/*
* Copyright (C) 2003-2011 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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 program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "pcm_convert.h"
#include "pcm_channels.h"
#include "pcm_format.h"
#include "pcm_byteswap.h"
#include "pcm_pack.h"
#include "audio_format.h"
#include "glib_compat.h"
#include <assert.h>
#include <string.h>
#include <math.h>
#include <glib.h>
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "pcm"
void pcm_convert_init(struct pcm_convert_state *state)
{
memset(state, 0, sizeof(*state));
pcm_dsd_init(&state->dsd);
pcm_resample_init(&state->resample);
pcm_dither_24_init(&state->dither);
pcm_buffer_init(&state->format_buffer);
pcm_buffer_init(&state->pack_buffer);
pcm_buffer_init(&state->channels_buffer);
pcm_buffer_init(&state->byteswap_buffer);
}
void pcm_convert_deinit(struct pcm_convert_state *state)
{
pcm_dsd_deinit(&state->dsd);
pcm_resample_deinit(&state->resample);
pcm_buffer_deinit(&state->format_buffer);
pcm_buffer_deinit(&state->pack_buffer);
pcm_buffer_deinit(&state->channels_buffer);
pcm_buffer_deinit(&state->byteswap_buffer);
}
void
pcm_convert_reset(struct pcm_convert_state *state)
{
pcm_dsd_reset(&state->dsd);
pcm_resample_reset(&state->resample);
}
static const void *
pcm_convert_channels(struct pcm_buffer *buffer, enum sample_format format,
uint8_t dest_channels,
uint8_t src_channels, const void *src,
size_t src_size, size_t *dest_size_r,
GError **error_r)
{
const void *dest = NULL;
switch (format) {
case SAMPLE_FORMAT_UNDEFINED:
case SAMPLE_FORMAT_S8:
case SAMPLE_FORMAT_S24:
case SAMPLE_FORMAT_FLOAT:
case SAMPLE_FORMAT_DSD:
case SAMPLE_FORMAT_DSD_LSBFIRST:
g_set_error(error_r, pcm_convert_quark(), 0,
"Channel conversion not implemented for format '%s'",
sample_format_to_string(format));
return NULL;
case SAMPLE_FORMAT_S16:
dest = pcm_convert_channels_16(buffer, dest_channels,
src_channels, src,
src_size, dest_size_r);
break;
case SAMPLE_FORMAT_S24_P32:
dest = pcm_convert_channels_24(buffer, dest_channels,
src_channels, src,
src_size, dest_size_r);
break;
case SAMPLE_FORMAT_S32:
dest = pcm_convert_channels_32(buffer, dest_channels,
src_channels, src,
src_size, dest_size_r);
break;
}
if (dest == NULL) {
g_set_error(error_r, pcm_convert_quark(), 0,
"Conversion from %u to %u channels "
"is not implemented",
src_channels, dest_channels);
return NULL;
}
return dest;
}
static const int16_t *
pcm_convert_16(struct pcm_convert_state *state,
const struct audio_format *src_format,
const void *src_buffer, size_t src_size,
const struct audio_format *dest_format, size_t *dest_size_r,
GError **error_r)
{
const int16_t *buf;
size_t len;
assert(dest_format->format == SAMPLE_FORMAT_S16);
buf = pcm_convert_to_16(&state->format_buffer, &state->dither,
src_format->format, src_buffer, src_size,
&len);
if (buf == NULL) {
g_set_error(error_r, pcm_convert_quark(), 0,
"Conversion from %s to 16 bit is not implemented",
sample_format_to_string(src_format->format));
return NULL;
}
if (src_format->channels != dest_format->channels) {
buf = pcm_convert_channels_16(&state->channels_buffer,
dest_format->channels,
src_format->channels,
buf, len, &len);
if (buf == NULL) {
g_set_error(error_r, pcm_convert_quark(), 0,
"Conversion from %u to %u channels "
"is not implemented",
src_format->channels,
dest_format->channels);
return NULL;
}
}
if (src_format->sample_rate != dest_format->sample_rate) {
buf = pcm_resample_16(&state->resample,
dest_format->channels,
src_format->sample_rate, buf, len,
dest_format->sample_rate, &len,
error_r);
if (buf == NULL)
return NULL;
}
if (dest_format->reverse_endian) {
