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rockbox/lib/rbcodec/dsp/dsp_misc.h

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Fundamentally rewrite much of the audio DSP. 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>
2012-03-27 23:52:15 +00:00
/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Miika Pekkarinen
* Copyright (C) 2005 Magnus Holmgren
* Copyright (C) 2007 Thom Johansen
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#ifndef DSP_MISC_H
#define DSP_MISC_H
/* Set the tri-pdf dithered output */
void dsp_dither_enable(bool enable); /* in dsp_sample_output.c */
Make DSP's replaygain independent of global_settings. 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 <jethead71@rockbox.org>
2012-04-30 20:27:01 +00:00
enum replaygain_types
{
REPLAYGAIN_TRACK = 0,
REPLAYGAIN_ALBUM,
REPLAYGAIN_SHUFFLE,
REPLAYGAIN_OFF
};
struct replaygain_settings
{
bool noclip; /* scale to prevent clips */
int type; /* 0=track gain, 1=album gain, 2=track gain if
shuffle is on, album gain otherwise, 4=off */
int preamp; /* scale replaygained tracks by this */
};
Fundamentally rewrite much of the audio DSP. 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>
2012-03-27 23:52:15 +00:00
/* Structure used with REPLAYGAIN_SET_GAINS message */
#define REPLAYGAIN_SET_GAINS (DSP_PROC_SETTING+DSP_PROC_MISC_HANDLER)
struct dsp_replay_gains
{
long track_gain;
long album_gain;
long track_peak;
long album_peak;
};
Make DSP's replaygain independent of global_settings. 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 <jethead71@rockbox.org>
2012-04-30 20:27:01 +00:00
void dsp_replaygain_set_settings(const struct replaygain_settings *settings);
Fundamentally rewrite much of the audio DSP. 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>
2012-03-27 23:52:15 +00:00
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 18:19:06 +00:00
#ifdef HAVE_PITCHCONTROL
void dsp_set_pitch(int32_t pitch);
int32_t dsp_get_pitch(void);
#endif /* HAVE_PITCHCONTROL */
Fundamentally rewrite much of the audio DSP. 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>
2012-03-27 23:52:15 +00:00
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>
2013-05-23 17:58:51 +00:00
/* Set output samplerate for all DSPs */
void dsp_set_all_output_frequency(unsigned int samplerate);
/* Return DSP's output samplerate */
struct dsp_config;
unsigned int dsp_get_output_frequency(struct dsp_config *dsp);
Fundamentally rewrite much of the audio DSP. 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>
2012-03-27 23:52:15 +00:00
#endif /* DSP_MISC_H */
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