alsa.c

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/*
mediastreamer2 library - modular sound and video processing and streaming
Copyright (C) 2006  Simon MORLAT ([e-mail])

This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.

This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
GNU General Public License for more details.

You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
*/



#include <alsa/asoundlib.h>


#include "mediastreamer2/msfilter.h"
#include "mediastreamer2/mssndcard.h"

/*uncomment the following line if you have problems with an alsa driver
having sound quality trouble:*/
/*#define EPIPE_BUGFIX 1*/

static MSSndCard * alsa_card_new(int id);
static MSSndCard *alsa_card_duplicate(MSSndCard *obj);
static MSFilter * ms_alsa_read_new(const char *dev);
static MSFilter * ms_alsa_write_new(const char *dev);


struct _AlsaData{
	char *pcmdev;
	char *mixdev;
};

typedef struct _AlsaData AlsaData;


static int alsa_set_params(snd_pcm_t *pcm_handle, int rw, int bits, int stereo, int rate)
{
	snd_pcm_hw_params_t *hwparams=NULL;
	snd_pcm_sw_params_t *swparams=NULL;
	int dir;
	uint exact_uvalue;
	unsigned long exact_ulvalue;
	int channels;
	int periods=8;
	int periodsize=256;
	int err;
	int format;
	
	/* Allocate the snd_pcm_hw_params_t structure on the stack. */
	snd_pcm_hw_params_alloca(&hwparams);
	
	/* Init hwparams with full configuration space */
	if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0) {
		ms_warning("alsa_set_params: Cannot configure this PCM device.\n");
		return -1;
	}
	
	if (snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
		ms_warning("alsa_set_params: Error setting access.\n");
		return -1;
	}
	/* Set sample format */
	format=SND_PCM_FORMAT_S16;
	if (snd_pcm_hw_params_set_format(pcm_handle, hwparams, format) < 0) {
		ms_warning("alsa_set_params: Error setting format.\n");
		return -1;
	}
	/* Set number of channels */
	if (stereo) channels=2;
	else channels=1;
	if (snd_pcm_hw_params_set_channels(pcm_handle, hwparams, channels) < 0) {
		ms_warning("alsa_set_params: Error setting channels.\n");
		return -1;
	}
	/* Set sample rate. If the exact rate is not supported */
	/* by the hardware, use nearest possible rate.         */ 
	exact_uvalue=rate;
	dir=0;
	if ((err=snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_uvalue, &dir))<0){
		ms_warning("alsa_set_params: Error setting rate to %i:%s",rate,snd_strerror(err));
		return -1;
	}
	if (dir != 0) {
		ms_warning("alsa_set_params: The rate %d Hz is not supported by your hardware.\n "
		"==> Using %d Hz instead.\n", rate, exact_uvalue);
	}
	/* choose greater period size when rate is high */
	periodsize=periodsize*(rate/8000);	
	
	/* Set buffer size (in frames). The resulting latency is given by */
	/* latency = periodsize * periods / (rate * bytes_per_frame)     */
	/* set period size */
	exact_ulvalue=periodsize;
	dir=0;
	if (snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, &exact_ulvalue, &dir) < 0) {
		ms_warning("alsa_set_params: Error setting period size.\n");
		return -1;
	}
	if (dir != 0) {
		ms_warning("alsa_set_params: The period size %d is not supported by your hardware.\n "
		"==> Using %d instead.\n", periodsize, (int)exact_ulvalue);
	}
	periodsize=exact_ulvalue;
	/* Set number of periods. Periods used to be called fragments. */ 
	exact_uvalue=periods;
	dir=0;
	if (snd_pcm_hw_params_set_periods_near(pcm_handle, hwparams, &exact_uvalue, &dir) < 0) {
		ms_warning("alsa_set_params: Error setting periods.\n");
		return -1;
	}
	if (dir != 0) {
		ms_warning("alsa_set_params: The number of periods %d is not supported by your hardware.\n "
		"==> Using %d instead.\n", periods, exact_uvalue);
	}
	/* Apply HW parameter settings to */
	/* PCM device and prepare device  */
	if ((err=snd_pcm_hw_params(pcm_handle, hwparams)) < 0) {
		ms_warning("alsa_set_params: Error setting HW params:%s",snd_strerror(err));
		return -1;
	}
	/*prepare sw params */
	if (rw){
		snd_pcm_sw_params_alloca(&swparams);
		snd_pcm_sw_params_current(pcm_handle, swparams);
		if ((err=snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams,periodsize*2 ))<0){
			ms_warning("alsa_set_params: Error setting start threshold:%s",snd_strerror(err));
			return -1;
		}
		if ((err=snd_pcm_sw_params(pcm_handle, swparams))<0){
			ms_warning("alsa_set_params: Error setting SW params:%s",snd_strerror(err));
			return -1;
		}
	}
	return 0;	
}

