| Commit message (Collapse) | Author | Age | Files | Lines |
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Change-Id: Ibe42bfa73816bbfeb7e652d435254d0171b89727
related-to-bug: 7638150
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in our upcoming wfd _sink_ implementation.
Change-Id: I4509c30d5a7b992bc841b73d63db902bbcf8f76a
related-to-bug: 7638155
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for WiFi display" into jb-mr1.1-dev
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display
The time interval between periodic neighboring IDR frames is increased from 1 second to 15 seconds.
o related-to-bug: 7524791
Change-Id: Ic32f37448f952f329549eda5e73637ee3b02f046
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o related-to-bug: 7524791
Change-Id: I95ac4ee925e2dbeb00b3cfb2e29c611698c5cc9f
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related-to-bug: 6870049
Squashed commit of the following:
commit eee2f3ba6bb7335f4e285632726db85645669929
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 15:02:01 2012 -0800
Make everything a lot less verbose by default.
Change-Id: I884d7a7901aa1e7d4ff590f065ca57a79d2af8b3
commit 6bbdb837ed5bd88008e45efb8faf595e4051ba26
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 14:34:46 2012 -0800
HLS now properly signals media time changes at discontinuities including
the start of playback (which may not necessarily be at time 0 if the playlist
is of type 'event' and hasn't completed yet).
Change-Id: I5ab747d024f9b8d0df72a4e06a12ebb29f62802e
commit 1555589832b1878a144a976a643e1af4d61f877c
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 14:32:28 2012 -0800
As part of a time discontinuity, clients of IStreamListener can now
signal the corresponding media time after the discontinuity, i.e. the first PTS
timestamp following the discontinuity will be considered equivalent to the
specified media time and media buffers timestamped accordingly.
Change-Id: Id7db7679b7faa6efd6270620ff52e34e884f3e92
commit 5c24c605c073a11c426d025b1e7478fc1ad8365a
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 13:00:56 2012 -0800
NuPlayer sources now expose flags() and can announce
that duration may change (increase) dynamically, in which case duration
will be polled at 1 second intervals and communicated to the upper layers.
Change-Id: I45102909b7a19eed0dda576747e3814d742a0eea
commit ecb71de8e281e61971a2cd73e7161a97540bc357
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 12:57:47 2012 -0800
Stop caching duration in MediaPlayer, duration could increase dynamically.
Change-Id: I7bb2f16c0abe49debdf45c776d2266aa069d7791
commit 544aec5823e6d7a3e97e15b6b23546616bcd343e
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 08:46:28 2012 -0800
An attempt to add support for "event" style HLS playlists.
Change-Id: I3dfb2e801ecaff8f5d8bdb3a4fca1b18aeeb2c60
Change-Id: I48cf7f65a654d33f2f49ded74f8be22aed9e3b98
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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: 7564718
Change-Id: Ie7821cdee57966d88af048759578439a3e6ecb2e
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Bug: 7528721
Change-Id: I10bc16a26f33dba6572b730a170cb3bf00e68e30
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jb-mr1.1-dev
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previously any error signaled by setupXXX inside ACodec::configureCodec
would be overwritten with the result of setMinBufferSize at the end
of the function.
Change-Id: Id4beb571ca52ea4646239d0af006e09ce1130268
related-to-bug: 7542181
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- manually prepend SPS/PPS if encoder doesn't support it
- latency improvements
- support for "our" method of optional RTP retransmission
- improvements to the wfd commandline tool for testing
- make it easier to turn on/off suspension of the video pipeline on idle
- fixes an issue where an error during encryption would cause a SEGV
- add HDCP descriptor if necessary
Squashed commit of the following:
commit 1115be0ebb3b885b4f1b7dba56761ca013d0ec4a
Author: Andreas Huber <andih@google.com>
Date: Fri Nov 9 11:32:23 2012 -0800
Better shutdown of wfd -l sessions.
Change-Id: Id898a14ae21efd3b065b00a729830063d39195a7
commit 0e7d106dfe4eb6e2640b0b66c65deaba265f7ff0
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 8 16:38:55 2012 -0800
No more sending delay, create rtp packets upfront.
