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audio_out.c
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/*
** Copyright (c) 1999-2016, Erik de Castro Lopo <[email protected]>
** All rights reserved.
**
** This code is released under 2-clause BSD license. Please see the
** file at : https://github.com/libsndfile/libsamplerate/blob/master/COPYING
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#ifdef HAVE_UNISTD_H
#include <unistd.h>
#endif
#ifdef _WIN32
#ifndef WIN32_LEAN_AN_MEAN
#define WIN32_LEAN_AN_MEAN
#endif
#include <windows.h>
#include <mmsystem.h>
#endif
#include "audio_out.h"
#if (HAVE_SNDFILE)
#include <math.h>
#include <sndfile.h>
#define BUFFER_LEN (2048)
#define MAKE_MAGIC(a,b,c,d,e,f,g,h) \
((a) + ((b) << 1) + ((c) << 2) + ((d) << 3) + ((e) << 4) + ((f) << 5) + ((g) << 6) + ((h) << 7))
/*------------------------------------------------------------------------------
** Linux (ALSA and OSS) functions for playing a sound.
*/
#if defined (__linux__)
#if HAVE_ALSA
#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#include <alsa/asoundlib.h>
#include <sys/time.h>
#define ALSA_MAGIC MAKE_MAGIC ('L', 'n', 'x', '-', 'A', 'L', 'S', 'A')
typedef struct AUDIO_OUT
{ int magic ;
snd_pcm_t * dev ;
int channels ;
} ALSA_AUDIO_OUT ;
static int alsa_write_float (snd_pcm_t *alsa_dev, float *data, int frames, int channels) ;
static AUDIO_OUT *
alsa_open (int channels, unsigned samplerate)
{ ALSA_AUDIO_OUT *alsa_out ;
const char * device = "default" ;
snd_pcm_hw_params_t *hw_params ;
snd_pcm_uframes_t buffer_size ;
snd_pcm_uframes_t alsa_period_size, alsa_buffer_frames ;
snd_pcm_sw_params_t *sw_params ;
int err ;
alsa_period_size = 1024 ;
alsa_buffer_frames = 4 * alsa_period_size ;
if ((alsa_out = calloc (1, sizeof (ALSA_AUDIO_OUT))) == NULL)
{ perror ("alsa_open : malloc ") ;
exit (1) ;
} ;
alsa_out->magic = ALSA_MAGIC ;
alsa_out->channels = channels ;
if ((err = snd_pcm_open (&alsa_out->dev, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
{ fprintf (stderr, "cannot open audio device \"%s\" (%s)\n", device, snd_strerror (err)) ;
goto catch_error ;
} ;
snd_pcm_nonblock (alsa_out->dev, 0) ;
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0)
{ fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_any (alsa_out->dev, hw_params)) < 0)
{ fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_access (alsa_out->dev, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
{ fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_format (alsa_out->dev, hw_params, SND_PCM_FORMAT_FLOAT)) < 0)
{ fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_rate_near (alsa_out->dev, hw_params, &samplerate, 0)) < 0)
{ fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_channels (alsa_out->dev, hw_params, channels)) < 0)
{ fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_buffer_size_near (alsa_out->dev, hw_params, &alsa_buffer_frames)) < 0)
{ fprintf (stderr, "cannot set buffer size (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_period_size_near (alsa_out->dev, hw_params, &alsa_period_size, 0)) < 0)
{ fprintf (stderr, "cannot set period size (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params (alsa_out->dev, hw_params)) < 0)
{ fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
/* 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) ;
goto catch_error ;
} ;
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)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_sw_params_current (alsa_out->dev, sw_params)) != 0)
{ fprintf (stderr, "%s: snd_pcm_sw_params_current: %s", __func__, snd_strerror (err)) ;
goto catch_error ;
} ;
/* note: set start threshold to delay start until the ring buffer is full */
