d37bf24d90
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>
210 lines
6.2 KiB
C
210 lines
6.2 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2006-2007 Thom Johansen
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* Copyright (C) 2012 Michael Sevakis
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#include "rbcodecconfig.h"
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#include "fixedpoint.h"
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#include "fracmul.h"
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#include "dsp_filter.h"
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#include "dsp_proc_entry.h"
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#include "dsp_core.h"
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#include "dsp_misc.h"
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#include "eq.h"
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#include "pga.h"
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#include "replaygain.h"
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#include <string.h>
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/**
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* Current setup is one lowshelf filters eight peaking filters and one
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* highshelf filter. Varying the number of shelving filters make no sense,
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* but adding peaking filters is possible. Check EQ_NUM_BANDS to have
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* 2 shelving filters and EQ_NUM_BANDS-2 peaking filters.
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*/
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#if EQ_NUM_BANDS < 3
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/* No good. Expect at least 1 peaking and low/high shelving filters */
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#error Band count must be greater than or equal to 3
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#endif
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/* Cached band settings */
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static struct eq_band_setting settings[EQ_NUM_BANDS];
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static struct eq_state
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{
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uint32_t enabled; /* Mask of enabled bands */
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uint8_t bands[EQ_NUM_BANDS+1]; /* Indexes of enabled bands */
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struct dsp_filter filters[EQ_NUM_BANDS]; /* Data for each filter */
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} eq_data IBSS_ATTR;
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#define FOR_EACH_ENB_BAND(b) \
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for (uint8_t *b = eq_data.bands; *b < EQ_NUM_BANDS; b++)
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/* Clear histories of all enabled bands */
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static void eq_flush(void)
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{
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if (eq_data.enabled == 0)
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return; /* Not initialized yet/no bands on */
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FOR_EACH_ENB_BAND(b)
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filter_flush(&eq_data.filters[*b]);
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}
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static void update_band_filter(int band, unsigned int fout)
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{
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/* Convert user settings to format required by coef generator
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functions */
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typeof (filter_pk_coefs) *coef_gen = filter_pk_coefs;
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/* Only first and last bands are not peaking filters */
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if (band == 0)
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coef_gen = filter_ls_coefs;
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else if (band == EQ_NUM_BANDS-1)
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coef_gen = filter_hs_coefs;
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const struct eq_band_setting *setting = &settings[band];
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struct dsp_filter *filter = &eq_data.filters[band];
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coef_gen(fp_div(setting->cutoff, fout, 32), setting->q ?: 1,
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setting->gain, filter);
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}
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/* Resync all bands to a new DSP output frequency */
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static void update_samplerate(unsigned int fout)
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{
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if (eq_data.enabled == 0)
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return; /* Not initialized yet/no bands on */
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FOR_EACH_ENB_BAND(b)
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update_band_filter(*b, fout);
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}
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/** DSP interface **/
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/* Set the precut gain value */
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void dsp_set_eq_precut(int precut)
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{
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pga_set_gain(PGA_EQ_PRECUT, get_replaygain_int(precut * -10));
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}
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/* Update the filter configuration for the band */
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void dsp_set_eq_coefs(int band, const struct eq_band_setting *setting)
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{
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if (band < 0 || band >= EQ_NUM_BANDS)
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return;
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settings[band] = *setting; /* cache setting */
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struct dsp_config *dsp = dsp_get_config(CODEC_IDX_AUDIO);
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/* NOTE: The coef functions assume the EMAC unit is in fractional mode,
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which it should be, since we're executed from the main thread. */
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uint32_t mask = eq_data.enabled;
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/* Assume a band is disabled if the gain is zero */
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mask &= ~BIT_N(band);
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if (setting->gain != 0)
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{
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mask |= BIT_N(band);
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update_band_filter(band, dsp_get_output_frequency(dsp));
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}
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if (mask == eq_data.enabled)
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return; /* No change in band-enable state */
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if (mask & BIT_N(band))
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filter_flush(&eq_data.filters[band]); /* Coming online */
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eq_data.enabled = mask;
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/* Only be active if there are bands to process - if EQ is off, then
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this call has no effect */
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dsp_proc_activate(dsp, DSP_PROC_EQUALIZER, mask != 0);
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/* Prepare list of enabled bands for efficient iteration */
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for (band = 0; mask != 0; mask &= mask - 1, band++)
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eq_data.bands[band] = (uint8_t)find_first_set_bit(mask);
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eq_data.bands[band] = EQ_NUM_BANDS;
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}
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/* Enable or disable the equalizer */
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void dsp_eq_enable(bool enable)
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{
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struct dsp_config *dsp = dsp_get_config(CODEC_IDX_AUDIO);
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bool enabled = dsp_proc_enabled(dsp, DSP_PROC_EQUALIZER);
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if (enable == enabled)
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return;
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dsp_proc_enable(dsp, DSP_PROC_EQUALIZER, enable);
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if (enable && eq_data.enabled != 0)
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dsp_proc_activate(dsp, DSP_PROC_EQUALIZER, true);
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}
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/* Apply EQ filters to those bands that have got it switched on. */
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static void eq_process(struct dsp_proc_entry *this,
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struct dsp_buffer **buf_p)
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{
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struct dsp_buffer *buf = *buf_p;
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int count = buf->remcount;
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unsigned int channels = buf->format.num_channels;
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FOR_EACH_ENB_BAND(b)
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filter_process(&eq_data.filters[*b], buf->p32, count, channels);
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(void)this;
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}
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/* DSP message hook */
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static intptr_t eq_configure(struct dsp_proc_entry *this,
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struct dsp_config *dsp,
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unsigned int setting,
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intptr_t value)
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{
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switch (setting)
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{
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case DSP_PROC_INIT:
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this->process = eq_process;
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/* Wouldn't have been getting frequency updates */
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update_samplerate(dsp_get_output_frequency(dsp));
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/* Fall-through */
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case DSP_PROC_CLOSE:
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pga_enable_gain(PGA_EQ_PRECUT, setting == DSP_PROC_INIT);
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break;
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case DSP_FLUSH:
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eq_flush();
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break;
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case DSP_SET_OUT_FREQUENCY:
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update_samplerate(value);
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break;
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}
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return 0;
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(void)dsp;
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}
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/* Database entry */
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DSP_PROC_DB_ENTRY(EQUALIZER,
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eq_configure);
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