Opus to Wav working example
Signed-off-by: Mahyar Koshkouei <deltabeard@users.noreply.github.com>
This commit is contained in:
@@ -17,6 +17,7 @@
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#if !defined(_POSIX_SOURCE)
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# define _POSIX_SOURCE 1
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#endif
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#include <3ds.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <errno.h>
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@@ -28,146 +29,118 @@
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# define fileno _fileno
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#endif
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static void put_le32(unsigned char *_dst,opus_uint32 _x){
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_dst[0]=(unsigned char)(_x&0xFF);
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_dst[1]=(unsigned char)(_x>>8&0xFF);
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_dst[2]=(unsigned char)(_x>>16&0xFF);
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_dst[3]=(unsigned char)(_x>>24&0xFF);
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}
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#define BUFFER_SIZE 120*48*2
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/*Make a header for a 48 kHz, stereo, signed, 16-bit little-endian PCM WAV.*/
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static void make_wav_header(unsigned char _dst[44],ogg_int64_t _duration){
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/*The chunk sizes are set to 0x7FFFFFFF by default.
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Many, though not all, programs will interpret this to mean the duration is
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"undefined", and continue to read from the file so long as there is actual
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data.*/
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static const unsigned char WAV_HEADER_TEMPLATE[44]={
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'R','I','F','F',0xFF,0xFF,0xFF,0x7F,
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'W','A','V','E','f','m','t',' ',
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0x10,0x00,0x00,0x00,0x01,0x00,0x02,0x00,
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0x80,0xBB,0x00,0x00,0x00,0xEE,0x02,0x00,
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0x04,0x00,0x10,0x00,'d','a','t','a',
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0xFF,0xFF,0xFF,0x7F
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};
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memcpy(_dst,WAV_HEADER_TEMPLATE,sizeof(WAV_HEADER_TEMPLATE));
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if(_duration>0){
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if(_duration>0x1FFFFFF6){
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fprintf(stderr,"WARNING: WAV output would be larger than 2 GB.\n");
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fprintf(stderr,
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"Writing non-standard WAV header with invalid chunk sizes.\n");
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}
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else{
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opus_uint32 audio_size;
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audio_size=(opus_uint32)(_duration*4);
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put_le32(_dst+4,audio_size+36);
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put_le32(_dst+40,audio_size);
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}
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}
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}
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int convOpus(const char* in, const char *outf){
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int convOpus(const char* in, const char *outf)
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{
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OggOpusFile *of;
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FILE *wav;
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ogg_int64_t duration;
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unsigned char wav_header[44];
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ogg_int64_t duration = 0;
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int ret;
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int output_seekable;
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ogg_int64_t nsamples = 0;
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opus_int16* pcm = malloc(BUFFER_SIZE*sizeof(opus_int16));
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unsigned char* out = malloc(BUFFER_SIZE*2*sizeof(unsigned char));
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of = op_open_file(in, &ret);
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if((wav = fopen(outf, "w+")) == NULL)
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{
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fprintf(stderr,"Failed to open file '%s': %i\n",outf,errno);
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fprintf(stderr,"Failed to open file '%s': %i %s\n",outf,errno, strerror(errno));
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return EXIT_FAILURE;
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}
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if(of==NULL){
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if(of == NULL)
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{
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fprintf(stderr,"Failed to open file '%s': %i\n",in,ret);
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return EXIT_FAILURE;
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}
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duration=0;
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output_seekable=fseek(wav,0,SEEK_CUR)!=-1;
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output_seekable = fseek(wav, 0, SEEK_CUR);
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if(op_seekable(of)){
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if(op_seekable(of))
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{
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fprintf(stderr,"Total number of links: %i\n",op_link_count(of));
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duration=op_pcm_total(of,-1);
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// Get PCM length of entire stream.
