Commit graph

167 commits

Author SHA1 Message Date
Chiwen Chang
572b36a51a fix surround & pbe dsp crash
check handle before clean up buffer in flush().

Change-Id: I36a130c45c9f5dce97aa723ef98922b6935ead75
2015-01-30 20:06:00 +01:00
Chiwen Chang
30784cc262 fix pbe/haas surround dsps: surround_enabled, redo flush functions.
surround_enabled was never true, end up dsp_surround_flush didn't work; Thats why a cracking noise occurs in right channel when moving track positions.

redo pbe/surround flush in a much simpler way suits the current single buffer style.

Change-Id: I394054ddfb164b82c90b3dcf49df4442db87d8d2
2015-01-22 13:28:34 +11:00
Frank Gevaerts
163ca14e58 Enable buflib and core_alloc for warble.
Most of the work comes from http://gerrit.rockbox.org/r/#/c/1088/
by Thomas Jarosch.

Change-Id: Iaa673dad2388d1e44fc95ffaa14bafadc6158101
2015-01-19 21:30:03 +01:00
Chiwen Chang
3ae0f32ac3 three new DSPs
perceptual bass enhancement
- a bbe-ish group delay corrction with Biophonic EQ boost.
- precut

auditory fatigue reduction
-reduce signal in frequency that may trigger temporary threshold shift

haas surround
-frequency between f(x1) and f(x2) is always bypassed.
-can apply to side only.

Change-Id: Icb6355ce9b1c99bf2c58c9385c3c411c0ae209d3
2015-01-19 19:34:01 +01:00
Thomas Jarosch
2a3e1628a5 Limit more variables to file scope
Change-Id: I30219d626316776eb73b4205d63376fa3dbc6361
2015-01-11 21:40:51 +01:00
Thomas Jarosch
e7c282fed7 More standard conforming codec_realloc()
- Leave original ptr untouched if allocation fails
  (bail out early)
- Behave like malloc() in case ptr is NULL

Change-Id: Ib854ca19bd0e069999b7780d2d9a533ece705add
2014-12-27 17:33:24 +01:00
Michael Giacomelli
d924c83066 Fix warning in WMA Pro and remove a c++ comment.
Change-Id: Id9b50c1fdeca4d87f158da717de8958330f027ef
2014-11-28 23:04:50 +01:00
Michael Giacomelli
aa2c55e105 Fix FS#13009.
This file revealed several problems with our ASF parser:

1)  The packet count in the ASF was actually a 64 bit value,
leading to overflow in very long files.

2)  Seeking blindly trusted the bitrate listed in the ASF header
rather than computing it from the packet size and number of packets.

Fix these problems and fix a few minor issues.

Change-Id: Ie0f68734e6423e837757528ddb155f3bdcc979f3
2014-11-28 22:30:05 +01:00
Marcin Bukat
da417ab93d fix yellow
Change-Id: Ie3aa9b208e3f4f17d4d02f11f69839e9b381217d
2014-09-22 10:55:11 +02:00
nialv7
d392da8002 metadata: Add cuesheet embedded in ape tags.
Change-Id: I5d9e731c3ea786fb910afbb0a5201fc68dcab9f9
Reviewed-on: http://gerrit.rockbox.org/965
Reviewed-by: Nick Peskett <rockbox@peskett.co.uk>
Tested: Nick Peskett <rockbox@peskett.co.uk>
Reviewed-by: Marcin Bukat <marcin.bukat@gmail.com>
2014-09-22 10:20:39 +02:00
Michael Sevakis
5d31d3c3bc Fix last warning for Warble
Unused result warnings will have to be dealt with separately.

Change-Id: I00c45e28d4d43a5376745036e650ff8df576c2db
2014-08-30 01:50:22 -04:00
Michael Sevakis
da4938d6ee Get the last errors I hope!
Change-Id: Ia285b95480cc9ac6494b745d80892c4b1b912341
2014-08-30 01:29:18 -04:00
Michael Sevakis
528715a672 Fix warnings from 6ed0087
Forgot to (void) an unused parameter when priorityless.

usb-drv-rl27xx.c was using a compound init to initialize a semaphore
but the structure changed so that it is no longer correct. Use
designated initializers to avoid having to complete all fields.

Forgot to break compatibility on all plugins and codecs since the
kernel objects are now different. Take care of that too and do the
sort thing.

