This complements offset-based resume and playback start funcionality.
The implementation is global on both HWCODEC and SWCODEC.
Basically, if either the specified elapsed or offset are non-zero,
it indicates a mid-track resume.
To resume by time only, set elapsed to nonzero and offset to zero.
To resume by offset only, set offset to nonzero and elapsed to zero.
Which one the codec uses and which has priority is up to the codec;
however, using an elapsed time covers more cases:
* Codecs not able to use an offset such as VGM or other atomic
formats
* Starting playback at a nonzero elapsed time from a source that
contains no offset, such as a cuesheet
The change re-versions pretty much everything from tagcache to nvram.
Change-Id: Ic7aebb24e99a03ae99585c5e236eba960d163f38
Reviewed-on: http://gerrit.rockbox.org/516
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested: Michael Sevakis <jethead71@rockbox.org>
As comment in code states:
"It is possible for our seek to land in the middle of audio
data that looks exactly like a frame header from a future
version of an encoder. When that happens, frame_sync() will
return false. But there is a remote possibility that it is
properly synced at such a "future-codec frame", so to make sure,
we wait to see several "unparseable" errors in a row before
bailing out."
Currently we wait for 10 "unparseable" errors. libFLAC waits for 20.
But I've got a valid flac+cue, wherein switching to certain track
gave me 24 "unparsaeable" errors. Therefore I increased
unparseable_count to 30.
Change-Id: I4e97a5385c729adf3d5075d41ea312622c69e548
Reviewed-on: http://gerrit.rockbox.org/658
Reviewed-by: Michael Giacomelli <giac2000@hotmail.com>
Reviewed-by: Boris Gjenero <boris.gjenero@gmail.com>
Tested-by: Andrew Ryabinin <ryabinin.a.a@gmail.com>
Reviewed-by: Andrew Ryabinin <ryabinin.a.a@gmail.com>
Although Linux accepts several implicit definitions of SEEK_END found in
stdio.h, the compiler on FreeBSD won't. Rockbox compilation will fail
without stdio.h included.
There is a precedent for including this header, see
lib/rbcodec/codecs/libtremor/ivorbisfile.h.
Change-Id: I58510101b59a354cd6601cb3f323f385a824d2e8
Reviewed-on: http://gerrit.rockbox.org/639
Tested-by: Kevin Zheng <kevinz5000@gmail.com>
Reviewed-by: Frank Gevaerts <frank@gevaerts.be>
This enables the encoders - i.e. to record audio -
to be loaded also on the simulator.
Change-Id: I54fdbeb75b89023c0d7824a34cf76301c02c3150
Reviewed-on: http://gerrit.rockbox.org/632
Reviewed-by: Thomas Martitz <kugel@rockbox.org>
Sync opus codec to upstream commit
02fed471a4568852d6618e041c4f2af0d7730ee2 (August 30 2013)
This brings in a lot of optimizations but also makes the diff
between our codec and the upstream much smaller as most of our
optimizations have been upstreamed or supeceded.
Speedups across the board for CELT mode files:
64kbps 128kbps
H300 9.82MHz 15.48MHz
c200 4.86MHz 9.63MHz
fuze v1 10.32MHz 15.92MHz
For the silk mode test file (16kbps) arm targets get a speedup
of about 2MHz while the H300 is 7.8MHz slower, likely because it's
now using the pseudostack more rather than the real stack which
is in iram. Patches to get around that are upcomming.
Change-Id: Ifecf963e461c51ac42e09dac1e91bc4bc3b12fa3
codec makefiles larger freedom in what they can do to it.
Use this in libopus to prepend the libopus searchpaths to
CODECFLAGS so that its internal config.h will be picked up before
our global one. This avoids having to do a s/config.h/opus_config.h/
when syncing which will be handy soon.
