Karaoke Machine
Dependencies: SDFileSystem mbed-rtos mbed
main.cpp
- Committer:
- myeh
- Date:
- 2015-12-07
- Revision:
- 0:1d4e1f997a95
File content as of revision 0:1d4e1f997a95:
#include "SDFileSystem.h" #include "rtos.h" #include "mbed.h" typedef struct uFMT_STRUCT { short comp_code; short num_channels; unsigned voice_rate; unsigned avg_Bps; short block_align; short sig_bps; } FMT_STRUCT; class wave_player { public: /** Create a wave player using a pointer to the given AnalogOut object. * * @param _dac pointer to an AnalogOut object to which the voices are sent. */ wave_player(AnalogOut *_dac); void play(FILE *wavefile); void set_verbosity(int v); /** the player function. * * @param wavefile A pointer to an opened wave file */ /** Set the printf verbosity of the wave player. A nonzero verbosity level * will put wave_player in a mode where the complete contents of the wave * file are echoed to the screen, including header values, and including * all of the voice values placed into the DAC FIFO, and the voice values * removed from the DAC FIFO by the ISR. The voice output frequency is * fixed at 2 Hz in this mode, so it's all very slow and the DAC output isn't * very useful, but it lets you see what's going on and may help for debugging * wave files that don't play correctly. * * @param v the verbosity level */ private: void dac_out(void); int verbosity; AnalogOut *wave_DAC; Ticker tick; unsigned short DAC_fifo[256]; short DAC_wptr; volatile short DAC_rptr; short DAC_on; }; wave_player::wave_player(AnalogOut *_dac) { wave_DAC=_dac; wave_DAC->write_u16(32768); //DAC is 0-3.3V, so idles at ~1.6V verbosity=0; } //----------------------------------------------------------------------------- // if verbosity is set then wave player enters a mode where the wave file // is decoded and displayed to the screen, including voice values put into // the DAC FIFO, and values read out of the DAC FIFO by the ISR. The DAC output // itself is so slow as to be unusable, but this might be handy for debugging // wave files that don't play //----------------------------------------------------------------------------- void wave_player::set_verbosity(int v) { verbosity=v; } //----------------------------------------------------------------------------- // player function. Takes a pointer to an opened wave file. The file needs // to be stored in a filesystem with enough bandwidth to feed the wave data. // LocalFileSystem isn't, but the SDcard is, at least for 22kHz files. The // SDcard filesystem can be hotrodded by increasing the SPI frequency it uses // internally. //----------------------------------------------------------------------------- void wave_player::play(FILE *wavefile) { unsigned chunk_id,chunk_size,channel; unsigned data,samp_int,i; short unsigned dac_data; long long slice_value; char *slice_buf; short *data_sptr; unsigned char *data_bptr; int *data_wptr; FMT_STRUCT wav_format; long slice,num_slices; DAC_wptr=0; DAC_rptr=0; for (i=0;i<256;i+=2) { DAC_fifo[i]=0; DAC_fifo[i+1]=3000; } DAC_wptr=4; DAC_on=0; fread(&chunk_id,4,1,wavefile); fread(&chunk_size,4,1,wavefile); while (!feof(wavefile)) { if (verbosity) printf("Read chunk ID 0x%x, size 0x%x\n",chunk_id,chunk_size); switch (chunk_id) { case 0x46464952: fread(&data,4,1,wavefile); if (verbosity) { printf("RIFF chunk\n"); printf(" chunk size %d (0x%x)\n",chunk_size,chunk_size); printf(" RIFF type 0x%x\n",data); } break; case 0x20746d66: fread(&wav_format,sizeof(wav_format),1,wavefile); if (verbosity) { printf("FORMAT chunk\n"); printf(" chunk size %d (0x%x)\n",chunk_size,chunk_size); printf(" compression code %d\n",wav_format.comp_code); printf(" %d channels\n",wav_format.num_channels); printf(" %d voices/sec\n",wav_format.voice_rate); printf(" %d bytes/sec\n",wav_format.avg_Bps); printf(" block align %d\n",wav_format.block_align); printf(" %d bits per voice\n",wav_format.sig_bps); } if (chunk_size > sizeof(wav_format)) fseek(wavefile,chunk_size-sizeof(wav_format),SEEK_CUR); break; case 0x61746164: // allocate a buffer big enough