didn't set the genre string to null on case entry
means read failure would leave genre string indeterminate
add explicit NULL & leave genre uninitialized
Change-Id: I1452a0b57985646c982ec80755d0df9d03745f1e
* direct use of memcpy() instead of ci->memcpy() in flac and mod
* uninitialized variable in mpegplayer
Change-Id: I2d08682d5f66c319780e69e3ff63d600c61d8f5a
Encountered a file with ~600K of ID3v1 headers.
Optimize the potential overhead by minimizing reverse seeks.
Change-Id: I972dbf1af1c36659f19c7ab4eed0b9149c642880
The old code would seek forward by the frame length, expecting to see a
frame header there, perform a validity check, and then seek back to the
current header.
Unfortunately this doesn't handle situations where there is extra padding
between the frames, leading us to potentially read garbage, causing the
validity tests to fail and rejecting the file outright.
Instead, keep track of the previous valid header/position, and if we find
"valid" headers in a row return the first after seeking back to it.
This change allows the file referenced in FS13299 to be properly parsed, but
further work is needed to get the file to be playable. (file reports itself
as layer 1, variable bit rate, variable sample rate!)
Change-Id: I85f61a6360cc041a172db4b7a6b5516e5b60ceee
This is an MIT-licensed library for reading and writing v7 format
tar files. The version here is my fork, which fixes security issues
in the original code (it hasn't been updated in 4 years, probably
abandoned by the author).
Change-Id: I86d41423dacc46e9fa0514b4fc7386a96c216e86
As per multiple user requests:
https://forums.rockbox.org/index.php/topic,53319.msg
The acceptable size for id3v2 fields currently maxes out at
240 bytes on targets with more than 2MB of memory.
The comments field, especially for Podcasts, can sometimes
contain significantly more characters than Rockbox allows.
The limit for devices with more than 8MB of memory
is increased to 500 bytes for individual fields, and
to 1800 bytes for the buffer containing all fields.
Change-Id: I4593372229158756f102f67bcc4a43e64f632d58
Since that encompasses _all_ of our native targets in a post-archos world,
either replace it with #if (CONFIG_PLATFORM & PLATFORM_NATIVE) or
delete it altogher as appropriate.
Change-Id: I9128a456e850d5c96a9e05806aad3acd923f90c5
Some flac encoded files contain junk that our decoder
picked up
upstream has some sign and overflow fixes too
Change-Id: I5857b2fe56906a48f04944cdfee8fe2306f2c3fd
I think the assumption is that during playback, the DSP is already running
so it assumes PBE needs to be flushed before it is properly enabled.
Change-Id: I2bac3d02c80f97c8d9ce26a575175f6344a8e86a
mp4 files can have multiple 'mdat' chunks. This is common for
audiobooks, where there is often a secondary mdat containing the
chapter names, but it's also legal to have multiple mdat chunks
for a single logical "track"
This confuses the mp4 metadata parser, which assumes there is
only a single mdat, and always uses the last mdat seen to
determine the "filesize" of the data we're trying to decode.
Work around this by appending each mdat's size to result in the final
"filesize"
Change-Id: I3e7a7efb0f05ef965e8d77f79e450c957524a480
GCC 4.9.4 was already used for MIPS and all hosted targets; this enables
it across the board for everything (ie m68k and arm native)
Other changes:
* Use '-Os' as the default optiomization for all targets
(was only disabled for arm native)
* Enable -funit-at-a-time and -Wextra
* Drop all obsolete toolchain patches
* Update ARM multilib/exception patch
* Bump toolchain libs (gmp, mpfr, mpc) to recommended versions, and
add 'isl' to enable better optimization & vectorization opportunities.
(Will revisit optimization for the codecs and plugins at a later date)
Confirmed working:
* armv4t (ipodmini2g and many other PP502x targets)
* arm >= v5 (sansaclipplus, ipod6g, ipodnano2g, sansafuzeplus)
* m68k (ihp100)
Change-Id: If9ed405ae0f289d9adea46d4cf46bfefc2f4250d
This codec requires floating point.
Original author: Peter Sovietov
Ported to Rockbox: Roman Skylarov
Further integration and bugfixes: Solomon Peachy
Change-Id: I781ecd3592dfcdbbc694063334350342534f1d6c
- Avoid use of uname, instead ask the compiler about the binaries it
creates.
- Move platform specific commands used into functions.
- Replace use of echo with make functions.
