mirror of
https://github.com/QB64Official/qb64.git
synced 2024-09-04 11:10:15 +00:00
285 lines
7.7 KiB
C
285 lines
7.7 KiB
C
#ifndef DEPENDENCY_AUDIO_DECODE
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//Stubs:
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//(none required)
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#else
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#define DEPENDENCY_AUDIO_DECODE_OGG
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#define DEPENDENCY_AUDIO_DECODE_MP3
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#define DEPENDENCY_AUDIO_DECODE_WAV
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#ifdef QB64_BACKSLASH_FILESYSTEM
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#ifdef DEPENDENCY_AUDIO_DECODE_MP3
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#include "mp3_mini\\src.c"
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#endif
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#ifdef DEPENDENCY_AUDIO_DECODE_WAV
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#include "wav\\src.c"
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#endif
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#ifdef DEPENDENCY_AUDIO_DECODE_OGG
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#include "ogg\\src.c"
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#endif
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#else
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#ifdef DEPENDENCY_AUDIO_DECODE_MP3
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#include "mp3_mini/src.c"
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#endif
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#ifdef DEPENDENCY_AUDIO_DECODE_WAV
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#include "wav/src.c"
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#endif
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#ifdef DEPENDENCY_AUDIO_DECODE_OGG
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#include "ogg/src.c"
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#endif
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#endif
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//forward refs:
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void sub__sndvol(int32 handle,float volume);
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void sub__sndclose(int32 handle);
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int32 func__sndopen(qbs* filename,qbs* requirements,int32 passed){
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sndsetup();
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if (new_error) return 0;
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static qbs *s1=NULL;
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if (!s1) s1=qbs_new(0,0);
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static qbs *req=NULL;
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if (!req) req=qbs_new(0,0);
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static qbs *s3=NULL;
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if (!s3) s3=qbs_new(0,0);
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static uint8 r[32];
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static int32 i,i2,i3;
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//check requirements
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memset(r,0,32);
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if (passed){
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if (requirements->len){
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i=1;
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qbs_set(req,qbs_ucase(requirements));//convert tmp str to perm str
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nextrequirement:
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i2=func_instr(i,req,qbs_new_txt(","),1);
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if (i2){
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qbs_set(s1,func_mid(req,i,i2-i,1));
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}else{
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qbs_set(s1,func_mid(req,i,req->len-i+1,1));
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}
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qbs_set(s1,qbs_rtrim(qbs_ltrim(s1)));
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if (qbs_equal(s1,qbs_new_txt("SYNC"))){r[0]++; goto valid;}
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if (qbs_equal(s1,qbs_new_txt("VOL"))){r[1]++; goto valid;}
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if (qbs_equal(s1,qbs_new_txt("PAUSE"))){r[2]++; goto valid;}
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if (qbs_equal(s1,qbs_new_txt("LEN"))){r[3]++; goto valid;}
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if (qbs_equal(s1,qbs_new_txt("SETPOS"))){r[4]++; goto valid;}
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error(5); return 0;//invalid requirements
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valid:
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if (i2){i=i2+1; goto nextrequirement;}
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for (i=0;i<32;i++) if (r[i]>1){error(5); return 0;}//cannot define requirements twice
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}//->len
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}//passed
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qbs_set(s1,qbs_add(filename,qbs_new_txt_len("\0",1)));//s1=filename+CHR$(0)
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if (!r[0]){//NOT SYNC
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if (snd_stream_handle){error(5); return 0;}//stream in use
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}
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//load file
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if (s1->len==1) return 0;//return invalid handle if null length string
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static int32 fh,result;
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static int64 lof;
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fh=gfs_open(s1,1,0,0);
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if (fh<0) return 0;
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lof=gfs_lof(fh);
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static uint8* content;
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content=(uint8*)malloc(lof); if (!content){gfs_close(fh); return 0;}
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result=gfs_read(fh,-1,content,lof);
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gfs_close(fh);
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if (result<0){free(content); return 0;}
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//identify file format
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static snd_sequence_struct *seq;
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//OGG?
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#ifdef DEPENDENCY_AUDIO_DECODE_OGG
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if (lof>=3){
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if (content[0]==79){ if (content[1]==103){ if (content[2]==103){//"Ogg"
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seq=snd_decode_ogg(content,lof);
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goto got_seq;
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}}}
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}//3
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#endif
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//WAV?
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#ifdef DEPENDENCY_AUDIO_DECODE_WAV
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if (lof>=12){
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if ((*(uint32*)&content[8])==0x45564157){//WAVE
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seq=snd_decode_wav(content,lof);
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goto got_seq;
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}//WAVE
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}
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#endif
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//assume mp3!
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//MP3?
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#ifdef DEPENDENCY_AUDIO_DECODE_MP3
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seq=snd_decode_mp3(content,lof);
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#endif
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got_seq:
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free(content);
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if (seq==NULL) return 0;
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//convert sequence (includes sample rate conversion etc etc)
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//just perform sample_rate fix for now...
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//1. 8->16bit conversion and/or edian conversion
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static int32 incorrect_format;
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incorrect_format=0;
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if (seq->bits_per_sample!=16) incorrect_format=1;
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if (seq->is_unsigned) incorrect_format=1;
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//todo... if (seq->endian==???)
