OSDN Git Service
Simon Wilson [Thu, 29 Nov 2012 23:26:48 +0000 (15:26 -0800)]
Merge "Use ATRACE macros instead of Tracer statics"
Dylan Powers [Thu, 29 Nov 2012 23:23:00 +0000 (15:23 -0800)]
am
2ee204af: am
47c1a5f7: Bug fix for the MediaPlayer::prepare() api.
* commit '
2ee204af3d715b983bc4806fb830feefd401fd4b':
Bug fix for the MediaPlayer::prepare() api.
Dylan Powers [Thu, 29 Nov 2012 23:20:31 +0000 (15:20 -0800)]
am
47c1a5f7: Bug fix for the MediaPlayer::prepare() api.
* commit '
47c1a5f7c13d82aa8834fd4543bd1d713b97808e':
Bug fix for the MediaPlayer::prepare() api.
Simon Wilson [Thu, 29 Nov 2012 23:18:50 +0000 (15:18 -0800)]
Use ATRACE macros instead of Tracer statics
ATRACE_BEGIN and ATRACE_END have replaced the static
Tracer::traceBegin and Tracer::traceEnd functions, so
use them instead.
Fixes compilation errors when tracing is enabled.
Change-Id: I4d1147d2f76afcdf113e9986f0544cb848802b15
Dylan Powers [Wed, 28 Nov 2012 00:06:38 +0000 (16:06 -0800)]
Bug fix for the MediaPlayer::prepare() api.
For an MP3 source, within the prepare command, ID3 tags are checked in search of
gapless playback info. This causes problems for streamed sources. If ID3v2 tags
aren't present, then a check is done for ID3v1 tags. This results in a read
command that asks the cache to jump to the end of the file, and subsequently
make an extra http call to request those bytes. For a streamed source, this
causes the file to not be downloaded automatically when MediaPlayer::prepare()
is called, and causes stuttering and extra buffering time to be needed when
start() is finally called.
The solution is to ignore the ID3v1 tags as the gapless info would never exist
there, and only check for ID3v2 tags.
Cherrypicked from external contribution
ffd6ffc5429c45577fd8e9f8fa90e79bb91b8a84
b/
7638165
Change-Id: I7d1b94cffbfe7c38ca094834dedbc92a58855e20
Glenn Kasten [Thu, 29 Nov 2012 22:32:22 +0000 (14:32 -0800)]
Andreas Huber [Thu, 29 Nov 2012 22:32:21 +0000 (14:32 -0800)]
am
2f6107ec: am
b64def9a: Merge "[wfd] Support a low(er) power state by triggering PAUSE/RESUME." into jb-mr1.1-dev
* commit '
2f6107ec8e67eea1f73b9558a5ad34caec52867f':
[wfd] Support a low(er) power state by triggering PAUSE/RESUME.
Andreas Huber [Thu, 29 Nov 2012 22:32:20 +0000 (14:32 -0800)]
am
5ea87edb: am
e5aed03d: Enable retransmission of UDP packets in case we want to use it
* commit '
5ea87edbc97cd693fb7a9a8e00e712978315693c':
Enable retransmission of UDP packets in case we want to use it
James Dong [Thu, 29 Nov 2012 22:32:19 +0000 (14:32 -0800)]
am
bd4f7b89: am
79c56d3f: Merge "Reduce the frequency of IDR frames and add intra-fresh mode support for WiFi display" into jb-mr1.1-dev
* commit '
bd4f7b8931a316ca43fae033d86167c83a8bac07':
Reduce the frequency of IDR frames and add intra-fresh mode support for WiFi display
James Dong [Thu, 29 Nov 2012 22:32:18 +0000 (14:32 -0800)]
am
296cb9dd: am
0dbe5a93: Added optional intra macroblock refresh support for encoding
* commit '
296cb9ddd916c43983bfb4ddab9c69ed555d4cc4':
Added optional intra macroblock refresh support for encoding
Glenn Kasten [Thu, 29 Nov 2012 22:28:51 +0000 (14:28 -0800)]
Andreas Huber [Thu, 29 Nov 2012 22:27:59 +0000 (14:27 -0800)]
am
b64def9a: Merge "[wfd] Support a low(er) power state by triggering PAUSE/RESUME." into jb-mr1.1-dev
* commit '
b64def9a555bfbf533a2da41ba0189b9842a76fb':
[wfd] Support a low(er) power state by triggering PAUSE/RESUME.