Change-Id: I365f656e02ddb90ee91ae60fec1230b4aaaf224f
Note: I left behind lcd_bitmap in features.txt, because removing it
would require considerable work in the manual and the translations.
Change-Id: Ia8ca7761f610d9332a0d22a7d189775fb15ec88a
'swcodec' is now always set (and recording_swcodec for recording-capable
units) in feature.txt so the manual and language strings don't need to
all be fixed up.
Change-Id: Ib2c9d5d157af8d33653e2d4b4a12881b9aa6ddb0
By moving three structures to the heap. None are in the hot decode
loop, instead having to do with file sync / header state.
Has neglible impact on performance (within measurement noise) on Clip+,
Rocker, and Xduoo X3.
On PP5022 (ipodmini2g) performance drops from 138.66% to 138.22% realtime.
(0.3%)
Unknown effect on Coldfire which lacks D$.
Stack savings are pretty significant especially on lowmem devices.
Change-Id: Ic8a1e93062ff5a46230e824134032053c4e1986d
apparently we should be doing this anyway
mark4o> The packets overlap and may reuse state set by other recent packets,
so if you seek to a different position,
resetting the state helps to ensure that the subsequent
packets won't use the state set by the unrelated packets
that were processed before the seek.
remove stack bump WORKAROUND_FS13060
Change-Id: I1c14e23b1721a360b91e3e55202c1557aef0fcc6
* Properly account for ID3v1 tags
* Play time computation fixes
* Add speech feedback
Patch by Igor Poretsky
Change-Id: Ia6df8fb171882a88527cfa9d3b76b705f09becdd
When the starting address of the plugin buffer
is not aligned to 8 bytes crashes occur in tlsf
(on ARM atleast)
Change-Id: I655500c25e1c8f84b4a2418e9ec5c5948e4bea82
opus requires the comment header to be a valid file our codec attemps to skip the comment data
in order to reduce the ram allocated originally it caused files with large album art to skip
the beginning of tracks my first attempt at fixing this then caused files with low bitrates
to do the same while fixing files with large album art
This patch should fix both although the initial start might be a bit slower but
this shouldn't cause too much of an issue
Change-Id: Ia1c3561347894cc45f24bb2659436914f8f03b43
knocks off about .5 second from decode time not a big change but might help a bit on
devices that barely achieve realtime
Change-Id: If6e822b7273613c9449c102ce7dd3543bf975d37
ogg_sync_reset() causes issues on the partial page boundary
due to the next page (already in buffer) being discarded
instead seek next page boundary past complete page
Change-Id: Ic05f188f489b015693d663f131e09cd92ad37ff7
Files with extension "aac" in ADTS or ADIF format are now playable.
Full credit goes to Igor Poretsky.
Change-Id: I413b34e15e5242fea60d3461966ae0984080f530
* More tolerance to the file format variations.
* AC3 coded files in realaudio format are now playable
Full credit to Igor Poretsky
Change-Id: Id24e94bc00623e89fb8c80403efa92f69ab1e5d7
In particular, this solves seeking glitches seen in ~6 hr mp3 files.
(Patch taken from Igor Poretsky's tree)
Change-Id: Id65b6726146b6d2d1a223e90b88e401d1b2d597a
On Classic, IRAM1 (second 128Kb of a total of 256KB available IRAM) is
slower than DRAM. Codecs that actually are using regions of IRAM1 runs
faster when DRAM is used, so IRAM1 is disabled and only IRAM0 remains
enabled: 48KB for core and 80KB for codecs/plugins.
The next test_codec results shows how decode time is decreased:
file boosted unboosted
*.ra ~1.5% ~0.5%
*.mpc ~21% ~4.5%
*.ogg ~0.5% ~0%
nero_he*.m4a ~8% ~1%
nero*.m4a ~25% ~7%
wmapro*.wma ~4.5% ~0%
wma*.wma ~25% ~7%
In addition there is a small power save when IRAM1 HW is disabled.
Change-Id: I102adee11458e82037f23076d5d5956e23235de8
Just use long so the compiler potentially doesn't complain about
use of other values not in the enum. It's also the type used
around the system for event ids.
Increase min codec API version.
No functional changes.
Change-Id: If4419b42912f5e4ef673adcdeb69313e503f94cc
The new vuprintf makes unnecessary workarounds due to formatting
limitations. I checked grep output for whatever appeared to fit
but it's possible I missed some instances because they weren't
so obvious.
