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[/] [or1k/] [trunk/] [mp3/] [sw/] [mad-xess/] [audio.c] - Rev 1773
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/* * mad - MPEG audio decoder * Copyright (C) 2000-2001 Robert Leslie * * 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 * * $Id: audio.c,v 1.3 2001-11-06 17:01:07 simons Exp $ */ # ifdef HAVE_CONFIG_H # include "config.h" # endif # include "audio.h" # include "mad.h" #ifndef EMBED # include <string.h> # include <stdio.h> extern FILE *fo; int printf(char *fmt, ...); #endif char const *audio_error; static mad_fixed_t left_err, right_err; #define AUDIO_DBG 0 #if AUDIO_DBG //unsigned short wave_dump[(0x40000/2)]; //int wave_seg_index[0x10000]; int wave_seg_nb = 0; int wave_index = 0; #endif /* * NAME: audio_linear_dither() * DESCRIPTION: generic linear sample quantize and dither routine */ inline signed long audio_linear_dither(unsigned int bits, mad_fixed_t sample, mad_fixed_t *error, struct audio_stats *stats) { mad_fixed_t quantized; /* dither */ sample += *error; # if 1 /* clip */ quantized = sample; if (sample >= stats->peak_sample) { if (sample >= MAD_F_ONE) { quantized = MAD_F_ONE - 1; ++stats->clipped_samples; if (sample - quantized > stats->peak_clipping && mad_f_abs(*error) < (MAD_F_ONE >> (MAD_F_FRACBITS + 1 - bits))) stats->peak_clipping = sample - quantized; } stats->peak_sample = quantized; } else if (sample < -stats->peak_sample) { if (sample < -MAD_F_ONE) { quantized = -MAD_F_ONE; ++stats->clipped_samples; if (quantized - sample > stats->peak_clipping && mad_f_abs(*error) < (MAD_F_ONE >> (MAD_F_FRACBITS + 1 - bits))) stats->peak_clipping = quantized - sample; } stats->peak_sample = -quantized; } # else /* clip */ quantized = sample; if (sample >= MAD_F_ONE) quantized = MAD_F_ONE - 1; else if (sample < -MAD_F_ONE) quantized = -MAD_F_ONE; # endif /* quantize */ quantized &= ~((1L << (MAD_F_FRACBITS + 1 - bits)) - 1); /* error */ *error = sample - quantized; /* scale */ return quantized >> (MAD_F_FRACBITS + 1 - bits); } static struct audio_stats stats; /* * NAME: audio_pcm_s16le() * DESCRIPTION: write a block of signed 16-bit little-endian PCM samples */ unsigned int audio_pcm_s16le(unsigned char *data, unsigned int nsamples, mad_fixed_t const *left, mad_fixed_t const *right) { unsigned int len; register signed int sample0, sample1; len = nsamples; #if AUDIO_DBG // wave_seg_index[wave_seg_nb] = wave_index; // wave_seg_nb++; report(wave_index); #endif while (len--) { sample0 = audio_linear_dither(16, *left++, &left_err, &stats); sample1 = audio_linear_dither(16, *right++, &right_err, &stats); #ifdef EMBED #ifdef OR1K #ifndef OR1K_SIM *(volatile unsigned short *)0x40000000 = sample0; // *(volatile unsigned long *)0x40000000 = sample0; // *(volatile unsigned long *)0x40000000 = sample1; // *(volatile unsigned long *)0x40000000 = sample1; #if AUDIO_DBG // wave_dump[wave_index] = (unsigned short)sample0; wave_index++; // wave_dump[wave_index] = (unsigned short)sample1; wave_index++; #endif #else // asm volatile("l.mtspr r0,%0,0x01234" : : "r" (sample0 | (sample1 << 16))); asm volatile("l.mtspr r0,%0,0x0FFFE" : : "r" (sample0 >> 0)); asm volatile("l.mtspr r0,%0,0x0FFFE" : : "r" (sample0 >> 8)); asm volatile("l.mtspr r0,%0,0x0FFFE" : : "r" (sample1 >> 0)); asm volatile("l.mtspr r0,%0,0x0FFFE" : : "r" (sample1 >> 8)); #endif #else printf("l.mtspr (0x0000FFFE) <- %x\n", (sample0 >> 0)); printf("l.mtspr (0x0000FFFE) <- %x\n", (sample0 >> 8)); printf("l.mtspr (0x0000FFFE) <- %x\n", (sample1 >> 0)); printf("l.mtspr (0x0000FFFE) <- %x\n", (sample1 >> 8)); #endif #else data[0] = sample0 >> 0; data[1] = sample0 >> 8; data[2] = sample1 >> 0; data[3] = sample1 >> 8; if (fo) { fputc (data[0], fo); fputc (data[1], fo); fputc (data[2], fo); fputc (data[3], fo); } data += 4; #endif } return nsamples * 2 * 2; }
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