Change-Id: I018d729aa0c8300fa3149f22a5a8c5668b339dfa
Reviewed-on: http://gerrit.rockbox.org/496
Reviewed-by: Nils Wallménius <nils@rockbox.org>
Distractions make logic fail. It only needs one more loop and should
not trigger further compression cycles after not feeding more data.
Change-Id: Ie0dbb34af92e0ca5718480dd4ab4719a141717ff
For mp3_enc, split encoding duties between COP and CPU.
For wavpack_enc, simply run the encoding on COP (splitting that one
needs more consideration) which keeps the it and the UI from running
on the same core.
As a result, at least they are now useable on PP at "normal" sample
rates.
mp3_enc in all this gets an extensive renovation and some optimizations
for speed, to reduce IRAM requirements and remove unneeded stuff.
Change-Id: I215578dbe36f14e516b05a5ca70880eb01ca0ec2
Basically, just give it a good rewrite.
Software codec recording can be implemented in a more straightforward
and simple manner and made more robust through the better codec
control now available.
Encoded audio buffer uses a packed format instead of fixed-size
chunks and uses smaller data headers leading to more efficient usage.
The greatest benefit is with a VBR format like wavpack which needs
to request a maximum size but only actually ends up committing part
of that request.
No guard buffers are used for either PCM or encoded audio. PCM is
read into the codec's provided buffer and mono conversion done at
that time in the core if required. Any highly-specialized sample
conversion is still done within the codec itself, such as 32-bit
(wavpack) or interleaved mono (mp3).
There is no longer a separate filename array. All metadata goes
onto the main encoded audio buffer, eliminating any predermined
file limit on the buffer as well as not wasting the space for
unused path queue slots.
The core and codec interface is less awkward and a bit more sensible.
Some less useful interface features were removed. Threads are kept
on narrow code paths ie. the audio thread never calls encoding
functions and the codec thread never calls file functions as before.
Codecs no longer call file functions directly. Writes are buffered
in the core and data written to storage in larger chunks to speed up
flushing of data. In fact, codecs are no longer aware of the stream
being a file at all and have no access to the fd.
SPDIF frequency detection no longer requires a restart of recording
or plugging the source before entering the screen. It will poll
for changes and update when stopped or prerecording (which does
discard now-invalid prerecorded data).
I've seen to it that writing a proper header on full disk works
when the format makes it reasonably practical to do so. Other cases
may have incorrect data sizes but sample info will be in tact. File
left that way may play anyway.
mp3_enc.codec acquires the ability to write 'Info' headers with LAME
tags to make it gapless (bonus).
Change-Id: I670685166d5eb32ef58ef317f50b8af766ceb653
Reviewed-on: http://gerrit.rockbox.org/493
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
Most SoCs are these days are fast enough for realtime BRR, gaussian
interpolation and echo processing.
Change-Id: I180ce8ad45242c67b5e573a406b9522098a3f12b
Affected BRR cached waveforms but not realtime BRR decode as far as
I could ascertain. BRR cached waves required loop points to be inside
the initial waveform but this change removes that restriction.
Change-Id: I0ef4db720e5c28bd7b2fb9ae255d27c0a7213f79
CPU optimization gets its own files in which to fill-in optimizable
routines.
Some pointless #if 0's for profiling need removal. Those macros are
empty if not profiling.
Force some functions that are undesirable to be force-inlined by the
compiler to be not inlined.
Change-Id: Ia7b7e45380d7efb20c9b1a4d52e05db3ef6bbaab
Use the tlsf malloc and friends instead of the silly
codec_malloc to get actually working free and saner
realloc that doesn't leak memory.
Makes files with moderately sized embedded AA play
on targets with large enough codec buffers and files
with too large AA are now skipped rather than crashing.
Fixes crash when playing example file in FS#12842.
Change-Id: I06562955c4d9a95bd90f55738214fba462092b71
DSP_SWITCH_FREQUENCY has been deprecated and the same enumerated value
as DSP_SET_FREQUENCY since major DSP revisions were committed. This
task should have been performed much earlier but, oh well, do it now.
