I think the original code was named "alsa wave".
If I could find out the author I would credit it. I just tweaked it to use the right devices (and a seggie)
It's public domain stuff anyways. just basic setup and header stuff.
Code:
// A simple C example to play a mono or stereo, 16-bit 44KHz
// WAVE file using ALSA. This goes directly to the first
// audio card (ie, its first set of audio out jacks). It
// uses the snd_pcm_writei() mode of outputting waveform data,
// blocking.
//
// Compile as so to create "salsa":
// gcc -lasound -B /usr/include -o salsa salsa.c
//
// Run it from a terminal, specifying the name of a WAVE file to play:
// ./salsa wave.wav
// Tweaked by Twobob to work on the kindle
//#include <stdio.h> or tcclib perhaps (however its not required)
#include <stdlib.h>
#include <unistd.h>
// Include the ALSA .H file that defines ALSA functions/data
#include <alsa/asoundlib.h>
#pragma pack (1)
/////////////////////// WAVE File Stuff /////////////////////
// An IFF file header looks like this
typedef struct _FILE_head
{
unsigned char ID[4];
// could be {'R', 'I', 'F', 'F'} or {'F', 'O', 'R', 'M'}
unsigned int Length;
// Length of subsequent file (including remainder of header).
//This is in Intel reverse byte order if RIFF, Motorola format if FORM.
unsigned char Type[4];
// {'W', 'A', 'V', 'E'} or {'A', 'I', 'F', 'F'}
} FILE_head;
// An IFF chunk header looks like this
typedef struct _CHUNK_head
{
unsigned char ID[4]; // 4 ascii chars that is the chunk ID
unsigned int Length; // Length of subsequent data within this chunk. This is in Intel reverse byte
// order if RIFF, Motorola format if FORM. Note: this doesn't include any
// extra byte needed to pad the chunk out to an even size.
} CHUNK_head;
// WAVE fmt chunk
typedef struct _FORMAT {
short wFormatTag;
unsigned short wChannels;
unsigned int dwSamplesPerSec;
unsigned int dwAvgBytesPerSec;
unsigned short wBlockAlign;
unsigned short wBitsPerSample;
// Note: there may be additional fields here, depending upon wFormatTag
} FORMAT;
#pragma pack()
// Size of the audio card hardware buffer. Here we want it
// set to 1024 16-bit sample points. This is relatively
// small in order to minimize latency. If you have trouble
// with underruns, you may need to increase this, and PERIODSIZE
// (trading off lower latency for more stability)
#define BUFFERSIZE (2*1024)
// How many sample points the ALSA card plays before it calls
// our callback to fill some more of the audio card's hardware
// buffer. Here we want ALSA to call our callback after every
// 64 sample points have been played
#define PERIODSIZE (2*64)
// Handle to ALSA (audio card's) playback port
snd_pcm_t *PlaybackHandle;
// Handle to our callback thread
snd_async_handler_t *CallbackHandle;
// Points to loaded WAVE file's data
unsigned char *WavePtr;
// Size (in frames) of loaded WAVE file's data
snd_pcm_uframes_t WaveSize;
// Sample rate
unsigned short WaveRate;
// Bit resolution
unsigned char WaveBits;
// Number of channels in the wave file
unsigned char WaveChannels;
// The name of the ALSA port we output to. In this case, we're
// directly writing to hardware card 0,0 (ie, first set of audio
// outputs on the first audio card)
static const char SoundCardPortName[] = "hw";
// For WAVE file loading
static const unsigned char Riff[4] = { 'R', 'I', 'F', 'F' };
static const unsigned char Wave[4] = { 'W', 'A', 'V', 'E' };
static const unsigned char Fmt[4] = { 'f', 'm', 't', ' ' };
static const unsigned char Data[4] = { 'd', 'a', 't', 'a' };
/********************** compareID() *********************
* Compares the passed ID str (ie, a ptr to 4 Ascii
* bytes) with the ID at the passed ptr. Returns TRUE if
* a match, FALSE if not.
*/
static unsigned char compareID(const unsigned char * id, unsigned char * ptr)
{
register unsigned char i = 4;
while (i--)
{
if ( *(id)++ != *(ptr)++ ) return(0);
}
return(1);
}
/********************** waveLoad() *********************
* Loads a WAVE file.
*
* fn = Filename to load.
*
* RETURNS: 0 if success, non-zero if not.
*
* NOTE: Sets the global "WavePtr" to an allocated buffer
* containing the wave data, and "WaveSize" to the size
* in sample points.