Andreas Huber [Thu, 29 Nov 2012 22:27:53 +0000 (14:27 -0800)]
am
e5aed03d: Enable retransmission of UDP packets in case we want to use it
* commit '
e5aed03d30ea0ce49728873c5b74f89ba05a9541':
Enable retransmission of UDP packets in case we want to use it
James Dong [Thu, 29 Nov 2012 22:25:49 +0000 (14:25 -0800)]
am
79c56d3f: Merge "Reduce the frequency of IDR frames and add intra-fresh mode support for WiFi display" into jb-mr1.1-dev
* commit '
79c56d3f17d3193a0a86eb3c9bfdea90b89ae3f9':
Reduce the frequency of IDR frames and add intra-fresh mode support for WiFi display
James Dong [Thu, 29 Nov 2012 22:25:47 +0000 (14:25 -0800)]
am
0dbe5a93: Added optional intra macroblock refresh support for encoding
* commit '
0dbe5a9321b24b6883fbb2fe97cd9d525128b0b5':
Added optional intra macroblock refresh support for encoding
Glenn Kasten [Thu, 29 Nov 2012 22:09:41 +0000 (14:09 -0800)]
Fix log spam
Change-Id: Ie6c982af906dcfd3cdea4b771dfab1f7e47745ca
Andreas Huber [Thu, 29 Nov 2012 22:08:24 +0000 (14:08 -0800)]
Merge "Clear the sticky EOS flags when transitioning to LOADED state"
Andreas Huber [Thu, 29 Nov 2012 21:49:07 +0000 (13:49 -0800)]
Clear the sticky EOS flags when transitioning to LOADED state
instead of transitioning _from_ UNINITIALIZED state. This makes codec instances
reusable.
Change-Id: I8f0c11923978ffee58b553a5ac59c740b0223c54
Marco Nelissen [Thu, 29 Nov 2012 19:35:12 +0000 (11:35 -0800)]
am
ce8dcdf5: am
031c93df: Merge "Bug fix for the MediaPlayer::prepare() api."
* commit '
ce8dcdf5361dd5de8c86cf5b0308c71d519f98ca':
Bug fix for the MediaPlayer::prepare() api.
Marco Nelissen [Thu, 29 Nov 2012 19:33:11 +0000 (11:33 -0800)]
am
031c93df: Merge "Bug fix for the MediaPlayer::prepare() api."
* commit '
031c93df74621dc2149876dc377aedee8930547f':
Bug fix for the MediaPlayer::prepare() api.
Marco Nelissen [Thu, 29 Nov 2012 19:08:16 +0000 (11:08 -0800)]
Merge "Bug fix for the MediaPlayer::prepare() api."
Andreas Huber [Thu, 29 Nov 2012 18:57:06 +0000 (10:57 -0800)]
Merge "[wfd] Support a low(er) power state by triggering PAUSE/RESUME." into jb-mr1.1-dev
Andreas Huber [Thu, 29 Nov 2012 17:54:42 +0000 (09:54 -0800)]
Enable retransmission of UDP packets in case we want to use it
in our upcoming wfd _sink_ implementation.
Change-Id: I4509c30d5a7b992bc841b73d63db902bbcf8f76a
related-to-bug:
7638155
Andreas Huber [Fri, 16 Nov 2012 18:38:11 +0000 (10:38 -0800)]
[wfd] Support a low(er) power state by triggering PAUSE/RESUME.