snd_pcm_sw_params_current (alsa_out->dev, sw_params) ;
if ((err = snd_pcm_sw_params_set_start_threshold (alsa_out->dev, sw_params, buffer_size)) < 0)
{ fprintf (stderr, "cannot set start threshold (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_sw_params (alsa_out->dev, sw_params)) != 0)
{ fprintf (stderr, "%s: snd_pcm_sw_params: %s", __func__, snd_strerror (err)) ;
goto catch_error ;
} ;
snd_pcm_sw_params_free (sw_params) ;
snd_pcm_reset (alsa_out->dev) ;
catch_error :
if (err < 0 && alsa_out->dev != NULL)
{ snd_pcm_close (alsa_out->dev) ;
return NULL ;
} ;
return (AUDIO_OUT *) alsa_out ;
} /* alsa_open */
static void
alsa_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
{ static float buffer [BUFFER_LEN] ;
ALSA_AUDIO_OUT *alsa_out ;
int read_frames ;
if ((alsa_out = (ALSA_AUDIO_OUT*) audio_out) == NULL)
{ printf ("alsa_close : AUDIO_OUT is NULL.\n") ;
return ;
} ;
if (alsa_out->magic != ALSA_MAGIC)
{ printf ("alsa_close : Bad magic number.\n") ;
return ;
} ;
while ((read_frames = callback (callback_data, buffer, BUFFER_LEN / alsa_out->channels)))
alsa_write_float (alsa_out->dev, buffer, read_frames, alsa_out->channels) ;
return ;
} /* alsa_play */
static int
alsa_write_float (snd_pcm_t *alsa_dev, float *data, int frames, int channels)
{ static int epipe_count = 0 ;
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
if (0)
{ snd_pcm_status_t *status ;
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") ;
} ;
#endif
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 */
static void
alsa_close (AUDIO_OUT *audio_out)
{ ALSA_AUDIO_OUT *alsa_out ;
if ((alsa_out = (ALSA_AUDIO_OUT*) audio_out) == NULL)
{ printf ("alsa_close : AUDIO_OUT is NULL.\n") ;
return ;
} ;
if (alsa_out->magic != ALSA_MAGIC)
{ printf ("alsa_close : Bad magic number.\n") ;
return ;
} ;
memset (alsa_out, 0, sizeof (ALSA_AUDIO_OUT)) ;
free (alsa_out) ;
return ;
} /* alsa_close */
#endif /* HAVE_ALSA */
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/soundcard.h>
#define OSS_MAGIC MAKE_MAGIC ('L', 'i', 'n', 'u', 'x', 'O', 'S', 'S')
typedef struct
{ int magic ;
int fd ;
int channels ;
} OSS_AUDIO_OUT ;
static AUDIO_OUT *opensoundsys_open (int channels, int samplerate) ;
static void opensoundsys_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ;
static void opensoundsys_close (AUDIO_OUT *audio_out) ;
static AUDIO_OUT *
opensoundsys_open (int channels, int samplerate)
{ OSS_AUDIO_OUT *opensoundsys_out ;
int stereo, fmt, error ;
if ((opensoundsys_out = calloc (1, sizeof (OSS_AUDIO_OUT))) == NULL)
{ perror ("opensoundsys_open : malloc ") ;
exit (1) ;
} ;
opensoundsys_out->magic = OSS_MAGIC ;
opensoundsys_out->channels = channels ;
if ((opensoundsys_out->fd = open ("/dev/dsp", O_WRONLY, 0)) == -1)
{ perror ("opensoundsys_open : open ") ;
exit (1) ;
} ;
stereo = 0 ;
if (ioctl (opensoundsys_out->fd, SNDCTL_DSP_STEREO, &stereo) == -1)
{ /* Fatal error */
perror ("opensoundsys_open : stereo ") ;
exit (1) ;
} ;
if (ioctl (opensoundsys_out->fd, SNDCTL_DSP_RESET, 0))
{ perror ("opensoundsys_open : reset ") ;
exit (1) ;
} ;
fmt = CPU_IS_BIG_ENDIAN ? AFMT_S16_BE : AFMT_S16_LE ;
if (ioctl (opensoundsys_out->fd, SNDCTL_DSP_SETFMT, &fmt) != 0)
{ perror ("opensoundsys_open_dsp_device : set format ") ;
exit (1) ;
} ;
if ((error = ioctl (opensoundsys_out->fd, SNDCTL_DSP_CHANNELS, &channels)) != 0)
{ perror ("opensoundsys_open : channels ") ;
exit (1) ;
} ;
if ((error = ioctl (opensoundsys_out->fd, SNDCTL_DSP_SPEED, &samplerate)) != 0)
{ perror ("opensoundsys_open : sample rate ") ;
exit (1) ;
} ;
if ((error = ioctl (opensoundsys_out->fd, SNDCTL_DSP_SYNC, 0)) != 0)
{ perror ("opensoundsys_open : sync ") ;