Also, this means sound settings can dynamically work with any
number of decimals rather than the current assumption of one or
two. Add an ipow() function to help and take advantage of dynamic
field width and precision. Consolidate string formatting of sound
settings.
Change-Id: I46caf534859dfd1916cd440cd25e5206b192fcd8
This call was not needed in the first place, but was causing crashes in
sgt-puzzles. Removing it fixes the crashes.
Change-Id: I1149d5600e1c97e0e848fdd34bf65d54c930adab
This should make it build cleanly under -Wcast-align, which should
hopefully avoid any alignment issues on ARM.
Change-Id: Ie147323d2d8cb980dcbb94710387b7ee80826c4d
Most importantly is surround shouldn't operate in mono mode. Have it
watch and (de)activate itself on relevant format changes as it should.
Other changes to better handle buffer allocation failure.
PBE was set internally at 100 by default; SBZ.
Change-Id: I328e0b674e56751a255eae817d7892d685796b06
Simply extends the current isqrt() to be able to do fractional bits
and improves the initial estimate using clz(). iqrt() itself is
no more and is equivalent to fp_sqrt(x, 0). The original also had
a small bug where the guess comparision should have been >=, not >.
Uses no large integer math or division and is very accurate
(simply returns a truncated fraction).
Change-Id: I2ae26e6505df1770dc01e56220f7385369f90ae9
SUPPORTED SERIES:
- NWZ-E450
- NWZ-E460
- NWZ-E470
- NWZ-E580
- NWZ-A10
NOTES:
- bootloader makefile convert an extra font to be installed alongside the bootloader
since sysfont is way too small
- the toolsicon bitmap comes from the Oxygen iconset
- touchscreen driver is untested
TODO:
- implement audio routing driver (pcm is handled by pcm-alsa)
- fix playback: it crashes on illegal instruction in DEBUG builds
- find out why the browser starts at / instead of /contents
- implement radio support
- implement return to OF for usb handling
- calibrate battery curve (NB: of can report a battery level on a 0-5 scale but
probabl don't want to use that ?)
- implement simulator build (we need a nice image of the player)
- figure out if we can detect jack removal
POTENTIAL TODOS:
- try to build a usb serial gadget and gdbserver
Change-Id: Ic77d71e0651355d47cc4e423a40fb64a60c69a80
On Windows 64-bit, the size of long is 32-bit, thus any pointer to long cast is
not valid. In any case, one should use intptr_t and ptrdiff_t when casting
to integers. This commit attempts to fix all instances reported by GCC.
When relevant, I replaced code by the macros PTR_ADD, ALIGN_UP from system.h
Change-Id: I2273b0e8465d3c4689824717ed5afa5ed238a2dc
The mingw linker uses strlen() in some cases, and codeclib.c redefines it, that
leads to mingw runtime init to call into our strlen() and then ci->strlen() which
of course crashes. Apply the same fix as for malloc and friends: rename the symbol.
The codeclib.h include is necessary for normal builds.
Change-Id: Ifa85901a3e4a31cc0e10b4b905df348a239d5c99
In DEBUG build, the codec API struct is consider with DEBUG flag in apps/
but without DEBUG flah in rbcodecs/, leading to unmatched structure and horrible
crashes in some cases (mostly encoders). I have no idea why the codecs Makefile
removes the DEBUG flag (maybe for performance reasons?) but it cannot be right.
Change-Id: Idb2c5f66741408ec2939624590fc39c4cf69fc2b
The codec wasn't calling ci->set_offset() while decoding; as a result,
the saved offset in ci.id3->offset was only updated at the start of the
file and when seeking.
To reproduce the problem in the simulator or on a real device:
- Start playing an Opus file.
- Let it play until 15s, then turn the player off.
- Turn back on and resume playback. This'll resume correctly from 15s
(using time-based resume, I think, as the offset was 0?).
- Let it play until 30s, then turn the player off again.
- Turn back on and resume playback. This'll resume from 15s, based on
the initial position from last time, when it should resume from 30s.
I believe this will also fix FS#12799 ("Resuming opus file from bookmark
is not working correctly").
Change-Id: Iba67368e0029c968ef802693767e0722719bc38b
ffmpeg_bitstream.c is included in libcodec, so there doesn't seem to
be any reason for individual codecs to also compile it (and clobber
any previous copy while they're at it, leading to broken builds)
Change-Id: I2bedc277ab109f44a6e8feb3d12ed01a720e00a6