Change-Id: Ibe42bfa73816bbfeb7e652d435254d0171b89727
related-to-bug:
7638150
Andreas Huber [Thu, 29 Nov 2012 17:49:35 +0000 (09:49 -0800)]
am
37ddc8fc: am
251c04b1: Merge "Unsolicited server responses cause RTSP streaming to crash"
* commit '
37ddc8fc0f78234b5b3b58886113560cdf98aadf':
Unsolicited server responses cause RTSP streaming to crash
Andreas Huber [Thu, 29 Nov 2012 17:14:30 +0000 (09:14 -0800)]
am
251c04b1: Merge "Unsolicited server responses cause RTSP streaming to crash"
* commit '
251c04b1f3d048f541832c93347c6604b314e0ea':
Unsolicited server responses cause RTSP streaming to crash
Andreas Huber [Thu, 29 Nov 2012 16:51:21 +0000 (08:51 -0800)]
Merge "Unsolicited server responses cause RTSP streaming to crash"
Lena Magnusson [Fri, 20 Jan 2012 08:39:38 +0000 (09:39 +0100)]
Unsolicited server responses cause RTSP streaming to crash
If the set up of the RTSP stream contains an incorrect or otherwise
problematic URL, some servers will send an unsolicited server response
that contains a negative number in the sequence number (CSeq) field.
This negative value is not returned from the function findPendingRequest(),
so the check in notifyResponseListener() will not work. Instead there will
be a crash when 0 is used as the index to find a matching request/response
pair that doesn’t exist.
The fix is to return the received sequence number also when it is an
unsolicited server-client message.
Change-Id: Iedaba8a63dece7b43bce007069baefbfd10970b8
James Dong [Wed, 28 Nov 2012 23:40:54 +0000 (15:40 -0800)]
Merge "Reduce the frequency of IDR frames and add intra-fresh mode support for WiFi display" into jb-mr1.1-dev
James Dong [Fri, 16 Nov 2012 02:31:50 +0000 (18:31 -0800)]
Reduce the frequency of IDR frames and add intra-fresh mode support for WiFi 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
James Dong [Thu, 15 Nov 2012 22:00:26 +0000 (14:00 -0800)]
Added optional intra macroblock refresh support for encoding
o related-to-bug:
7524791
Change-Id: I95ac4ee925e2dbeb00b3cfb2e29c611698c5cc9f
Andreas Huber [Wed, 28 Nov 2012 17:44:19 +0000 (09:44 -0800)]
am
7b7f17dc: am
b7c8e918: Add support for HLS playlists of type \'event\'.
* commit '
7b7f17dc9b30ff4ecdf0aea9bcfa1c518d4ac1e7':
Add support for HLS playlists of type 'event'.
Andreas Huber [Wed, 28 Nov 2012 17:42:18 +0000 (09:42 -0800)]
am
b7c8e918: Add support for HLS playlists of type \'event\'.
* commit '
b7c8e91880463ff4981e3e53e98e45d68e2fe374':
Add support for HLS playlists of type 'event'.
Ben Murdoch [Wed, 28 Nov 2012 13:58:29 +0000 (13:58 +0000)]
Fix master build.
Change-Id: Ia362f74d8cd7df76292473c26c112dffe190c599
Glenn Kasten [Wed, 28 Nov 2012 02:43:32 +0000 (18:43 -0800)]
am
5d7b2778: resolved conflicts for merge of
41829f30 to jb-mr1-dev-plus-aosp
* commit '
5d7b2778d0e9849fa601d722ec2efcee7d032d4f':
New VHQ resampler
Dylan Powers [Wed, 28 Nov 2012 00:06:38 +0000 (16:06 -0800)]
Bug fix for the MediaPlayer::prepare() api.
For an MP3 source, within the prepare command, ID3 tags are checked in search of
gapless playback info. This causes problems for streamed sources. If ID3v2 tags
aren't present, then a check is done for ID3v1 tags. This results in a read
command that asks the cache to jump to the end of the file, and subsequently
make an extra http call to request those bytes. For a streamed source, this
causes the file to not be downloaded automatically when MediaPlayer::prepare()
is called, and causes stuttering and extra buffering time to be needed when
start() is finally called.
The solution is to ignore the ID3v1 tags as the gapless info would never exist
there, and only check for ID3v2 tags.
Change-Id: I7d1b94cffbfe7c38ca094834dedbc92a58855e20
Igor Murashkin [Wed, 28 Nov 2012 01:34:23 +0000 (17:34 -0800)]
am
b8fa1240: am
aa9e3e01: Camera: Play shutter sound iff enableShutterSound(true) && ShutterCallback !null
* commit '
b8fa1240eedeb487a7ee0cf7d60e17bed9b25cf4':
Camera: Play shutter sound iff enableShutterSound(true) && ShutterCallback !null
Andreas Huber [Tue, 27 Nov 2012 23:02:53 +0000 (15:02 -0800)]
Add support for HLS playlists of type 'event'.