exit (1) ;
} ;
return (AUDIO_OUT*) opensoundsys_out ;
} /* opensoundsys_open */
static void
opensoundsys_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
{ OSS_AUDIO_OUT *opensoundsys_out ;
static float float_buffer [BUFFER_LEN] ;
static short buffer [BUFFER_LEN] ;
int k, read_frames ;
if ((opensoundsys_out = (OSS_AUDIO_OUT*) audio_out) == NULL)
{ printf ("opensoundsys_play : AUDIO_OUT is NULL.\n") ;
return ;
} ;
if (opensoundsys_out->magic != OSS_MAGIC)
{ printf ("opensoundsys_play : Bad magic number.\n") ;
return ;
} ;
while ((read_frames = callback (callback_data, float_buffer, BUFFER_LEN / opensoundsys_out->channels)))
{ for (k = 0 ; k < read_frames * opensoundsys_out->channels ; k++)
buffer [k] = lrint (32767.0 * float_buffer [k]) ;
if (write (opensoundsys_out->fd, buffer, read_frames * opensoundsys_out->channels * sizeof (short))) {}
} ;
return ;
} /* opensoundsys_play */
static void
opensoundsys_close (AUDIO_OUT *audio_out)
{ OSS_AUDIO_OUT *opensoundsys_out ;
if ((opensoundsys_out = (OSS_AUDIO_OUT*) audio_out) == NULL)
{ printf ("opensoundsys_close : AUDIO_OUT is NULL.\n") ;
return ;
} ;
if (opensoundsys_out->magic != OSS_MAGIC)
{ printf ("opensoundsys_close : Bad magic number.\n") ;
return ;
} ;
memset (opensoundsys_out, 0, sizeof (OSS_AUDIO_OUT)) ;
free (opensoundsys_out) ;
return ;
} /* opensoundsys_close */
#endif /* __linux__ */
/*------------------------------------------------------------------------------
** Mac OS X functions for playing a sound.
*/
#if (defined (__MACH__) && defined (__APPLE__)) /* MacOSX */
#include <AvailabilityMacros.h>
#include <CoreAudio/AudioHardware.h>
#ifndef MAC_OS_VERSION_12_0
#define kAudioObjectPropertyElementMain kAudioObjectPropertyElementMaster
#endif
#define MACOSX_MAGIC MAKE_MAGIC ('M', 'a', 'c', ' ', 'O', 'S', ' ', 'X')
typedef struct
{ int magic ;
AudioStreamBasicDescription format ;
UInt32 buf_size ;
AudioDeviceID device ;
int channels ;
int samplerate ;
int buffer_size ;
int done_playing ;
get_audio_callback_t callback ;
void *callback_data ;
AudioDeviceIOProcID ioprocid;
} MACOSX_AUDIO_OUT ;
static AUDIO_OUT *macosx_open (int channels, int samplerate) ;
static void macosx_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ;
static void macosx_close (AUDIO_OUT *audio_out) ;
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) ;
static AUDIO_OUT *
macosx_open (int channels, int samplerate)
{ MACOSX_AUDIO_OUT *macosx_out ;
OSStatus err ;
UInt32 count ;
AudioObjectPropertyAddress propertyAddress ;
if ((macosx_out = calloc (1, sizeof (MACOSX_AUDIO_OUT))) == NULL)
{ perror ("macosx_open : malloc ") ;
exit (1) ;
} ;
macosx_out->magic = MACOSX_MAGIC ;
macosx_out->channels = channels ;
macosx_out->samplerate = samplerate ;
macosx_out->device = kAudioDeviceUnknown ;
/* get the default output device for the HAL */
propertyAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
propertyAddress.mScope = kAudioDevicePropertyScopeOutput;
propertyAddress.mElement = kAudioObjectPropertyElementMain;
count = sizeof (AudioDeviceID) ;
if ((err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, NULL,
&count, &(macosx_out->device))) != noErr)
{ printf ("AudioObjectGetPropertyData (kAudioHardwarePropertyDefaultOutputDevice) failed.\n") ;
free (macosx_out) ;
return NULL ;
} ;
/* get the buffersize that the default device uses for IO */
count = sizeof (UInt32) ;
propertyAddress.mSelector = kAudioDevicePropertyBufferSize ;
if ((err = AudioObjectGetPropertyData (macosx_out->device, &propertyAddress, 0, NULL,
&count, &(macosx_out->buffer_size))) != noErr)
{ printf ("AudioObjectGetPropertyData (kAudioDevicePropertyBufferSize) (AudioDeviceGetProperty) failed.\n") ;
