/* * wav5n.c - writes wav file, appends blocks of sound, then updates length in header * usage: ..\tcc -run wav5n.c * usage: gcc wav5n.c -o wav5n && ./wav5n && aplay serialout.wav * * This have been modified to convert wav9 output to wav files * wav9 converts a binary file to CUTS 1200 baud. * * This puts a wav header onto raw audio in 8000 samples per second mono little endian * usage: wav5n inputRawfilename outputWavFilename.WAV * e.g. * wav5 pcmtx_dhrdlin30.out op2.wav * * idea use fseek to position and update header. * 1) write header first - final length is unknown for now. * 2) append data * 3) seek END to seek to final length, use ftell to get length * 4) seek to "data" and update length. */ #ifndef MAKE_WAV_H #define MAKE_WAV_H void write_wav(char * filename, unsigned long num_samples, short int * data, int s_rate); /* open a file named filename, write signed 16-bit values as a monoaural WAV file at the specified sampling rate and close the file */ #endif /* http://karplus4arduino.wordpress.com/2011/10/08/making-wav-files-from-c-programs/ karplus4arduino http://gasstationwithoutpumps.wordpress.com/2011/10/08/making-wav-files-from-c-programs/ 2011 October 8 Making WAV files from C programs Filed under: Digital music — gasstationwithoutpumps @ 21:03 Tags: software, WAV I’m going to try embedding some short pieces of code in this post, and later work out a better way of distributing larger blocks of code. The code I want to share today consists of a small C file and the associated header for creating WAV files for producing sounds on a laptop or desktop computer. This is not a full-featured WAV file module: the format has a lot of options and I didn’t want the complexity of dealing with all of them. I just wanted to output monophonic sound with a reasonable sampling rate. */ /* make_wav.c * Creates a WAV file from an array of ints. * Output is monophonic, signed 16-bit samples * copyright * Fri Jun 18 16:36:23 PDT 2010 Kevin Karplus * Creative Commons license Attribution-NonCommercial * http://creativecommons.org/licenses/by-nc/3.0/ */ #include #include #include #define S_RATE (8000) #define BUF_SIZE (S_RATE/2) /* 2 second buffer */ short int buffer[BUF_SIZE]; #ifndef M_PI #define M_PI 3.14 #endif int fillBuffer( int t1, int t2, int level ){ int i; int x1,y1; int x2,y2; int n; float t; float amplitude = 32000; float freq_Hz = 440; float phase=0; float freq_radians_per_sample = freq_Hz*2*M_PI/S_RATE; x1 = level ; y1 = 0; x2 = level ; y2 = 0; /* fill buffer with a sine wave */ for (i=0; i0) { buf = word & 0xff; fwrite(&buf, 1,1, wav_file); num_bytes--; word >>= 8; } } /* information about the WAV file format from http://ccrma.stanford.edu/courses/422/projects/WaveFormat/ */ void write_wav(char * filename, unsigned long num_samples, short int * data, int s_rate) { FILE* wav_file; unsigned int sample_rate; unsigned int num_channels; unsigned int bytes_per_sample; unsigned int byte_rate; unsigned long i; /* counter for samples */ num_channels = 1; /* monoaural */ bytes_per_sample = 2; if (s_rate<=0) sample_rate = S_RATE; else sample_rate = (unsigned int) s_rate; byte_rate = sample_rate*num_channels*bytes_per_sample; wav_file = fopen(filename, "wb"); assert(wav_file); /* make sure it opened */ /* write RIFF header */ fwrite("RIFF", 1, 4, wav_file); write_little_endian(36 + bytes_per_sample* num_samples*num_channels, 4, wav_file); fwrite("WAVE", 1, 4, wav_file); /* write fmt subchunk */ fwrite("fmt ", 1, 4, wav_file); write_little_endian(16, 4, wav_file); /* SubChunk1Size is 16 */ write_little_endian(1, 2, wav_file); /* PCM is format 1 */ write_little_endian(num_channels, 2, wav_file); write_little_endian(sample_rate, 4, wav_file); write_little_endian(byte_rate, 4, wav_file); write_little_endian(num_channels*bytes_per_sample, 2, wav_file); /* block align */ write_little_endian(8*bytes_per_sample, 2, wav_file); /* bits/sample */ /* write data subchunk */ fwrite("data", 1, 4, wav_file); i = bytes_per_sample* num_samples*num_channels; i = 0x5555; printf("%ld\n",i); write_little_endian(bytes_per_sample* num_samples*num_channels, 4, wav_file); for (i=0; i< num_samples; i++) { write_little_endian((unsigned int)(data[i]),bytes_per_sample, wav_file); } fclose(wav_file); } void write_wav_header(char * filename, unsigned long num_samples, short int * data, int s_rate) { FILE* wav_file; unsigned int sample_rate; unsigned int num_channels; unsigned int bytes_per_sample; unsigned int byte_rate; unsigned long i; /* counter for samples */ num_channels = 1; /* monoaural */ bytes_per_sample = 2; if (s_rate<=0) sample_rate = S_RATE; else sample_rate = (unsigned int) s_rate; byte_rate = sample_rate*num_channels*bytes_per_sample; wav_file = fopen(filename, "wb+"); assert(wav_file); /* make sure it opened */ fseek( wav_file,0,SEEK_SET); /* write RIFF header */ fwrite("RIFF", 1, 4, wav_file); write_little_endian(36 + bytes_per_sample* num_samples*num_channels, 4, wav_file); fwrite("WAVE", 1, 4, wav_file); /* write fmt