Change-Id: I809a225f664fdb485c7d9a49a27886601a6a26b2
commit d399e8571b77353d59afb57508dfd2a82c1ef93a
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 8 14:19:43 2012 -0800
Restore AudioSource buffer size, factor out TimeSeries, make
suspending video optional.
Change-Id: Ifdfe4d447b901e714abf52856b4641d1d55a5d41
commit f8b649f0b8f917d59f4b8a2e8e6d7db61a684a78
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 8 09:34:06 2012 -0800
Pull 480 frames at a time from AudioSource/AudioRecord
Change-Id: I1e215abd329faec3da026631122c0f4c800c0ac4
commit 1bc13452eb35eebbba00f5da93fa86535be5db59
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 8 08:50:30 2012 -0800
fixed bitrate traffic simulation
Change-Id: Ic5efb7cbb0b5d3b4917bc77b8ba73d447249e695
commit 016cdff18e74bdc631a5679e97192645ed095aa2
Author: Andreas Huber <andih@google.com>
Date: Wed Nov 7 14:00:03 2012 -0800
resurrected "our" style of retransmission.
Change-Id: I34d757aba67428437cb39b8293a9651750ad20d9
commit 384cf1a3c8fb4ec410bdf8fa5722c298e6028f3e
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 6 09:38:55 2012 -0800
Changes to make wfd work on manta.
Change-Id: I7a4e00cf16581fe2146edd1b359af195774090e4
commit 9628f24b22b35f28630d99dda3614babf51bc07e
Author: Andreas Huber <andih@google.com>
Date: Wed Nov 7 09:15:44 2012 -0800
Patch up rtp timestamps to more accurately measure network jitter.
Change-Id: I9502a4615575f97f98a215a13131a89a6ae93c6d
commit 7c891a1a24f08bbd50f55be13f7d05f43e807eb8
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 6 09:37:24 2012 -0800
Additions to the "wfd" tool to create a local wfd source.
Change-Id: I99558653a70fdc703f9d13990b3ce1c4d3ae227a
Change-Id: Ia94c63fc390f597014531073485f0cfc53b3418a
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Change-Id: Idf4c25115d89397ba668fc290344b2e7c1ea1993
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This happens occasionally when taking a bugreport.
Bug: 6447319
Change-Id: Ia6531a4a3658461f8fd3f7106e7996da7cc5933a
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* Was breaking on 8660 due to previous commit.
Change-Id: Ia9f5c45552cc933db336a66b6d1214b65e810488
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Change-Id: Ic73cb2f4152ac77a83fcda7e89b4a8202df440c6
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Change-Id: Ib6cc3401c889a8ec52dc83e3bea367f94a582a40
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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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Change-Id: Ie52b6207293abe5a57d3bbe9d9c43e8c5b070af5
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Creating interface and framework for using FM Radio
RX and TX from different vendors.
Change-Id: I1a71aed01bfffdddfabf1cdfbfa3707cb1ed016b
Signed-off-by: Benn Porscke <benn.porscke@stericsson.com>
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This patch enables the TI ducati H264 encoder profile
via BoardConfig setting:
BOARD_USE_TI_DUCATI_H264_PROFILE := true
Allows correct video decoding on Motorola OMAP4 / Kindle Fires
and other devices using newer DOMX libs.
Effectively a cherry-pick of omapzoom commit (with creative
board setting name):
http://www.omapzoom.org/?p=platform/frameworks/av.git;a=commit;h=e28784d5c68c8699cfd9ebe0231e7132d8b13dad
Change-Id: Idc49b00030558a22a9e50e8798e5814ad54fe841
Signed-off-by: Hashcode <hashcode0f@gmail.com>
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DOMX default
Part 3 of 3 patches: To allow omap4 devices to use custom "domx" source
via a new BoardConfig.mk item:
TI_CUSTOM_DOMX_PATH := device/<manufacturer>/<device-name>/domx
This setting provides for 3 changes during the build:
1. In hardware/ti/omap4xxx this settings stops standard Google domx
source from being built and changes the domx reference for tiutils.