Change-Id: Ie2ab8da152d40be0c69dc573ced8d697d94b0674
2014-08-16 06:00:36 -04:00
Nils Wallménius
8e8671a73e libopus: use iram for more constants
Speeds up decoding of the 64 kbps test file by 2.59 MHz and the
128 kbps test file by 4.31 MHz on H300 (cf). Decoding the same
files on c200 is sped up by 0.33 MHz and 0.55 MHz respectively.

Change-Id: I0f9f9ef6a7293581cf45e3201b33c65504c95c81
2014-07-13 14:19:54 +02:00
Nils Wallménius
888e05ec12 libopus: asm C_MUL for coldfire
The recent merge of upstream changed the fft to use C_MUL which
wasn't implemented in asm for coldfire.

Speeds up decoding 64 kbps test file by 2.68 MHz and 128 kbps
test file by 2.80 MHz on H300.

Change-Id: I8b61fc0f9568d6350431e311a12e44fe4f60f72e
2014-07-13 11:49:34 +02:00
Nils Wallménius
9b7ec42403 Sync to upstream libopus
Sync to commit bb4b6885a139644cf3ac14e7deda9f633ec2d93c

This brings in a bunch of optimizations to decode speed
and memory usage. Allocations are switched from using
the pseudostack to using the real stack. Enabled hacks
to reduce stack usage.

This should fix crashes on sansa clip, although some
files will not play due to failing allocations in the
codec buffer.

Speeds up decoding of the following test files:

                 H300 (cf)   C200 (arm7tdmi)  ipod classic (arm9e)
16 kbps (silk)   14.28 MHz   4.00 MHz         2.61 MHz
64 kbps (celt)   4.09 MHz    8.08 MHz         6.24 MHz
128 kbps (celt)  1.93 MHz    8.83 MHz         6.53 MHz

Change-Id: I851733a8a5824b61feb363a173091bc7e6629b58
2014-07-13 11:12:40 +02:00
Thomas Martitz
466441dc14 libmad: Use 32bit unsigned for requantize table.
Implicit promotion of integer literals to unsigned long introduced a subtle bug
on 64-bit systems due to weird sign extensions (leads to audible glitches in a
few files). The table is originally designed for unsigned 32bit integers, and
it works with those so use them. As a consequence the lookup table size is
halved as well.

Change-Id: I35d878d6df03300387f0e403e0f3c3bdc73eea00
2014-04-15 23:49:07 +02:00
Michael Sevakis
31b7122867 Implement time-based resume and playback start.
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>
2014-03-10 04:12:30 +01:00
Thomas Martitz
68768260e8 Fix more reds.
Change-Id: I4b58dda0953b7f9799238c32b78037b0a5403c04
2014-03-03 20:26:08 +01:00
Thomas Martitz
c245de029d Fix various reds. Some includes needed fixup.
Change-Id: I4327740bae17054131feb917abdd58846c451988
2014-03-03 19:10:48 +01:00
Jack Whitham
ca423ed0e3 Proposed fix for FS#12878: Zero-length embedded album art prevents mp3 playback
see http://www.rockbox.org/tracker/task/12878

Change-Id: Ib4233c06e18d1d193dfb9e73e745ca5d174e40b2
Reviewed-on: http://gerrit.rockbox.org/507
Reviewed-by: Michael Giacomelli <giac2000@hotmail.com>
Reviewed-by: Thomas Martitz <kugel@rockbox.org>
2013-12-23 17:55:15 +01:00
Nils Wallménius
e3c2ed7a71 Sync libopus to upstream release 1.1
Change-Id: I9fea7460fc33f60faff961b3389dd97b5191463c
2013-12-16 21:13:23 +01:00
Ryan Billing
d0918b98fa DSP Compressor: Sidechain, Exponential Atk/Rls
This is an improvement to the current compressor which I have added
to my own Sansa Fuze V2 build.  I am submitting here in case others
find it interesting.

Features added to the existing compressor:
Attack, Look-ahead, Sidechain Filtering.
Exponential attack and release characteristic response.

Benefits from adding missing features:
Attack:
Preserve perceived "brightness" of tone by letting onset transients
come through at a higher level than the rest of the compressed program
material.

Look-ahead:
With Attack comes clipping on the leading several cycles of a transient
onset.  With look-ahead function, this can be pre-emptively mitigated with
a slower gain change (less distortion).  Look-ahead limiting is implemented
to prevent clipping while keeping gain change ramp to an interval near 3ms
instead of instant attack.

The existing compressor implementation distorts the leading edge of a
transient by causing instant gain change, resulting in log() distortion.
This sounds "woofy" to me.