#ifdef EPIPE_BUGFIX
static void alsa_fill_w (snd_pcm_t *pcm_handle)
{
	snd_pcm_hw_params_t *hwparams=NULL;
	int channels;
        snd_pcm_uframes_t buffer_size;
	int buffer_size_bytes;
	void *buffer;

	/* Allocate the snd_pcm_hw_params_t structure on the stack. */
	snd_pcm_hw_params_alloca(&hwparams);
	snd_pcm_hw_params_current(pcm_handle, hwparams);

	/* get channels */
	snd_pcm_hw_params_get_channels (hwparams, &channels);

	/* get buffer size */
	snd_pcm_hw_params_get_buffer_size (hwparams, &buffer_size);

	/* fill half */
	buffer_size /= 2;

	/* allocate buffer assuming 2 bytes per sample */
	buffer_size_bytes = buffer_size * channels * 2;
	buffer = alloca (buffer_size_bytes);
	memset (buffer, 0, buffer_size_bytes);

	/* write data */
	snd_pcm_writei(pcm_handle, buffer, buffer_size);
}
#endif

static snd_pcm_t * alsa_open_r(const char *pcmdev,int bits,int stereo,int rate)
{
	snd_pcm_t *pcm_handle;
	int err;

	ms_message("alsa_open_r: opening %s at %iHz, bits=%i, stereo=%i",pcmdev,rate,bits,stereo);


	if (snd_pcm_open(&pcm_handle, pcmdev,SND_PCM_STREAM_CAPTURE,SND_PCM_NONBLOCK) < 0) {
		ms_warning("alsa_open_r: Error opening PCM device %s\n",pcmdev );
		return NULL;
	}
	if (alsa_set_params(pcm_handle,0,bits,stereo,rate)<0){
		snd_pcm_close(pcm_handle);
		return NULL;
	}
	err=snd_pcm_start(pcm_handle);
	if (err<0){
		ms_warning("snd_pcm_start() failed: %s", snd_strerror(err));
	}
	return pcm_handle;
}

static snd_pcm_t * alsa_open_w(const char *pcmdev,int bits,int stereo,int rate)
{
	snd_pcm_t *pcm_handle;
	
	if (snd_pcm_open(&pcm_handle, pcmdev,SND_PCM_STREAM_PLAYBACK,SND_PCM_NONBLOCK) < 0) {
		ms_warning("alsa_open_w: Error opening PCM device %s\n",pcmdev );
		return NULL;
	}
	if (alsa_set_params(pcm_handle,1,bits,stereo,rate)<0){
		snd_pcm_close(pcm_handle);
		return NULL;
	}
	return pcm_handle;
}


static int alsa_read(snd_pcm_t *handle,unsigned char *buf,int nsamples)
{
	int err;
	err=snd_pcm_readi(handle,buf,nsamples);
	if (err<0) {
		if (err==-EPIPE){
			snd_pcm_prepare(handle);
			err=snd_pcm_readi(handle,buf,nsamples);
			if (err<0) ms_warning("alsa_read: snd_pcm_readi() failed:%s.",snd_strerror(err));
		}else if (err!=-EWOULDBLOCK){
			ms_warning("alsa_read: snd_pcm_readi() failed:%s.",snd_strerror(err));
		}
	}else if (err==0){
		ms_warning("alsa_read: snd_pcm_readi() returned 0");
	}
	return err;
}


static int alsa_write(snd_pcm_t *handle,unsigned char *buf,int nsamples)
{
	int err;
	if ((err=snd_pcm_writei(handle,buf,nsamples))<0){
		if (err==-EPIPE){
			snd_pcm_prepare(handle);
#ifdef EPIPE_BUGFIX
			alsa_fill_w (handle);
#endif
			err=snd_pcm_writei(handle,buf,nsamples);
			if (err<0) ms_warning("alsa_card_write: Error writing sound buffer (nsamples=%i):%s",nsamples,snd_strerror(err));
		}else if (err!=-EWOULDBLOCK){
			ms_warning("alsa_card_write: snd_pcm_writei() failed:%s.",snd_strerror(err));
		}
	}else if (err!=nsamples) {
		ms_debug("Only %i samples written instead of %i",err,nsamples);
	}
	return err;
}


