Martin Yeh / Mbed 2 deprecated Karaoke

Dependencies:   SDFileSystem mbed-rtos mbed

Files at this revision

API Documentation at this revision

Comitter:
myeh
Date:
Mon Dec 07 20:36:22 2015 +0000
Commit message:
This is the code for a karaoke machine

Changed in this revision

SDFileSystem.lib Show annotated file Show diff for this revision Revisions of this file
main.cpp Show annotated file Show diff for this revision Revisions of this file
mbed-rtos.lib Show annotated file Show diff for this revision Revisions of this file
mbed.bld Show annotated file Show diff for this revision Revisions of this file
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/SDFileSystem.lib	Mon Dec 07 20:36:22 2015 +0000
@@ -0,0 +1,1 @@
+http://developer.mbed.org/users/neilt6/code/SDFileSystem/#84b2958bbcae
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/main.cpp	Mon Dec 07 20:36:22 2015 +0000
@@ -0,0 +1,290 @@
+#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;
+  }
+}
\ No newline at end of file
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/mbed-rtos.lib	Mon Dec 07 20:36:22 2015 +0000
@@ -0,0 +1,1 @@
+http://developer.mbed.org/users/mbed_official/code/mbed-rtos/#c825593ece39
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/mbed.bld	Mon Dec 07 20:36:22 2015 +0000
@@ -0,0 +1,1 @@
+http://mbed.org/users/mbed_official/code/mbed/builds/9296ab0bfc11
\ No newline at end of file