| Commit message (Collapse) | Author | Age | Files | Lines |
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- Handle new ADSP status parameter
- media/libmedia: Add new ADSP status audio parameter
- framework/av: Add handling of new key-pair value in
Audio Flinger
- Handle Tunnel mode SubSys Restart
- framework/av: Post SSR event to Audio Flinger
- media/libmedia: Post SSR event to AudioTrack
- media/libmediaplayerservice: Post SSR event to
MediaPlayerService
- media/libstagefright: Post SSR event to TunnelPlayer
Change-Id: I8c8385af45be91caf7d7160ab2e0236d6591b159
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- After a pause and resume, tunnel playback volume is always
set to maximum irrespective of the volume value before pause.
- The cause for this is, the stream volume is not used to set
the volume in directaudiotrack.
- Fix is to use the stream volume to set volume during tunnel
playback.
Change-Id: I59cda146ed88bd5c4186aeb9ae5d165f4a27493f
CRs-fixed: 452285
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- Call stream->set_volume to apply volume on
particular stream of DirectOutputs
Change-Id: I98e3b856ee508d407f893afad66caade5eda3e4a
CRs-Fixed: 445953
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https://android.googlesource.com/platform/frameworks/av into 1.1
Android 4.2.2 release 1
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Bug: 6490974
Change-Id: Ib926a9258bde4ee05ed42eea662dff68e426a997
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Change-Id: Ie6c982af906dcfd3cdea4b771dfab1f7e47745ca
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Squashed commit of the following:
commit 12b6952da9f25e94d06dd7185bce255924e7e791
Author: Mathias Agopian <mathias@google.com>
Date: Mon Nov 19 15:27:26 2012 -0800
fix a typo in SINC resampler that prevented tracks to be mixed
we were always erasing the current mix instead of mixing into it.
Change-Id: Ib229245f9e5a0d384f1727640a59e9f0469211a2
commit 0019ce082df430278f14ab922e900ce33b64897d
Author: Dave Bort <dbort@google.com>
Date: Tue Nov 13 01:30:32 2007 -0800
Rename "TARGET" to "MODULE" in the build system.
Part one of the grand renaming.
API_CHANGE: Third parties may need to update their makefiles.
Any variables with "LOCAL" and "TARGET" in their names
should now use "MODULE" instead of "TARGET"; e.g., LOCAL_MODULE,
LOCAL_MODULE_TAGS.
PRESUBMIT=passed
OCL=39840
Change-Id: Ica9a7937d3d9552ab84db46ac6eea8a290e404fe
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit f01adc0cef0e39e75c76d9195ac26a94cac0a100
Author: Glenn Kasten <gkasten@google.com>
Date: Wed Nov 14 08:32:08 2012 -0800
Fix build warnings
Change-Id: Ic43bcca166a529a6431711b05a7fa21849b6a38b
commit 9bb031a565c753a03d9c9397edea318947d80528
Author: Mathias Agopian <mathias@google.com>
Date: Sat Nov 10 04:44:30 2012 -0800
more optimizations...
calculate the offsets from the phase differently, this happens
to reduce the register pressure in the main loop, which in turns
allows the compiler to generate much better code (doesn't need
to spill a lot of stuff on the stack).
this gives another 15% performance increase
Change-Id: I2ce3479dd48b9e6941adb80e6d443d6e14d64d96
commit 5a951598f31217b8cd2babd0720c9608ee17291a
Author: Mathias Agopian <mathias@google.com>
Date: Sat Nov 10 03:26:39 2012 -0800
refactor code to improve neon code
we want to make sure we don't transfer data from the
neon unit to the arm register file, as this can be quite
slow. instead we do all the calculation on the neon side
and write the result directly to main memory.
Change-Id: Ibb56664d3ab03098ae2798b75e2b6927ac900187
commit b381ee9e83bc9fd18986e79c7809841514ed590e
Author: Mathias Agopian <mathias@google.com>
Date: Sun Nov 4 15:16:13 2012 -0800
NEON optimized SINC resampler
this currently gives us a 60% to 80% boost depending
on the quality level selected.
