sndfile-play.c

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/*
** Copyright (C) 1999-2004 Erik de Castro Lopo <[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 "config.h"

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <errno.h>

#if HAVE_UNISTD_H
#include <unistd.h>
#endif

#if HAVE_ALSA_ASOUNDLIB_H
	#define ALSA_PCM_NEW_HW_PARAMS_API
	#define ALSA_PCM_NEW_SW_PARAMS_API
	#include <alsa/asoundlib.h>
	#include <sys/time.h>
#endif

#if defined (__linux__)
	#include 	<fcntl.h>
	#include 	<sys/ioctl.h>
	#include 	<sys/soundcard.h>

#elif (defined (__MACH__) && defined (__APPLE__))
	#include <Carbon.h>
	#include <CoreAudio/AudioHardware.h>

#elif (defined (sun) && defined (unix))
	#include <fcntl.h>
	#include <sys/ioctl.h>
	#include <sys/audioio.h>

#elif (OS_IS_WIN32 == 1)
	#include <windows.h>
	#include <mmsystem.h>

#endif

#include	<sndfile.h>

#define	SIGNED_SIZEOF(x)	((int) sizeof (x))
#define	BUFFER_LEN			(2048)

/*------------------------------------------------------------------------------
**	Linux/OSS functions for playing a sound.
*/

#if HAVE_ALSA_ASOUNDLIB_H

static snd_pcm_t * alsa_open (int channels, int srate) ;
static int alsa_write_float (snd_pcm_t *alsa_dev, float *data, int frames, int channels) ;

static void
alsa_play (int argc, char *argv [])
{	static float buffer [BUFFER_LEN] ;
	SNDFILE *sndfile ;
	SF_INFO sfinfo ;
	snd_pcm_t * alsa_dev ;
	int		k, readcount, subformat ;

	for (k = 1 ; k < argc ; k++)
	{	memset (&sfinfo, 0, sizeof (sfinfo)) ;

		printf ("Playing %s\n", argv [k]) ;
		if (! (sndfile = sf_open (argv [k], SFM_READ, &sfinfo)))
		{	puts (sf_strerror (NULL)) ;
			continue ;
			} ;

		if (sfinfo.channels < 1 || sfinfo.channels > 2)
		{	printf ("Error : channels = %d.\n", sfinfo.channels) ;
			continue ;
			} ;

		alsa_dev = alsa_open (sfinfo.channels, sfinfo.samplerate) ;

		subformat = sfinfo.format & SF_FORMAT_SUBMASK ;

		if (subformat == SF_FORMAT_FLOAT || subformat == SF_FORMAT_DOUBLE)
		{	double	scale ;
			int 	m ;

			sf_command (sndfile, SFC_CALC_SIGNAL_MAX, &scale, sizeof (scale)) ;
			if (scale < 1e-10)
				scale = 1.0 ;
			else
				scale = 32700.0 / scale ;

			while ((readcount = sf_read_float (sndfile, buffer, BUFFER_LEN)))
			{	for (m = 0 ; m < readcount ; m++)
					buffer [m] *= scale ;
				alsa_write_float (alsa_dev, buffer, BUFFER_LEN / sfinfo.channels, sfinfo.channels) ;
				} ;
			}
		else
		{	while ((readcount = sf_read_float (sndfile, buffer, BUFFER_LEN)))
				alsa_write_float (alsa_dev, buffer, BUFFER_LEN / sfinfo.channels, sfinfo.channels) ;
			} ;

		snd_pcm_close (alsa_dev) ;

		sf_close (sndfile) ;
		} ;

	return ;
} /* alsa_play */

static snd_pcm_t *
alsa_open (int channels, int samplerate)
{	const char * device = "plughw:0" ;
	snd_pcm_t *alsa_dev ;
	snd_pcm_hw_params_t *hw_params ;
	snd_pcm_uframes_t buffer_size, xfer_align, start_threshold ;
	snd_pcm_uframes_t alsa_period_size, alsa_buffer_frames ;
	snd_pcm_sw_params_t *sw_params ;

