/* the Music Player Daemon (MPD) * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com) * This project's homepage is: 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_utils.h" #include "pcm_channels.h" #include "conf.h" #include "audio_format.h" #include #include #include #include #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); } static void pcm_convert_8_to_16(int16_t *out, const int8_t *in, unsigned num_samples) { while (num_samples > 0) { *out++ = *in++ << 8; --num_samples; } } static void pcm_convert_24_to_16(struct pcm_dither_24 *dither, int16_t *out, const int32_t *in, unsigned num_samples) { pcm_dither_24_to_16(dither, out, in, num_samples); } static const int16_t * pcm_convert_to_16(struct pcm_convert_state *convert, uint8_t bits, const void *src, size_t src_size, size_t *dest_size_r) { static int16_t *buf; static size_t len; unsigned num_samples; switch (bits) { case 8: num_samples = src_size; *dest_size_r = src_size << 1; if (*dest_size_r > len) { len = *dest_size_r; buf = g_realloc(buf, len); } pcm_convert_8_to_16((int16_t *)buf, (const int8_t *)src, num_samples); return buf; case 16: *dest_size_r = src_size; return src; case 24: num_samples = src_size / 4; *dest_size_r = num_samples * 2; if (*dest_size_r > len) { len = *dest_size_r; buf = g_realloc(buf, len); } pcm_convert_24_to_16(&convert->dither, (int16_t *)buf, (const int32_t *)src, num_samples); return buf; } g_warning("only 8 or 16 bits are supported for conversion!\n"); return NULL; } static void pcm_convert_8_to_24(int32_t *out, const int8_t *in, unsigned num_samples) { while (num_samples > 0) { *out++ = *in++ << 16; --num_samples; } } static void pcm_convert_16_to_24(int32_t *out, const int16_t *in, unsigned num_samples) { while (num_samples > 0) { *out++ = *in++ << 8; --num_samples; } } static const int32_t * pcm_convert_to_24(uint8_t bits, const void *src, size_t src_size, size_t *dest_size_r) { static int32_t *buf; static size_t len; unsigned num_samples; switch (bits) { case 8: num_samples = src_size; *dest_size_r = src_size * 4; if (*dest_size_r > len) { len = *dest_size_r; buf = g_realloc(buf, len); } pcm_convert_8_to_24(buf, (const int8_t *)src, num_samples); return buf; case 16: num_samples = src_size / 2; *dest_size_r = num_samples * 4; if (*dest_size_r > len) { len = *dest_size_r; buf = g_realloc(buf, len); } pcm_convert_16_to_24(buf, (const int16_t *)src, num_samples); return buf; case 24: *dest_size_r = src_size; return src; } g_warning("only 8 or 24 bits are supported for conversion!\n"); return NULL; } 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, 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(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; } /* outFormat bits must be 16 and channels must be 1 or 2! */ size_t pcm_convert(const struct audio_format *inFormat, const char *src, size_t src_size, const struct audio_format *outFormat, char *outBuffer, struct pcm_convert_state *convState) { switch (outFormat->bits) { case 16: return pcm_convert_16(inFormat, src, src_size, outFormat, (int16_t*)outBuffer, convState); case 24: return pcm_convert_24(inFormat, src, src_size, outFormat, (int32_t*)outBuffer, 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; }