Community Projects

freeaudio.bmx
?MacOS
Import "-framework AudioUnit"
Import "-framework AudioToolbox"
?
?Linux
Import "-lasound"
?

freeaudio.cpp
#ifdef __linux

#include <stdio.h>
#include <stdlib.h>
#include <alsa/asoundlib.h>
#include <pthread.h>

void *mixerthread(void *dev);

#define LINUXFRAG 2048

struct linuxaudio:audiodevice
{
	pthread_t	audiothread;
	int			threadid;
	snd_pcm_t 	*audio_fd;
	int 		playback;
	int			playing;
	int			fragsize,fragcount;
	int			buffersize;	//in bytes
	short		*buffer;
	
	int reset()
	{
		playing=0;
		mix=new mixer(LINUXFRAG);
		mix->freq=44100;
		mix->channels=2;
		buffer=new short[LINUXFRAG];
		buffersize=LINUXFRAG*2;
		pthread_attr_t	attr;
		pthread_attr_init(&attr);
		threadid=pthread_create(&audiothread,&attr,mixerthread,(void*)this);	
		return 0;
	}
	
	int close()
	{	
		int		timeout;
		playing=0;
		timeout=5;
		while (timeout-- && playback) sleep(1);
		return 0;
	}
};

audiodevice *OpenSystemAudio()
{
	return new linuxaudio();
}

void *mixerthread(void *v)
{
	sched_param		sched;	
	linuxaudio 		*dev;
	int			policy;
	unsigned int val;
	snd_pcm_uframes_t periodsize;
	snd_pcm_hw_params_t *hwparams;
	snd_pcm_hw_params_alloca(&hwparams);
	int output_rate;
	int channels;
	int fragment_size;
	int fragment_count;
	fragment_size=LINUXFRAG;  //overall buffer size
	fragment_count=2; //2 - 16 fragment count - 2 minimum, the lower it is potentially the lower the latency

	pthread_getschedparam(pthread_self(),&policy,&sched);
	sched.sched_priority++;//policy=SCHED_RR;
	pthread_setschedparam(pthread_self(),policy,&sched);

	//open audio device	
	dev=(linuxaudio*)v;	
	dev->playback=snd_pcm_open(&dev->audio_fd, strdup("default"), SND_PCM_STREAM_PLAYBACK, 0); //uses default audio device..
	if (dev->playback==-1)
	{
		printf("linuxaudio failed\n");
		dev->playback=0;
		return 0;
	}		
	//configure device
	if (snd_pcm_hw_params_any(dev->audio_fd, hwparams) < 0) {
		printf("linuxaudio failed at params any\n");
		return 0;
	}	
	if (snd_pcm_hw_params_set_access(dev->audio_fd, hwparams,SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
		printf("linuxaudio failed at set access\n");
		return 0;
	}	
	
	if (snd_pcm_hw_params_set_format(dev->audio_fd, hwparams,SND_PCM_FORMAT_S16_LE) < 0) {
		printf("linuxaudio failed at set format\n");
		return 0;
	}
	val = 44100;
	if (snd_pcm_hw_params_set_rate_near(dev->audio_fd, hwparams,&val, 0) < 0) {
		/* Try 48KHZ too */
		printf("linuxaudio - %d HZ not available, trying 48000HZ\n", output_rate);
		val = 48000;
		if (snd_pcm_hw_params_set_rate_near(dev->audio_fd, hwparams,&val, 0) < 0) {
			printf("linuxaudio failed at setting output rate (%d)\n", output_rate);
			return 0;
		}
		dev->mix->freq=val;		
	}
	channels=2;
	if (snd_pcm_hw_params_set_channels(dev->audio_fd, hwparams, channels) < 0) {
		printf("linuxaudio failed at set channels (%d)\n", channels);
		return 0;
	}
	periodsize = (fragment_size) / 4; // bytes -> frames for 16-bit,stereo - should be a minimum of 512
	if (snd_pcm_hw_params_set_period_size_near(dev->audio_fd, hwparams,&periodsize, 0) < 0) {
		printf("linuxaudio failed at set period size (%d)\n", (int)periodsize);			
		return 0;
	}
	val = fragment_count;
	if (snd_pcm_hw_params_set_periods_near(dev->audio_fd, hwparams,&val, 0) < 0) {
		printf("linuxaudio failed at set periods (%d)\n", val);			
		//should attempt a one by one period increase up to 16?
		return 0;
	}
	if (snd_pcm_hw_params(dev->audio_fd, hwparams) < 0) {
		printf("linuxaudio failed at installing hw params\n");
		return 0;
	}
	//loop while playing sound
	dev->playing=1;
	while (dev->playing)
	{
		dev->mix->mix16(dev->buffer);
		if ((snd_pcm_writei (dev->audio_fd, dev->buffer,LINUXFRAG/2)) < 0) {	//Half buffer for two channels?
			printf ("linuxaudio warning: buffer underrun occurred\n");
			if (snd_pcm_prepare(dev->audio_fd) < 0) {
				printf ("linuxaudio failed at preparing pcm\n");
				dev->playing=0; //die gracefully
			}
		}	
	}
	snd_pcm_drop(dev->audio_fd);
	snd_pcm_close (dev->audio_fd);
	dev->audio_fd=0;
	dev->playback=0;
	return 0;
}