subchunk */ fwrite("fmt ", 1, 4, wav_file); write_little_endian(16, 4, wav_file); /* SubChunk1Size is 16 */ write_little_endian(1, 2, wav_file); /* PCM is format 1 */ write_little_endian(num_channels, 2, wav_file); write_little_endian(sample_rate, 4, wav_file); write_little_endian(byte_rate, 4, wav_file); write_little_endian(num_channels*bytes_per_sample, 2, wav_file); /* block align */ write_little_endian(8*bytes_per_sample, 2, wav_file); /* bits/sample */ /* write data subchunk */ fwrite("data", 1, 4, wav_file); i = bytes_per_sample* num_samples*num_channels; i = 0x55555555; i = 0x10; //printf("write_wav_header: %ld\n",i); write_little_endian( i , 4, wav_file); /* for (i=0; i< num_samples; i++) { write_little_endian((unsigned int)(data[i]),bytes_per_sample, wav_file); } */ fclose(wav_file); } void write_wav_length(char * filename, unsigned long num_samples, short int * data, int s_rate) { FILE* wav_file; unsigned long i; /* counter for samples */ wav_file = fopen( filename, "rb+"); assert(wav_file); /* make sure it opened */ /* find the end of the file in bytes */ fseek( wav_file,0,SEEK_END); i = ftell(wav_file); /* now update data length at the start of the file */ fseek( wav_file,36,SEEK_SET); /* write data subchunk */ fwrite("data", 1, 4, wav_file); //printf("write_wav_header: %ld 0x%lX\n",i,i); write_little_endian( i , 4, wav_file); fseek( wav_file,0,SEEK_END); fclose(wav_file); } void write_wav_data(char * filename, unsigned long num_samples, short int * data, int s_rate) { FILE* wav_file; unsigned int sample_rate; unsigned int num_channels; unsigned int bytes_per_sample; unsigned int byte_rate; unsigned long i; /* counter for samples */ num_channels = 1; /* monoaural */ bytes_per_sample = 2; if (s_rate<=0) sample_rate = S_RATE; else sample_rate = (unsigned int) s_rate; byte_rate = sample_rate*num_channels*bytes_per_sample; /* apend data samples onto the end of the file, but it does not update length */ wav_file = fopen(filename, "ab+"); assert(wav_file); /* make sure it opened */ /* if we use "ab+" we append, so we do not need to seek */ // fseek( wav_file,36,SEEK_SET); for (i=0; i< num_samples; i++) { write_little_endian((unsigned int)(data[i]),bytes_per_sample, wav_file); } fclose(wav_file); } void copy_wav_data(char * filename,char * filename_in ) { FILE* wav_file; FILE* wav_file_in; FILE* wav_file_fjs; size_t got; short sample; short samplesA[20]; //opFP = stdout; /* apend data samples onto the end of the file, but it does not update length */ wav_file = fopen(filename, "ab+"); wav_file_in = fopen(filename_in, "rb" ); wav_file_fjs= fopen( "f.js" , "w+" ); if ( ! wav_file ){ printf( "error: output file "); } if ( ! wav_file_in ){ printf( "error: Input file "); } // printf("%p %p",wav_file,wav_file_in ); if ( wav_file_in ){ while(1){ /* 44.1 / 8 is 5.5, but convert stereo to mono as well */ got = fread( samplesA, sizeof( short ), 1, wav_file_in ); sample = samplesA[0]; if ( sample != 0 ){ if( wav_file_fjs ){ fprintf( wav_file_fjs , "f( %d,0,%d,0);\n", sample, sizeof( samplesA )); }; } if( feof(wav_file_in) ) { break ; } if ( wav_file ){ fwrite( samplesA, sizeof( short ), 1, wav_file); } //printf( "%d", got ); }; if ( wav_file ){ fclose( wav_file ); } if ( wav_file_in ){ fclose( wav_file_in ); } if ( wav_file_fjs ){ fclose( wav_file_fjs ); } } } /* Note that the above code is not a full program, just a tiny library routine that can be used to generate WAV files. Here is a test program to generate sine waves (inefficiently) to see if the program works. Once you have the basic idea, you can write arbitrary programs to generate WAV files and play them back with QuickTime Player, RealPlayer, Audacity, or any of several other sound-playing programs. This is really old-school computer music, from the days when computers were so slow that quality sounds could not be generated in real time. It provides a good way to get started in learning the algorithms, though, as you don’t need to worry about efficiency or all the details needed to compute samples and provide them to a DAC at exactly the right rate. */ /* test_make_wav.c * Fri Jun 18 17:13:19 PDT 2010 Kevin Karplus * Test program for the write_wav function in make_wav.c */ int main(int argc, char ** argv) { int i; if( argc < 2 ){ printf("\nThis puts a wav header onto raw audio in 8000ms mono little endian\n usage: wav5 inputRawfilename outputWavFilename \n"); } /* wave file HEADER "data"+length data if the data length is not known the length needs to be updated. */ char * ipFileName = "serialout.raw"; char * opFileName = "serialout.wav"; if( argc > 1 ){ ipFileName = argv[1]; } if( argc > 2 ){ opFileName = argv[2]; } printf("Using: %s > %s \n",ipFileName,opFileName ); //write_wav_header( "pcmtx_dhrdlin30.wav", BUF_SIZE, buffer, S_RATE ); write_wav_header( opFileName, BUF_SIZE, buffer, S_RATE ); copy_wav_data(opFileName,ipFileName); /* copy output input */ /* update length */ write_wav_length( opFileName, BUF_SIZE, buffer, S_RATE ); return 0; }