GCC 4.9 always emits assembly with divided syntax. Setting unified
syntax in inline assembly causes the assembler to complain about
GCC's generated code, because the directive extends past the scope
of the inline asm. Fix this by setting divided mode at the end of
the inline assembly block.
The assembler directives are hidden behind macros because later
versions of GCC won't need this workaround: they can be told to
use the unified syntax with -masm-syntax-unified.
Change-Id: Ic09e729e5bbb6fd44d08dac348daf6f55c75d7d8
Replace the minimum version bound with a check on the size of
the API struct. The version only needs to be incremented for
ABI breaking changes. Additions to the API won't need to touch
the version number, resulting in fewer merge conflicts.
Change-Id: I916a04a7bf5890dcf5d615ce30087643165f8e1f
Plain import of the library parts first. Adaptions to Rockbox will
follow. A *lot* of kudos go to Mihaly Horvath for creating this library
from his already lightweight cSID-light, mainly for Rockbox. Besides a
lot of other things, he made his algorithms integer-only and
significantly improved the C64 emulation, so finally RSIDs could be
played as well as PSIDs. TinySID was nice for what it is, but this is a
quantum leap in SID playback quality for Rockbox. Check for example:
https://hvsc.csdb.dk/MUSICIANS/P/Page_Jason/Eighth.sidhttps://hvsc.csdb.dk/MUSICIANS/J/Jeff/Blowing.sid
Change-Id: I353e12fbfd7cd8696b834616e55743e7b844a73e
It turns out removing DSP_INIT broke the codec ABI and caused
old codecs to crash; the loop and mdelay() was a red herring.
This reverts commit 541960a110.
Change-Id: I020d826e7b4beb006d093d9c3d4f45fa5eaac717
Commit 6bcd830490 ported an optimization to decode_subframe_fixed()
from FFmpeg (upstream commit 08965b22e2). This contains an out of
bounds read, which doesn't affect the decoder output, but makes ASAN
complain.
FFmpeg fixed the out of bounds read (upstream commit 0ec7b71de8) but
that appears to increase code size a lot.
Inlining the initialization of a, b, c, d into the switch produces
similar code as the non-bounds-checked version with only a handful
of instructions of overhead (checked on MIPS & ARM).
Change-Id: I053fac4efc4676b133eb7545c80e23f37fb00d86
Resume by offset was obviously inaccurate for ALAC -- it tried
to convert the offset to an elapsed time using the approximate
bitrate, which is going to be wrong for VBR files. This became
a problem since commit 26ffcd8f9f restored the ability to resume
by offset.
It turns out that m4a_seek_raw() has terrible resolution since
it can only seek to chunk boundaries, and lies about the real
sample position; basically the same issue that affected seeking
described in commit 4dd3c2b33e. Resuming by offset is still not
very accurate because of this. Prefer to resume by time first,
which is normally highly accurate (and never worse than offset)
but use the file offset if it's the only thing we have.
There were a couple time calculations still using 32-bit math,
so clean those up too to reduce issues due to rounding errors.
Change-Id: Idd3bccd67505f4e59e784d92e45ea80a273975bb
The codec used 32-bit math for elapsed time <-> file position
calculations. The rounding errors seem to be the cause of poor
seek/resume accuracy on long VBR files; switching to 64-bit math
makes things much better.
Change-Id: Iba638d9e031a891022510c31c141cc4541e3f149
It fixes Playback/Bookmarks Resume for long vbr mp3 files
It also fixes resume by time for asf files.
As a replacement for https://gerrit.rockbox.org/r/c/rockbox/+/4750
Change-Id: Iaa59b5862385f5fe91fdc2fb0b1fde8ce75c0b54
Seeking doesn't work well in M4A files with very few chunks due to
the seek method used (chunk based using the info in the 'stco' atom).
According to libm4a/demux.c the expected seek resolution using this
method is 1/4 to 1/2 seconds. However, ffmpeg generates files with a
1 megabyte chunk size, so the resolution is much worse than expected
on some files: around 30-40 seconds at 256kbps.
There was a bug with the seek position reported back to Rockbox: the
codec pretended it could seek exactly to the requested sample, but it
would only seek to the start of a chunk. This could leave the UI in a
confusing state because the real playback position was different from
what the elapsed time showed. Fix this by recalculating the reported
sample position using the chunk start.
To fix the low seek accuracy, use the table in the 'stsz' atom to skip
individual packets within a chunk. This is very accurate, but it takes
a lot of RAM to allocate the table. Currently the table is not allowed
to use more than half of the codec RAM, which should suffice for short
files on most targets. On files where the table is too large the codec
will fall back to the less accurate chunk-based seek method.
Change-Id: Ide38ea846c1cdd69691e9b1e1cd87eb0fa11cf78
* direct use of memcpy() instead of ci->memcpy() in flac and mod
* uninitialized variable in mpegplayer
Change-Id: I2d08682d5f66c319780e69e3ff63d600c61d8f5a
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
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
'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
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