free (macosx_out) ;
return NULL ;
} ;
/* get a description of the data format used by the default device */
count = sizeof (AudioStreamBasicDescription) ;
propertyAddress.mSelector = kAudioDevicePropertyStreamFormat ;
if ((err = AudioObjectGetPropertyData (macosx_out->device, &propertyAddress, 0, NULL,
&count, &(macosx_out->format))) != noErr)
{ printf ("AudioObjectGetPropertyData (kAudioDevicePropertyStreamFormat) failed.\n") ;
free (macosx_out) ;
return NULL ;
} ;
macosx_out->format.mSampleRate = samplerate ;
macosx_out->format.mChannelsPerFrame = channels ;
propertyAddress.mSelector = kAudioDevicePropertyStreamFormat ;
count = sizeof (AudioStreamBasicDescription) ;
if ((err = AudioObjectGetPropertyData (macosx_out->device, &propertyAddress, 0, NULL,
&count, &(macosx_out->format))) != noErr)
{ printf ("AudioObjectGetPropertyData (kAudioDevicePropertyStreamFormat) failed.\n") ;
free (macosx_out) ;
return NULL ;
} ;
/* we want linear pcm */
if (macosx_out->format.mFormatID != kAudioFormatLinearPCM)
{ free (macosx_out) ;
return NULL ;
} ;
macosx_out->done_playing = 0 ;
/* Fire off the device. */
if ((err = AudioDeviceCreateIOProcID (macosx_out->device, macosx_audio_out_callback,
(void *) macosx_out, &macosx_out->ioprocid)) != noErr)
{ printf ("AudioDeviceAddIOProc failed.\n") ;
free (macosx_out) ;
return NULL ;
} ;
return (AUDIO_OUT *) macosx_out ;
} /* macosx_open */
static void
macosx_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
{ MACOSX_AUDIO_OUT *macosx_out ;
OSStatus err ;
if ((macosx_out = (MACOSX_AUDIO_OUT*) audio_out) == NULL)
{ printf ("macosx_play : AUDIO_OUT is NULL.\n") ;
return ;
} ;
if (macosx_out->magic != MACOSX_MAGIC)
{ printf ("macosx_play : Bad magic number.\n") ;
return ;
} ;
/* Set the callback function and callback data. */
macosx_out->callback = callback ;
macosx_out->callback_data = callback_data ;
err = AudioDeviceStart (macosx_out->device, macosx_audio_out_callback) ;
if (err != noErr)
printf ("AudioDeviceStart failed.\n") ;
while (macosx_out->done_playing == SF_FALSE)
usleep (10 * 1000) ; /* 10 000 milliseconds. */
return ;
} /* macosx_play */
static void
macosx_close (AUDIO_OUT *audio_out)
{ MACOSX_AUDIO_OUT *macosx_out ;
OSStatus err ;
if ((macosx_out = (MACOSX_AUDIO_OUT*) audio_out) == NULL)
{ printf ("macosx_close : AUDIO_OUT is NULL.\n") ;
return ;
} ;
if (macosx_out->magic != MACOSX_MAGIC)
{ printf ("macosx_close : Bad magic number.\n") ;
return ;
} ;
if ((err = AudioDeviceStop (macosx_out->device, macosx_audio_out_callback)) != noErr)
{ printf ("AudioDeviceStop failed.\n") ;
return ;
} ;
err = AudioDeviceDestroyIOProcID(macosx_out->device,
macosx_out->ioprocid);
if (err != noErr)
{ printf ("AudioDeviceRemoveIOProc failed.\n") ;
return ;
} ;
} /* macosx_close */
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)
{ MACOSX_AUDIO_OUT *macosx_out ;
int size, frame_count, read_count ;
float *buffer ;
/* unused params: */
(void) device;
(void) current_time;
(void) data_in;
(void) time_in;
(void) time_out;
if ((macosx_out = (MACOSX_AUDIO_OUT*) client_data) == NULL)
{ printf ("macosx_play : AUDIO_OUT is NULL.\n") ;
return 42 ;
} ;
if (macosx_out->magic != MACOSX_MAGIC)
{ printf ("macosx_play : Bad magic number.\n") ;
return 42 ;
} ;
size = data_out->mBuffers [0].mDataByteSize ;
frame_count = size / sizeof (float) / macosx_out->channels ;
buffer = (float*) data_out->mBuffers [0].mData ;
read_count = macosx_out->callback (macosx_out->callback_data, buffer, frame_count) ;
if (read_count < frame_count)
{ memset (&(buffer [read_count]), 0, (frame_count - read_count) * sizeof (float)) ;
macosx_out->done_playing = 1 ;
} ;
return noErr ;