Change-Id: I3f30d651b894136a07c7e17f78fc16a7d98631ff
Will need it soon enough.
Combine the contents of all the various fixedpoint.h files.
Not moving fixedpoint.c for now since I'm not sure where it
should be and it causes some dependency issues.
Change-Id: Ideacbca2ca78f9158c2b114b113c274f68e908d5
Prevents cutoff of tracks, especially short ones:
* Extend looped tracks by fade length to fade at start of loop repeat.
* No fade occurs for non-repeating track only having an intro.
* Uses id3.tail_trim field to store fade duration.
Use libGME built-in elapsed time reporting instead of custom calculation:
* libGME already reports in milliseconds.
* Don't advance time counter when Repeat == One. It just runs the progress
over the length limit.
Fix a comment about sample rate and set the reported bitrate to be
accurate for 44.1 kHz stereo.
Change-Id: I3ede22bda0f9a941a3fef751f4d678eb0027344c
Flush decoder state and frame out buffer upon a forced stop to prevent
a short burst of stale audio from the previously decoding track from
playing when skipping from one WMA track to another.
Change-Id: I24c910c5dbd83caed2510db68d9e39a474332a79
Reviewed-on: http://gerrit.rockbox.org/406
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
Speeds up decoding of 128k opus files by 1.2MHz on AMSv2. Rounding
error is 1 bit due to KissFFT using a 15 bit shift instead of a 16 bit shift.
Also, change an LDMIA in the armv4 code to LDM as the pointer should not
increment.
Change-Id: I626a207c6a056a1984e33cfe89415c35d0caed93
Reviewed-on: http://gerrit.rockbox.org/377
Reviewed-by: Michael Giacomelli <giac2000@hotmail.com>
Tested-by: Michael Giacomelli <giac2000@hotmail.com>
I'm not 100% sure that the rounding of denormals is correct. As compared to foobar2000,
some samples are off by +1 LSB. However, since I can't output 24 bit PCM easily with
rockbox, I'm not sure if this is due to a bug or just how rockbox rounds. In practice
I don't think it matters so I'm just going to commit this for now.
Change-Id: Ic0792fcb172e4369a5512d202121c2b918b36079
avoids complicated index calculations in the loops.
saves 0.3MHz decoding a 64kbps test file on h300 (cf) and
0.2MHz on c200 (pp)
Change-Id: I1918912d9a4502f89980c6bb270ec2ef10a07010
Signed-off-by: Nils Wallménius <nils@rockbox.org>
on a target with a disk.
Change-Id: I37c875c9cd014eb61fe5232dab0f4b8f15f057dd
Reviewed-on: http://gerrit.rockbox.org/319
Tested-by: Thiago Okada <thiago.mast3r@gmail.com>
Reviewed-by: Frederik Vestre <freqmod@gmail.com>
Tested-by: Frederik Vestre <freqmod@gmail.com>
speeds up decoding of a 64kbps test_file by 1.5MHz on c200 (pp)
and 1.9MHz on fuzev1 (amsv1)
Change-Id: I1db460b634eba608c3e00541d96fc93d5a05710b
Signed-off-by: Nils Wallménius <nils@rockbox.org>
speeds up decoding of a 64kbps test file by 0.5Hz on h300 (cf)
0.9MHz on c200 (pp) and 0.2MHz on fuzev1 (amsv1)
Change-Id: Ib537c2393fa6dca0b61e4e9f80eef5e688c2c2bd
Signed-off-by: Nils Wallménius <nils@rockbox.org>
Unroll overlap add loop by four and use memcpy for copying
instead of loops.
Change-Id: I17114626a395d5972130251d892f851bc86e3a6a
Signed-off-by: Nils Wallménius <nils@rockbox.org>
Replace complicated macro doing three 16*16 muls and add an inline
asm implementation for arm, speeds up decoding a 64kbps test file
by 0.5MHz on c200 (pp) and gives slightly better precision.