Exponential Attack/Release:
eMore natural sounding.  On attack, this is a true straight line of 10dB per
attack interval.  Release is a little different, however, sounds natural as
an analog compressor.

Sidechain Filtering:
Mild high-pass filter reduces response to low frequency onsets.  For example,
a hard kick drum is less likely to make the whole of the program material
appear to fade in and out.  Combined with a moderate attack time, such a
transient will ride through with minimal audible artifact.

Overall these changes make dynamic music sound more "open", more natural.  The
goal of a compressor is to make dyanamic music sound louder without necessarily
sounding as though it has been compressed.  I believe these changes come closer to this goal.

Enjoy.  If not, I am enjoying it

Change-Id: I664eace546c364b815b4dc9ed4a72849231a0eb2
Reviewed-on: http://gerrit.rockbox.org/626
Tested: Purling Nayuki <cyq.yzfl@gmail.com>
Reviewed-by: Michael Giacomelli <giac2000@hotmail.com>
2013-12-15 22:24:08 +01:00
Albert Song
f633d5ed48 Add support for flac embeded album art.
Change-Id: I077768f7d80b57976f9a7278b640ef67cf4f2af2
Reviewed-on: http://gerrit.rockbox.org/694
Reviewed-by: Thomas Martitz <kugel@rockbox.org>
2013-12-13 12:37:20 +01:00
Andrew Ryabinin
b770f63934 flac: fix seeking.
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>
2013-11-18 07:45:59 +01:00
Kevin Zheng
4626b1770b Add missing #include statements.
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>
2013-10-20 16:52:46 +02:00
Lorenzo Miori
0f1d44dba2 Simulator - encoders can now be loaded
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>
2013-10-05 12:25:13 +02:00
Nils Wallménius
b97cdc8f5e Opus: delete two files that were moved into a subdir
Change-Id: I54ef0dfd57fbb493ad38855767a8f5e724e5bc65
2013-09-01 18:36:12 +02:00
Nils Wallménius
3000ca32f9 Fix warning from a35c1b3
Change-Id: I0e9b2c265a6a2355dc39b1696df4c8f266d9a638
2013-09-01 17:54:10 +02:00
Nils Wallménius
a35c1b3595 Opus: Coldfire inline asm for comb_filter_const
Speeds up decoding a 64kbps test file by 2.6MHz

Change-Id: Ibeb30f37cc00a4a6f65b64851009753f40e06fc1
2013-09-01 17:39:15 +02:00
Nils Wallménius
516f7fbd6c Add cf asm inline for multiplication commonly used in silk.
Speeds up decoding a 16kbps test file by 4.9MHz on h300.

Change-Id: I8c25431c98dfa9a1c3806a84055e0847eb77a9f1
2013-08-31 17:57:33 +02:00
Nils Wallménius
b592a7a8a5 Put two hot silk arrays on real stack (iram)
Speeds up decoding of 16kbps test file by 16.7MHz on H300.

Change-Id: I39c90e3b423ae8e2ee5c2b88c5dcec8d48807f77
2013-08-31 17:14:58 +02:00
Nils Wallménius
a602ea3d3d Silence spurious warning
Change-Id: I856c722e959314c0a86e9c0a3a31cb824ddb41cc
2013-08-31 09:00:13 +02:00
Nils Wallménius
580b307fd7 Sync opus codec to upstream git
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
2013-08-31 08:30:51 +02:00
Nils Wallménius
b2e80edd16 Change CODECFLAGS to a "simply-expanded" var to give the individual
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>
2013-08-23 18:34:30 +02:00
Michael Sevakis
b1209d4789 Fix FS#12889 : Audible pop right after setting Repeat/Shuffle
The quickscreen calls settings_apply() and the crossfeed code wasn't
checking that the right crossfeed was set before updating the filter
for the custom setting, which was overwriting the Meier crossfeed
data (custom and Meier share the same data space).

Change-Id: Ifaa2f46fe062d4497681a2dd0d5068ec906c96a3
2013-08-16 09:28:36 -04:00
Michael Sevakis
e04e29d017 mp3_enc: Fix early snafu with stream finish on COP
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
2013-07-11 04:50:27 -04:00
Michael Sevakis
95bc93194e Multithread compressing encoders on multicore targets.
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
2013-07-09 06:28:33 -04:00
Michael Sevakis
d37bf24d90 Enable setting of global output samplerate on certain targets.
Replaces the NATIVE_FREQUENCY constant with a configurable frequency.

The user may select 48000Hz if the hardware supports it. The default is
still 44100Hz and the minimum is 44100Hz. The setting is located in the
playback settings, under "Frequency".