2. In frameworks/base it changes the default openmax references for
frameworks/base/media/jni/mediaeditor/Android.mk to the new location
3. In frameworks/av changes the openmax references in 5 places, and adds
new includes in ACodec.cpp, CameraSource.cpp and OMXCodec.cpp
This is a combination of cherry-picks from omapzoom (with a more descriptive
BoardConfig setting name):
http://www.omapzoom.org/?p=platform/frameworks/av.git;a=commit;h=8044105ca117c2e99b35ad9f341d630fc5a9d2e0
http://www.omapzoom.org/?p=platform/frameworks/av.git;a=commit;h=4adf712d1f3f2050fe0010652bbba7ecb8468870
http://www.omapzoom.org/?p=platform/frameworks/av.git;a=commit;h=fa37231ca59872ac491461ca3c14e019834848e5
Change-Id: I53dbf120d515eaf5ec82688dcea4c670c173ed01
Signed-off-by: Hashcode <hashcode0f@gmail.com>
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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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Change-Id: I28a1a909aa8267ba77c2b44d4191f7a92e8ef533
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Without this exception samsung JB stock audio HAL crashes
Change-Id: I596e6e077f5bd4236359de48db754100e0a133e4
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Change-Id: I96ec5b79c08e37c9bca59470addb5a9f7869eaea
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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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Change-Id: Idb8d5c98fbdc01e6063ad0eb224baba66acc813d
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Change-Id: I311f608d761987d0c714be9ab81188a3ca8eef61
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Based on commit 7036d66bec7697160ba1b2e255f1e7fae0c8c435 from CAF,
branch ics_chocolate.
Change-Id: I652d63c33db9a8da4a9f8f613be12877d032de37
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Bug: 7342813
Change-Id: I98852f036e4679a653723fc2bf1435d6e9d7d34c
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Change-Id: I79b7b4dd8a2ba71d7c95f067d13620260a2e0f63
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codeworkx pulled the Samsung colorformat patch from Insignal's
exynos-jb branch, which is an invalid source for exynos4 patches.
It was checking for an OMX codec name that didn't exist.
This still doesn't stop Netflix JPlayer failing in 10.1, but
does fix Netflix failing if you disable JPlayer by convincing
it that it's on an older Android version.
Change-Id: Ia55878dc08f81e2900307558e35c4e0c5b9cac2f
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These have been around since early stagefright, and were dropped for
JB. Unfortunately, they're still necessary with for this encoder to
work.
Change-Id: I8a251bf195a24b166db7464a90a822d6e69b644d
libstagefright: Add support for the 720P OMAP3 encoders
Bring back some more OMAP code that was removed by Google in JB,
and a couple of omapzoom patches.
This may stop being necessary if TI publishes JB-specific OMAP3
code, but as long as we're using the ICS domx, these need to
be here
Change-Id: Ia29f8c9f9ed769ba07b09c07260486f6502841d6
libstagefright: Unbreak OMAP4 encoders
The "manual" construction of the h264 codec data is only needed
on OMAP3. Execution of this code on OMAP4 breaks the mpeg4 header
generation
Change-Id: I3ae52f2e685e2d9097796685c98dffa93cfa6430
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libstagefright: Add support for VC1 clips
- configure decoder in frame-by-frame or arbitrary mode
based on the codec type.
Change-Id: I6404e5b7ee217045e6456f51f914dbd8a651d98a
CRs-Fixed: 432847
stagefright: Miscellaneous fixes for LPA and Tunnel playback
1. Fix for AV sync issue with Tunnel playback
APIs for returning correct timestamps were implemented
2. Crash while exiting TunnelPlayer
Check whether sink is open before flushing or closing it.
Check for mIsAudioRouted is good to know if we are closing it
3. Seekbar freezes after seek and pause
Check pause status before writing
Change-Id: Id8ab7b258e9c05b20e121bdf3c4dc30d519f6c15
frameworks/av: Add support for surround sound recording
- Add 5.1 channel as supported input channel
Change-Id: I50fcd87245c5c855ede8f09ea1a7c5be2e684640
stagefright: Add tunnel decode for amr-wb/wb+
When tunnel.decode is true, playback of AMR-WB
and AMR-WB+ uses tunnel decode.
Change-Id: Ic06cb3faaac18f605402c98a27cb4dfa4b7faee2
stagefright: Skip LPA if #channels > 2
Skip LPA for Multi channel playback
Change-Id: I5b97471ef55aea260d1c02db672eb4bd64c3edef
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https://android.googlesource.com/platform/frameworks/av into aosp
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