hw

Dependents:   wave_player_mp3 4180_lab2_part9 4180_Lab3_rtos_basic PianoTiles

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API Documentation at this revision

Comitter:
zchen311
Date:
Sun Jul 14 20:29:36 2013 +0000
Commit message:
done

Changed in this revision

wave_player.cpp Show annotated file Show diff for this revision Revisions of this file
wave_player.h Show annotated file Show diff for this revision Revisions of this file
diff -r 000000000000 -r 353c78110e44 wave_player.cpp
--- /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;
+    }
+    
+
+}
+
diff -r 000000000000 -r 353c78110e44 wave_player.h
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/wave_player.h	Sun Jul 14 20:29:36 2013 +0000
@@ -0,0 +1,74 @@
+#include <mbed.h>
+
+typedef struct uFMT_STRUCT {
+  short comp_code;
+  short num_channels;
+  unsigned sample_rate;
+  unsigned avg_Bps;
+  short block_align;
+  short sig_bps;
+} FMT_STRUCT;
+
+
+/** wave file player class.
+ *
+ * Example:
+ * @code
+ * #include <mbed.h>
+ * #include <wave_player.h>
+ *
+ * AnalogOut DACout(p18);
+ * wave_player waver(&DACout);
+ *
+ * int main() {
+ *  FILE *wave_file;
+ *  
+ *  printf("\n\n\nHello, wave world!\n");
+ *  wave_file=fopen("/sd/44_8_st.wav","r");
+ *  waver.play(wave_file);
+ *  fclose(wave_file); 
+ * }
+ * @endcode
+ */
+class wave_player {
+
+public:
+int* getVolumeMod();
+/** Create a wave player using a pointer to the given AnalogOut object.
+ *
+ * @param _dac pointer to an AnalogOut object to which the samples are sent.
+ */
+wave_player(AnalogOut *_dac);
+
+/** the player function.
+ *
+ * @param wavefile  A pointer to an opened wave file
+ */
+void play(FILE *wavefile,bool *ptr);
+
+/** 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 sample values placed into the DAC FIFO, and the sample values
+ * removed from the DAC FIFO by the ISR.  The sample 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
+ */
+void set_verbosity(int v);
+
+private:
+int  volumeMod;
+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;
+};
+
+