Change-Id: I7db385007e811ed7bffe5fd3403b44e300894f5b
commit bea077354210242ea193a50b0dbab0fedab25df3
Author: Mathias Agopian <mathias@google.com>
Date: Mon Nov 5 01:51:37 2012 -0800
minor cleanups
Change-Id: Ia12ee4fb59e90221761bec85e6450db29197591f
commit 8f4ed7decbe161a5ff38200b218f5216d80aba46
Author: Mathias Agopian <mathias@google.com>
Date: Sun Nov 4 18:49:14 2012 -0800
improve resample test
- handle stereo input
- input file can now be ommited, in this case
a linear chirp will be used automatically
- better usage information
Change-Id: I5d62a6c26a9054a1c1a517a065b4df5a2cdcda22
commit 5fcd634ea6cb4df27c495abe20f5f9b8ff55d128
Author: Mathias Agopian <mathias@google.com>
Date: Sun Nov 4 02:03:49 2012 -0800
change how we store the FIR coefficients
The coefficient table is now transposed and shows
much better its polyphase nature: we now have a FIR
per line, each line corresponding to a phase.
This doesn't change at all the results produced by
the filter, but allows us to make slightly better
use of the data cache and improves performance a bit
(although not as much as I thought it would).
The main benefit is that it is the first step
before we can make much larger optimizations
(like using NEON).
Change-Id: Iebf7695825dcbd41f25861efcaefbaa3365ecb43
commit d652231abf4c7e2ea1fc89caae730cec1f7259a1
Author: Mathias Agopian <mathias@google.com>
Date: Sat Nov 3 23:37:53 2012 -0700
improve SINC resampler performance
The improvement is about 60% by just tweaking a few
things to help the compiler generate better code.
It turns out that inlining too much stuff manually was hurting us.
Change-Id: I8068f0f75051f95ac600e50ce552572dd1e8c304
commit 9dc68ef5b94c700c4ee68790e8cbb334c90a538d
Author: Mathias Agopian <mathias@google.com>
Date: Thu Nov 1 21:03:46 2012 -0700
new coefficients for the vhq resampler
previous coefficients were provided by a 3rd party and didn't have a
way to re-generate them. we're now using the 'fir' utility.
the performance of the filter is virtually identical, except for
the down-sampling case which seems slightly better now:
It looks like both the previous and new coefficients are generating
some sort of clipping for full-scale signals in the down-sampling case
(although the new ones seem better), the reason for that is
unknown (see bug: 7453062)
Also updated the HQ coefficients for the down-samplers, previous ones
were a little bit too conservative -- the new ones push the cut-off
frequency up by about 1 KHz.
Change-Id: I54a827b5c707c7cc41268ed01283758dce1d7647
commit 38e0b8560a6fc1b7124e22e0e09a84a285182f8e
Author: Mathias Agopian <mathias@google.com>
Date: Tue Oct 30 13:51:44 2012 -0700
fix SINC resampler on non ARM architectures
make sure the C version of the code generates the same
output than the ARM assemply version.
Change-Id: Ide218785c35d02598b2d7278e646b1b178148698
commit a1878128b182696ba508569b4d211d0dfae92463
Author: Mathias Agopian <mathias@google.com>
Date: Tue Oct 30 12:49:07 2012 -0700
fix another issue with generating FIR coefficients
the impulse response of a low-pass is 2*f*sinc(2*pi*f*k), we were
missing the 2*f scale factor. This explains why we were seeing
clipping and had to manually scale the filter down.
Change-Id: I86d0bb82ecdd99681c8ba5a8112a8257bf6f0186
commit 1a0fb993430acc9f601e6c538305bc407c20ac5d
Author: Mathias Agopian <mathias@google.com>
Date: Mon Oct 29 17:13:20 2012 -0700
fir a typo that caused up-sampling coefficiens to be wrong
up-sample coefficient were generated with a cut-off frequency of 24KHz
intead of ~20KHz, which caused more aliasing in the audible band.
also increased the attenuation to 1.3 dB on both up and down
sampling coefficient to avoid clipping.
