rockbox/lib/rbcodec/dsp/resample.c
Michael Sevakis fce81a8a74 Rename all the "lin_resample..." stuff to simply "resample_...".
Change-Id: I79f44f0dcc1b23b33a5040795220713660a1d18a
2013-05-07 00:35:46 -04:00

300 lines
8.7 KiB
C

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Miika Pekkarinen
* Copyright (C) 2012 Michael Sevakis
*
* 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.
*
****************************************************************************/
#include "config.h"
#include "system.h"
#include "fracmul.h"
#include "fixedpoint.h"
#include "dsp_proc_entry.h"
#include <string.h>
/**
* Linear interpolation resampling that introduces a one sample delay because
* of our inability to look into the future at the end of a frame.
*/
#if 0 /* Set to '1' to enable debug messages */
#include <debug.h>
#else
#undef DEBUGF
#define DEBUGF(...)
#endif
#define RESAMPLE_BUF_COUNT 192 /* Per channel, per DSP */
/* CODEC_IDX_AUDIO = left and right, CODEC_IDX_VOICE = mono */
static int32_t resample_out_bufs[3][RESAMPLE_BUF_COUNT] IBSS_ATTR;
/* Data for each resampler on each DSP */
static struct resample_data
{
uint32_t delta; /* 00h: Phase delta for each step */
uint32_t phase; /* 04h: Current phase [pos16|frac16] */
int32_t last_sample[2]; /* 08h: Last samples for interpolation (L+R) */
/* 10h */
int32_t frequency; /* Virtual samplerate */
struct dsp_buffer resample_buf; /* Buffer descriptor for resampled data */
int32_t *resample_out_p[2]; /* Actual output buffer pointers */
} resample_data[DSP_COUNT] IBSS_ATTR;
/* Actual worker function. Implemented here or in target assembly code. */
int resample_linear(struct resample_data *data, struct dsp_buffer *src,
struct dsp_buffer *dst);
static void resample_flush_data(struct resample_data *data)
{
data->phase = 0;
data->last_sample[0] = 0;
data->last_sample[1] = 0;
}
static void resample_flush(struct dsp_proc_entry *this)
{
struct resample_data *data = (void *)this->data;
data->resample_buf.remcount = 0;
resample_flush_data(data);
}
static bool resample_new_delta(struct resample_data *data,
struct sample_format *format)
{
int32_t frequency = format->frequency; /* virtual samplerate */
data->frequency = frequency;
data->delta = fp_div(frequency, NATIVE_FREQUENCY, 16);
if (frequency == NATIVE_FREQUENCY)
{
/* NOTE: If fully glitch-free transistions from no resampling to
resampling are desired, last_sample history should be maintained
even when not resampling. */
resample_flush_data(data);
return false;
}
return true;
}
#if !defined(CPU_COLDFIRE) && !defined(CPU_ARM)
/* Where the real work is done */
int resample_linear(struct resample_data *data, struct dsp_buffer *src,
struct dsp_buffer *dst)
{
int ch = src->format.num_channels - 1;
uint32_t count = MIN(src->remcount, 0x8000);
uint32_t delta = data->delta;
uint32_t phase, pos;
int32_t *d;
do
{
const int32_t *s = src->p32[ch];
d = dst->p32[ch];
int32_t *dmax = d + dst->bufcount;
phase = data->phase;
pos = phase >> 16;
pos = MIN(pos, count);
int32_t last = pos > 0 ? s[pos - 1] : data->last_sample[ch];
if (pos < count)
{
while (1)
{
*d++ = last + FRACMUL((phase & 0xffff) << 15, s[pos] - last);
phase += delta;
pos = phase >> 16;
if (pos >= count || d >= dmax)
break;
if (pos > 0)
last = s[pos - 1];
}
if (pos > 0)
{
pos = MIN(pos, count);
last = s[pos - 1];
}
}
data->last_sample[ch] = last;
}
while (--ch >= 0);
/* Wrap phase accumulator back to start of next frame. */
data->phase = phase - (pos << 16);
dst->remcount = d - dst->p32[0];
return pos;
}
#endif /* CPU */
/* Resample count stereo samples or stop when the destination is full.