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
Glenn Kasten [Tue, 27 Nov 2012 23:05:48 +0000 (15:05 -0800)]
resolved conflicts for merge of
41829f30 to jb-mr1-dev-plus-aosp
Change-Id: I952d333c2e88b6b28663793046a136822d1b6838
Glenn Kasten [Tue, 27 Nov 2012 22:51:19 +0000 (14:51 -0800)]
Merge "New VHQ resampler" into jb-mr1.1-dev
Glenn Kasten [Mon, 19 Nov 2012 17:00:47 +0000 (09:00 -0800)]
New VHQ resampler
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
Igor Murashkin [Tue, 27 Nov 2012 19:39:22 +0000 (11:39 -0800)]
am
aa9e3e01: Camera: Play shutter sound iff enableShutterSound(true) && ShutterCallback !null
* commit '
aa9e3e01b86bd9bfb5ac36c0f360d5fe478cbb2d':
Camera: Play shutter sound iff enableShutterSound(true) && ShutterCallback !null
Andreas Huber [Tue, 27 Nov 2012 17:41:12 +0000 (09:41 -0800)]
am
8b5985d9: am
5768fa03: Merge "Crash in android::MyHandler::parsePlayResponse"
* commit '
8b5985d99becc2d5126b8a26afb6f5798b560007':
Crash in android::MyHandler::parsePlayResponse
Andreas Huber [Tue, 27 Nov 2012 17:41:11 +0000 (09:41 -0800)]
am
9abbd0fb: am
3eb46d17: Merge "Fix of uninitialized mIsDrm variable."
* commit '
9abbd0fb822aa0076f6de03f2092db47760b924d':
Fix of uninitialized mIsDrm variable.
Andreas Huber [Tue, 27 Nov 2012 17:39:12 +0000 (09:39 -0800)]
am
5768fa03: Merge "Crash in android::MyHandler::parsePlayResponse"
* commit '
5768fa034ede834656697d3612c525595ff85ef9':
Crash in android::MyHandler::parsePlayResponse
Andreas Huber [Tue, 27 Nov 2012 17:39:11 +0000 (09:39 -0800)]
am
3eb46d17: Merge "Fix of uninitialized mIsDrm variable."
* commit '
3eb46d179b1f62cde21077fde466925d4c5c79ad':
Fix of uninitialized mIsDrm variable.
Andreas Huber [Tue, 27 Nov 2012 16:50:52 +0000 (08:50 -0800)]
Merge "Crash in android::MyHandler::parsePlayResponse"
Andreas Huber [Tue, 27 Nov 2012 16:49:34 +0000 (08:49 -0800)]
Merge "Fix of uninitialized mIsDrm variable."
Patric Frederiksen [Mon, 26 Sep 2011 08:51:35 +0000 (10:51 +0200)]
Crash in android::MyHandler::parsePlayResponse
This fix handles problems with several asynchronous calls
within streaming. This case is when the phone has sent a
request to the server and while the response is being sent
back by the server the request is aborted by the user.
The fix is an if case that checks if we have aborted while
waiting for a response from the server. If we have aborted
we should ignore the late response instead of continuing.
Change-Id: I1264bb992f6abcaee1f10a89479e08b54a95e3c2
Henrik B Andersson [Wed, 31 Oct 2012 12:02:47 +0000 (13:02 +0100)]
Fix of uninitialized mIsDrm variable.
The mIsDrm is a bool that isn't initialized.
This causes it to be true in most default cases.