#endif

freeaudio.bmx
?MacOS
Import "-framework AudioUnit"
Import "-framework AudioToolbox"
?
?Linux
Import "-lasound"
?

#ifdef __linux

#include <sys/ioctl.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/soundcard.h>
#include <pthread.h>
#include <alsa/asoundlib.h>

void *mixerthread(void *dev);

#define LINUXFRAG 2048

struct linuxaudio:audiodevice
{
	pthread_t	audiothread;
	int			threadid;
	int			audio_fd;
	snd_pcm_t 	*alsaaudio_fd;
	int			playing;
	int			fragsize,fragcount;
	int			buffersize;	//in bytes
	short		*buffer;
	
	int reset()
	{
		playing=0;
		mix=new mixer(LINUXFRAG);
		mix->freq=44100;
		mix->channels=2;
		buffer=new short[LINUXFRAG];
		buffersize=LINUXFRAG*2;
		pthread_attr_t	attr;
		pthread_attr_init(&attr);
		threadid=pthread_create(&audiothread,&attr,mixerthread,(void*)this);	
		return 0;
	}
	
	int close()
	{	
		int		timeout;
		playing=0;
		timeout=5;
		while (timeout-- && audio_fd) sleep(1);
		return 0;
	}
};

audiodevice *OpenSystemAudio()
{
	return new linuxaudio();
}

void initAudioALSA(linuxaudio *dev)
{
	if (snd_pcm_open(&dev->alsaaudio_fd, strdup("default"), SND_PCM_STREAM_PLAYBACK, 0)==-1) 	//uses default audio device..
	{
	dev->alsaaudio_fd=0;
	}
}

void initAudioOSS(linuxaudio *dev)
{
	dev->audio_fd=open("/dev/dsp",O_WRONLY,0);		//|O_NONBLOCK
	if (dev->audio_fd==-1)
	{
	dev->audio_fd=0;
	}
}

void *playAudioOSS(linuxaudio *dev)
{
	int				data,res;

 		data=0x03000c;		//2 fragments of 4096 samples
		res=ioctl(dev->audio_fd,SNDCTL_DSP_SETFRAGMENT,&data);		
		data=AFMT_S16_LE;
		res=ioctl(dev->audio_fd,SNDCTL_DSP_SETFMT,&data);
		data=2;
		res=ioctl(dev->audio_fd,SNDCTL_DSP_CHANNELS,&data);		
		data=44100;
		res=ioctl(dev->audio_fd,SNDCTL_DSP_SPEED,&data);
		
		dev->playing=1;
		while (dev->playing)
		{
			dev->mix->mix16(dev->buffer);
			int n=write(dev->audio_fd,dev->buffer,dev->buffersize);
		}
		
		close(dev->audio_fd);
}

void *playAudioALSA(linuxaudio *dev)
{
	unsigned int val;
	snd_pcm_uframes_t periodsize;
	snd_pcm_hw_params_t *hwparams;
	snd_pcm_hw_params_alloca(&hwparams);
	int output_rate;
	int channels;
	int fragment_size;
	int fragment_count;
	fragment_size=LINUXFRAG;  //overall buffer size
	fragment_count=2; //2 - 16 fragment count - 2 minimum, the lower it is potentially the lower the latency

