/*** Copyright (c) 1999-2016, Erik de Castro Lopo <erikd@mega-nerd.com>** 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*/ #include <stdio.h>#include <stdlib.h>#include <string.h>#include <unistd.h> #ifdef _WIN32#define WIN32_LEAN_AN_MEAN#include <windows.h>#include <mmsystem.h>#endif #include "src_config.h" #include "audio_out.h" #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>#endif #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_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 voidalsa_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 intalsa_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 voidalsa_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 voidopensoundsys_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 voidopensoundsys_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 <CoreAudio/AudioHardware.h> #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 OSStatusmacosx_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 = kAudioObjectPropertyElementMaster; 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 ; } ; Lines 1–500 of 1109. 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/* ** Copyright (c) 1999-2016, Erik de Castro Lopo <erikd@mega-nerd.com> ** 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 */ #include <stdio.h> #include <stdlib.h> #include <string.h> #include <unistd.h> #ifdef _WIN32 #define WIN32_LEAN_AN_MEAN #include <windows.h> #include <mmsystem.h> #endif #include "src_config.h" #include "audio_out.h" #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> #endif #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_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 <CoreAudio/AudioHardware.h> #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 = kAudioObjectPropertyElementMaster; 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 ; } ;