mpd/src/pcm_convert.c
Max Kellermann 5fe7e3bc14 pcm_format: use the pcm_buffer library
Replace a "static" buffer with the PCM buffer library.
2009-01-07 23:56:34 +01:00

167 lines
4.6 KiB
C

/*
* Copyright (C) 2003-2009 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "pcm_convert.h"
#include "pcm_channels.h"
#include "pcm_format.h"
#include "audio_format.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_resample_init(&state->resample);
pcm_dither_24_init(&state->dither);
pcm_buffer_init(&state->format_buffer);
}
void pcm_convert_deinit(struct pcm_convert_state *state)
{
pcm_resample_deinit(&state->resample);
pcm_buffer_deinit(&state->format_buffer);
}
static size_t
pcm_convert_16(const struct audio_format *src_format,
const void *src_buffer, size_t src_size,
const struct audio_format *dest_format,
int16_t *dest_buffer,
struct pcm_convert_state *state)
{
const int16_t *buf;
size_t len;
size_t dest_size = pcm_convert_size(src_format, src_size, dest_format);
assert(dest_format->bits == 16);
buf = pcm_convert_to_16(&state->format_buffer, &state->dither,
src_format->bits, src_buffer, src_size,
&len);
if (!buf)
g_error("pcm_convert_to_16() failed");
if (src_format->channels != dest_format->channels) {
buf = pcm_convert_channels_16(dest_format->channels,
src_format->channels,
buf, len, &len);
if (!buf)
g_error("pcm_convert_channels_16() failed");
}
if (src_format->sample_rate == dest_format->sample_rate) {
assert(dest_size >= len);
memcpy(dest_buffer, buf, len);
} else {
len = pcm_resample_16(dest_format->channels,
src_format->sample_rate, buf, len,
dest_format->sample_rate,
dest_buffer, dest_size,
&state->resample);
}
return len;
}
static size_t
pcm_convert_24(const struct audio_format *src_format,
const void *src_buffer, size_t src_size,
const struct audio_format *dest_format,
int32_t *dest_buffer,
struct pcm_convert_state *state)
{
const int32_t *buf;
size_t len;
size_t dest_size = pcm_convert_size(src_format, src_size, dest_format);
assert(dest_format->bits == 24);
buf = pcm_convert_to_24(&state->format_buffer, src_format->bits,
src_buffer, src_size, &len);
if (!buf)
g_error("pcm_convert_to_24() failed");
if (src_format->channels != dest_format->channels) {
buf = pcm_convert_channels_24(dest_format->channels,
src_format->channels,
buf, len, &len);
if (!buf)
g_error("pcm_convert_channels_24() failed");
}
if (src_format->sample_rate == dest_format->sample_rate) {
assert(dest_size >= len);
memcpy(dest_buffer, buf, len);
} else {
len = pcm_resample_24(dest_format->channels,
src_format->sample_rate, buf, len,
dest_format->sample_rate,
(int32_t*)dest_buffer, dest_size,
&state->resample);
}
return len;
}
size_t pcm_convert(const struct audio_format *inFormat,
const void *src, size_t src_size,
const struct audio_format *outFormat,
void *dest,
struct pcm_convert_state *convState)
{
switch (outFormat->bits) {
case 16:
return pcm_convert_16(inFormat, src, src_size,
outFormat, (int16_t*)dest,
convState);
case 24:
return pcm_convert_24(inFormat, src, src_size,
outFormat, (int32_t*)dest,
convState);
default:
g_error("cannot convert to %u bit\n", outFormat->bits);
}
}
size_t pcm_convert_size(const struct audio_format *inFormat, size_t src_size,
const struct audio_format *outFormat)
{
const double ratio = (double)outFormat->sample_rate /
(double)inFormat->sample_rate;
size_t dest_size = src_size;
/* no partial frames allowed */
assert((src_size % audio_format_frame_size(inFormat)) == 0);
dest_size /= audio_format_frame_size(inFormat);
dest_size = ceil((double)dest_size * ratio);
dest_size *= audio_format_frame_size(outFormat);
return dest_size;
}