/* ******************************************************************** * File: aufileutil.c * Project: Audio Delay * Module: Audio File Untilities * SubModule: * Purpose: Read audio (.au, .aiff, .snd) data file and decode data * Data Format: AE_PCMU, AE_PCMA, AE_L8 (sign?), AE_L16, AE_L32(sign?) * Frequency: Any * Call: None * Called by: delay.c * Author: Hao Huang * Date: 1/1/98 * Revision: * Notes: * * machine bits max sampling rate #output channels * ------------------------------------------------------------------------- * Mac (all types) 8 22k 1 * Mac (newer ones) 16 64k 4(128) * Apple IIgs 8 32k / >70k 16(st) * PC/soundblaster pro 8 ?/(22k st, 44.1k mo) 1(st) * PC/soundblaster 16 16 44.1k 1(st) * PC/pas 8 44.1k st, 88.2k mo 1(st) * PC/pas-16 16 44.1k st, 88.2k mo 1(st) * PC/turtle beach multisound 16 44.1k 1(st) * PC/cards with aria chipset 16 44.1k 1(st) * PC/roland rap-10 16 44.1k 1(st) * PC/gravis ultrasound 8/16 44.1k 14-32(st) * Atari ST 8 22k 1 * Atari STE,TT 8 50k 2 * Atari Falcon 030 16 50k 8(st) * Amiga 8 varies above 29k 4(st) * Sun Sparc U-LAW 8k 1 * Sun Sparcst. 10 U-LAW,8,16 48k 1(st) * NeXT U-LAW,8,16 44.1k 1(st) * SGI Indigo 8,16 48k 4(st) * SGI Indigo2,Indy 8,16 48k 16(st,4-channel) * Acorn Archimedes ~U-LAW ~180k 8(st) * Sony NWS-3xxx U,A,8,16 8-37.8k 1(st) * Sony NWS-5xxx U,A,8,16 8-48k 1(st) * VAXstation 4000 U-LAW 8k 1 * DEC 3000 U-LAW 8k 1 * DEC 5000/20-25 U-LAW 8k 1 * Tandy 1000L 8 >=44k 1 * Tandy 2500 8 >=44k 1 * HP9000/705,710,425e U,A-LAW,16 8k 1 * HP9000/715,725,735 U,A-LAW,16 48k 1(st) * HP9000/755 option: U,A-LAW,16 48k 1(st) * NCD MCX terminal U,A,8,16 52k 1(st) * * Inverse Mu-law * mp ( y ) * m = --- ( (u+1)e - 1 ) * u ( ) * Mu-Law: mp is the peak value, u = 255 * sgn(m) ( |m |) |m | * y = ------- ln( 1+ u|--|) |--| =< 1 * ln(1+u) ( |mp|) |mp| * Mu-Law simplified with x = m / mp * ln(1+ u|x|) * y = sgn(x) ------------- * ln(u+1) * ******************************************************************** */ #include #include #include // ntohl() and ntohs() #include "audio_generic.h" #include "util.h" #include "aufileutil.h" const char * ENCODING[] = { "PCMU", "PCMA", "G721", "DVI4", "G723", "GSM", "1016", "LPC", "Linear 8-bit", "Linear 16-bit", "Linear 24-bit", "Linear 32-bit", "G728", "TRUE" }; const char * OPEN_AUDIO_FILE_ERROR[] = { "The channel count is not 2", "Insufficeint samples in the file\n", }; /* * Given the encoding, return the data unit size in byte * This is incomplete. Most default to 1 */ int getUnitSize(int encoding) { switch (encoding) { case AE_PCMU: return(1); case AE_PCMA: return(1); case AE_G721: return(1); // unknown case AE_DVI4: return(1); // unknown case AE_G723: return(1); // unknown case AE_GSM: return(1); // unknown case AE_1016: return(1); // unknown case AE_LPC: return(1); // unknown case AE_L8: return(1); case AE_L16: return(2); case AE_L24: return(3); case AE_L32: return(4); case AE_G728: return(1); // unknown case AE_TRUE: return(1); // unknown case AE_MAX: return(1); // unknown default: return(1); } } // int getUnitSize(int encoding) /* * Open the audio file and read in header * time is in seconds * Return NULL if failed * ErrorCode: -1 failed to open the file * 0 the channel number is not 2 * 1 insufficient sample data in file */ FILE* openAudioFile(const char * name, float time, audio_file_t * af_p, int * error_p) { FILE * fp; int n; if (!