X-Git-Url: http://git.asbjorn.biz/?a=blobdiff_plain;f=lib%2Fmodules%2Fswfsound.c;h=d76baec513275bd69f53906d29e331ed0f85f3f2;hb=879d0eec420fe0fd5ddcd56c8fe62b82a6744edd;hp=ba5e93593cb1bf6cf09976377fb95cfed6aa4542;hpb=5bb773e10a99dea6e224030bda4b58aba1679869;p=swftools.git diff --git a/lib/modules/swfsound.c b/lib/modules/swfsound.c index ba5e935..d76baec 100644 --- a/lib/modules/swfsound.c +++ b/lib/modules/swfsound.c @@ -7,19 +7,34 @@ Copyright (c) 2001, 2002 Matthias Kramm - 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 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 + 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 @@ -45,11 +60,10 @@ void swf_SetSoundStreamBlock(TAG*tag, S16*samples, int numsamples, char first) char*buf; int len = 0; + buf = rfx_alloc(init->bufferSize); if(!buf) return; - buf = malloc(init->bufferSize); - len = codecEncodeChunk (numsamples, samples, buf); len += codecFlush (&buf[len]); len += codecExit (&buf[len]); @@ -59,6 +73,261 @@ void swf_SetSoundStreamBlock(TAG*tag, S16*samples, int numsamples, char first) swf_SetU16(tag, 0); // seek } swf_SetBlock(tag, buf, len); - free(buf); + 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 +#include + +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;tstop?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;tenvelopes;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); +}