutputThread and RecordThread
- AFEffects.h: EffectModule, EffectChain and EffectHandle
AFThreads.h includes the follownig headers inline:
- AFTrackBase.h: TrackBase
- AFPlaybackTracks: Track, TimedTrack, OutputTrack
- AFRecordTracks: RecordTrack
Change-Id: I512ebc3a51813ab7a4afccc9a538b18125165c4c
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Change-Id: I4f5e95a5ddf016530d1b2747a0a5ca0962caabda
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Bug: 6654403
Change-Id: I3993d62987cf0dd85db10bf002a5cce53d4f01bd
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Change-Id: I66852d90d384f1d9e77b51ad1a1ebdbaf61d0607
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cubic resampler was disabled because it hadn't been qualified,
however after I did some tests, it does improve significantly
the sound quality over the order-1 resampler, even if it is
still quite bad.
also HIGH_QUALITY resampler was partially disabled, it's now
fully enabled. It's a big improvement over the cubic resampler
in terms of aliasing noise (it's not as good in the pass-band).
Change-Id: I70e3658c255896588642697be9eb594ff4ec0f8b
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Bug: 7229644
Change-Id: I93bde36be1c3ec84174a4c98423e28f8b3d8782f
Signed-off-by: ty.lee <ty.lee@lge.com>
Signed-off-by: Iliyan Malchev <malchev@google.com>
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Summary:
Very high quality is enabled only for 44.1 -> 48 or 48 -> 44.1,
and uses low quality for all other use cases.
Track estimated CPU load and throttles the quality based on load;
as currently configured it should allow up to 2 instances of very high quality.
Medium quality and high quality are currently disabled unless explicitly requested.
Details:
Only load .so the first time it is needed.
Cleanup code style: formatting, indentation, whitespace.
Restore medium quality resampler, but it is not used (see next line).
Fix memory leak for sinc resampler.
Check sample rate in resampler constructor.
Add logs for debugging.
Rename DEFAULT to DEFAULT_QUALITY for consistency with other quality levels.
Renumber VERY_HIGH_QUALITY from 255 to 4.
Use enum src_quality consistently.
Improve parsing of property af.resampler.quality.
Fix reentrancy bug - allow an instance of high quality and an instance
of very high quality to both be active concurrently.
Bug: 7229644
Change-Id: I0ce6b913b05038889f50462a38830b61a602a9f7
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It's not critical, and is wasting power
Bug: 7241714
Change-Id: I6ad4375f0000c92529688723dbe0ff0caa809c5d
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-Add a separate quality VERY_HIGH_QUALITY in resampler
-Use resample coefficients audio-resampler library for
quality VERY_HIGH_QUALITY.
-This improves the quality of resampled output.
Bug: 7024293
Change-Id: Ia44142413bed5f5963d7eab7846eec877a2415e4
Signed-off-by: Iliyan Malchev <malchev@google.com>
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libnbaio is now a separate shared library from AudioFlinger, rather
than a static library used only by AudioFlinger.
AudioBufferProvider interface is now also independent of AudioFlinger,
moved to include/media/
Change-Id: I9bb62ffbc38d42a38b0af76e66da5e9ab1e0e21b
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Change-Id: I73a2afe72d8acb53e57e6b4e6fb5133e22b7875a
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Certain CPUs with dynamic cluster swapping and hotplug
don't report CPU frequency accurately. The file descriptors
used to read the frequency become stale and report bogus data.
So make this feature a build time option for debugging only.
This will also improve performance of the fast mixer loop.
Change-Id: I602f81ec3281a37992769208be08084ed1469e8c
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