} /* macosx_audio_out_callback */
#endif /* MacOSX */
/*------------------------------------------------------------------------------
** Win32 functions for playing a sound.
**
** This API sucks. Its needlessly complicated and is *WAY* too loose with
** passing pointers arounf in integers and and using char* pointers to
** point to data instead of short*. It plain sucks!
*/
#if (defined (_WIN32) || defined (WIN32))
#define WIN32_BUFFER_LEN (1<<15)
#define WIN32_MAGIC MAKE_MAGIC ('W', 'i', 'n', '3', '2', 's', 'u', 'x')
typedef struct
{ int magic ;
HWAVEOUT hwave ;
WAVEHDR whdr [2] ;
HANDLE Event ;
short short_buffer [WIN32_BUFFER_LEN / sizeof (short)] ;
float float_buffer [WIN32_BUFFER_LEN / sizeof (short) / 2] ;
int bufferlen, current ;
int channels ;
get_audio_callback_t callback ;
void *callback_data ;
} WIN32_AUDIO_OUT ;
static AUDIO_OUT *win32_open (int channels, int samplerate) ;
static void win32_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ;
static void win32_close (AUDIO_OUT *audio_out) ;
static DWORD CALLBACK
win32_audio_out_callback (HWAVEOUT hwave, UINT msg, DWORD_PTR data, DWORD_PTR param1, DWORD_PTR param2) ;
static AUDIO_OUT*
win32_open (int channels, int samplerate)
{ WIN32_AUDIO_OUT *win32_out ;
WAVEFORMATEX wf ;
int error ;
if ((win32_out = calloc (1, sizeof (WIN32_AUDIO_OUT))) == NULL)
{ perror ("win32_open : malloc ") ;
exit (1) ;
} ;
win32_out->magic = WIN32_MAGIC ;
win32_out->channels = channels ;
win32_out->current = 0 ;
win32_out->Event = CreateEvent (0, FALSE, FALSE, 0) ;
wf.nChannels = channels ;
wf.nSamplesPerSec = samplerate ;
wf.nBlockAlign = (WORD) (channels * sizeof (short)) ;
wf.wFormatTag = WAVE_FORMAT_PCM ;
wf.cbSize = 0 ;
wf.wBitsPerSample = 16 ;
wf.nAvgBytesPerSec = wf.nBlockAlign * wf.nSamplesPerSec ;
error = waveOutOpen (&(win32_out->hwave), WAVE_MAPPER, &wf, (DWORD_PTR) win32_audio_out_callback,
(DWORD_PTR) win32_out, CALLBACK_FUNCTION) ;
if (error)
{ puts ("waveOutOpen failed.") ;
free (win32_out) ;
return NULL ;
} ;
waveOutPause (win32_out->hwave) ;
return (AUDIO_OUT *) win32_out ;
} /* win32_open */
static void
win32_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
{ WIN32_AUDIO_OUT *win32_out ;
int error ;
if ((win32_out = (WIN32_AUDIO_OUT*) audio_out) == NULL)
{ printf ("win32_play : AUDIO_OUT is NULL.\n") ;
return ;
} ;
if (win32_out->magic != WIN32_MAGIC)
{ printf ("win32_play : Bad magic number (%d %d).\n", win32_out->magic, WIN32_MAGIC) ;
return ;
} ;
/* Set the callback function and callback data. */
win32_out->callback = callback ;
win32_out->callback_data = callback_data ;
win32_out->whdr [0].lpData = (char*) win32_out->short_buffer ;
win32_out->whdr [1].lpData = ((char*) win32_out->short_buffer) + sizeof (win32_out->short_buffer) / 2 ;
win32_out->whdr [0].dwBufferLength = sizeof (win32_out->short_buffer) / 2 ;
win32_out->whdr [1].dwBufferLength = sizeof (win32_out->short_buffer) / 2 ;
win32_out->bufferlen = sizeof (win32_out->short_buffer) / 2 / sizeof (short) ;
/* Prepare the WAVEHDRs */
if ((error = waveOutPrepareHeader (win32_out->hwave, &(win32_out->whdr [0]), sizeof (WAVEHDR))))