to hold a slice slice_buf=(char *)malloc(wav_format.block_align); if (!slice_buf) { printf("Unable to malloc slice buffer"); exit(1); } num_slices=chunk_size/wav_format.block_align; samp_int=1000000/(wav_format.voice_rate); if (verbosity) { printf("DATA chunk\n"); printf(" chunk size %d (0x%x)\n",chunk_size,chunk_size); printf(" %d slices\n",num_slices); printf(" Ideal voice interval=%d\n",(unsigned)(1000000.0/wav_format.voice_rate)); printf(" programmed interrupt tick interval=%d\n",samp_int); } // starting up ticker to write voices out -- no printfs until tick.detach is called if (verbosity) tick.attach_us(this,&wave_player::dac_out, 500000); else tick.attach_us(this,&wave_player::dac_out, samp_int); DAC_on=1; // start reading slices, which contain one voice each for however many channels // are in the wave file. one channel=mono, two channels=stereo, etc. Since // mbed only has a single AnalogOut, all of the channels present are averaged // to produce a single voice value. This summing and averaging happens in // a variable of type signed long long, to make sure that the data doesn't // overflow regardless of voice size (8 bits, 16 bits, 32 bits). // // note that from what I can find that 8 bit wave files use unsigned data, // while 16 and 32 bit wave files use signed data // for (slice=0;slice<num_slices;slice+=1) { fread(slice_buf,wav_format.block_align,1,wavefile); if (feof(wavefile)) { printf("Oops -- not enough slices in the wave file\n"); exit(1); } data_sptr=(short *)slice_buf; // 16 bit voices data_bptr=(unsigned char *)slice_buf; // 8 bit voices data_wptr=(int *)slice_buf; // 32 bit voices slice_value=0; for (channel=0;channel<wav_format.num_channels;channel++) { switch (wav_format.sig_bps) { case 16: if (verbosity) printf("16 bit channel %d data=%d ",channel,data_sptr[channel]); slice_value+=data_sptr[channel]; break; case 32: if (verbosity) printf("32 bit channel %d data=%d ",channel,data_wptr[channel]); slice_value+=data_wptr[channel]; break; case 8: if (verbosity) printf("8 bit channel %d data=%d ",channel,(int)data_bptr[channel]); slice_value+=data_bptr[channel]; break; } } slice_value/=wav_format.num_channels; // slice_value is now averaged. Next it needs to be scaled to an unsigned 16 bit value // with DC offset so it can be written to the DAC. switch (wav_format.sig_bps) { case 8: slice_value<<=8; break; case 16: slice_value+=32768; break; case 32: slice_value>>=16; slice_value+=32768; break; } dac_data=(short unsigned)slice_value; if (verbosity) printf("voice %d wptr %d slice_value %d dac_data %u\n",slice,DAC_wptr,(int)slice_value,dac_data); DAC_fifo[DAC_wptr]=dac_data; DAC_wptr=(DAC_wptr+1) & 0xff; while (DAC_wptr==DAC_rptr) { } } DAC_on=0; tick.detach(); free(slice_buf); break; case 0x5453494c: if (verbosity) printf("INFO chunk, size %d\n",chunk_size); fseek(wavefile,chunk_size,SEEK_CUR); break; default: printf("unknown chunk type 0x%x, size %d\n",chunk_id,chunk_size); data=fseek(wavefile,chunk_size,SEEK_CUR); break; } fread(&chunk_id,4,1,wavefile); fread(&chunk_size,4,1,wavefile); } } BusOut myleds(LED1,LED2,LED3,LED4); SDFileSystem sd(p5, p6, p7, p8, "sd"); AnalogOut speaker(p18); wave_player waver(&speaker); Serial pc(USBTX, USBRX); DigitalOut P15(p15); DigitalOut P16(p16); DigitalOut P19(p19); DigitalOut P20(p20); unsigned short data; class microphone { public : microphone(PinName pin); unsigned short read_u16(); operator unsigned short (); private : AnalogIn _pin; }; microphone::microphone (PinName pin): _pin(pin) { } unsigned short microphone::read_u16() { return _pin.read_u16(); } inline microphone::operator unsigned short () { return _pin.read_u16(); } microphone mymicrophone(p17); int main() { while(1) { FILE *wave_file; wave_file=fopen("/sd/wavfiles/Yakko's World.wav","r"); waver.play(wave_file); fclose(wave_file); } } void wave_player::dac_out() { if (DAC_on) { #ifdef VERBOSE printf("ISR rdptr %d got %u\n",DAC_rptr,DAC_fifo[DAC_rptr]); #endif wave_DAC->write_u16((mymicrophone + DAC_fifo[DAC_rptr])/2); DAC_rptr=(DAC_rptr+1) & 0xff; } }