"Frequency" was duplicated in english.lang for now to avoid having to
fix every .lang file for the moment and throwing everything out of sync
because of the new play_frequency feature in features.txt. The next
cleanup should combine it with the one included for recording and
generalize the ID label.

If the hardware doesn't support 48000Hz, no setting will be available.

On particular hardware where very high rates are practical and desireable,
the upper bound can be extended by patching.

The PCM mixer can be configured to play at the full hardware frequency
range. The DSP core can configure to the hardware minimum up to the
maximum playback setting (some buffers must be reserved according to
the maximum rate).

If only 44100Hz is supported or possible on a given target for playback,
using the DSP and mixer at other samperates is possible if the hardware
offers them.

Change-Id: I6023cf0c0baa8bc6292b6919b4dd3618a6a25622
Reviewed-on: http://gerrit.rockbox.org/479
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
2013-07-06 04:22:04 +02:00
Michael Sevakis
4888131972 Update software recording engine to latest codec interface.
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>
2013-06-30 00:40:27 +02:00
Michael Sevakis
a9ea1a4269 Fix some whitespace in files changed in following commit.
Change-Id: Ie3f43e43076e0dcae9a10f1b0b9e4698b398acee
Reviewed-on: http://gerrit.rockbox.org/492
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
2013-06-30 00:40:09 +02:00
Michael Giacomelli
d475dd36a3 Remove old EQ presets.
The old presets never made sense for Rockbox's EQ.  They were apparently
copied from some other software.  We have a parametric EQ, that means that
EQ bands can be made wider or narrower.  Putting two identical bands side
by side just wastes battery life and adds rounding error.  Replacement
presets are on gerrit but they need more work.  In the mean time, users
should probably not be using these.

Change-Id: I85213100129fafd3ac0fa1a9438cb4d651bb94cb
2013-06-21 16:53:02 +02:00
Frank Gevaerts
d4061a46d8 Silence some (harmless) warnings.
Change-Id: I8d1278b8cfaa376d2ad5a99dd552dc980c66e1da
2013-06-16 18:23:18 +02:00
Dominik Riebeling
b6ddbc41a5 Fix id3v2 album art if more than one image is present.
Rockbox only uses the first album art image (APIC / PIC frame) found in id3v2
tags. When a file contains more than one image the second one is ignored but
the parsealbumart() callback overwrites the already set data. This causes the
metadata structure to contain an invalid pointer to the image data, resulting
in no image shown.

Make parsealbumart() aware of this and skip parsing when an albumart image has
already been found. Fixes FS#12870.

Change-Id: Id8164f319cd5e1ee868b581f8f4ad3ea69c17f77
2013-06-15 21:04:13 +02:00
Michael Sevakis
46688a60db Missed removing a couple unwanted includes in previous commit.
Get those too.

Change-Id: Id2a39afe7a61d6ec0cea38633b94fe1b7122204f
2013-05-27 03:40:02 -04:00
Michael Sevakis
b5a6517e9d Remove explicit config.h and system.h includes from DSP code.
Replace with rbcodecconfig.h and platform.h includes. Remove now-
unneeded ones as well.

Change-Id: I6111b71e90bf86d9fe272a7916f2d34a5c6dd724
2013-05-27 03:23:33 -04:00
Michael Sevakis
30fe6eb66c SPC Codec ARMv5: I didn't have fast gauss quite right.
Fix wrapping hazard which did eventually manifest on the right file.

Change-Id: I996a6efd3181b56fd172b5c3a526c7434f88bbbe
2013-05-26 00:33:30 -04:00
Michael Sevakis
6e211ab3ac Remove dsp_callback because DSP is now library code, not app code.
Yep, nope, not necessary anymore. Just call functions directly.

Change-Id: I21dc35f8d674c2a9c8379b7cebd5613c1f05b5eb
2013-05-23 14:25:37 -04:00
Michael Sevakis
33f3af2b8d SPC Codec: Add ARMv5 optimized code. Easy peasy.
Why? Why not? Cuts a few MHz.

Change-Id: Ied5c70b1aedd255cbe5d42b7d3028bbe47aad01d
2013-05-23 03:15:12 -04:00
Michael Sevakis
9b43f14165 SPC Codec: Simplify configuration and assume nothing need be disabled.
Most SoCs are these days are fast enough for realtime BRR, gaussian
interpolation and echo processing.

Change-Id: I180ce8ad45242c67b5e573a406b9522098a3f12b
2013-05-21 20:39:22 -04:00