/*
* 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 "PcmConvert.hxx"
#include "PcmChannels.hxx"
#include "PcmFormat.hxx"
#include "pcm_pack.h"
#include "audio_format.h"
#include <glib.h>
#include <assert.h>
#include <math.h>
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "pcm"
PcmConvert::PcmConvert()
{
}
PcmConvert::~PcmConvert()
{
}
void
PcmConvert::Reset()
{
dsd.Reset();
resampler.Reset();
}
inline const int16_t *
PcmConvert::Convert16(const audio_format *src_format,
const void *src_buffer, size_t src_size,
const 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(format_buffer, dither,
sample_format(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(sample_format(src_format->format)));
return NULL;
}
if (src_format->channels != dest_format->channels) {
buf = pcm_convert_channels_16(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 = resampler.Resample16(dest_format->channels,
src_format->sample_rate, buf, len,
dest_format->sample_rate, &len,
error_r);
if (buf == NULL)
return NULL;
}
*dest_size_r = len;
return buf;
}
inline const int32_t *
PcmConvert::Convert24(const audio_format *src_format,
const void *src_buffer, size_t src_size,
const 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(format_buffer,
sample_format(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(sample_format(src_format->format)));
return NULL;
}
if (src_format->channels != dest_format->channels) {
buf = pcm_convert_channels_24(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 = resampler.Resample24(dest_format->channels,
src_format->sample_rate, buf, len,
dest_format->sample_rate, &len,
error_r);
if (buf == NULL)
return NULL;
}
*dest_size_r = len;
return buf;
}
inline const int32_t *
PcmConvert::Convert32(const audio_format *src_format,
const void *src_buffer, size_t src_size,
const 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(format_buffer,
sample_format(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(sample_format(src_format->format)));
return NULL;
}
if (src_format->channels != dest_format->channels) {
buf = pcm_convert_channels_32(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 = resampler.Resample32(dest_format->channels,
src_format->sample_rate, buf, len,
dest_format->sample_rate, &len,
error_r);
if (buf == NULL)
return buf;
}
*dest_size_r = len;
return buf;
}
inline const float *
PcmConvert::ConvertFloat(const audio_format *src_format,
const void *src_buffer, size_t src_size,
const audio_format *dest_format, size_t *dest_size_r,
GError **error_r)
{
const float *buffer = (const float *)src_buffer;
size_t size = src_size;
assert(dest_format->format == SAMPLE_FORMAT_FLOAT);
/* convert to float now */
buffer = pcm_convert_to_float(format_buffer,
sample_format(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(sample_format(src_format->format)));
return NULL;
}
/* convert channels */
if (src_format->channels != dest_format->channels) {
buffer = pcm_convert_channels_float(channels_buffer,
dest_format->channels,
src_format->channels,
buffer, size, &size);
if (buffer == 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;
}
}
/* resample with float, because this is the best format for
libsamplerate */
if (src_format->sample_rate != dest_format->sample_rate) {
buffer = resampler.ResampleFloat(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 *
PcmConvert::Convert(const audio_format *src_format,
const void *src, size_t src_size,
const audio_format *dest_format,
size_t *dest_size_r,
GError **error_r)
{
struct audio_format float_format;
if (src_format->format == SAMPLE_FORMAT_DSD) {
size_t f_size;
const float *f = dsd.ToFloat(src_format->channels,
false, (const uint8_t *)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 (sample_format(dest_format->format)) {
case SAMPLE_FORMAT_S16:
return Convert16(src_format, src, src_size,
dest_format, dest_size_r,
error_r);
case SAMPLE_FORMAT_S24_P32:
return Convert24(src_format, src, src_size,
dest_format, dest_size_r,
error_r);
case SAMPLE_FORMAT_S32:
return Convert32(src_format, src, src_size,
dest_format, dest_size_r,
error_r);
case SAMPLE_FORMAT_FLOAT:
return ConvertFloat(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(sample_format(dest_format->format)));
return NULL;
}
}