ze chen
hw
Dependents: wave_player_mp3 4180_lab2_part9 4180_Lab3_rtos_basic PianoTiles
Diff: wave_player.cpp
- Revision:
- 0:353c78110e44
--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/wave_player.cpp Sun Jul 14 20:29:36 2013 +0000
@@ -0,0 +1,229 @@
+//-----------------------------------------------------------------------------
+// a sample mbed library to play back wave files.
+//
+// explanation of wave file format.
+// https://ccrma.stanford.edu/courses/422/projects/WaveFormat/
+
+// if VERBOSE is uncommented then the wave player will enter a verbose
+// mode that displays all data values as it reads them from the file
+// and writes them to the DAC. Very slow and unusable output on the DAC,
+// but useful for debugging wave files that don't work.
+//#define VERBOSE
+
+
+#include <mbed.h>
+#include <stdio.h>
+#include <wave_player.h>
+
+
+//-----------------------------------------------------------------------------
+// constructor -- accepts an mbed pin to use for AnalogOut. Only p18 will work
+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;
+ volumeMod=0;
+}
+int * wave_player:: getVolumeMod()
+{
+
+ return & volumeMod;
+
+ }
+
+//-----------------------------------------------------------------------------
+// if verbosity is set then wave player enters a mode where the wave file
+// is decoded and displayed to the screen, including sample 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,bool *ptr)
+{
+ 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 samples/sec\n",wav_format.sample_rate);
+ printf(" %d bytes/sec\n",wav_format.avg_Bps);
+ printf(" block align %d\n",wav_format.block_align);
+ printf(" %d bits per sample\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.sample_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 sample interval=%d\n",(unsigned)(1000000.0/wav_format.sample_rate));
+ printf(" programmed interrupt tick interval=%d\n",samp_int);
+ }
+
+// starting up ticker to write samples 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 sample 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 sample 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 sample 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) {
+ if(*ptr==0)
+ break;
+ 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 samples
+ data_bptr=(unsigned char *)slice_buf; // 8 bit samples
+ data_wptr=(int *)slice_buf; // 32 bit samples
+ 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("sample %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);
+ }
+}
+
+
+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(DAC_fifo[DAC_rptr] * (16 - volumeMod)>>4);
+
+ DAC_rptr=(DAC_rptr+1) & 0xff;
+ }
+
+
+}
+