buf = pcm_byteswap_16(&state->byteswap_buffer, buf, len);
assert(buf != NULL);
}
*dest_size_r = len;
return buf;
}
static const int32_t *
pcm_convert_24(struct pcm_convert_state *state,
const struct audio_format *src_format,
const void *src_buffer, size_t src_size,
const struct audio_format *dest_format, size_t *dest_size_r,
GError **error_r)
{
const int32_t *buf;
size_t len;
assert(dest_format->format == SAMPLE_FORMAT_S24_P32);
buf = pcm_convert_to_24(&state->format_buffer, src_format->format,
src_buffer, src_size, &len);
if (buf == NULL) {
g_set_error(error_r, pcm_convert_quark(), 0,
"Conversion from %s to 24 bit is not implemented",
sample_format_to_string(src_format->format));
return NULL;
}
if (src_format->channels != dest_format->channels) {
buf = pcm_convert_channels_24(&state->channels_buffer,
dest_format->channels,
src_format->channels,
buf, len, &len);
if (buf == NULL) {
g_set_error(error_r, pcm_convert_quark(), 0,
"Conversion from %u to %u channels "
"is not implemented",
src_format->channels,
dest_format->channels);
return NULL;
}
}
if (src_format->sample_rate != dest_format->sample_rate) {
buf = pcm_resample_24(&state->resample,
dest_format->channels,
src_format->sample_rate, buf, len,
dest_format->sample_rate, &len,
error_r);
if (buf == NULL)
return NULL;
}
if (dest_format->reverse_endian) {
buf = pcm_byteswap_32(&state->byteswap_buffer, buf, len);
assert(buf != NULL);
}
*dest_size_r = len;
return buf;
}
/**
* Convert to 24 bit packed samples (aka S24_3LE / S24_3BE).
*/
static const void *
pcm_convert_24_packed(struct pcm_convert_state *state,
const struct audio_format *src_format,
const void *src_buffer, size_t src_size,
const struct audio_format *dest_format,
size_t *dest_size_r,
GError **error_r)
{
assert(dest_format->format == SAMPLE_FORMAT_S24);
/* use the normal 24 bit conversion first */
struct audio_format audio_format;
audio_format_init(&audio_format, dest_format->sample_rate,
SAMPLE_FORMAT_S24_P32, dest_format->channels);
const int32_t *buffer;
size_t buffer_size;
buffer = pcm_convert_24(state, src_format, src_buffer, src_size,
&audio_format, &buffer_size, error_r);
if (buffer == NULL)
return NULL;
/* now convert to packed 24 bit */
unsigned num_samples = buffer_size / 4;
size_t dest_size = num_samples * 3;
uint8_t *dest = pcm_buffer_get(&state->pack_buffer, dest_size);
pcm_pack_24(dest, buffer, buffer + num_samples,
dest_format->reverse_endian);
*dest_size_r = dest_size;
return dest;
}
static const int32_t *
pcm_convert_32(struct pcm_convert_state *state,
const struct audio_format *src_format,
const void *src_buffer, size_t src_size,
const struct audio_format *dest_format, size_t *dest_size_r,
GError **error_r)
{
const int32_t *buf;
size_t len;
assert(dest_format->format == SAMPLE_FORMAT_S32);
buf = pcm_convert_to_32(&state->format_buffer, src_format->format,
src_buffer, src_size, &len);
if (buf == NULL) {
g_set_error(error_r, pcm_convert_quark(), 0,
"Conversion from %s to 32 bit is not implemented",
sample_format_to_string(src_format->format));
return NULL;
}
if (src_format->channels != dest_format->channels) {
buf = pcm_convert_channels_32(&state->channels_buffer,
dest_format->channels,
src_format->channels,
buf, len, &len);
if (buf == NULL) {
g_set_error(error_r, pcm_convert_quark(), 0,
"Conversion from %u to %u channels "
"is not implemented",
src_format->channels,
dest_format->channels);
return NULL;
}
}
if (src_format->sample_rate != dest_format->sample_rate) {
buf = pcm_resample_32(&state->resample,
dest_format->channels,
src_format->sample_rate, buf, len,
dest_format->sample_rate, &len,
error_r);
if (buf == NULL)