	int err ;

	alsa_period_size = 512 ;
	alsa_buffer_frames = 3 * alsa_period_size ;

	if ((err = snd_pcm_open (&alsa_dev, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
	{	fprintf (stderr, "cannot open audio device \"%s\" (%s)\n", device, snd_strerror (err)) ;
		return NULL ;
		} ;

	snd_pcm_nonblock (alsa_dev, 0) ;

	if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0)
	{	fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err)) ;
		return NULL ;
		} ;

	if ((err = snd_pcm_hw_params_any (alsa_dev, hw_params)) < 0)
	{	fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err)) ;
		return NULL ;
		} ;

	if ((err = snd_pcm_hw_params_set_access (alsa_dev, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
	{	fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err)) ;
		return NULL ;
		} ;

	if ((err = snd_pcm_hw_params_set_format (alsa_dev, hw_params, SND_PCM_FORMAT_FLOAT)) < 0)
	{	fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err)) ;
		return NULL ;
		} ;

	if ((err = snd_pcm_hw_params_set_rate_near (alsa_dev, hw_params, &samplerate, 0)) < 0)
	{	fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err)) ;
		return NULL ;
		} ;

	if ((err = snd_pcm_hw_params_set_channels (alsa_dev, hw_params, channels)) < 0)
	{	fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror (err)) ;
		return NULL ;
		} ;

	if ((err = snd_pcm_hw_params_set_buffer_size_near (alsa_dev, hw_params, &alsa_buffer_frames)) < 0)
	{	fprintf (stderr, "cannot set buffer size (%s)\n", snd_strerror (err)) ;
		return NULL ;
		} ;


	if ((err = snd_pcm_hw_params_set_period_size_near (alsa_dev, hw_params, &alsa_period_size, 0)) < 0)
	{	fprintf (stderr, "cannot set period size (%s)\n", snd_strerror (err)) ;
		return NULL ;
		} ;

	if ((err = snd_pcm_hw_params (alsa_dev, hw_params)) < 0)
	{	fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err)) ;
		return NULL ;
		} ;

	/* extra check: if we have only one period, this code won't work */
	snd_pcm_hw_params_get_period_size (hw_params, &alsa_period_size, 0) ;
	snd_pcm_hw_params_get_buffer_size (hw_params, &buffer_size) ;
	if (alsa_period_size == buffer_size)
	{	fprintf (stderr, "Can't use period equal to buffer size (%lu == %lu)", alsa_period_size, buffer_size) ;
		return NULL ;
		} ;

	snd_pcm_hw_params_free (hw_params) ;

	if ((err = snd_pcm_sw_params_malloc (&sw_params)) != 0)
	{	fprintf (stderr, "%s: snd_pcm_sw_params_malloc: %s", __func__, snd_strerror (err)) ;
		return NULL ;
		} ;

	if ((err = snd_pcm_sw_params_current (alsa_dev, sw_params)) != 0)
	{	fprintf (stderr, "%s: snd_pcm_sw_params_current: %s", __func__, snd_strerror (err)) ;
		return NULL ;
		} ;

	/* note: set start threshold to delay start until the ring buffer is full */
	snd_pcm_sw_params_current (alsa_dev, sw_params) ;
	if ((err = snd_pcm_sw_params_get_xfer_align (sw_params, &xfer_align)) < 0)
	{	fprintf (stderr, "cannot get xfer align (%s)\n", snd_strerror (err)) ;
		return NULL ;
		} ;

