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 "../rfxswf.h"
+#ifdef BLADEENC
+fjokjklj
+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 = 0; // 0 = 8 bit, 1 = 16 bit
- U8 playbacktype = 1; // 0 = mono, 1 = stereo
- U8 compression = 2; // 0 = raw, 1 = ADPCM, 2 = mp3
+ 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 = 0; // 0 = 8 bit, 1 = 16 bit
- U8 type = 1; // 0 = mono, 1 = stereo
+ 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, U16*samples, int numsamples)
+void swf_SetSoundStreamBlock(TAG*tag, S16*samples, int numsamples, char first)
{
- CodecInitOut * init;
- CodecInitIn params;
char*buf;
int len = 0;
+ buf = malloc(init->bufferSize);
if(!buf)
return;
+
+ len = codecEncodeChunk (numsamples, samples, buf);
+ len += codecFlush (&buf[len]);
+ len += codecExit (&buf[len]);
- memset(¶ms, 0, sizeof(params));
- params.frequency = 44100; //48000, 44100 or 32000
- params.mode = 0; //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)
+ if(first) {
+ swf_SetU16(tag, numsamples); // number of samples
+ swf_SetU16(tag, 0); // seek
+ }
+ swf_SetBlock(tag, buf, len);
+ free(buf);
+}
+#endif
- init = codecInit(¶ms);
- printf("nSamples:%d\n", init->nSamples);
- printf("bufferSize:%d\n", init->bufferSize);
-
- buf = malloc(init->bufferSize);
+
+#ifdef LAME
+
+#include "../lame/lame.h"
+
+/* TODO: find a 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;
+
+static lame_global_flags*lame_flags;
+
+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);
+
+ /* 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);
+}
+
+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);
- len = codecEncodeChunk(numsamples, samples, buf);
- len += codecFlush (&buf[len]);
- len += codecExit(&buf[len]);
+ 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 = malloc(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);
+ }*/
free(buf);
}
+void swf_SetSoundStreamEnd(TAG*tag)
+{
+ 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 = (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 = malloc(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;
+ }
+
+ free(buf);
+}
+
+#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]);
+ }
+ }
+}
+
+#endif
+
+#endif // RFXSWF_DISABLESOUND
git.asbjorn.biz / swftools.git / blobdiff