Copyright (c) 2001, 2002 Matthias Kramm <kramm@quiss.org>
- This file is distributed under the GPL, see file COPYING for details
+ 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 */
#ifndef RFXSWF_DISABLESOUND
#include "../lame/lame.h"
/* TODO: find a way to set these from the outside */
-int swf_mp3_samplerate = 44100;
+int swf_mp3_in_samplerate = 44100;
+int swf_mp3_out_samplerate = 11025;
int swf_mp3_channels = 1;
int swf_mp3_bitrate = 32;
static lame_global_flags*lame_flags;
-static void initlame(unsigned char*buf, int bufsize)
+static void initlame()
{
+ unsigned char buf[4096];
+ int bufsize = 1152*2;
+
lame_flags = lame_init();
- lame_set_in_samplerate(lame_flags, swf_mp3_samplerate);
+ lame_set_in_samplerate(lame_flags, swf_mp3_in_samplerate);
lame_set_num_channels(lame_flags, swf_mp3_channels);
lame_set_scale(lame_flags, 0);
// MPEG1 32, 44.1, 48khz
// MPEG2 16, 22.05, 24
// MPEG2.5 8, 11.025, 12
- // (not used by decoding routines)
- lame_set_out_samplerate(lame_flags, 11025);
+ lame_set_out_samplerate(lame_flags, swf_mp3_out_samplerate);
lame_set_quality(lame_flags, 0);
lame_set_mode(lame_flags, MONO/*3*/);
void swf_SetSoundStreamHead(TAG*tag, int avgnumsamples)
{
- unsigned char buf[4096];
- int bufsize = 1152*2;
int len;
- short int samples[1152*2];
U8 playbackrate = 1; // 0 = 5.5 Khz, 1 = 11 Khz, 2 = 22 Khz, 3 = 44 Khz
U8 playbacksize = 1; // 0 = 8 bit, 1 = 16 bit
U8 rate = 1; // 0 = 5.5 Khz, 1 = 11 Khz, 2 = 22 Khz, 3 = 44 Khz
U8 size = 1; // 0 = 8 bit, 1 = 16 bit
U8 type = 0; // 0 = mono, 1 = stereo
-
- memset(samples,0,sizeof(samples));
- initlame(buf, bufsize);
+ if(swf_mp3_out_samplerate == 5512) playbackrate = rate = 0; // lame doesn't support this
+ else if(swf_mp3_out_samplerate == 11025) playbackrate = rate = 1;
+ else if(swf_mp3_out_samplerate == 22050) playbackrate = rate = 2;
+ else if(swf_mp3_out_samplerate == 44100) playbackrate = rate = 3;
+ else fprintf(stderr, "Invalid samplerate: %d\n", swf_mp3_out_samplerate);
+
+ initlame();
swf_SetU8(tag,(playbackrate<<2)|(playbacksize<<1)|playbacktype);
swf_SetU8(tag,(compression<<4)|(rate<<2)|(size<<1)|type);
void swf_SetSoundStreamBlock(TAG*tag, S16*samples, int seek, char first)
{
char*buf;
- int oldlen=0,len = 0;
+ int len = 0;
int bufsize = 16384;
- int numsamples = 1152*2;
+ int numsamples = (int)(((swf_mp3_out_samplerate > 22050) ? 1152 : 576) * ((double)swf_mp3_in_samplerate/swf_mp3_out_samplerate));
+ int fs = 0;
buf = malloc(bufsize);
if(!buf)
return;
if(first) {
- int fs = lame_get_framesize(lame_flags);
- //printf("framesize:%d\n", fs);
+ fs = lame_get_framesize(lame_flags);
swf_SetU16(tag, fs * first); // samples per mp3 frame
swf_SetU16(tag, seek); // seek
}
len += lame_encode_buffer(lame_flags, samples, samples, numsamples, &buf[len], bufsize-len);
- //printf("block: %d (+%d)\n", len, len-oldlen);
- oldlen = len;
-
len += lame_encode_flush_nogap(lame_flags, &buf[len], bufsize-len);
- //printf("flush: %d (+%d)\n", len, len-oldlen);
- oldlen = len;
-
swf_SetBlock(tag, buf, len);
-
- /* len += lame_encode_flush(lame_flags, &buf[len], bufsize-len);
- printf("flush! %d (+%d)\n", len, len-oldlen);*/
-
+ if(len == 0) {
+ fprintf(stderr, "error: mp3 empty block, %d samples, first:%d, framesize:%d\n",
+ numsamples, first, fs);
+ }/* else {
+ fprintf(stderr, "ok: mp3 nonempty block, %d samples, first:%d, framesize:%d\n",
+ numsamples, first, fs);
+ }*/
free(buf);
}
lame_close (lame_flags);
}
+void swf_SetSoundDefineRaw(TAG*tag, S16*samples, int num, int samplerate)
+{
+ //swf_SetU8(tag,(/*compression*/0<<4)|(/*rate*/3<<2)|(/*size*/1<<1)|/*mono*/0);
+ //swf_SetU32(tag, numsamples); // 44100 -> 11025
+ //swf_SetBlock(tag, wav2.data, numsamples*2);
+}
void swf_SetSoundDefine(TAG*tag, S16*samples, int num)
{
char*buf;
int oldlen=0,len = 0;
int bufsize = 16384;
- int blocksize = 1152*2;
+ int blocksize = (int)(((swf_mp3_out_samplerate > 22050) ? 1152 : 576) * ((double)swf_mp3_in_samplerate/swf_mp3_out_samplerate));
int t;
int blocks;
U8 rate = 1; // 0 = 5.5 Khz, 1 = 11 Khz, 2 = 22 Khz, 3 = 44 Khz
U8 size = 1; // 0 = 8 bit, 1 = 16 bit
U8 type = 0; // 0 = mono, 1 = stereo
+
+ if(swf_mp3_out_samplerate == 5512) rate = 0;
+ else if(swf_mp3_out_samplerate == 11025) rate = 1;
+ else if(swf_mp3_out_samplerate == 22050) rate = 2;
+ else if(swf_mp3_out_samplerate == 44100) rate = 3;
+ else fprintf(stderr, "Invalid samplerate: %d\n", swf_mp3_out_samplerate);
blocks = num / (blocksize);
swf_SetU8(tag,(compression<<4)|(rate<<2)|(size<<1)|type);
- swf_SetU32(tag,blocks*blocksize / 4); // 44100 -> 11025
+ swf_SetU32(tag, (int)(tag,blocks*blocksize /
+ ((double)swf_mp3_in_samplerate/swf_mp3_out_samplerate)) // account for resampling
+ );
buf = malloc(bufsize);
if(!buf)
return;
- initlame(buf, bufsize);
+ initlame();
swf_SetU16(tag, 0); //delayseek
for(t=0;t<blocks;t++) {
git.asbjorn.biz / swftools.git / blobdiff