Change-Id: I41b534514bf6a3ca88a9f0994b814d55fcd7453b
Igor Murashkin [Mon, 26 Nov 2012 18:50:55 +0000 (10:50 -0800)]
Camera: Play shutter sound iff enableShutterSound(true) && ShutterCallback !null
Bug:
7564718
Change-Id: Ie7821cdee57966d88af048759578439a3e6ecb2e
Andreas Huber [Mon, 26 Nov 2012 22:39:20 +0000 (14:39 -0800)]
am
5bc5bf39: am
79fd6853: Merge "MediaCodec: Add a method for getting the component name"
* commit '
5bc5bf39120ae27ef46a8b13f85bf44ea19c7d5e':
MediaCodec: Add a method for getting the component name
Andreas Huber [Mon, 26 Nov 2012 22:37:22 +0000 (14:37 -0800)]
am
79fd6853: Merge "MediaCodec: Add a method for getting the component name"
* commit '
79fd685323e34e0fde22d17fd6848d33f171f4ae':
MediaCodec: Add a method for getting the component name
Andreas Huber [Mon, 26 Nov 2012 21:48:41 +0000 (13:48 -0800)]
Merge "MediaCodec: Add a method for getting the component name"
Marco Nelissen [Mon, 26 Nov 2012 19:20:33 +0000 (11:20 -0800)]
am
21006fa5: (-s ours) am
5865ddf7: Merge "AudioTrack::dump null mCblk check test"
* commit '
21006fa5fa180d1eb1513a5ae297211a24312021':
AudioTrack::dump null mCblk check test
Marco Nelissen [Mon, 26 Nov 2012 18:00:09 +0000 (10:00 -0800)]
am
5865ddf7: Merge "AudioTrack::dump null mCblk check test"
* commit '
5865ddf769d368d714af630aba18392ea1387bc7':
AudioTrack::dump null mCblk check test
Marco Nelissen [Mon, 26 Nov 2012 16:59:50 +0000 (08:59 -0800)]
Merge "AudioTrack::dump null mCblk check test"
Zbigniew Mazur [Thu, 11 Oct 2012 11:56:41 +0000 (13:56 +0200)]
AudioTrack::dump null mCblk check test
This fix is protecting AudioTrack::dump from SIGSEGEV
when calling dumpsys shell command.
Change-Id: I30d136e195a12d6fdad41e14f557b0ad9e59b8a2
Marco Nelissen [Tue, 20 Nov 2012 20:49:52 +0000 (12:49 -0800)]
am
b96c4b4c: am
2da6e4ae: Merge "Avoid memory leaks when handling metadata strings"
* commit '
b96c4b4ce8eb90a6bdb162681affe1e50fe6aafc':
Avoid memory leaks when handling metadata strings
Marco Nelissen [Tue, 20 Nov 2012 20:47:56 +0000 (12:47 -0800)]
am
2da6e4ae: Merge "Avoid memory leaks when handling metadata strings"
* commit '
2da6e4ae488896df10b22166b0aa0b2cc15492f1':
Avoid memory leaks when handling metadata strings
Marco Nelissen [Tue, 20 Nov 2012 20:28:27 +0000 (12:28 -0800)]
Merge "Avoid memory leaks when handling metadata strings"
David Williams [Mon, 19 Nov 2012 08:52:16 +0000 (09:52 +0100)]
Avoid memory leaks when handling metadata strings
Don't duplicate strings when retrieveing metadata from media
files. As any requests for metadata strings would pass through
the binder, this would cause the reference to the duplicate string
to be lost, causing a memory leak as the duplicate would not be
freed.
Change-Id: I2593733472b1bb589bc502b2c11080f581015bb5
Eric Laurent [Tue, 20 Nov 2012 18:37:47 +0000 (10:37 -0800)]
Merge "AudioFlinger files reorganization"
Andreas Huber [Tue, 20 Nov 2012 17:27:22 +0000 (09:27 -0800)]
am
14dda623: am
7013209c: Merge "Handle large AVCC chunks"
* commit '
14dda623c8db5f991b8a22dce4f19f8d8b47fea2':
Handle large AVCC chunks
Marco Nelissen [Tue, 20 Nov 2012 17:27:21 +0000 (09:27 -0800)]
am
343f9c81: am
dca2b5d7: Merge "Changed parsing of trkn and disk from 8 bits to 16 bits"
* commit '
343f9c81f293f56e09b1cc1921844ecd3372e435':
Changed parsing of trkn and disk from 8 bits to 16 bits
Andreas Huber [Tue, 20 Nov 2012 17:24:25 +0000 (09:24 -0800)]
am
7013209c: Merge "Handle large AVCC chunks"
* commit '
7013209cdf393b3d958ddd46ed50394349378826':
Handle large AVCC chunks
Marco Nelissen [Tue, 20 Nov 2012 17:24:24 +0000 (09:24 -0800)]
am
dca2b5d7: Merge "Changed parsing of trkn and disk from 8 bits to 16 bits"
* commit '
dca2b5d7c29ee06f3c82527dd7264fcc21cac9a6':
Changed parsing of trkn and disk from 8 bits to 16 bits
Andreas Huber [Tue, 20 Nov 2012 16:52:44 +0000 (08:52 -0800)]
Merge "Handle large AVCC chunks"
Marco Nelissen [Tue, 20 Nov 2012 16:18:38 +0000 (08:18 -0800)]
Merge "Changed parsing of trkn and disk from 8 bits to 16 bits"
Eric Laurent [Mon, 19 Nov 2012 22:55:58 +0000 (14:55 -0800)]
AudioFlinger files reorganization
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
Mathias Agopian [Mon, 19 Nov 2012 23:27:26 +0000 (15:27 -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
Glenn Kasten [Mon, 19 Nov 2012 19:53:52 +0000 (11:53 -0800)]
Merge "Clean up channel count and channel mask"
Glenn Kasten [Wed, 14 Nov 2012 20:47:55 +0000 (12:47 -0800)]
Clean up channel count and channel mask
Channel count is uint32_t.