{ printf ("waveOutPrepareHeader [0] failed : %08X\n", error) ;
waveOutClose (win32_out->hwave) ;
return ;
} ;
if ((error = waveOutPrepareHeader (win32_out->hwave, &(win32_out->whdr [1]), sizeof (WAVEHDR))))
{ printf ("waveOutPrepareHeader [1] failed : %08X\n", error) ;
waveOutUnprepareHeader (win32_out->hwave, &(win32_out->whdr [0]), sizeof (WAVEHDR)) ;
waveOutClose (win32_out->hwave) ;
return ;
} ;
waveOutRestart (win32_out->hwave) ;
/* Fake 2 calls to the callback function to queue up enough audio. */
win32_audio_out_callback (0, MM_WOM_DONE, (DWORD_PTR) win32_out, 0, 0) ;
win32_audio_out_callback (0, MM_WOM_DONE, (DWORD_PTR) win32_out, 0, 0) ;
/* Wait for playback to finish. The callback notifies us when all
** wave data has been played.
*/
WaitForSingleObject (win32_out->Event, INFINITE) ;
waveOutPause (win32_out->hwave) ;
waveOutReset (win32_out->hwave) ;
waveOutUnprepareHeader (win32_out->hwave, &(win32_out->whdr [0]), sizeof (WAVEHDR)) ;
waveOutUnprepareHeader (win32_out->hwave, &(win32_out->whdr [1]), sizeof (WAVEHDR)) ;
waveOutClose (win32_out->hwave) ;
win32_out->hwave = 0 ;
return ;
} /* win32_play */
static void
win32_close (AUDIO_OUT *audio_out)
{ WIN32_AUDIO_OUT *win32_out ;
if ((win32_out = (WIN32_AUDIO_OUT*) audio_out) == NULL)
{ printf ("win32_close : AUDIO_OUT is NULL.\n") ;
return ;
} ;
if (win32_out->magic != WIN32_MAGIC)
{ printf ("win32_close : Bad magic number.\n") ;
return ;
} ;
memset (win32_out, 0, sizeof (WIN32_AUDIO_OUT)) ;
free (win32_out) ;
} /* win32_close */
static DWORD CALLBACK
win32_audio_out_callback (HWAVEOUT hwave, UINT msg, DWORD_PTR data, DWORD_PTR param1, DWORD_PTR param2)
{
UNREFERENCED_PARAMETER (hwave) ;
UNREFERENCED_PARAMETER (param1) ;
UNREFERENCED_PARAMETER (param2) ;
WIN32_AUDIO_OUT *win32_out ;
int read_count, frame_count, k ;
short *sptr ;
/*
** I consider this technique of passing a pointer via an integer as
** fundamentally broken but thats the way microsoft has defined the
** interface.
*/
if ((win32_out = (WIN32_AUDIO_OUT*) data) == NULL)
{ printf ("win32_audio_out_callback : AUDIO_OUT is NULL.\n") ;
return 1 ;
} ;
if (win32_out->magic != WIN32_MAGIC)
{ printf ("win32_audio_out_callback : Bad magic number (%d %d).\n", win32_out->magic, WIN32_MAGIC) ;
return 1 ;
} ;
if (msg != MM_WOM_DONE)
return 0 ;
/* Do the actual audio. */
frame_count = win32_out->bufferlen / win32_out->channels ;
read_count = win32_out->callback (win32_out->callback_data, win32_out->float_buffer, frame_count) ;
sptr = (short*) win32_out->whdr [win32_out->current].lpData ;
for (k = 0 ; k < read_count ; k++)
sptr [k] = (short) lrint (32767.0 * win32_out->float_buffer [k]) ;
if (read_count > 0)
{ /* Fix buffer length is only a partial block. */
if (read_count * (int) sizeof (short) < win32_out->bufferlen)
win32_out->whdr [win32_out->current].dwBufferLength = read_count * sizeof (short) ;
/* Queue the WAVEHDR */
waveOutWrite (win32_out->hwave, (LPWAVEHDR) &(win32_out->whdr [win32_out->current]), sizeof (WAVEHDR)) ;
}
else
{ /* Stop playback */
waveOutPause (win32_out->hwave) ;
SetEvent (win32_out->Event) ;
} ;
win32_out->current = (win32_out->current + 1) % 2 ;
return 0 ;