return buf;
}
if (dest_format->reverse_endian) {
buf = pcm_byteswap_32(&state->byteswap_buffer, buf, len);
assert(buf != NULL);
}
*dest_size_r = len;
return buf;
}
static const float *
pcm_convert_float(struct pcm_convert_state *state,
const struct audio_format *src_format,
const void *src_buffer, size_t src_size,
const struct audio_format *dest_format, size_t *dest_size_r,
GError **error_r)
{
const float *buffer = src_buffer;
size_t size = src_size;
assert(dest_format->format == SAMPLE_FORMAT_FLOAT);
if (src_format->reverse_endian || dest_format->reverse_endian) {
g_set_error_literal(error_r, pcm_convert_quark(), 0,
"Reverse endian not supported");
return NULL;
}
/* convert channels first, hoping the source format is
supported (float is not) */
if (dest_format->channels != src_format->channels) {
buffer = pcm_convert_channels(&state->channels_buffer,
src_format->format,
dest_format->channels,
src_format->channels,
buffer, size, &size, error_r);
if (buffer == NULL)
return NULL;
}
/* convert to float now */
buffer = pcm_convert_to_float(&state->format_buffer,
src_format->format,
buffer, size, &size);
if (buffer == NULL) {
g_set_error(error_r, pcm_convert_quark(), 0,
"Conversion from %s to float is not implemented",
sample_format_to_string(src_format->format));
return NULL;
}
/* resample with float, because this is the best format for
libsamplerate */
if (src_format->sample_rate != dest_format->sample_rate) {
buffer = pcm_resample_float(&state->resample,
dest_format->channels,
src_format->sample_rate,
buffer, size,
dest_format->sample_rate, &size,
error_r);
if (buffer == NULL)
return NULL;
}
*dest_size_r = size;
return buffer;
}
const void *
pcm_convert(struct pcm_convert_state *state,
const struct audio_format *src_format,
const void *src, size_t src_size,
const struct audio_format *dest_format,
size_t *dest_size_r,
GError **error_r)
{
if (src_format->reverse_endian) {
/* convert to host byte order, because all of our
conversion libraries assume host byte order */
src = pcm_byteswap(&state->byteswap_buffer, src_format->format,
src, src_size);
if (src == NULL) {
g_set_error(error_r, pcm_convert_quark(), 0,
"PCM byte order change of format '%s' is not implemented",
sample_format_to_string(src_format->format));
return NULL;
}
}
struct audio_format float_format;
if (src_format->format == SAMPLE_FORMAT_DSD ||
src_format->format == SAMPLE_FORMAT_DSD_LSBFIRST) {
size_t f_size;
const bool lsbfirst =
src_format->format == SAMPLE_FORMAT_DSD_LSBFIRST;
const float *f = pcm_dsd_to_float(&state->dsd,
src_format->channels,
lsbfirst, src, src_size,
&f_size);
if (f == NULL) {
g_set_error_literal(error_r, pcm_convert_quark(), 0,
"DSD to PCM conversion failed");
return NULL;
}
float_format = *src_format;
float_format.format = SAMPLE_FORMAT_FLOAT;
src_format = &float_format;
src = f;
src_size = f_size;
}
switch (dest_format->format) {
case SAMPLE_FORMAT_S16:
return pcm_convert_16(state,
src_format, src, src_size,
dest_format, dest_size_r,
error_r);
case SAMPLE_FORMAT_S24:
return pcm_convert_24_packed(state,
src_format, src, src_size,
dest_format, dest_size_r,
error_r);
case SAMPLE_FORMAT_S24_P32:
return pcm_convert_24(state,
src_format, src, src_size,
dest_format, dest_size_r,
error_r);
case SAMPLE_FORMAT_S32:
return pcm_convert_32(state,
src_format, src, src_size,
dest_format, dest_size_r,
error_r);
case SAMPLE_FORMAT_FLOAT:
return pcm_convert_float(state,
src_format, src, src_size,
dest_format, dest_size_r,
error_r);
default:
g_set_error(error_r, pcm_convert_quark(), 0,
"PCM conversion to %s is not implemented",
sample_format_to_string(dest_format->format));
return NULL;
}
}