Remove redundant mask parameter to AudioTrack::createTrack_l()
and AudioRecord::openRecord_l().
Change-Id: I5dc2b18eb609b2c0dc3091994cbaa4628062c17f
Marco Nelissen [Mon, 19 Nov 2012 17:49:18 +0000 (09:49 -0800)]
delete -> free
Strings duplicated with strdup() should be free()d, not deleted.
Change-Id: I42bb3df9625bb8d35f80b02d15364b94c36496f8
Martin Storsjo [Tue, 25 Sep 2012 08:43:02 +0000 (11:43 +0300)]
MediaCodec: Add a method for getting the component name
If the codec was chosen based on mime type, the caller does
not know what component actually was chosen. This allows
getting essential information (such as supported color formats,
for a video encoder) for this component.
Change-Id: Ie471f40f8104b37d27ced3dba5a54facc6504b1b
Andreas Lillvik [Wed, 13 Oct 2010 13:37:01 +0000 (15:37 +0200)]
Changed parsing of trkn and disk from 8 bits to 16 bits
The MPEG4Extractor was parsing 8 bits instead of 16 bits when parsing
'trkn' and 'disk'. Also added support for 16 bytes size 'disk'.
Change-Id: I22b4de2ac800881884d5759776cb380917522a87
Jan Olof Svensson [Wed, 26 Sep 2012 07:08:11 +0000 (09:08 +0200)]
Handle large AVCC chunks
If enabling seq_scaling_matrix_present_flag = 1 the AVCC chunk can
be larger than the original buffer size. Changed to using ABuffer
instead.
Change-Id: Idacc14b45ea2634c5e608919f3ce567f23363135
Glenn Kasten [Fri, 16 Nov 2012 23:47:51 +0000 (15:47 -0800)]
Merge ""if" statements use curly braces per media style"
Glenn Kasten [Fri, 16 Nov 2012 23:38:29 +0000 (15:38 -0800)]
Merge "Fix time vs. bytes units bug in getRenderPosition"
Glenn Kasten [Fri, 16 Nov 2012 23:23:42 +0000 (15:23 -0800)]
Merge "Don't use control block frame count after create"
Glenn Kasten [Fri, 16 Nov 2012 23:03:51 +0000 (15:03 -0800)]
Merge "Don't explicitly log tid"
Glenn Kasten [Wed, 14 Nov 2012 21:42:25 +0000 (13:42 -0800)]
Don't use control block frame count after create
This is part of a series to clean up the control block.
