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DEFINITIONS
This source file includes following definitions.
- swf_SetSoundStreamHead
- swf_SetSoundStreamBlock
- swf_SetSoundDefineRaw
- null_errorf
- initlame
- swf_SetSoundStreamHead
- swf_SetSoundStreamBlock
- swf_SetSoundStreamEnd
- swf_SetSoundDefine
- swf_SetSoundStreamHead
- swf_SetSoundStreamBlock
- swf_SetSoundStreamEnd
- swf_SetSoundDefine
- swf_SetSoundInfo
- swf_SetSoundDefineMP3
/* swfaction.c
SWF Sound handling routines
Extension module for the rfxswf library.
Part of the swftools package.
Copyright (c) 2001, 2002 Matthias Kramm <kramm@quiss.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 */
#ifndef NO_MP3
#include "../rfxswf.h"
#ifdef BLADEENC
#define HAVE_SOUND
CodecInitOut * init = 0;
void swf_SetSoundStreamHead(TAG*tag, U16 avgnumsamples)
{
U8 playbackrate = 3; // 0 = 5.5 Khz, 1 = 11 Khz, 2 = 22 Khz, 3 = 44 Khz
U8 playbacksize = 1; // 0 = 8 bit, 1 = 16 bit
U8 playbacktype = 0; // 0 = mono, 1 = stereo
U8 compression = 2; // 0 = raw, 1 = ADPCM, 2 = mp3, 3 = raw le, 6 = nellymoser
U8 rate = 3; // 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
CodecInitIn params;
memset(¶ms, 0, sizeof(params));
params.frequency = 44100; //48000, 44100 or 32000
params.mode = 3; //0 = Stereo, 2 = Dual Channel, 3 = Mono
params.emphasis = 0; //0 = None, 1 = 50/15 microsec, 3 = CCITT J.17
params.bitrate = 128; //default is 128 (64 for mono)
init = codecInit(¶ms);
swf_SetU8(tag,(playbackrate<<2)|(playbacksize<<1)|playbacktype);
swf_SetU8(tag,(compression<<4)|(rate<<2)|(size<<1)|type);
swf_SetU16(tag,avgnumsamples);
printf("numSamples:%d\n",init->nSamples);
printf("bufferSize:%d\n",init->bufferSize);
}
void swf_SetSoundStreamBlock(TAG*tag, S16*samples, int numsamples, char first)
{
char*buf;
int len = 0;
buf = rfx_alloc(init->bufferSize);
if(!buf)
return;
len = codecEncodeChunk (numsamples, samples, buf);
len += codecFlush (&buf[len]);
len += codecExit (&buf[len]);
if(first) {
swf_SetU16(tag, numsamples); // number of samples
swf_SetU16(tag, 0); // seek
}
swf_SetBlock(tag, buf, len);
rfx_free(buf);
}
#endif
void swf_SetSoundDefineRaw(TAG*tag, S16*samples, int numsamples)
{
swf_SetU8(tag,(/*compression*/0<<4)|(/*rate*/3<<2)|(/*size*/1<<1)|/*mono*/0);
swf_SetU32(tag, numsamples); // 44100 -> 11025
swf_SetBlock(tag, (U8*)samples, numsamples*2);
}
/* TODO: find a better way to set these from the outside */
int swf_mp3_in_samplerate = 44100;
int swf_mp3_out_samplerate = 11025;
int swf_mp3_channels = 1;
int swf_mp3_bitrate = 32;
#ifdef HAVE_LAME
#define HAVE_SOUND
#include <stdarg.h>
#include <lame.h>
static lame_global_flags*lame_flags;
void null_errorf(const char *format, va_list ap)
{
}
static void initlame()
{
unsigned char buf[4096];
int bufsize = 1152*2;
lame_flags = lame_init();
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
lame_set_out_samplerate(lame_flags, swf_mp3_out_samplerate);
lame_set_quality(lame_flags, 0);
lame_set_mode(lame_flags, MONO/*3*/);
lame_set_brate(lame_flags, swf_mp3_bitrate);
//lame_set_compression_ratio(lame_flags, 11.025);
lame_set_bWriteVbrTag(lame_flags, 0);
lame_init_params(lame_flags);
lame_init_bitstream(lame_flags);
lame_set_errorf(lame_flags, null_errorf);
/* The first two flush calls to lame always fail, for
some reason. Do them here where they cause no damage. */
lame_encode_flush_nogap(lame_flags, buf, bufsize);
//printf("init:flush_nogap():%d\n", len);
lame_encode_flush(lame_flags, buf, bufsize);
//printf("init:flush():%d\n", len);
lame_set_errorf(lame_flags, 0);
}
void swf_SetSoundStreamHead(TAG*tag, int avgnumsamples)
{
int len;
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 playbacktype = 0; // 0 = mono, 1 = stereo