Change-Id: Ie8aeecf1429190541b656810c6716b6aae5ece2e
commit 9520ad6862bd682ad075a9d9e3e94ada9f6e58b6
Author: Mathias Agopian <mathias@google.com>
Date: Mon Oct 29 17:13:16 2012 -0700
test-resample: clip instead of overflowing
Change-Id: I550e5a59e51c11e1095ca338222b094f92b96878
commit ba36656300f250f7f1fdeb75149749344260e6cb
Author: Mathias Agopian <mathias@google.com>
Date: Sun Oct 21 01:01:38 2012 -0700
a test app for the resamplers
Change-Id: I66852d90d384f1d9e77b51ad1a1ebdbaf61d0607
commit 056a08b9bfd33cf27228c992adc8293a56b01be8
Author: Mathias Agopian <mathias@google.com>
Date: Fri Oct 26 14:11:01 2012 -0700
reenable the cubic resampler
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
commit 8c0241d3ff50ae85167f69b3bd369244894cfa44
Author: Mathias Agopian <mathias@google.com>
Date: Fri Oct 26 13:48:42 2012 -0700
improve SINC resampler coefficients
- we increase the interpolation precision from 4 to 7 bits
this doesn't increase CPU power required, it only increases the
size of the filter table but significantly reduces the noise
introduced by the quantization of the impulse response.
- the parameters of the filter are set such that aliasing is
rejected at 80 dB below 20 KHz. Because we don't use a lot of
coefficient (to save compute power), there are quite a bit of
attenuation in the pass-band: starting at 9KHz for the
down-sampler (48 to 44.1), and starting at 13 KHz for the
up-sampler (44.1 to 48) -- the transition band is about 15 KHz.
Change-Id: I855548d2aab8a0fb0d2a2da3a364b6842d7d3838
commit 69e7dab2192adc1f780464146810629ebd01b145
Author: Pixelflinger <mathias.agopian@gmail.com>
Date: Thu Oct 25 19:43:49 2012 -0700
improve fir tool: cleanup, better default, bug fixes
- all parameters can be changed on the command-line
- added float output
- added debug option
- added an option to generate a polyphase filter coefficients
- added an attenuation option in dBFS
- added a lot of comments and references
- fixed kaiser window parameter
also the default should generate a filter with 80 dB rejection
(of the 24 KHz aliasing) above 20 KHz and a 15 KHz transition
band around ~20 KHz (for 48 KHz sampling rate).
It's not very good but corresponds to the current code.
commit 8347499d105a50257c18e9dac652e750b06428b1
Author: Glenn Kasten <gkasten@google.com>
Date: Mon Oct 22 17:09:27 2012 -0700
Increase allowed number of VHQ resamplers to 3
Bug: 7378660
Change-Id: I69e33ca2eb4bb9bd38e2c63df62cd1130d68baf6
commit f91cf3cad7f5c4d52614271c89ab468741c5d24c
Author: Mathias Agopian <mathias@google.com>
Date: Sun Oct 21 03:04:05 2012 -0700
Fix a typo that caused the high quality resampler to produce garbage
the problem is that if libaudio_resampler is present, it is those
coefficients that will always be selected, but the correct
meta-data.
Bug: 7385994
Change-Id: Ieebeb37b4dfb62a1a051bc29fae2ce056dbc6621
commit e158a9e4262a174c59469a205658bc3ca4078234
Author: Dan Bornstein <danfuzz@google.com>
Date: Fri Oct 3 10:34:57 2008 -0700
Manually merge change #111620 from tc3 to mainline, to keep the
automerger from choking on it.
p4 sync
p4 integrate -r -b android_to_tc3 //...@111620,111620
p4 resolve -a
p4 resolve # resolve a couple merge travesties
PRESUBMIT=passed
BUG=1399648
TBR=edheyl
OCL=111902
Change-Id: I854b01553dd92bbf9c864f5a9bd51a3d665f0ac2
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit b9f3c26032be7a6ea01a10d93d94826f449e68ab
Author: Dave Bort <dbort@google.com>
Date: Fri Jan 18 14:51:05 2008 -0800
Rename "Makefile" to "Android.mk" throughout the tree.
