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>
345 lines
11 KiB
C
345 lines
11 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) 2005 Miika Pekkarinen
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* Copyright (C) 2012 Michael Sevakis
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* Copyright (C) 2013 Michael Giacomelli
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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 "platform.h"
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#include "fracmul.h"
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#include "fixedpoint.h"
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#include "dsp_proc_entry.h"
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#include "dsp_misc.h"
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#include <string.h>
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/**
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* Linear interpolation resampling that introduces a one sample delay because
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* of our inability to look into the future at the end of a frame.
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*/
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#if 1 /* Set to '0' to enable debug messages */
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#undef DEBUGF
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#define DEBUGF(...)
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#endif
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#define RESAMPLE_BUF_COUNT 192 /* Per channel, per DSP */
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/* CODEC_IDX_AUDIO = left and right, CODEC_IDX_VOICE = mono */
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static int32_t resample_out_bufs[3][RESAMPLE_BUF_COUNT] IBSS_ATTR;
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/* Data for each resampler on each DSP */
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static struct resample_data
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{
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uint32_t delta; /* 00h: Phase delta for each step in s15.16*/
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uint32_t phase; /* 04h: Current phase [pos16|frac16] */
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int32_t history[2][3]; /* 08h: Last samples for interpolation (L+R)
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0 = oldest, 2 = newest */
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/* 20h */
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unsigned int frequency; /* Virtual input samplerate */
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unsigned int frequency_out; /* Resampler output samplerate */
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struct dsp_buffer resample_buf; /* Buffer descriptor for resampled data */
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int32_t *resample_out_p[2]; /* Actual output buffer pointers */
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} resample_data[DSP_COUNT] IBSS_ATTR;
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/* Actual worker function. Implemented here or in target assembly code. */
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int resample_hermite(struct resample_data *data, struct dsp_buffer *src,
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struct dsp_buffer *dst);
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static void resample_flush_data(struct resample_data *data)
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{
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data->phase = 0;
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memset(&data->history, 0, sizeof (data->history));
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}
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static void resample_flush(struct dsp_proc_entry *this)
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{
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struct resample_data *data = (void *)this->data;
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data->resample_buf.remcount = 0;
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resample_flush_data(data);
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}
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static bool resample_new_delta(struct resample_data *data,
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struct sample_format *format,
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unsigned int fout)
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{
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unsigned int frequency = format->frequency; /* virtual samplerate */
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data->frequency = frequency;
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data->frequency_out = fout;
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data->delta = fp_div(frequency, fout, 16);
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if (frequency == data->frequency_out)
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{
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/* NOTE: If fully glitch-free transistions from no resampling to
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resampling are desired, history should be maintained even when
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not resampling. */
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resample_flush_data(data);
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return false;
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}
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return true;
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}
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#if !defined(CPU_COLDFIRE) && !defined(CPU_ARM)
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int resample_hermite(struct resample_data *data, struct dsp_buffer *src,
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struct dsp_buffer *dst)
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{
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int ch = src->format.num_channels - 1;
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uint32_t count = MIN(src->remcount, 0x8000);
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uint32_t delta = data->delta;
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uint32_t phase, pos;
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int32_t *d;
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do
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{
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const int32_t *s = src->p32[ch];
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d = dst->p32[ch];
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int32_t *dmax = d + dst->bufcount;
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/* Restore state */
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phase = data->phase;
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pos = phase >> 16;
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pos = MIN(pos, count);
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while (pos < count && d < dmax)
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{
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int x0, x1, x2, x3;
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if (pos < 3)
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{
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x3 = data->history[ch][pos+0];
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x2 = pos < 2 ? data->history[ch][pos+1] : s[pos-2];
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x1 = pos < 1 ? data->history[ch][pos+2] : s[pos-1];
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}
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else
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{
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x3 = s[pos-3];
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x2 = s[pos-2];
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x1 = s[pos-1];
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}
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x0 = s[pos];
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int32_t frac = (phase & 0xffff) << 15;
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/* 4-point, 3rd-order Hermite/Catmull-Rom spline (x-form):
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* c1 = -0.5*x3 + 0.5*x1
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* = 0.5*(x1 - x3) <--
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*
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* v = x1 - x2, -v = x2 - x1
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* c2 = x3 - 2.5*x2 + 2*x1 - 0.5*x0
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* = x3 + 2*(x1 - x2) - 0.5*(x0 + x2)
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* = x3 + 2*v - 0.5*(x0 + x2) <--
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*
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* c3 = -0.5*x3 + 1.5*x2 - 1.5*x1 + 0.5*x0
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* = 0.5*x0 - 0.5*x3 + 0.5*(x2 - x1) + (x2 - x1)
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* = 0.5*(x0 - x3 - v) - v <--
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*
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* polynomial coefficients */
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int32_t c1 = (x1 - x3) >> 1;
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int32_t v = x1 - x2;
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int32_t c2 = x3 + 2*v - ((x0 + x2) >> 1);
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int32_t c3 = ((x0 - x3 - v) >> 1) - v;
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/* Evaluate polynomial at time 'frac'; Horner's rule. */
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int32_t acc;
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acc = FRACMUL(c3, frac) + c2;
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acc = FRACMUL(acc, frac) + c1;
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acc = FRACMUL(acc, frac) + x2;
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*d++ = acc;
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phase += delta;
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pos = phase >> 16;
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}
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pos = MIN(pos, count);
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/* Save delay samples for next time. Must do this even if pos was
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* clamped before loop in order to keep record up to date. */
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data->history[ch][0] = pos < 3 ? data->history[ch][pos+0] : s[pos-3];
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data->history[ch][1] = pos < 2 ? data->history[ch][pos+1] : s[pos-2];
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data->history[ch][2] = pos < 1 ? data->history[ch][pos+2] : s[pos-1];
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}
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while (--ch >= 0);
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/* Wrap phase accumulator back to start of next frame. */
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data->phase = phase - (pos << 16);
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dst->remcount = d - dst->p32[0];
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return pos;
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}
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#endif /* CPU */
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/* Resample count stereo samples or stop when the destination is full.