(fp = af_open(open_file(0, name, 0, "r", 0), "r", af_p))) { *error_p = -1; return(NULL); } if (2 != af_p->ad.channels) { fclose(fp); *error_p = 0; return(NULL); } n = time * af_p->ad.sample_rate; if (af_p->data_size < n) { //fprintf(stderr, "datasize: %d\n", af_p->data_size); //fprintf(stderr, "sample_rate:%d\n", af_p->ad.sample_rate); fclose(fp); *error_p = 1; return(NULL); } return(fp); } /* openAudioFile */ /* Print the audio description */ void printAudioDescr(FILE * fp, const audio_descr_t * p) { fprintf(fp, "Encoding = %s\nSampleRate = %d\nChannels = %d\n", p->encoding < AE_MAX ? ENCODING[p->encoding] : "Unkown", p->sample_rate, p->channels); } /* printAudioDescr */ /* Print the info of the audio file */ void printAudioFileInfo(FILE * fp, const audio_file_t * p) { char * s; printAudioDescr(fp, &p->ad); switch (p->type) { case AF_au: s = ".au"; break; case AF_snd: s = ".snd"; break; case AF_aiff: s = ".aiff"; break; case AF_other: s = "others"; break; default: s = "Unknown"; break; } fprintf(fp, "Type = %s\nDataOffset = %d\nDataSize = %d\n", s, p->data_offset, p->data_size); if (p->info) { fprintf(fp, "Info = %s\n", p->info); } } /* printAudioFileInfo */ /* * Decode audio data in buffer to data1_a and data2_a for given encoding * Return non-zero if error occured */ int decodeAudioData(void * buffer, float *data1_a, float *data2_a, int len, audio_encoding_t encoding) { union { signed char * b1; signed short * b2; signed long * b4; unsigned char * ub1; unsigned short * ub2; unsigned long * ub4; } buf; int i; //fprintf(stderr, "Decoding ... \n"); // Data are interleaved, so put them in data1 and data2 buf.b1 = (char*)buffer; switch (encoding) { case AE_PCMU: // U-LAW 8-bit signed case AE_PCMA: // A-LAW 8-bit signed //fprintf(stderr, "Decoding U-LAW or A-LAW\n"); // Not really decode it since the waveform does not change. for (i = 0; i < 2*len; i+=2) { data1_a[1+i/2] = buf.b1[i]; data2_a[1+i/2] = buf.b1[i+1]; } break; case AE_G721: return(-1); // unknown case AE_DVI4: return(-1); // unknown case AE_G723: return(-1); // unknown case AE_GSM: return(-1); // unknown case AE_1016: return(-1); // unknown case AE_LPC: return(-1); // unknown case AE_L8: // b-bit unsigned for (i = 0; i < 2*len; i+=2) { data1_a[1+i/2] = buf.b1[i]; data2_a[1+i/2] = buf.b1[i+1]; } break; case AE_L16: // 16-bit signed! //fprintf(stderr, "Decoding 16-bit linear\n"); for (i = 0; i < 2*len; i+=2) { data1_a[1+i/2] = buf.b2[i]; data2_a[1+i/2] = buf.b2[i+1]; } break; case AE_L24: return(3); case AE_L32: // 32-bit unsigned? for (i = 0; i < 2*len; i+=2) { data1_a[1+i/2] = buf.b4[i]; data2_a[1+i/2] = buf.b4[i+1]; } break; case AE_G728: return(-1); // unknown case AE_TRUE: return(-1); // unknown case AE_MAX: // unsupported default: // unsupported return(-1); } return(0); } /* decodeAudioData */