Rory Hand
does not work yet. Sound starts but then stops after a few seconds, whole thing hangs. Published so as I can import through mbed CLI.
Dependencies: mbed sinelookup SDFileSystem_Copy_of_mbed_version I2S
Revision 5:5b120a2d9249, committed 2018-10-23
- Comitter:
- roryhand
- Date:
- Tue Oct 23 19:02:45 2018 +0000
- Parent:
- 4:1a9482a07ee2
- Child:
- 6:fd6ae534d120
- Commit message:
- chsnged to work with 8khz file (does not work yet)
Changed in this revision
--- a/I2S.lib Sat Oct 06 11:51:23 2018 +0000 +++ b/I2S.lib Tue Oct 23 19:02:45 2018 +0000 @@ -1,1 +1,1 @@ -http://os.mbed.com/users/p07gbar/code/I2S/#455d5826751b +http://os.mbed.com/users/p07gbar/code/I2S/#9dd9537d8ab6
--- a/WOLFSON_config.h Sat Oct 06 11:51:23 2018 +0000 +++ b/WOLFSON_config.h Tue Oct 23 19:02:45 2018 +0000 @@ -27,11 +27,15 @@ const int _WM8731_RIGHT_LINEIN = 0X0180; // MIC SETTINGS: ENABLE MUTE, ENABLE SIMULTANEOUS LOAD TO LEFT AND RIGHT CHANNELS const int _WM8731_LEFT_HP = 0X00F9; // HEADPHONE SETTINGS : 0DB const int _WM8731_RIGHT_HP = 0X00F9; // HEADPHONE SETTINGS : 0DB +//const int _WM8731_LEFT_HP = 0X00E1; // HEADPHONE SETTINGS : -6?DB +//const int _WM8731_RIGHT_HP = 0X00E1; // HEADPHONE SETTINGS : -6?DB const int _WM8731_ANALOGAUDIO = 0XD0; // DAC SELECT -const int _WM8731_DIGITALAUDIO = 0X00; +const int _WM8731_DIGITALAUDIO = 0X00; //11011 const int _WM8731_POWER = 0X00; // DISABLE POWER DOWN const int _WM8731_DAIF = 0X02; // ENABLE SLAVE MODE, 16BIT DATA AND I2S DATA -const int _WM8731_SAMPLING = 0X00; // 48000HZ,12.288MHz oscillator. +//const int _WM8731_SAMPLING = 0X00; // 48000HZ,12.288MHz oscillator. +//const int _WM8731_SAMPLING = 0X2E; // 8000HZ,12.288MHz oscillator. +const int _WM8731_SAMPLING = 0X18; // 8000HZ,12.288MHz oscillator. const int _WM8731_ACTIVATE = 0X01; // MODULE IS ON const int _WM8731_DEACTIVATE = 0X00; // MODULE IS OFF const int _WM8731_RESET = 0X00; // RESET VALUE
--- a/wolfson_3_wav.cpp Sat Oct 06 11:51:23 2018 +0000
+++ b/wolfson_3_wav.cpp Tue Oct 23 19:02:45 2018 +0000
@@ -44,7 +44,7 @@
int buffer[1];
int AudioFormat, NumChannels, SampleRate, BitsPerSample ;
char *slice_buf;
-int *data_sptr;
+short *data_sptr;
unsigned char *data_bptr;
int *data_wptr;
unsigned channel;
@@ -52,7 +52,7 @@
long slice, num_slices;
int verbosity = 0;
//long long slice_value;
-int slice_value;
+int slice_value[1];
FILE *my_wav;
FMT_STRUCT wav_format;
DATA_STRUCT wav_data;
@@ -69,8 +69,10 @@
buffer[0] = (*data_sptr)>>1;
myled = 1;
int bufflen = sizeof(buffer);
+ if(verbosity){
printf("buffer data: %d \n\r",buffer);
printf("data_sptr data: %d \n\r",data_sptr);
+ }
//t.start();
//buffer[0] = (int*)data_sptr;
//buffer[0] = newvar;
@@ -261,7 +263,10 @@
data[1] = 0x00;
FILE *my_wav;
//my_wav = fopen("/sd/mydir/gp40.wav", "rb");//i think that it is "w" for write, "r" for read.
- my_wav = fopen("/sd/mydir/gp40_48K.wav", "rb");//i think that it is "w" for write, "r" for read.
+ //my_wav = fopen("/sd/mydir/gp40_48K.wav", "rb");//i think that it is "w" for write, "r" for read.
+ //my_wav = fopen("/sd/mydir/gp40_48K_quieter.wav", "rb");//i think that it is "w" for write, "r" for read.
+ my_wav = fopen("/sd/mydir/gp40_8K.wav","rb");
+ //my_wav = fopen("/sd/mydir/emd_645.wav", "rb");
fseek(my_wav, 20, SEEK_SET); // set pointer to byte 20
fread(&AudioFormat, 2, 1, my_wav); // check file is PCM
if (AudioFormat==0x01) {
@@ -331,6 +336,7 @@
printf("sizeof(wav_format): %d\n\r",sizeof(wav_format));
printf("do we egt to this point:\n\r");
//sampletick.attach(&isr,1/96000);
+ //sampletick.attach(&isr,1000000/48000);
//while(!feof(my_wav)){
//while(1){
//if (wav_data.subchunk2_size > sizeof(wav_format)){
@@ -354,43 +360,53 @@
printf("Oops -- not enough slices in the wave file\n");
exit(1);
}
- data_sptr=(int *)slice_buf; // 16 bit samples
+ 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;
+ slice_value[1]=0;
for (channel=0;channel<wav_format.num_channels;channel++) {
switch (wav_format.sig_bps) {
case 16:
+ int mybuffer[1];
+ mybuffer[0] = data_sptr[channel];// - 32768;
+ i2s.write(data_sptr,2);
+ wait_us(0.1);
if (verbosity)
- printf("16 bit channel %d data=%d ",channel,data_sptr[channel]);
- slice_value+=data_sptr[channel];
+ printf("16 bit channel %d data=%d\n\r ",channel,data_sptr[channel]);
+ slice_value[1]+=data_sptr[channel];
break;
case 32:
if (verbosity)
printf("32 bit channel %d data=%d ",channel,data_wptr[channel]);
- slice_value+=data_wptr[channel];
+ slice_value[1]+=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];
+ slice_value[1]+=data_bptr[channel];
break;
}
}
- slice_value/=wav_format.num_channels;
+ slice_value[1]/=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;
+ case 8: slice_value[1]<<=8;
break;
- case 16: slice_value+=32768;
+ case 16: slice_value[1]+=32768;
break;
- case 32: slice_value>>=16;
- slice_value+=32768;
+ case 32: slice_value[1]>>=16;
+ slice_value[1]+=32768;
break;
}
- i2s.write(data_sptr,2);
+ if (verbosity)
+ {
+ printf("data_sptr data: %d \n\r",data_sptr);
+ printf("slice_value data: %d \n\r",slice_value);
+ }
+ int mynewvar[1];
+ //i2s.write(data_sptr,2);
//t.stop();
//printf("The time taken was %f seconds\n", t.read());
}
@@ -405,7 +421,7 @@
- sampletick.detach();
+ //sampletick.detach();
i2s.stop();
fclose(my_wav);