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* Updates the src buffer and changes to its own output buffer to refer to
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* the resampled data. */
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static void resample_process(struct dsp_proc_entry *this,
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struct dsp_buffer **buf_p)
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{
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struct resample_data *data = (void *)this->data;
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struct dsp_buffer *src = *buf_p;
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struct dsp_buffer *dst = &data->resample_buf;
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*buf_p = dst;
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if (dst->remcount > 0)
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return; /* data still remains */
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dst->remcount = 0;
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dst->p32[0] = data->resample_out_p[0];
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dst->p32[1] = data->resample_out_p[1];
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if (src->remcount > 0)
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{
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dst->bufcount = RESAMPLE_BUF_COUNT;
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int consumed = resample_hermite(data, src, dst);
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/* Advance src by consumed amount */
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if (consumed > 0)
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dsp_advance_buffer32(src, consumed);
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}
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/* else purged resample_buf */
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/* Inherit in-place processed mask from source buffer */
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dst->proc_mask = src->proc_mask;
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}
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/* Finish draining old samples then switch format or shut off */
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static intptr_t resample_new_format(struct dsp_proc_entry *this,
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struct dsp_config *dsp,
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struct sample_format *format)
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{
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struct resample_data *data = (void *)this->data;
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struct dsp_buffer *dst = &data->resample_buf;
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if (dst->remcount > 0)
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return PROC_NEW_FORMAT_TRANSITION;
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DSP_PRINT_FORMAT(DSP_PROC_RESAMPLE, *format);
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unsigned int frequency = data->frequency;
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unsigned int fout = dsp_get_output_frequency(dsp);
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bool active = dsp_proc_active(dsp, DSP_PROC_RESAMPLE);
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if ((unsigned int)format->frequency != frequency ||
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data->frequency_out != fout)
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{
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DEBUGF(" DSP_PROC_RESAMPLE- new settings: %u %u\n",
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format->frequency, fout);
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active = resample_new_delta(data, format, fout);
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dsp_proc_activate(dsp, DSP_PROC_RESAMPLE, active);
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}
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/* Everything after us is fout */
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dst->format = *format;
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dst->format.frequency = fout;
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dst->format.codec_frequency = fout;
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if (active)
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return PROC_NEW_FORMAT_OK;
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/* No longer needed */
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DEBUGF(" DSP_PROC_RESAMPLE- deactivated\n");
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return PROC_NEW_FORMAT_DEACTIVATED;
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}
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static void INIT_ATTR resample_dsp_init(struct dsp_config *dsp,
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enum dsp_ids dsp_id)
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{
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int32_t *lbuf, *rbuf;
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switch (dsp_id)
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{
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case CODEC_IDX_AUDIO:
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lbuf = resample_out_bufs[0];
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rbuf = resample_out_bufs[1];
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break;
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case CODEC_IDX_VOICE:
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lbuf = rbuf = resample_out_bufs[2]; /* Always mono */
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break;
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default:
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/* huh? */
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DEBUGF("DSP_PROC_RESAMPLE- unknown DSP %d\n", (int)dsp_id);
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return;
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}
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/* Always enable resampler so that format changes may be monitored and
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* it self-activated when required */
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dsp_proc_enable(dsp, DSP_PROC_RESAMPLE, true);
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resample_data[dsp_id].resample_out_p[0] = lbuf;
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resample_data[dsp_id].resample_out_p[1] = rbuf;
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}
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static void INIT_ATTR resample_proc_init(struct dsp_proc_entry *this,
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struct dsp_config *dsp)
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{
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struct resample_data *data = &resample_data[dsp_get_id(dsp)];
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this->data = (intptr_t)data;
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dsp_proc_set_in_place(dsp, DSP_PROC_RESAMPLE, false);
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data->frequency_out = DSP_OUT_DEFAULT_HZ;
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this->process = resample_process;
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}
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/* DSP message hook */
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static intptr_t resample_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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intptr_t retval = 0;
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switch (setting)
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{
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case DSP_INIT:
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resample_dsp_init(dsp, (enum dsp_ids)value);
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break;
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case DSP_FLUSH:
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resample_flush(this);
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break;
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case DSP_PROC_INIT:
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resample_proc_init(this, dsp);
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break;
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case DSP_PROC_CLOSE:
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/* This stage should be enabled at all times */
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DEBUGF("DSP_PROC_RESAMPLE- Error: Closing!\n");
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break;
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case DSP_PROC_NEW_FORMAT:
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retval = resample_new_format(this, dsp, (struct sample_format *)value);
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break;
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case DSP_SET_OUT_FREQUENCY:
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dsp_proc_want_format_update(dsp, DSP_PROC_RESAMPLE);
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break;
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}
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return retval;
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}
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/* Database entry */
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DSP_PROC_DB_ENTRY(RESAMPLE,
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resample_configure);
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