U8 compression = 2; // 0 = raw, 1 = ADPCM, 2 = mp3, 3 = raw le, 6 = nellymoser
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) 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);
swf_SetU16(tag,avgnumsamples);
}
void swf_SetSoundStreamBlock(TAG*tag, S16*samples, int seek, char first)
{
char*buf;
int len = 0;
int bufsize = 16384;
int numsamples = (int)(((swf_mp3_out_samplerate > 22050) ? 1152 : 576) * ((double)swf_mp3_in_samplerate/swf_mp3_out_samplerate));
int fs = 0;
buf = rfx_alloc(bufsize);
if(!buf)
return;
if(first) {
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);
len += lame_encode_flush_nogap(lame_flags, &buf[len], bufsize-len);
swf_SetBlock(tag, buf, len);
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);
}*/
rfx_free(buf);
}
void swf_SetSoundStreamEnd(TAG*tag)
{
lame_close (lame_flags);
}
void swf_SetSoundDefine(TAG*tag, S16*samples, int num)
{
char*buf;
int oldlen=0,len = 0;
int bufsize = 16384;
int blocksize = (int)(((swf_mp3_out_samplerate > 22050) ? 1152 : 576) * ((double)swf_mp3_in_samplerate/swf_mp3_out_samplerate));
int t;
int blocks;
U8 compression = 2; // 0 = raw, 1 = ADPCM, 2 = mp3, 3 = raw le, 6 = nellymoser
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, (int)(tag,blocks*blocksize /
((double)swf_mp3_in_samplerate/swf_mp3_out_samplerate)) // account for resampling
);
buf = rfx_alloc(bufsize);
if(!buf)
return;
initlame();
swf_SetU16(tag, 0); //delayseek
for(t=0;t<blocks;t++) {
int s;
U16*pos;
pos= &samples[t*blocksize];
len += lame_encode_buffer(lame_flags, pos, pos, blocksize, &buf[len], bufsize-len);
len += lame_encode_flush_nogap(lame_flags, &buf[len], bufsize-len);
swf_SetBlock(tag, buf, len);
len = 0;
}
rfx_free(buf);
}
#endif
#endif
#ifndef HAVE_SOUND
// supply stubs
void swf_SetSoundStreamHead(TAG*tag, int avgnumsamples)
{
fprintf(stderr, "Error: no mp3 soundstream support compiled in.\n");exit(1);
}
void swf_SetSoundStreamBlock(TAG*tag, S16*samples, int seek, char first)
{
fprintf(stderr, "Error: no mp3 soundstream support compiled in.\n");exit(1);
}
void swf_SetSoundStreamEnd(TAG*tag)
{
fprintf(stderr, "Error: no mp3 soundstream support compiled in.\n");exit(1);
}
void swf_SetSoundDefine(TAG*tag, S16*samples, int num)
{
swf_SetSoundDefineRaw(tag, samples,num);
}
#endif
#define SOUNDINFO_STOP 32
#define SOUNDINFO_NOMULTIPLE 16
#define SOUNDINFO_HASENVELOPE 8
#define SOUNDINFO_HASLOOPS 4
#define SOUNDINFO_HASOUTPOINT 2
#define SOUNDINFO_HASINPOINT 1
void swf_SetSoundInfo(TAG*tag, SOUNDINFO*info)
{
U8 flags = (info->stop?SOUNDINFO_STOP:0)
|(info->nomultiple?SOUNDINFO_NOMULTIPLE:0)
|(info->envelopes?SOUNDINFO_HASENVELOPE:0)
|(info->loops?SOUNDINFO_HASLOOPS:0)
|(info->outpoint?SOUNDINFO_HASOUTPOINT:0)
|(info->inpoint?SOUNDINFO_HASINPOINT:0);
swf_SetU8(tag, flags);
if(flags&SOUNDINFO_HASINPOINT)
swf_SetU32(tag, info->inpoint);
if(flags&SOUNDINFO_HASOUTPOINT)
swf_SetU32(tag, info->outpoint);
if(flags&SOUNDINFO_HASLOOPS)
swf_SetU16(tag, info->loops);
if(flags&SOUNDINFO_HASENVELOPE) {
int t;
swf_SetU8(tag, info->envelopes);
for(t=0;t<info->envelopes;t++) {
swf_SetU32(tag, info->pos[t]);
swf_SetU16(tag, info->left[t]);
swf_SetU16(tag, info->right[t]);
}
}
}
void swf_SetSoundDefineMP3(TAG*tag, U8* data, unsigned length,
unsigned SampRate,
unsigned Channels,
unsigned NumFrames)
{
U8 compression = 2; // 0 = raw, 1 = ADPCM, 2 = mp3, 3 = raw le, 6 = nellymoser
U8 rate; // 0 = 5.5 Khz, 1 = 11 Khz, 2 = 22 Khz, 3 = 44 Khz
U8 size = 1; // 0 = 8 bit, 1 = 16 bit
U8 type = Channels==2; // 0=mono, 1=stereo
rate = (SampRate >= 40000) ? 3
: (SampRate >= 19000) ? 2
: (SampRate >= 8000) ? 1
: 0;
swf_SetU8(tag,(compression<<4)|(rate<<2)|(size<<1)|type);
swf_SetU32(tag, NumFrames * 576);
swf_SetU16(tag, 0); //delayseek
swf_SetBlock(tag, data, length);
}