Change-Id: I7f4cb05aef63053f8e2ab05b286d302260ef4758
Glenn Kasten [Fri, 16 Nov 2012 22:46:42 +0000 (14:46 -0800)]
am
abae71d3: am
d983364b: Static AudioTrack plays twice initially
* commit '
abae71d37d4860e297de7ee06f49efa5254b90ee':
Static AudioTrack plays twice initially
Glenn Kasten [Fri, 16 Nov 2012 22:44:05 +0000 (14:44 -0800)]
am
d983364b: Static AudioTrack plays twice initially
* commit '
d983364b3655a547b55bb11dbe148103198c011d':
Static AudioTrack plays twice initially
James Dong [Fri, 16 Nov 2012 22:31:15 +0000 (14:31 -0800)]
Fix a crash when the stop might be called due to some error before start in RTSPSource
o related-to-bug:
7507224
Change-Id: Ic8bfec13097b824ba337a01c9b00c98af2a33f43
Glenn Kasten [Sat, 23 Jun 2012 00:21:07 +0000 (17:21 -0700)]
"if" statements use curly braces per media style
Change-Id: I130e7849fd1da7a0b7fe56c3c53919d26e3843b8
Glenn Kasten [Fri, 2 Nov 2012 17:00:06 +0000 (10:00 -0700)]
Don't explicitly log tid
If needed, it can be obtained with adb logcat -v threadtime
Change-Id: I91b3911d20f7bcfc3361db4052db21ff9181f1cf
Glenn Kasten [Fri, 16 Nov 2012 20:01:44 +0000 (12:01 -0800)]
Fix time vs. bytes units bug in getRenderPosition
Rename correctLatency since it requires thread to be locked.
Use size_t for byte and frame counts.
Change-Id: I178fdd18bdb823813b9563927bdff8c0d28ca5a5
Glenn Kasten [Thu, 15 Nov 2012 22:13:16 +0000 (14:13 -0800)]
Static AudioTrack plays twice initially
Bug:
7528721
Change-Id: I10bc16a26f33dba6572b730a170cb3bf00e68e30
Andreas Huber [Fri, 16 Nov 2012 21:00:45 +0000 (13:00 -0800)]
Merge "Only pass the surface to the video decoder."
Marco Nelissen [Fri, 16 Nov 2012 19:30:43 +0000 (11:30 -0800)]
resolved conflicts for merge of
205d7249 to master
Change-Id: I3df408b6e30e0c0b2a19a3336134ce49fb73a8bb
Andreas Huber [Fri, 16 Nov 2012 19:15:44 +0000 (11:15 -0800)]
Only pass the surface to the video decoder.
Change-Id: Ice0cfc0021fdd9fe053be6ee324cbc64226ed122
Marco Nelissen [Fri, 16 Nov 2012 17:34:19 +0000 (09:34 -0800)]
am
da33d66e: Merge "Add .mpga to acceptable file name extensions list."
* commit '
da33d66e68791d0bfeccebc8253a59467b5ef670':
Add .mpga to acceptable file name extensions list.
Marco Nelissen [Fri, 16 Nov 2012 16:59:14 +0000 (08:59 -0800)]
Merge "Add .mpga to acceptable file name extensions list."
Marco Nelissen [Fri, 16 Nov 2012 16:19:30 +0000 (08:19 -0800)]
Merge "Add GSM 6.10 decoder"
Glenn Kasten [Fri, 16 Nov 2012 16:14:40 +0000 (08:14 -0800)]
Merge "Use size_t for frame counts"
Jan Bjernler [Fri, 16 Nov 2012 15:40:42 +0000 (16:40 +0100)]
Add .mpga to acceptable file name extensions list.
The *.mpga files are playable, but are not correctly scanned.
This is because they are prevented from being scanned in
StagefrightMediaScanner.cpp.
What this fix does is to add the extension to the list of valid
file extensions so that the scanner handles the filetype properly.
We have previously added the .mpga extension to the framework to
make it playable, but not added it so that the scanner scans it.
Change-Id: I02a44d770adb80d38e8bed77d0d76efa1b28a861
Marco Nelissen [Thu, 15 Nov 2012 22:31:56 +0000 (14:31 -0800)]
Add GSM 6.10 decoder
Supports Microsoft frame packing only, since that's what the sample
file used.
b/
6620569
Change-Id: Ia89d95bcbf0f8dcbaad42148a7401728f60e079d
Glenn Kasten [Wed, 14 Nov 2012 20:54:39 +0000 (12:54 -0800)]
Use size_t for frame counts
Also fix typo: bufferCount should be frameCount.
Change-Id: Ibed539504db75ef99dc21c8ff1bf2987122063a5
Glenn Kasten [Thu, 15 Nov 2012 23:05:51 +0000 (15:05 -0800)]
Merge "Static AudioTrack plays twice initially"