For <http://b/issue?id=960416>.
I've tested this as much as I can, but 1500 open files =
easy to mess things up. Please let me know if there's
a problem rather than rolling back this change.
PRESUBMIT=passed
BUG=960416
TBR=joeo
OCL=46537
Change-Id: I5a404caf0f398a7afa7ae7abaf2f2a1c6ab490eb
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit 0c22a9a44c4103483fba1d944acf1354c5eb1617
Author: Mathias Agopian <mathias@google.com>
Date: Mon Oct 29 23:44:25 2007 -0700
Tweak the SINC resampler parameters and double the performance. It's using about 10% CPU in the worse case now.
Change-Id: I50ac7e6c6702a427fa36ab6d976c507155057507
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit b85e41487983ad085b859acf8251e7e54480308a
Author: Mathias Agopian <mathias@google.com>
Date: Mon Oct 29 04:34:36 2007 -0700
A sinc resampler for Audioflinger. It's not enabled yet, but fully functional and apparently working. It need more "quality" tests. In the 48->44 KHz, it takes about 25% of the CPU time.
Change-Id: I80eb5185e13ebdb907e0b85c49ba1272c23d60ec
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit ba3949ef17cac2ba71cc3096c413782a49c922e5
Author: Mathias Agopian <mathias@google.com>
Date: Thu Aug 23 21:01:28 2007 -0700
fix a few small typos in the FIR computation
Change-Id: I6e56b514fe520f30f7487f85c64ea5d2a7c19b40
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit 7474bfa7de2604021963794dddfe44985648db6a
Author: Mathias Agopian <mathias@google.com>
Date: Thu Aug 23 03:16:02 2007 -0700
This is a tool to compute the the reconstruction filter coefficients for a sinc audio resampler.
Change-Id: I99be2505139b8e0e7647200e1647509d4f7e6067
Signed-off-by: Glenn Kasten <gkasten@google.com>
Bug: 7577965
Change-Id: I2c84a9283a1668723bad83e1a119c849c88c3e6b
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Bug: 7528721
Change-Id: I10bc16a26f33dba6572b730a170cb3bf00e68e30
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auido: Add amr-wb+ codec to ACodec.
-Add an entry for amr-wb+ decoder in ACodec.
-amr-wb+ non tunnel will be enabled by default.
Change-Id: Ied8902eb83da29a3164eb99e88630570a43f681e
libstagefright: Create MP3 decoder libraries without OMX layer
- With the current MP3 OMX SW decoders, the decoding time
is increased w.r.t the libraries without OMX layer that are
present in GB. This increase in decoding time results reduction in
power savings in LPA mode.
- This commit is to remove OMX layer for MP3 to reduce the
power consumption in LPA mode
Change-Id: I835ab6d013a326f111e513586f884bacd5f7106a
audioflinger: EffectModules are updated with device change
Issue: Effects modules are not updated with the device change
information
Fix: 1) Add setDevice information to mLPAEffectChain
2) Remove the return after sending the device route information to
Direct track so that mixer thread is also aware of the device
change for EffectsChain
Change-Id: I82936cd47290946a5e4772e448669d81e0e4d6f5
libmedia : Add a NULL pointer check
- Print frame count in AudioTrack::dump() only if the control
block is valid
Change-Id: Icf594eb721b48795c43d7bd165f6086031ce6efd
CRs-Fixed: 435050
libstagefright: Query AudioSystem for suggested record mute duration
- AudioSource mutes a pre-defined duration (defined by kAutoRampStartUs)
at the beginning of a recording.
- Instead, query the audio system for any ongoing playback streams
and use its output latency to calculate the duration to mute the incoming PCM stream.
- This assumes all current playback threads will be paused once recording
is started.