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//this section does not fix the frequency, only the bits per sample
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//and signed-ness of the data
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if (incorrect_format){
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static int32 bps; bps=seq->bits_per_sample/8;
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static int32 samples; samples=seq->data_size/bps;
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static uint8 *new_data;
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if (bps!=2){
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new_data=(uint8*)malloc(samples*2);
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}else{
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new_data=seq->data;
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}
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static int32 i,v;
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for (i=0;i<samples;i++){
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//read original value
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v=0;
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if (bps==1){
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if (seq->is_unsigned){
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v=*(uint8*)(seq->data+i*1);
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v=(v-128)*256;
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}else{
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v=*(int8*)(seq->data+i*1);
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v=v*128;
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}
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}
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if (bps==2){
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if (seq->is_unsigned){
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v=*(uint16*)(seq->data+i*2);
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v=v-32768;
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}else{
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v=*(int16*)(seq->data+i*2);
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}
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}
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//place new value into array
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((int16*)new_data)[i]=v;
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}//i
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if (bps!=2){free(seq->data); seq->data=new_data; seq->data_size=samples*2;}
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//update seq info
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seq->bits_per_sample=16;
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seq->is_unsigned=0;
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}//incorrect format
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//2. samplerate conversion
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if (seq->sample_rate != snd_frequency) { //need to resample seq->data
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//create new resampler
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SpeexResamplerState *state;
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state = speex_resampler_init(seq->channels, seq->sample_rate, snd_frequency, SPEEX_RESAMPLER_QUALITY_MIN, NULL);
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if (!state) { //NULL means failure
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free(seq->data);
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return 0;
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}
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//allocate new memory for output
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int32 out_samples_max = ((double)seq->data_size / seq->channels / 2) * ((((double)snd_frequency) / ((double)seq->sample_rate)) + 0.1) + 100;//10%+100 extra samples as a buffer-zone
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int16 *resampled = (int16 *)malloc(out_samples_max * seq->channels * sizeof(int16));
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if (!resampled) {
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free(seq->data);
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return 0;
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}
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//establish data sizes
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//in_len will be set by the resampler to number of samples processed
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spx_uint32_t in_len = seq->data_size / seq->channels / 2; // divide by 2 because 2byte samples, divive by #channels because function wants it per-channel
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//out_len will be set to the number of samples written
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spx_uint32_t out_len;
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//resample!
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if (speex_resampler_process_interleaved_int(state, (spx_int16_t *)seq->data, &in_len, (spx_int16_t *)resampled, &out_len) != RESAMPLER_ERR_SUCCESS) {
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//Error
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free(resampled);
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free(seq->data);
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speex_resampler_destroy(state);
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return 0;
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}
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//destroy the resampler anyway
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speex_resampler_destroy(state);
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//establish real size of new data and update seq
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free(seq->data); //That was the old data
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seq->data_size = out_len * seq->channels * 2; //remember out_len is perchannel, and each sample is 2 bytes
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seq->data = (uint8_t *)realloc(resampled, seq->data_size); //we overestimated the array size before, so make it the correct size now
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if (!seq->data) { //realloc could fail
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free(resampled);
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return 0;
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}
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seq->sample_rate = snd_frequency;
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}
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if (seq->channels==1){
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seq->data_mono=seq->data;
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seq->data_mono_size=seq->data_size;
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}
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if (seq->channels==2){
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seq->data_stereo=seq->data;
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seq->data_stereo_size=seq->data_size;
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}
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if (seq->channels>2) return 0;
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//attach sequence to handle (& inc. refs)
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//create snd handle
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static int32 handle; handle=list_add(snd_handles);
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static snd_struct *snd; snd=(snd_struct*)list_get(snd_handles,handle);
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snd->internal=0;
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snd->type=2;
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snd->seq=seq;
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snd->volume=1.0;
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snd->capability=r[0]*SND_CAPABILITY_SYNC+r[1]*SND_CAPABILITY_VOL+r[2]*SND_CAPABILITY_PAUSE+r[3]*SND_CAPABILITY_LEN+r[4]*SND_CAPABILITY_SETPOS;
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if (!r[0]){
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snd->streamed=1;//NOT SYNC
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snd_stream_handle=handle;
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}
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return handle;
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}
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void sub__sndplayfile(qbs *filename,int32 sync,double volume,int32 passed){
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if (new_error) return;
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sndsetup();
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static int32 handle;
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static int32 setvolume;
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static qbs *syncstr=NULL; if (!syncstr) syncstr=qbs_new(0,0);
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setvolume=0;
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if (passed&2){
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if ((volume<0)||(volume>1)){error(5); return;}
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if (volume!=1) setvolume=1;
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}
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if ((!setvolume)&&(!sync)) syncstr->len=0;
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if ((setvolume)&&(!sync)) qbs_set(syncstr,qbs_new_txt("VOL"));
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if ((!setvolume)&&(sync)) qbs_set(syncstr,qbs_new_txt("SYNC"));
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if ((setvolume)&&(sync)) qbs_set(syncstr,qbs_new_txt("SYNC,VOL"));
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if (syncstr->len){
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handle=func__sndopen(filename,syncstr,1);
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}else{
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handle=func__sndopen(filename,NULL,0);
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}
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if (handle==0) return;
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if (setvolume) sub__sndvol(handle,volume);
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sub__sndplay(handle);
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sub__sndclose(handle);
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}
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#endif
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