	/* round up to closest transfer boundary */
	start_threshold = (buffer_size / xfer_align) * xfer_align ;
	if (start_threshold < 1)
		start_threshold = 1 ;
	if ((err = snd_pcm_sw_params_set_start_threshold (alsa_dev, sw_params, start_threshold)) < 0)
	{	fprintf (stderr, "cannot set start threshold (%s)\n", snd_strerror (err)) ;
		return NULL ;
		} ;

	if ((err = snd_pcm_sw_params (alsa_dev, sw_params)) != 0)
	{	fprintf (stderr, "%s: snd_pcm_sw_params: %s", __func__, snd_strerror (err)) ;
		return NULL ;
		} ;
	snd_pcm_sw_params_free (sw_params) ;

	snd_pcm_reset (alsa_dev) ;

	return alsa_dev ;
} /* alsa_open */

static int
alsa_write_float (snd_pcm_t *alsa_dev, float *data, int frames, int channels)
{	static	int epipe_count = 0 ;

	snd_pcm_status_t *status ;
	int total = 0 ;
	int retval ;

	if (epipe_count > 0)
		epipe_count -- ;

	while (total < frames)
	{	retval = snd_pcm_writei (alsa_dev, data + total * channels, frames - total) ;

		if (retval >= 0)
		{	total += retval ;
			if (total == frames)
				return total ;

			continue ;
			} ;

		switch (retval)
		{	case -EAGAIN :
					puts ("alsa_write_float: EAGAIN") ;
					continue ;
					break ;

			case -EPIPE :
					if (epipe_count > 0)
					{	printf ("alsa_write_float: EPIPE %d\n", epipe_count) ;
						if (epipe_count > 140)
							return retval ;
						} ;
					epipe_count += 100 ;

					if (0)
					{	snd_pcm_status_alloca (&status) ;
						if ((retval = snd_pcm_status (alsa_dev, status)) < 0)
							fprintf (stderr, "alsa_out: xrun. can't determine length\n") ;
						else if (snd_pcm_status_get_state (status) == SND_PCM_STATE_XRUN)
						{	struct timeval now, diff, tstamp ;

							gettimeofday (&now, 0) ;
							snd_pcm_status_get_trigger_tstamp (status, &tstamp) ;
							timersub (&now, &tstamp, &diff) ;

							fprintf (stderr, "alsa_write_float xrun: of at least %.3f msecs. resetting stream\n",
									diff.tv_sec * 1000 + diff.tv_usec / 1000.0) ;
							}
						else
							fprintf (stderr, "alsa_write_float: xrun. can't determine length\n") ;
						} ;

					snd_pcm_prepare (alsa_dev) ;
					break ;

			case -EBADFD :
					fprintf (stderr, "alsa_write_float: Bad PCM state.n") ;
					return 0 ;
					break ;

			case -ESTRPIPE :
					fprintf (stderr, "alsa_write_float: Suspend event.n") ;
					return 0 ;
					break ;

			case -EIO :
					puts ("alsa_write_float: EIO") ;
					return 0 ;

			default :
					fprintf (stderr, "alsa_write_float: retval = %d\n", retval) ;
					return 0 ;
					break ;
			} ; /* switch */
		} ; /* while */

	return total ;
} /* alsa_write_float */

#endif /* HAVE_ALSA_ASOUNDLIB_H */

/*------------------------------------------------------------------------------
**	Linux/OSS functions for playing a sound.
*/

#if defined (__linux__)

static	int	linux_open_dsp_device (int channels, int srate) ;

static void
linux_play (int argc, char *argv [])
{	static short buffer [BUFFER_LEN] ;
	SNDFILE *sndfile ;
	SF_INFO sfinfo ;
	int		k, audio_device, readcount, subformat ;

	for (k = 1 ; k < argc ; k++)
	{	memset (&sfinfo, 0, sizeof (sfinfo)) ;

		printf ("Playing %s\n", argv [k]) ;
		if (! (sndfile = sf_open (argv [k], SFM_READ, &sfinfo)))
		{	puts (sf_strerror (NULL)) ;
			continue ;
			} ;

		if (sfinfo.channels < 1 || sfinfo.channels > 2)
		{	printf ("Error : channels = %d.\n", sfinfo.channels) ;
			continue ;
			} ;

		audio_device = linux_open_dsp_device (sfinfo.channels, sfinfo.samplerate) ;