Change-Id: Ie9b1d62e7be803ef1d8a59127b95c73e03fa5ce6
CRs-Fixed: 438149
libstagefright: Convert mono to stereo for LPA clips
- Sound effects are not supported for mono clips
- Repetative calling of effects_configure and effect_process for
mono clips is resulting in crash in the sound effects library.
- So, Mono clips are now converted to stereo by copying the left
sample to right.
- This is same as what Resampler does in Non-LPA Playback.
This commit is a port of fcc0647fab20ceaf1c07bc10bb243f14c48b114c
CRs-Fixed: 421639
Change-Id: Ie579c8d11afe3db8d42a35956e8bf23eeb88cfe6
audioflinger: Fix to set volume from MediaPlayer in Tunnel mode
Issue: MediaPlayer.setVolume does not have effect on Playback
volume in TunnelPlayer mode
Fix: the left and right volume parameters of setVolume are
hardcoded and defaulted in DirectAudioTrack. Updating the
parameters from the input arguments fixes the issue
Change-Id: I8a107ce57284b225b17d95fed0f69e3adc5fb131
CRs-Fixed: 441849
libstagefright: Enable Tunnel Decode for select formats
- Enable tunnel mode decode only if the audio mime type
matches a supported list.
Change-Id: I32afd83e5fda1e90cb671dd747f17cb83bb84fc1
CRs-Fixed:437651
framework/av:: Add support to decode mp3 data in mp4 container
- Added support to decode mp3 data in mp4 container packed as mp4a
atom and .mp3 atom as well.
Port of 8fa3774adf9259b33ee721cfaeff26da42c29928
Change-Id: I1a04022f30a9f6516575440aba7652986ab7dc58
CRs-Fixed: 439897
audiomixer: Use High Quality resampler
Use very high quality resampler to upsample to 48KHz sample
rate.
Change-Id: I1ba5b839f1e74ae71b405538d970e6a966bd1d47
CRs-fixed: 416730
audioflinger: Fix a deadlock
- A deadlock will happen if the obit recipient
registered by the DirectAudioTrack is called.
- Fix this by moving the lock acquisition in DirectAudioTrack::clearPowerManager()
to after DirectAudioTrack::releaseWakeLock() is called.
- Also synchronize use of mPowerManager in the DirectAudioTrack
destructor with DirectAudioTrack::clearPowerManager()
Change-Id: Ib127db1406c4a61a4054ca0cf30f4c7347a5c92a
CRs-Fixed: 444093
libstagefright: TunnelPlayer: update condition to send SEEK_COMPLETE
- If the client tries to seek to 0 (e.g as a result of LOOPING)
without ever calling getPosition(), we will always sent an immediate
seek notification without seeking.
Change-Id: Id2b9d00c611278d0521cb6fd402710f0ec37bbdd
CRs-Fixed: 441411
libstagefright: Remove unnecessary code from TunnelPlayer
- TunnelPlayer tries to mimick AudioPlayer when trying
to delete the extractor source.
- It is needed for AudioPlayer as the OMXCodec object
is referenced by the CallbackDispatcher as well as AudioPlayer.
- This condition is not true for TunnelPlayer, so why do it.
Change-Id: I79c4e17d01910e73ad01c5640ef374626313a18e
CRs-Fixed: 442365
Add MediaDebug header from CAF
Change-Id: I68dbe72f86a49685b82b64927d1aa80231647a7a
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-Handling of EOS, and triggering EOS was wrong
in TunnelPlayer. Seeking when EOS was posted
to the HAL was wrong. EOS should
not be posted till seek is complete
-Also, EOS should not be posted to
the app if we are seeking
-Player should wake up when seeked,
even after EOS was posted from player
to HAL
-Fixed this issue by cleaning up the code for EOS
-Disable tunnel mode playback for streaming use cases
to avoid jittery playback
Change-Id: I21699d2d5874bde6cbfe549ce0251b252e9a4090
CRs-Fixed: 433346
CRs-Fixed: 432233
CRs-Fixed: 429868
libstagefright: Add new mime for QC TS container
- Add new mime type for TS container that is sniffed by extended
extractor. This is needed for media extractor to determine which
parser to create.