	//configure device
		if (snd_pcm_hw_params_any(dev->alsaaudio_fd, hwparams) < 0) {
			//printf("linuxaudio failed at params any\n");
			return 0;
		}	
		if (snd_pcm_hw_params_set_access(dev->alsaaudio_fd, hwparams,SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
			//printf("linuxaudio failed at set access\n");
			return 0;
		}	
		
		if (snd_pcm_hw_params_set_format(dev->alsaaudio_fd, hwparams,SND_PCM_FORMAT_S16_LE) < 0) {
			//printf("linuxaudio failed at set format\n");
			return 0;
		}
		val = 44100;
		if (snd_pcm_hw_params_set_rate_near(dev->alsaaudio_fd, hwparams,&val, 0) < 0) {
			/* Try 48KHZ too */
			//printf("linuxaudio - %d HZ not available, trying 48000HZ\n", output_rate);
			val = 48000;
			if (snd_pcm_hw_params_set_rate_near(dev->alsaaudio_fd, hwparams,&val, 0) < 0) {
				//printf("linuxaudio failed at setting output rate (%d)\n", output_rate);
				return 0;
			}
			dev->mix->freq=val;		
		}
		channels=2;
		if (snd_pcm_hw_params_set_channels(dev->alsaaudio_fd, hwparams, channels) < 0) {
			//printf("linuxaudio failed at set channels (%d)\n", channels);
			return 0;
		}
		periodsize = (fragment_size) / 4; // bytes -> frames for 16-bit,stereo - should be a minimum of 512
		if (snd_pcm_hw_params_set_period_size_near(dev->alsaaudio_fd, hwparams,&periodsize, 0) < 0) {
			//printf("linuxaudio failed at set period size (%d)\n", (int)periodsize);			
			return 0;
		}
		val = fragment_count;
		if (snd_pcm_hw_params_set_periods_near(dev->alsaaudio_fd, hwparams,&val, 0) < 0) {
			//printf("linuxaudio failed at set periods (%d)\n", val);			
			//should attempt a one by one period increase up to 16?
			return 0;
		}
		if (snd_pcm_hw_params(dev->alsaaudio_fd, hwparams) < 0) {
			//printf("linuxaudio failed at installing hw params\n");
			return 0;
		}
		//loop while playing sound
		dev->playing=1;
		while (dev->playing)
		{
			dev->mix->mix16(dev->buffer);
			if ((snd_pcm_writei (dev->alsaaudio_fd, dev->buffer,LINUXFRAG/2)) < 0) {	//Half buffer for two channels?
				//printf ("linuxaudio warning: buffer underrun occurred\n");
				if (snd_pcm_prepare(dev->alsaaudio_fd) < 0) {
					//printf ("linuxaudio failed at preparing pcm\n");
					dev->playing=0; //die gracefully
				}
			}	
		}
		snd_pcm_drop(dev->alsaaudio_fd);
		snd_pcm_close (dev->alsaaudio_fd);
		dev->alsaaudio_fd=0; 
}

void *mixerthread(void *v)
{
	sched_param		sched;	
	linuxaudio 		*dev;
	int				policy;
	bool		usealsa;
	usealsa=1;
	pthread_getschedparam(pthread_self(),&policy,&sched);
	sched.sched_priority++;//policy=SCHED_RR;
	pthread_setschedparam(pthread_self(),policy,&sched);
	
	dev=(linuxaudio*)v;
	//try alsa
	initAudioALSA(dev);
	if (dev->alsaaudio_fd==0) 	//uses default audio device..
	{
		//try OSS
		initAudioOSS(dev);
		if (dev->audio_fd==0)
		{
			printf("linuxaudio failed\n");
			return 0;
		}
		usealsa=0;				
	}		
	if (usealsa=1) {
	playAudioALSA(dev);
	}
	else {
	playAudioOSS(dev);
	}
	dev->audio_fd=0;
	return 0;
}

#endif