} /* win32_audio_out_callback */
#endif /* Win32 */
/*------------------------------------------------------------------------------
** Solaris.
*/
#if (defined (sun) && defined (unix)) /* ie Solaris */
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/audioio.h>
#define SOLARIS_MAGIC MAKE_MAGIC ('S', 'o', 'l', 'a', 'r', 'i', 's', ' ')
typedef struct
{ int magic ;
int fd ;
int channels ;
int samplerate ;
} SOLARIS_AUDIO_OUT ;
static AUDIO_OUT *solaris_open (int channels, int samplerate) ;
static void solaris_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ;
static void solaris_close (AUDIO_OUT *audio_out) ;
static AUDIO_OUT *
solaris_open (int channels, int samplerate)
{ SOLARIS_AUDIO_OUT *solaris_out ;
audio_info_t audio_info ;
int error ;
if ((solaris_out = calloc (1, sizeof (SOLARIS_AUDIO_OUT))) == NULL)
{ perror ("solaris_open : malloc ") ;
exit (1) ;
} ;
solaris_out->magic = SOLARIS_MAGIC ;
solaris_out->channels = channels ;
solaris_out->samplerate = channels ;
/* open the audio device - write only, non-blocking */
if ((solaris_out->fd = open ("/dev/audio", O_WRONLY | O_NONBLOCK)) < 0)
{ perror ("open (/dev/audio) failed") ;
exit (1) ;
} ;
/* Retrive standard values. */
AUDIO_INITINFO (&audio_info) ;
audio_info.play.sample_rate = samplerate ;
audio_info.play.channels = channels ;
audio_info.play.precision = 16 ;
audio_info.play.encoding = AUDIO_ENCODING_LINEAR ;
audio_info.play.gain = AUDIO_MAX_GAIN ;
audio_info.play.balance = AUDIO_MID_BALANCE ;
if ((error = ioctl (solaris_out->fd, AUDIO_SETINFO, &audio_info)))
{ perror ("ioctl (AUDIO_SETINFO) failed") ;
exit (1) ;
} ;
return (AUDIO_OUT*) solaris_out ;
} /* solaris_open */
static void
solaris_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
{ SOLARIS_AUDIO_OUT *solaris_out ;
static float float_buffer [BUFFER_LEN] ;
static short buffer [BUFFER_LEN] ;
int k, read_frames ;
if ((solaris_out = (SOLARIS_AUDIO_OUT*) audio_out) == NULL)
{ printf ("solaris_play : AUDIO_OUT is NULL.\n") ;
return ;
} ;
if (solaris_out->magic != SOLARIS_MAGIC)
{ printf ("solaris_play : Bad magic number.\n") ;
return ;
} ;
while ((read_frames = callback (callback_data, float_buffer, BUFFER_LEN / solaris_out->channels)))
{ for (k = 0 ; k < read_frames * solaris_out->channels ; k++)
buffer [k] = psf_lrint (32767.0 * float_buffer [k]) ;
write (solaris_out->fd, buffer, read_frames * solaris_out->channels * sizeof (short)) ;
} ;
return ;
} /* solaris_play */
static void
solaris_close (AUDIO_OUT *audio_out)
{ SOLARIS_AUDIO_OUT *solaris_out ;
if ((solaris_out = (SOLARIS_AUDIO_OUT*) audio_out) == NULL)
{ printf ("solaris_close : AUDIO_OUT is NULL.\n") ;
return ;
} ;
if (solaris_out->magic != SOLARIS_MAGIC)
{ printf ("solaris_close : Bad magic number.\n") ;
return ;
} ;
memset (solaris_out, 0, sizeof (SOLARIS_AUDIO_OUT)) ;
free (solaris_out) ;
return ;
} /* solaris_close */
#endif /* Solaris */
/*==============================================================================
** Main function.
*/