Change-Id: I18dcebbbf3b31cea7db29a4dd65385638343bec1
libstagefright: Use software decoder for ADTS content.
Use software decoder for widevine content which uses ADTS
format.
CRs-fixed: 431096
(cherry picked from commit 3edf2e703bcdc47f122864865056d5cb65b7ab43)
Change-Id: I50eba673ddd6ec2bbb737577978e61902ff68d13
audioflinger: Fix to release wakelock after closeoutput
-In DirectTrack destructor, closeOutput is called after
releaseWakelock is done. This may sometimes result in
power collapse happening even before actual close
sequence of Audio path is completed and will result
in high power consumption.
-Release wakelock only after closeOutput is done
in directtrack destructor.
CRs-fixed: 438179
Change-Id: Ibe103804daf2cb09bade998d6d34c3a34508dd09
libstagefright: Add support to change clip duration to enable LPA
Added support to change the clip duration threshold value for LPA
playback. A new system property 'lpa.min_duration' has been added
which controls the minimum clip length for enabling LPA.
The default threshold value has been retained as 60sec.
Change-Id: I6a8be6d6bf67495977d8c75e5be14723a31353b1
frameworks/av: Skip tunnel mode for playback through AudioCache
In the use case of playback using SoundPool, decoded data is
cached from player and further rendered through AudioCache.
Avoid Tunnel mode for the use case AAC format through SoundPool
Change-Id: I21005a5b39f9fb480ae0d525ecb560fec4382620
CRs-Fixed: 437539
frameworks/base: dumpsys rendering statistics for Stagefright
- this adds extra fps statisticis
- report via dumpsys
Change-Id: I7b4d4582c4eb2ccf2d11557844dade92f9e587c0
CRs-Fixed: 435013
libstagefright: Stop extractor source after start in TunnelPlayer
Issue: In the use case of frequent suspend resume during Video
Playback with HDMI Connected, we encounter a scenario where tunnel
player is created and destroyed without the extractor source
started. In such use case, stopping the source in reset during the
tunnel player destruction leads to failure during assertion check.
Fix: Check for mStarted flag to ensure that stop on source is
called only after they are started.
Change-Id: Ib18e7ee3d10b2cc706944e358046f163d156706c
CRs-Fixed: 440239
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-AAC and MP3 clips are not playing.
-AAC and MP3 are not using LPA path.
-Enable LPA decoder path and implement LPAPlayer class.
Change-Id: I76438319fc72c4898fad5910f8de874f89287687
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Without this exception samsung JB stock audio HAL crashes
Change-Id: I596e6e077f5bd4236359de48db754100e0a133e4
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Old policies don't address the modules by handle and expect them
all to be scanned for supported devices; new modules stopped
listing supported devices. The combination of a new A2DP with
an old policy left policy unable to find A2DP outputs, so the
new HALs couldn't be used.
So... if an old policy is in use, searching for an A2DP output,
and finds a non-primary HAL that doesn't list any device, assume
it's A2DP and use it.
Change-Id: Ie946c065b5c325afd02ba3437077f634e079b386
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libstagefright: Add QC specific media format
- Add QC specific media extensions
- Add QC specific media definitions
Change-Id: I7dca90be3b977701d9537f5e017117790a030f1f
audio: Compile AudioParameter as shared library
- AudioParameter as shared lib is needed by BT
support in WFD source.
Change-Id: I464b428ace0cbb57ce6bf7bf3b57d51a7d56f032
libstagefright: Send flush on both i/p and o/p ports together
- ANR occurs in music due to race condition in OMX component if
flush is issued separately for i/p and o/p ports as DSP only
handles simultaneous flush on i/p and o/p ports.
Change-Id: I5b16cd5a9b57c857dc8bed489d2663b8f54769e3
libstagefright: Enable extended A\V format
- Add new files to support extended A\V format
Change-Id: I1e61d78d35b868d55fd8e99f95de8cab9c465db4
libstagefright: Framework to plug-in propritory parser
- Extend the current framework to plug-in propritory
parser
Change-Id: Ia586a3048420ddf1515261f20035589447263b7b
audio: add support for QCOM audio formats
- Add support for EVRC, QCELP, and WMA formats.