Change-Id: I6fc5b83c210f01bffdc38aec54cc5a8b646d8169
Signed-off-by: Nils Wallménius <nils@rockbox.org>
Hoist load of coefficients out of the loop.
Speeds up decoding of a 64kbps test file by 0.6MHz on h300 (cf)
0.2MHz on c200 (pp) and 0.1MHz on fuzev1 (amsv1)
Signed-off-by: Nils Wallménius <nils@rockbox.org>
Change-Id: I4be0059fc2a77748575f5fc9378f7f348d64f1c4
Skip expensive multiply-accumulate loop when gains are 0 and
just copy using memcpy if soure and destination are not the same
Speeds up decoding of a 64kbps test file by 6MHz on h300 (cf)
7MHz on c200 (pp) and 6MHz on fuzev1 (amsv1)
Change-Id: Ibbc9ddfd45a9ac661467b1327b8c67761924fb8b
Signed-off-by: Nils Wallménius <nils@rockbox.org>
Reorder operands to take advantage of the early termination of
multiplications. Saves 2.5MHz decoding a 64kbps opus test file
on c200 (pp).
Change-Id: I470266dc870ab183ece3b23426d41e2a64342a71
Speeds up decoding of 64kbps test file by 6.3MHz on h300 (cf)
and 1.2MHz on c200 (pp).
Signed-off-by: Nils Wallménius <nils@rockbox.org>
Change-Id: I08c2c332153abcbef9447c81986777fd2fcc73fe
speeds up decoding of 64kbps test file by 19MHz on h300 (cf)
and 2.5MHz on c200 (pp)
Change-Id: Idacd2f8962c20c518055d586daeec6b932b7ded2
Signed-off-by: Nils Wallménius <nils@rockbox.org>
Speeds up decoding of a 64kbps test file 26MHz on H300 (cf) and
2MHz on c200 (pp)
Change-Id: I2fb4fe6c0a29321087e02fbd17fd1b1eb84e7b57
Signed-off-by: Nils Wallménius <nils@rockbox.org>
Remove downsampling code from deemphasis loop as we don't use
it and remove multiplications that are not relevant when
not using custom modes. Saves 1.4MHz on h300 (cf), 4.3MHz on
c200 (pp) and 4.6 on fuzev1 (amsv1).
Change-Id: Iab3f1d737a656a563aaa351d50db987a9cff2287
Saves about 30MHz on h300 (cf) and 1.5MHz on c200 (pp) decoding a
64kbps test file. Stack usage is still below 70%.
Change-Id: Ib13df9011adb4eef4bb91a52e5a32741c8bf8988
Speeds up decoding of a 64kbps opus test file by 34MHz on h300 (cf),
24MHz on c200 (pp) and 13MHz on fuzev1 (amsv1)
Change-Id: I0dce6b3bfe6c81d0a722dfebb13891b9a428c6ba
Synchronised with opus repo on github (https://github.com/freqmod/rockbox-opus)
Status:
* Seeking ported from speex, but fails on some cases (e.g. seek to granule 0)
* ReplayGain parsing needs to be reworked, we do vorbis-style replaygain now.
http://wiki.xiph.org/OggOpus#Comment_Header explicitly forbids these in
favour of R128_TRACK_GAIN tag.
* No optimisation yet, source files still nearly identical to opus upstream
* Multi-stream opus files may not be parsed correctly
Change-Id: Ia66f1027dc1d288083e3c57b2816700078376f9a
Reviewed-on: http://gerrit.rockbox.org/300
Reviewed-by: Bertrik Sikken <bertrik@sikken.nl>
Tested-by: Bertrik Sikken <bertrik@sikken.nl>
Since gcc 4.4 the MIPS port no longer recognizes the "h" asm constraint.
It was necessary to remove this constraint in order to avoid generating
unpredictable code sequences. We can achieve the same effect using
128-bit types.