* Updates the src buffer and changes to its own output buffer to refer to
* the resampled data. */
static void resample_process(struct dsp_proc_entry *this,
struct dsp_buffer **buf_p)
{
struct resample_data *data = (void *)this->data;
struct dsp_buffer *src = *buf_p;
struct dsp_buffer *dst = &data->resample_buf;
*buf_p = dst;
if (dst->remcount > 0)
return; /* data still remains */
dst->remcount = 0;
dst->p32[0] = data->resample_out_p[0];
dst->p32[1] = data->resample_out_p[1];
if (src->remcount > 0)
{
dst->bufcount = RESAMPLE_BUF_COUNT;
int consumed = resample_linear(data, src, dst);
/* Advance src by consumed amount */
if (consumed > 0)
dsp_advance_buffer32(src, consumed);
}
/* else purged resample_buf */
/* Inherit in-place processed mask from source buffer */
dst->proc_mask = src->proc_mask;
}
/* Finish draining old samples then switch format or shut off */
static intptr_t resample_new_format(struct dsp_proc_entry *this,
struct dsp_config *dsp,
struct sample_format *format)
{
struct resample_data *data = (void *)this->data;
struct dsp_buffer *dst = &data->resample_buf;
if (dst->remcount > 0)
return PROC_NEW_FORMAT_TRANSITION;
DSP_PRINT_FORMAT(DSP_PROC_RESAMPLE, *format);
int32_t frequency = data->frequency;
bool active = dsp_proc_active(dsp, DSP_PROC_RESAMPLE);
if (format->frequency != frequency)
{
DEBUGF(" DSP_PROC_RESAMPLE- new delta\n");
active = resample_new_delta(data, format);
dsp_proc_activate(dsp, DSP_PROC_RESAMPLE, active);
}
/* Everything after us is NATIVE_FREQUENCY */
dst->format = *format;
dst->format.frequency = NATIVE_FREQUENCY;
dst->format.codec_frequency = NATIVE_FREQUENCY;
if (active)
return PROC_NEW_FORMAT_OK;
/* No longer needed */
DEBUGF(" DSP_PROC_RESAMPLE- deactivated\n");
return PROC_NEW_FORMAT_DEACTIVATED;
}
static void INIT_ATTR resample_dsp_init(struct dsp_config *dsp,
enum dsp_ids dsp_id)
{
int32_t *lbuf, *rbuf;
switch (dsp_id)
{
case CODEC_IDX_AUDIO:
lbuf = resample_out_bufs[0];
rbuf = resample_out_bufs[1];
break;
case CODEC_IDX_VOICE:
lbuf = rbuf = resample_out_bufs[2]; /* Always mono */
break;
default:
/* huh? */
DEBUGF("DSP_PROC_RESAMPLE- unknown DSP %d\n", (int)dsp_id);
return;
}
/* Always enable resampler so that format changes may be monitored and
* it self-activated when required */
dsp_proc_enable(dsp, DSP_PROC_RESAMPLE, true);
resample_data[dsp_id].resample_out_p[0] = lbuf;
resample_data[dsp_id].resample_out_p[1] = rbuf;
}
static void INIT_ATTR resample_proc_init(struct dsp_proc_entry *this,
struct dsp_config *dsp)
{
dsp_proc_set_in_place(dsp, DSP_PROC_RESAMPLE, false);
this->data = (intptr_t)&resample_data[dsp_get_id(dsp)];
this->process = resample_process;
}
/* DSP message hook */
static intptr_t resample_configure(struct dsp_proc_entry *this,
struct dsp_config *dsp,
unsigned int setting,
intptr_t value)
{
intptr_t retval = 0;
switch (setting)
{
case DSP_INIT:
resample_dsp_init(dsp, (enum dsp_ids)value);
break;
case DSP_FLUSH:
resample_flush(this);
break;
case DSP_PROC_INIT:
resample_proc_init(this, dsp);
break;
case DSP_PROC_CLOSE:
/* This stage should be enabled at all times */
DEBUGF("DSP_PROC_RESAMPLE- Error: Closing!\n");
break;
case DSP_PROC_NEW_FORMAT:
retval = resample_new_format(this, dsp, (struct sample_format *)value);
break;
}
return retval;
}
/* Database entry */
DSP_PROC_DB_ENTRY(RESAMPLE,
resample_configure);