Change-Id: Iaf80f982fc8b08617132dbd7d524a1748866745c
frameworks/av: Support Tunnel Playback
- Implement DirectTrack and DirectTrackClient
- DirectTrack exposes API to client so it can create a direct
output.
- DirectTrackClient allows notifications to be sent to the
client from DirectTrack
- DirectTrack is being used for Tunnel Audio
Change-Id: I2fbb18a781d8e44b8d65da9a357f6e39375f063a
frameworks/av: Support LPA Playback
Add support to enable Playback in LPA mode
Change-Id: I1b8ac4904f4735017d62f3757ede7bbb56e62fd3
audio: Send correct channel mask in voice call recording.
-Using popCount function to get channel count gives incorrect value on
voice call recording.
-Only STEREO and MONO bits to be considered to count
channels on input
Change-Id: I04c2c802422e868bdba0538ff8623dbf9eb659fe
libstagefright: Thumbnail mode initial commit
- use sync frame decoding mode when kClientNeedsFrameBuffer
is set for hardware decoders
- hardware decoder will only expect I frames, OMXCodec will
set EOS on first ETB to stop more frames from being pulled
- skip EOS check on FTB so that the first frame will be
handled
Change-Id: I0e8974e088fdcc468e27764861c128cfe291499f
audio: Add support for QCOM's VOIP solution
Change-Id: I1150f536fa204b535ca4019fdaa84f33f4695d93
audio: define QCOM audio parameters
- Define QCOM audio paramters for FM, VOIP,
fluence, SSR, and A2DP
Change-Id: I29d02e37685846f6d4f00dee02e2726b015eaae7
Add ifdefs for QCOM enhanced features
Change-Id: Ic8e5fe6ecc058466ced71030883b1af6c2bc055c
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Usage :
add "COMMON_GLOBAL_CFLAGS += -DMR0_AUDIO_BLOB" to
BoardConfig.mk
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Enable with the ICS_AUDIO_BLOB CFLAG
Change-Id: Ie174d5997202b8931c1f11db62b6ec2e377f096a
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Bug: 7378660
Change-Id: I69e33ca2eb4bb9bd38e2c63df62cd1130d68baf6
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the problem is that if libaudio_resampler is present, it is those
coefficients that will always be selected, but the correct
meta-data.
Bug: 7385994
Change-Id: Ieebeb37b4dfb62a1a051bc29fae2ce056dbc6621
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Bug: 7369232
Change-Id: I7ff9f525dad4be0aef562a53015b06ee7d3d50f1
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Audio tracks were not using the right latency estimation for
signalling the completion of their presetation. This caused
the synchronization mechanism between playback and record to be
off, and a synchronized recording would contain some of the audio
that was meant to be over once recording would start.
Use the playback thread's latency reporting which takes the audio
pipe into account.
Bug 7237669
Change-Id: I23a907a53ad0b0d68d246789ec595a77a79fced5
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Add support for querying whether there is currently a recording
underway from the specified audio source.
Bug 7314859
Change-Id: I986b231a10ffd368b08ec2f9c7f348d28eaeb892
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The volume and routing policy of AUDIO_STREAM_ENFORCED_AUDIBLE is
now controlled by AudioService.
Do not read ro.camera.sound.forced is not needed anymore.
Bug 7032634.
Change-Id: Ic0a6396fc4b6efb91cdb4dffe0c8eb035d0440bd
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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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An output can only be closed if there is no lock contention that
prevents ThreadBase::exit() from being blocked. If an output
device is waiting for an operation to complete (here a write
in the remote_submix module, because the pipe is full), signal
the module that it's entering the "exiting" state.