		subformat = sfinfo.format & SF_FORMAT_SUBMASK ;

		if (subformat == SF_FORMAT_FLOAT || subformat == SF_FORMAT_DOUBLE)
		{	static float float_buffer [BUFFER_LEN] ;
			double	scale ;
			int 	m ;

			sf_command (sndfile, SFC_CALC_SIGNAL_MAX, &scale, sizeof (scale)) ;
			if (scale < 1e-10)
				scale = 1.0 ;
			else
				scale = 32700.0 / scale ;

			while ((readcount = sf_read_float (sndfile, float_buffer, BUFFER_LEN)))
			{	for (m = 0 ; m < readcount ; m++)
					buffer [m] = scale * float_buffer [m] ;
				write (audio_device, buffer, readcount * sizeof (short)) ;
				} ;
			}
		else
		{	while ((readcount = sf_read_short (sndfile, buffer, BUFFER_LEN)))
				write (audio_device, buffer, readcount * sizeof (short)) ;
			} ;

		close (audio_device) ;

		sf_close (sndfile) ;
		} ;

	return ;
} /* linux_play */

static int
linux_open_dsp_device (int channels, int srate)
{	int fd, stereo, temp, error ;

	if ((fd = open ("/dev/dsp", O_WRONLY, 0)) == -1 &&
		(fd = open ("/dev/sound/dsp", O_WRONLY, 0)) == -1)
	{	perror ("linux_open_dsp_device : open ") ;
		exit (1) ;
		} ;

	stereo = 0 ;
	if (ioctl (fd, SNDCTL_DSP_STEREO, &stereo) == -1)
	{ 	/* Fatal error */
		perror ("linux_open_dsp_device : stereo ") ;
		exit (1) ;
		} ;

	if (ioctl (fd, SNDCTL_DSP_RESET, 0))
	{	perror ("linux_open_dsp_device : reset ") ;
		exit (1) ;
		} ;

	temp = 16 ;
	if ((error = ioctl (fd, SOUND_PCM_WRITE_BITS, &temp)) != 0)
	{	perror ("linux_open_dsp_device : bitwidth ") ;
		exit (1) ;
		} ;

	if ((error = ioctl (fd, SOUND_PCM_WRITE_CHANNELS, &channels)) != 0)
	{	perror ("linux_open_dsp_device : channels ") ;
		exit (1) ;
		} ;

	if ((error = ioctl (fd, SOUND_PCM_WRITE_RATE, &srate)) != 0)
	{	perror ("linux_open_dsp_device : sample rate ") ;
		exit (1) ;
		} ;

	if ((error = ioctl (fd, SNDCTL_DSP_SYNC, 0)) != 0)
	{	perror ("linux_open_dsp_device : sync ") ;
		exit (1) ;
		} ;

	return 	fd ;
} /* linux_open_dsp_device */

#endif /* __linux__ */

/*------------------------------------------------------------------------------
**	Mac OS X functions for playing a sound.
*/

#if (defined (__MACH__) && defined (__APPLE__)) /* MacOSX */

typedef struct
{	AudioStreamBasicDescription		format ;

	UInt32 			buf_size ;
	AudioDeviceID 	device ;

	SNDFILE 		*sndfile ;
	SF_INFO 		sfinfo ;

	int				fake_stereo ;
	int				done_playing ;
} MacOSXAudioData ;

#include <math.h>

static OSStatus
macosx_audio_out_callback (AudioDeviceID device, const AudioTimeStamp* current_time,
	const AudioBufferList* data_in, const AudioTimeStamp* time_in,
	AudioBufferList*	data_out, const AudioTimeStamp* time_out,
	void* client_data)
{	MacOSXAudioData	*audio_data ;
	int		size, sample_count, read_count, k ;
	float	*buffer ;

	/* Prevent compiler warnings. */
	device = device ;
	current_time = current_time ;
	data_in = data_in ;
	time_in = time_in ;
	time_out = time_out ;