rockbox/lib/rbcodec/dsp/dsp_misc.h, branch master

Enable setting of global output samplerate on certain targets.

2013-07-06T02:22:04+00:00

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 
Tested-by: Michael Sevakis

Remove dsp_callback because DSP is now library code, not app code.

2013-05-23T18:25:37+00:00

Yep, nope, not necessary anymore. Just call functions directly.

Change-Id: I21dc35f8d674c2a9c8379b7cebd5613c1f05b5eb

Straighten out the mad twisted state of sound.c and related areas.

2013-04-15T16:02:05+00:00

This is going right in since it's long overdue. If anything is goofed,
drop me a line or just tweak it yourself if you know what's wrong. :-)

Make HW/SW codec interface more uniform when emulating HW functionality
on SWCODEC for functions such as "audiohw_set_pitch". The firmware-to-
DSP plumbing is in firmware/drivers/audiohw-swcodec.c. "sound_XXX"
APIs are all in sound.c with none in DSP code any longer.

Reduce number of settings definitions needed by each codec by providing
defaults for common ones like balance, channels and SW tone controls.

Remove need for separate SIM code and tables and add virtual codec header
for hosted targets.

Change-Id: I3f23702bca054fc9bda40f49824ce681bb7f777b

Rename HAVE_PITCHSCREEN to HAVE_PITCHCONTROL

2012-05-09T12:32:38+00:00

Also move the definition to config.h

Change-Id: I36bb5020c5e06b2344292bc05e8c13ccc7a6a1ff
Reviewed-on: http://gerrit.rockbox.org/234
Reviewed-by: Nils Wallménius

Should've had dsp_replaygain_set_gains as static for now...

2012-04-30T21:51:05+00:00

...because currently gains are only set through dsp_configure.

Change-Id: I2866473a82fdd5f41de4705b45928daa7e43f8eb

Make DSP's replaygain independent of global_settings.

2012-04-30T20:47:37+00:00

Moves replaygain definitions to lib/rbcodec/dsp/dsp_misc.h.
Intermediate functions in misc.c handle any adjustment and calling
the rbcodec APIs.

Change-Id: I9f03561bca9aedd13760cf19c4e19aa3c68e7024
Reviewed-on: http://gerrit.rockbox.org/140
Reviewed-by: Michael Sevakis

Fundamentally rewrite much of the audio DSP.

2012-04-29T08:00:56+00:00

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 
Tested-by: Michael Sevakis