/* ** AudioPWM.c ** ** 07/22/06 v1.1 LDJ PIC24 ** 07/14/07 v2.0 LDJ PIC32 porting ** 11/23/07 V2.1 LDJ merging wav and audio PWM */ #include #include #include #include #include #include #include #define B_SIZE 512 // audio buffer size // audio configuration typedef struct { char stereo; // 0 - mono 1- stereo char fix; // sign fix 0x00 8-bit, 0x80 16-bit char skip; // advance pointer to next sample char size; // sample size (8 or 16-bit) } AudioCfg; // chunk IDs #define RIFF_DWORD 0x46464952UL #define WAVE_DWORD 0x45564157UL #define DATA_DWORD 0x61746164UL #define FMT_DWORD 0x20746d66UL #define WAV_DWORD 0x00564157UL typedef struct { // data chunk unsigned int dlength; // actual data size char data[4]; // "data" // format chunk unsigned short bitpsample; // bit per sample unsigned short bpsample; // bytes per sample // (4=16bit stereo) unsigned int bps; // bytes per second unsigned int srate; // sample rate in Hz unsigned short channels; // # of channels // (1= mono,2= stereo) unsigned short subtype; // always 01 unsigned int flength; // size of this block (16) char fmt_[4]; // "fmt_" char type[4]; // file type name "WAVE" unsigned int tlength; // size of encapsulated block char riff[4]; // envelope "RIFF" } WAVE; // global definitions char ABuffer[ 2][ B_SIZE]; // double data buffer volatile char CurBuf; // index of buffer in use volatile char AEmptyFlag; // buffer needs to be filled // audio configuration parameters AudioCfg ACfg; // local definitions unsigned char *BPtr; // pointer inside active buffer short BCount; // count bytes used void initAudio( void) { // configures peripherals for Audio playback // 1. activate the PWM modules // CH1 and CH2 in PWM mode, TMR2 based OpenOC1( OC_ON | OC_TIMER2_SRC | OC_PWM_FAULT_PIN_DISABLE, 0, 0); OpenOC2( OC_ON | OC_TIMER2_SRC | OC_PWM_FAULT_PIN_DISABLE, 0, 0); // 2. init the timebase // enable TMR2, prescale 1:1, internal clock, period OpenTimer2(T2_ON | T2_PS_1_1 | T2_SOURCE_INT, 0); mT2SetIntPriority( 4); // set TMR2 interrupt priority } // initAudio void startAudio( int bitrate, int position, int count) { // begins the audio playback // 1. init pointers and flags CurBuf = 0; // buffer 0 active first BPtr = ABuffer[ CurBuf] + position; AEmptyFlag = FALSE; // 2. number of actual samples to be played BCount = count/ACfg.skip; // 3. set the period for the given bitrate PR2 = FPB / bitrate-1; // 4. enable the interrupt state machine mT2ClearIntFlag(); // clear interrupt flag mT2IntEnable( 1); // enable TMR2 interrupt } // startAudio void haltAudio( void) { // stops playback state machine mT2IntEnable( 0); } // halt audio void __ISR( _TIMER_2_VECTOR, ipl4 ) T2Interrupt( void) { // 0. allow interrupt nesting asm( "ei"); // 1. load the new samples for the next cycle OC1RS = 30+(*BPtr ^ ACfg.fix); if ( ACfg.stereo) OC2RS = 30 + (*(BPtr + ACfg.size) ^ ACfg.fix); else // mono OC2RS = OC1RS; // 2. skip samples to reduce the bitrate BPtr += ACfg.skip; // 3. check if buffer emptied if ( --BCount == 0) { // 3.1 swap buffers CurBuf = 1- CurBuf; // 3.2. place pointer on first sample BPtr = &ABuffer[ CurBuf][ACfg.size-1]; // 3.3 restart counter BCount = B_SIZE/ACfg.skip; // 3.4 flag a new buffer needs to be filled AEmptyFlag = 1; } // 4. clear interrupt flag and exit mT2ClearIntFlag(); } // T2Interrupt /*------------------------------------------------------------- ** Wave ** ** Wave File Player ** Uses 2 x 8 bit PWM channels at 44kHz ** ** revision: ** 05/02/03 v1.00 LDJ PIC18 ** 07/08/06 v2.00 LDJ porting to PIC24/C30 ** 07/14/07 v3.00 LDJ PIC32 porting */ int playWAV( char *name) { WAVE wav; MFILE *f; unsigned int lc, r; int wi, pos, rate, period, last; char s[16]; // 1. open the file if ( (f = fopenM( name, "r")) == NULL) { // failed to open return FALSE; } // 2. verify it is a RIFF formatted file if ( ReadL( f->buffer, 0) != RIFF_DWORD) { fcloseM( f); return FALSE; } // 3. look for the WAVE chunk signature if ( (ReadL( f->buffer, 8)) != WAVE_DWORD) { fcloseM( f); return FALSE; } // 4. look for the chunk containing the wave format data if ( ReadL( f->buffer, 12) != FMT_DWORD) { fcloseM( f); return FALSE; } wav.channels = ReadW( f->buffer, 22); wav.bitpsample = ReadW( f->buffer, 34); wav.srate = ReadL( f->buffer, 24); wav.bps = ReadL( f->buffer, 28); wav.bpsample = ReadW( f->buffer, 32); // 5. search for the data chunk wi = 20 + ReadW( f->buffer, 16); while ( wi < 512) { if (ReadL( f->buffer, wi) == DATA_DWORD) break; wi += 8 + ReadW( f->buffer, wi+4); } if ( wi >= 512) // could not find in current sector { fcloseM( f); return FALSE; } // 6. find the data size (actual wave content) wav.dlength = ReadL( f->buffer, wi+4); // 7. if sample rate too high, skip rate = wav.bps / wav.bpsample; // rate = samples per second ACfg.skip = wav.bpsample; // skip to reduce bandwith while ( rate > 48000) { rate >>= 1; // divide sample rate by two ACfg.skip <<= 1; // multiply skip by two } // 8. check if sample rate too low period = (FPB/rate)-1; if ( period > ( 65536L)) // max timer period 16 bit { // period too long fcloseM( f); return FALSE; } // 9. init the Audio state machine CurBuf = 0; pos = wi+8; // data begin ACfg.stereo = (wav.channels == 2); ACfg.size = 1; // #bytes per channel ACfg.fix = 0; // sign fix / 16 bit file if ( wav.bitpsample == 16) { // if 16-bit pos++; // add 1 to get the MSB ACfg.size = 2; // two bytes per sample ACfg.fix = 0x80; // fix the sign } // 10 # of bytes composing the wav data chunk lc = wav.dlength; // 11. pre-load both buffer r = freadM( ABuffer[0], B_SIZE*2, f); lc -= r; AEmptyFlag = FALSE; // 12. configure Player state machine and start initAudio(); startAudio( rate, pos, r-pos); // 13. keep feeding the buffers in the playing loop // as long as entire buffers can be filled while (lc > 0) { // 13.1 check user input to stop playback if ( readKEY()) // if any button pressed { lc = 0; // playback completed break; } // 13.2 check if a buffer needs a refill if ( AEmptyFlag) { r = freadM( ABuffer[1-CurBuf], B_SIZE, f); lc-= r; // decrement byte count AEmptyFlag = FALSE; // refilled // <> putsLCD("\n"); // on the second line sprintf( s, "%dKB", (wav.dlength-lc)/1024); putsLCD( s); // byte count } } // while wav data available // 14. pad the rest of the buffer last = ABuffer[1-CurBuf][r-1]; while( r