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duration = op_pcm_total(of, -1);
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fprintf(stderr,"Total duration: ");
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fprintf(stderr," (%li samples @ 48 kHz)\n",(long)duration);
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}
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else if(!output_seekable){
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else if(output_seekable < 0)
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{
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fprintf(stderr,"WARNING: Neither input nor output are seekable.\n");
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fprintf(stderr,
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"Writing non-standard WAV header with invalid chunk sizes.\n");
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}
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make_wav_header(wav_header,duration);
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for(;;)
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{
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int si;
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if(!fwrite(wav_header,sizeof(wav_header),1,wav)){
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fprintf(stderr,"Error writing WAV header: %s\n",strerror(errno));
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ret=EXIT_FAILURE;
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/*Although we would generally prefer to use the float interface, WAV
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files with signed, 16-bit little-endian samples are far more
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universally supported, so that's what we output.*/
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ret = op_read_stereo(of, pcm, sizeof(pcm)/sizeof(*pcm));
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if(ret < 0)
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{
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fprintf(stderr, "\nError decoding '%s': %i\n", in, ret);
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ret = EXIT_FAILURE;
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break;
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}
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if(ret == 0)
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{
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ret = EXIT_SUCCESS;
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break;
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}
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/*Ensure the data is little-endian before writing it out.*/
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for(si=0; si<2*ret; si++)
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{
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out[2*si+0]=(unsigned char)(pcm[si]&0xFF);
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out[2*si+1]=(unsigned char)(pcm[si]>>8&0xFF);
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}
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if(fwrite(out, sizeof(u32), 1, wav) == 0)
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{
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fprintf(stderr, "\nError writing decoded audio data: %s\n",
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strerror(errno));
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ret = EXIT_FAILURE;
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break;
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}
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nsamples += ret;
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if(nsamples % 1000 == 0)
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fprintf(stderr, "\rRet: %d, Sample: %li", ret, (long)nsamples);
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}
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else{
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ogg_int64_t nsamples;
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opus_int16* pcm = malloc(120*48*2*sizeof(opus_int16));
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unsigned char* out = malloc(120*48*2*2*sizeof(unsigned char));
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nsamples=0;
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for(;;){
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int si;
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/*Although we would generally prefer to use the float interface, WAV
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files with signed, 16-bit little-endian samples are far more
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universally supported, so that's what we output.*/
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ret=op_read_stereo(of,pcm,sizeof(pcm)/sizeof(*pcm));
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if(ret<0){
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fprintf(stderr,"\nError decoding '%s': %i\n",in,ret);
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ret=EXIT_FAILURE;
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break;
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}
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if(ret<=0){
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ret=EXIT_SUCCESS;
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break;
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}
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/*Ensure the data is little-endian before writing it out.*/
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for(si=0;si<2*ret;si++){
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out[2*si+0]=(unsigned char)(pcm[si]&0xFF);
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out[2*si+1]=(unsigned char)(pcm[si]>>8&0xFF);
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}
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if(!fwrite(out,sizeof(*out)*4*ret,1,wav)){
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fprintf(stderr,"\nError writing decoded audio data: %s\n",
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strerror(errno));
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ret=EXIT_FAILURE;
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break;
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}
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nsamples+=ret;
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gfxFlushBuffers();
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gfxSwapBuffers();
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gspWaitForVBlank();
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if(nsamples % 1000 == 0)
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fprintf(stderr, "\rSample: %li", (long)nsamples);
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//End of for loop.
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}
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free(pcm);
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free(out);
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if(ret==EXIT_SUCCESS){
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fprintf(stderr,"\nDone: played ");
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fprintf(stderr," (%li samples @ 48 kHz).\n",(long)nsamples);
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}
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if(op_seekable(of)&&nsamples!=duration){
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fprintf(stderr,"\nWARNING: "
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"Number of output samples does not match declared file duration.\n");
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if(!output_seekable)fprintf(stderr,"Output WAV file will be corrupt.\n");
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}
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if(output_seekable&&nsamples!=duration){
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make_wav_header(wav_header,nsamples);
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if(fseek(wav,0,SEEK_SET)||
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!fwrite(wav_header,sizeof(wav_header),1,wav)){
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fprintf(stderr,"Error rewriting WAV header: %s\n",strerror(errno));
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ret=EXIT_FAILURE;
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}
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}
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free(pcm);
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free(out);
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if(ret == EXIT_SUCCESS)
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{
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fprintf(stderr, "\nDone: played ");
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fprintf(stderr, " (%li samples @ 48 kHz).\n", (long)nsamples);
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}
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fclose(wav);
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if(op_seekable(of) && nsamples != duration)
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{
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fprintf(stderr, "\nWARNING: Number of output samples does not match "
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"declared file duration.\n");
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if(output_seekable < 0)
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fprintf(stderr, "Output WAV file will be corrupt.\n");
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}
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if(fclose(wav) != 0)
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fprintf(stderr, "\nError closing file: %d - %s\n", errno, strerror(errno));
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op_free(of);
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return ret;
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}
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