See also:GCC 4.4 release notes at http://gcc.gnu.org/gcc-4.4/
Change-Id: I713cdf57cde1a989ad960aa441ab1ccf51f1cdc6
Several of the problem samples on the tracker use values outside this
range. Trying the larger table doesn't quite seem to fix things, but
its only a small amount of additional memory and looking at ffmpeg,
I think the larger table is correct.
Change-Id: Id046e62b68550701aa1f80c9abd0a1dcd711bd0d
No known samples are fixed by this problem, but I haven't tested many.
Backport of ffmpeg revision 26388.
Change-Id: Ife9654b7477a432834e3cab2cb43d16da071445a
It was only needed by the old arm toolchain that we no longer use or support.
Change-Id: Id0e6c67477f8834a637079b03cde5fbf9da68b1c
Reviewed-on: http://gerrit.rockbox.org/233
Reviewed-by: Nils Wallménius <nils@rockbox.org>
This would zero the first 4 or 8 bytes of the array because it is declared as a pointer
rockbox/lib/rbcodec/codecs/libasap/asap.c:1229:44: warning: argument to 'sizeof' in 'memset' call is the same expression as the destination; did you mean to provide an explicit length? [-Wsizeof-pointer-memaccess]
memset(ast -> memory, 0, sizeof(ast -> memory));
~~~~~~~~~~~~~ ~~~~~~~^~~~~~
librbcodec users must provide these two files when the library is built.
rbcodecconfig.h provides configuration #defines and basic types, and
will be included by public librbcodec headers, so it must not conflict
with the user's code. rbcodecplatform.h provides various OS functions,
and will only be included by source files and private headers. This
system is intended to provide maximum flexibility for use on embedded
systems, where no operating system headers are included. Unix systems
can just copy rbcodecconfig-example.h and rbcodecplatform-unix.h with
minimal changes.
Change-Id: I350a2274d173da391fd1ca00c4202e9760d91def
Reviewed-on: http://gerrit.rockbox.org/143
Reviewed-by: Nils Wallménius <nils@rockbox.org>
Tested-by: Nils Wallménius <nils@rockbox.org>
The LSP feature in WMA requires that the noise table values be
doubled verses when it is not used. Unfortunately, the previous
code would double the same values every time a LSP file was
decoded without first resetting them to their original values.
Change the code to check if the values are already doubled, and
then double/halve them as needed. This is still a bit ugly,
in the future consider using the built in rockbox dither instead
of a lookup table.
Fixes playback when skipping back and forth between low and high
bitrate WMA.
Change-Id: I4c393092e4a789bc8f98d74274fe207400b9550e
Reviewed-on: http://gerrit.rockbox.org/226
Reviewed-by: Michael Giacomelli <giac2000@hotmail.com>
Tested-by: Michael Giacomelli <giac2000@hotmail.com>
Creates a standard buffer passing, local data passing and messaging
system for processing stages. Stages can be moved to their own source
files to reduce clutter and ease assimilation of new ones. dsp.c
becomes dsp_core.c which supports an engine and framework for effects.
Formats and change notifications are passed along with the buffer so
that they arrive at the correct time at each stage in the chain
regardless of the internal delays of a particular one.
Removes restrictions on the number of samples that can be processed at
a time and it pays attention to destination buffer size restrictions
without having to limit input count, which also allows pcmbuf to
remain fuller and safely set its own buffer limits as it sees fit.
There is no longer a need to query input/output counts given a certain
number of input samples; just give it the sizes of the source and
destination buffers.
Works in harmony with stages that are not deterministic in terms of
sample input/output ratio (like both resamplers but most notably
the timestretch). As a result it fixes quirks with timestretch hanging
up with certain settings and it now operates properly throughout its
full settings range.
Change-Id: Ib206ec78f6f6c79259c5af9009fe021d68be9734
Reviewed-on: http://gerrit.rockbox.org/200
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