Change-Id: I8248add60da543e90c25a4c809866cdb26255651
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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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Change-Id: I34f5d36ae60010ec64222d6660d10a84da3bf566
Bug: 7024293
Signed-off-by: Iliyan Malchev <malchev@google.com>
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This reverts commit 44cda3a4e7ca3de0db9cb49145def3803b03ebb4
Change-Id: I7fd29b77690dab057ac966a42fb198b2772f092c
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Bug: 6635041
Change-Id: I3386a4a6c226bc4eceaf65556119e4fb15f73224
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Use broadcast() instead of signal() on the
thread wake up condition when starting record or requesting thread
exit to make sure that if another thread is waiting for the same
condition (e.g binder thread calling setParameters()) the mixer
thread will be woken up.
Bug 7184317.
Change-Id: I3154a4509ca7af6ffae5236e522b0fab8e75ed06
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Bug: 7100774
Change-Id: I15a84a19bb6d6ef1d9dac4beaa03587638196404
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The code detecting the end of an audio track presentation before
removing it from the active track list is based on the
count of audio frames sent to audio HAL. When an output stream
is suspended (e.g. A2DP when SCO is active), this count does not
change and a track in stopped state will never be removed from
active track list causing the mixer thread to never release
the wake lock.
The fix consists in incrementing the audio HAL frame count even
if the output is suspended.
Also fix a problem in getRenderPosition() when the output is suspended.
Bug 7167534.
Change-Id: I3be836cbbea29b65dc087199cac6a1cd84c0a41d
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When creating a fast AudioTrack, a request is sent to SchedulingPolicyService
to elevate the requesting thread priority. This generates a binder
call into system_server process and to a JAVA service via JNI.
If the thread from which the track was created is in the system_server
process and does not have the "can call java" attribute, a crash occurs because
the binder optimization reuses the same thread to process the returning binder
call and no JNI env is present.
The fix consists in sending the priority change request from the AudioFlinger
mixer thread, not from the binder thread.
This also reverts the workaround in commit 73431968
Bug 7126707.
Change-Id: I3347adf71ffbb56ed8436506d4357eab693078a3
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Enter standby when HAL returns an error, but also consider 0 bytes
returned as NOT_ENOUGH_DATA.
Change-Id: Ica83142310e9c176f936e0440571a6034cbc575f
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When calling start() on an AudioRecord with a HAL that
returns 0 on a read() operation, the start blocking
condition was never unblocked.
Add a boolean to track the first read operation so the returned
number of bytes (mBytesRead) is only evaluated after that
first read.
Change-Id: I8c735a00d48cd6a0da467ccdf75d3616b38f6afa
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The audio downmixer effect might need the audio session Id, pass it
from the track creation in AudioFlinger to the downmix effect
creation in AudioMixer.
Change-Id: I5e29540542ae89cf4a0cdb537b3e67f04442a20a
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Change-Id: I31c964caeb8b5d9ae0a426224f030cdcb01114a0
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The ThreadBase class now has a separate member for input
and output devices (mInDevice, mOutDevice).
Only query get_supported_devices() from audio HAL if the function
is exposed and if the audio policy manager did not specify the
audio module to open.
Also fixed bug in AEC preprocessing that would reset
to default output device when an input device was given.
Change-Id: I19d4d06aeb920b068e3ef31e6e6be6345ce5d67a
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Added support for EFFECT_CMD_SET_AUDIO_SOURCE audio effect
command to inform preprocessings of current audio source
selection for capture.
Change-Id: Ib2418a9aa8114e8457fe828ecd43b230ed86cdd6
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Change-Id: Ie7504d0ddb252f7e4d4f99ed0b44cfc7b1049816
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RecordThread::isValidSyncEvent() returns false, so most of
RecordThread::setSyncEvent() is never executed.
Change-Id: I0cf848beb46a367a45126d2df3073c5afa2ca59c
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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: Ie865ceeabaeb698a62c11eac9f96e303c44c28ae
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Change-Id: I92e32ee16274c032c9d0ce910676be2a7fa52471
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Change-Id: I3e4af69b929d4ca04afaac26c7e41c89fce25b9c
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setSyncEvent() returns a status_t which is sometimes ignored.
Emphasize this is intentional by casting to void.
Change-Id: Ic614988347cba36bd2504d7ad321594a355b0d9d
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