*/
static unsigned char waveLoad(const char *fn)
{
const char *message;
FILE_head head;
register int inHandle;
if ((inHandle = open(fn, O_RDONLY)) == -1)
message = "didn't open";
// Read in IFF File header
else
{
if (read(inHandle, &head, sizeof(FILE_head)) == sizeof(FILE_head))
{
// Is it a RIFF and WAVE?
if (!compareID(&Riff[0], &head.ID[0]) || !compareID(&Wave[0], &head.Type[0]))
{
message = "is not a WAVE file";
goto bad;
}
// Read in next chunk header
while (read(inHandle, &head, sizeof(CHUNK_head)) == sizeof(CHUNK_head))
{
// ============================ Is it a fmt chunk? ===============================
if (compareID(&Fmt[0], &head.ID[0]))
{
FORMAT format;
// Read in the remainder of chunk
if (read(inHandle, &format.wFormatTag, sizeof(FORMAT)) != sizeof(FORMAT)) break;
// Can't handle compressed WAVE files
if (format.wFormatTag != 1)
{
message = "compressed WAVE not supported";
goto bad;
}
WaveBits = (unsigned char)format.wBitsPerSample;
WaveRate = (unsigned short)format.dwSamplesPerSec;
WaveChannels = format.wChannels;
}
// ============================ Is it a data chunk? ===============================
else if (compareID(&Data[0], &head.ID[0]))
{
// Size of wave data is head.Length. Allocate a buffer and read in the wave data
if (!(WavePtr = (unsigned char *)malloc(head.Length)))
{
message = "won't fit in RAM";
goto bad;
}
if (read(inHandle, WavePtr, head.Length) != head.Length)
{
free(WavePtr);
break;
}
// Store size (in frames)
WaveSize = (head.Length * 8) / ((unsigned int)WaveBits * (unsigned int)WaveChannels);
close(inHandle);
return(0);
}
// ============================ Skip this chunk ===============================
else
{
if (head.Length & 1) ++head.Length; // If odd, round it up to account for pad byte
lseek(inHandle, head.Length, SEEK_CUR);
}
}
}
message = "is a bad WAVE file";
bad: close(inHandle);
}
printf("%s %s\n", fn, message);
return(1);
}
/********************** play_audio() **********************
* Plays the loaded waveform.
*
* NOTE: ALSA sound card's handle must be in the global
* "PlaybackHandle". A pointer to the wave data must be in
* the global "WavePtr", and its size of "WaveSize".
*/
static void play_audio(void)
{
register snd_pcm_uframes_t count, frames;
// Output the wave data
count = 0;
do
{
frames = snd_pcm_writei(PlaybackHandle, WavePtr + count, WaveSize - count);
// If an error, try to recover from it
if (frames < 0)
frames = snd_pcm_recover(PlaybackHandle, frames, 0);
if (frames < 0)
{
printf("Error playing wave: %s\n", snd_strerror(frames));
break;
}
// Update our pointer
count += frames;
} while (count < WaveSize);
// Wait for playback to completely finish
if (count == WaveSize)
snd_pcm_drain(PlaybackHandle);
}
/*********************** free_wave_data() *********************
* Frees any wave data we loaded.
*
* NOTE: A pointer to the wave data be in the global
* "WavePtr".
*/
static void free_wave_data(void)
{
if (WavePtr) free(WavePtr);
WavePtr = 0;
}
int main(int argc, char **argv)
{
// No wave data loaded yet
WavePtr = 0;
if (argc < 2)
printf("You must supply the name of a 16-bit mono WAVE file to play\n");
// Load the wave file
else if (!waveLoad(argv[1]))
{
register int err;
// Open audio card we wish to use for playback
if ((err = snd_pcm_open(&PlaybackHandle, &SoundCardPortName[0], SND_PCM_STREAM_PLAYBACK, 0)) < 0)
printf("Can't open audio %s: %s\n", &SoundCardPortName[0], snd_strerror(err));
else
{
switch (WaveBits)
{
case 8:
err = SND_PCM_FORMAT_U8;
break;
case 16:
err = SND_PCM_FORMAT_S16;
break;
case 24:
err = SND_PCM_FORMAT_S24;
break;
case 32:
err = SND_PCM_FORMAT_S32;
break;
}
// Set the audio card's hardware parameters (sample rate, bit resolution, etc)
if ((err = snd_pcm_set_params(PlaybackHandle, err, SND_PCM_ACCESS_RW_INTERLEAVED, WaveChannels, WaveRate, 1, 500000)) < 0)
printf("Can't set sound parameters: %s\n", snd_strerror(err));
// Play the waveform
else
play_audio();
// Close sound card
snd_pcm_close(PlaybackHandle);
}
}
// Free the WAVE data
free_wave_data();
return(0);
}
so
[root@kindle us]#
aplay -D plughw:0,0 mono.wav
Playing WAVE 'mono.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono
and
bash-4.1$
./salsa ~/mono.wav provide audio functionality, via alsa, except this we can control the guts of. (the port used to play through is "hw")
Many thanks. Heartening however to get something, functional, in c, that emulates the current access to sound (aplay) in a programmable way.
Your thoughts so far. thanks