static snd_mixer_t *alsa_mixer_open(const char *mixdev){
	snd_mixer_t *mixer=NULL;
	int err;
	err=snd_mixer_open(&mixer,0);
	if (err<0){
		ms_warning("Could not open alsa mixer: %s",snd_strerror(err));
		return NULL;
	}
	if ((err = snd_mixer_attach (mixer, mixdev)) < 0){
		ms_warning("Could not attach mixer to card: %s",snd_strerror(err));
		snd_mixer_close(mixer);
		return NULL;
	}
	if ((err = snd_mixer_selem_register (mixer, NULL, NULL)) < 0){
		ms_warning("snd_mixer_selem_register: %s",snd_strerror(err));
		snd_mixer_close(mixer);
		return NULL;
	}
	if ((err = snd_mixer_load (mixer)) < 0){
		ms_warning("snd_mixer_load: %s",snd_strerror(err));
		snd_mixer_close(mixer);
		return NULL;
	}
	return mixer;
}

static void alsa_mixer_close(snd_mixer_t *mix){
	snd_mixer_close(mix);
}

typedef enum {CAPTURE, PLAYBACK, CAPTURE_SWITCH, PLAYBACK_SWITCH} MixerAction;

static int get_mixer_element(snd_mixer_t *mixer,const char *name, MixerAction action){
	long value=0;
	const char *elemname;
	snd_mixer_elem_t *elem;
	int err;
	long sndMixerPMin=0;
	long sndMixerPMax=0;
	long newvol=0;
	elem=snd_mixer_first_elem(mixer);
	while (elem!=NULL){
		elemname=snd_mixer_selem_get_name(elem);
		//g_message("Found alsa mixer element %s.",elemname);
		if (strcmp(elemname,name)==0){
			switch (action){
				case CAPTURE:
				if (snd_mixer_selem_has_capture_volume(elem)){
					snd_mixer_selem_get_capture_volume_range(elem, &sndMixerPMin, &sndMixerPMax);
					err=snd_mixer_selem_get_capture_volume(elem,SND_MIXER_SCHN_UNKNOWN,&newvol);
					newvol-=sndMixerPMin;
					value=(100*newvol)/(sndMixerPMax-sndMixerPMin);
					if (err<0) ms_warning("Could not get capture volume for %s:%s",name,snd_strerror(err));
					//else ms_message("Succesfully get capture level for %s.",elemname);
					break;
				}
				break;
				case PLAYBACK:
				if (snd_mixer_selem_has_playback_volume(elem)){
					snd_mixer_selem_get_playback_volume_range(elem, &sndMixerPMin, &sndMixerPMax);
					err=snd_mixer_selem_get_playback_volume(elem,SND_MIXER_SCHN_FRONT_LEFT,&newvol);
					newvol-=sndMixerPMin;
					value=(100*newvol)/(sndMixerPMax-sndMixerPMin);
					if (err<0) ms_warning("Could not get playback volume for %s:%s",name,snd_strerror(err));
					//else g_message("Succesfully get playback level for %s.",elemname);
					break;
				}
				break;
				case CAPTURE_SWITCH:
				
				break;
				case PLAYBACK_SWITCH:

				break;
			}
		}
		elem=snd_mixer_elem_next(elem);
	}
	
	return value;
}


static void set_mixer_element(snd_mixer_t *mixer,const char *name, int level,MixerAction action){
	const char *elemname;
	snd_mixer_elem_t *elem;
	long sndMixerPMin=0;
	long sndMixerPMax=0;
	long newvol=0;
	
	elem=snd_mixer_first_elem(mixer);
	
	while (elem!=NULL){
		elemname=snd_mixer_selem_get_name(elem);
		//g_message("Found alsa mixer element %s.",elemname);
		if (strcmp(elemname,name)==0){
			switch(action){
				case CAPTURE:
				if (snd_mixer_selem_has_capture_volume(elem)){
					snd_mixer_selem_get_capture_volume_range(elem, &sndMixerPMin, &sndMixerPMax);
					newvol=(((sndMixerPMax-sndMixerPMin)*level)/100)+sndMixerPMin;
					snd_mixer_selem_set_capture_volume_all(elem,newvol);
					//g_message("Succesfully set capture level for %s.",elemname);
					return;
				}
				break;
				case PLAYBACK:
				if (snd_mixer_selem_has_playback_volume(elem)){
					snd_mixer_selem_get_playback_volume_range(elem, &sndMixerPMin, &sndMixerPMax);
					newvol=(((sndMixerPMax-sndMixerPMin)*level)/100)+sndMixerPMin;
					snd_mixer_selem_set_playback_volume_all(elem,newvol);
					//g_message("Succesfully set playback level for %s.",elemname);
					return;
				}
				break;
				case CAPTURE_SWITCH:
				if (snd_mixer_selem_has_capture_switch(elem)){
					snd_mixer_selem_set_capture_switch_all(elem,level);
					//g_message("Succesfully set capture switch for %s.",elemname);
				}
				break;
				case PLAYBACK_SWITCH: