--- /dev/null
+/*
+** 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/erikd/libsamplerate/blob/master/COPYING
+*/
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <unistd.h>
+
+#include <config.h>
+
+#include "audio_out.h"
+
+#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 (HAVE_SNDFILE)
+
+#include <float_cast.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))
+
+typedef struct AUDIO_OUT_s
+{ int magic ;
+} AUDIO_OUT ;
+
+
+/*------------------------------------------------------------------------------
+** Linux (ALSA and OSS) functions for playing a sound.
+*/
+
+#if defined (__linux__)
+
+#if HAVE_ALSA_ASOUNDLIB_H
+
+#define ALSA_MAGIC MAKE_MAGIC ('L', 'n', 'x', '-', 'A', 'L', 'S', 'A')
+
+typedef struct
+{ 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_ASOUNDLIB_H */
+
+#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]) ;
+ (void) 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 <Carbon.h>
+#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 ;
+} 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 ;
+ size_t count ;
+
+ 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 */
+ count = sizeof (AudioDeviceID) ;
+ if ((err = AudioHardwareGetProperty (kAudioHardwarePropertyDefaultOutputDevice,
+ &count, (void *) &(macosx_out->device))) != noErr)
+ { printf ("AudioHardwareGetProperty failed.\n") ;
+ free (macosx_out) ;
+ return NULL ;
+ } ;
+
+ /* get the buffersize that the default device uses for IO */
+ count = sizeof (UInt32) ;
+ if ((err = AudioDeviceGetProperty (macosx_out->device, 0, false, kAudioDevicePropertyBufferSize,
+ &count, &(macosx_out->buffer_size))) != noErr)
+ { printf ("AudioDeviceGetProperty (AudioDeviceGetProperty) failed.\n") ;
+ free (macosx_out) ;
+ return NULL ;
+ } ;
+
+ /* get a description of the data format used by the default device */
+ count = sizeof (AudioStreamBasicDescription) ;
+ if ((err = AudioDeviceGetProperty (macosx_out->device, 0, false, kAudioDevicePropertyStreamFormat,
+ &count, &(macosx_out->format))) != noErr)
+ { printf ("AudioDeviceGetProperty (kAudioDevicePropertyStreamFormat) failed.\n") ;
+ free (macosx_out) ;
+ return NULL ;
+ } ;
+
+ macosx_out->format.mSampleRate = samplerate ;
+ macosx_out->format.mChannelsPerFrame = channels ;
+
+ if ((err = AudioDeviceSetProperty (macosx_out->device, NULL, 0, false, kAudioDevicePropertyStreamFormat,
+ sizeof (AudioStreamBasicDescription), &(macosx_out->format))) != noErr)
+ { printf ("AudioDeviceSetProperty (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 = AudioDeviceAddIOProc (macosx_out->device, macosx_audio_out_callback,
+ (void *) macosx_out)) != noErr)
+ { printf ("AudioDeviceAddIOProc failed.\n") ;
+ free (macosx_out) ;
+ return NULL ;
+ } ;
+
+ return (MACOSX_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 = AudioDeviceRemoveIOProc (macosx_out->device, macosx_audio_out_callback) ;
+ 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 k, size, frame_count, read_count ;
+ float *buffer ;
+
+ 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))
+
+#include <windows.h>
+#include <mmsystem.h>
+
+#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 data, DWORD param1, DWORD 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 = 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) win32_audio_out_callback,
+ (DWORD) win32_out, CALLBACK_FUNCTION) ;
+ if (error)
+ { puts ("waveOutOpen failed.") ;
+ free (win32_out) ;
+ return NULL ;
+ } ;
+
+ waveOutPause (win32_out->hwave) ;
+
+ return (WIN32_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) win32_out, 0, 0) ;
+ win32_audio_out_callback (0, MM_WOM_DONE, (DWORD) 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 data, DWORD param1, DWORD 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. */
+ sample_count = win32_out->bufferlen ;
+ frame_count = sample_count / 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] = lrint (32767.0 * win32_out->float_buffer [k]) ;
+
+ if (read_count > 0)
+ { /* Fix buffer length is only a partial block. */
+ if (read_count * 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] = 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.
+*/
+
+AUDIO_OUT *
+audio_open (int channels, int samplerate)
+{
+#if defined (__linux__)
+ #if HAVE_ALSA_ASOUNDLIB_H
+ if (access ("/proc/asound/cards", R_OK) == 0)
+ return alsa_open (channels, samplerate) ;
+ #endif
+ return opensoundsys_open (channels, samplerate) ;
+#elif (defined (__MACH__) && defined (__APPLE__))
+ return macosx_open (channels, samplerate) ;
+#elif (defined (sun) && defined (unix))
+ return solaris_open (channels, samplerate) ;
+#elif (defined (_WIN32) || defined (WIN32))
+ return win32_open (channels, samplerate) ;
+#else
+ #warning "*** Playing sound not yet supported on this platform."
+ #warning "*** Please feel free to submit a patch."
+ printf ("Error : Playing sound not yet supported on this platform.\n") ;
+ return NULL ;
+#endif
+
+
+ return NULL ;
+} /* audio_open */
+
+void
+audio_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
+{
+
+ if (callback == NULL)
+ { printf ("Error : bad callback pointer.\n") ;
+ return ;
+ } ;
+
+ if (audio_out == NULL)
+ { printf ("Error : bad audio_out pointer.\n") ;
+ return ;
+ } ;
+
+ if (callback_data == NULL)
+ { printf ("Error : bad callback_data pointer.\n") ;
+ return ;
+ } ;
+
+#if defined (__linux__)
+ #if HAVE_ALSA_ASOUNDLIB_H
+ if (audio_out->magic == ALSA_MAGIC)
+ alsa_play (callback, audio_out, callback_data) ;
+ #endif
+ opensoundsys_play (callback, audio_out, callback_data) ;
+#elif (defined (__MACH__) && defined (__APPLE__))
+ macosx_play (callback, audio_out, callback_data) ;
+#elif (defined (sun) && defined (unix))
+ solaris_play (callback, audio_out, callback_data) ;
+#elif (defined (_WIN32) || defined (WIN32))
+ win32_play (callback, audio_out, callback_data) ;
+#else
+ #warning "*** Playing sound not yet supported on this platform."
+ #warning "*** Please feel free to submit a patch."
+ printf ("Error : Playing sound not yet supported on this platform.\n") ;
+ return ;
+#endif
+
+ return ;
+} /* audio_play */
+
+void
+audio_close (AUDIO_OUT *audio_out)
+{
+#if defined (__linux__)
+ #if HAVE_ALSA_ASOUNDLIB_H
+ if (audio_out->magic == ALSA_MAGIC)
+ alsa_close (audio_out) ;
+ #endif
+ opensoundsys_close (audio_out) ;
+#elif (defined (__MACH__) && defined (__APPLE__))
+ macosx_close (audio_out) ;
+#elif (defined (sun) && defined (unix))
+ solaris_close (audio_out) ;
+#elif (defined (_WIN32) || defined (WIN32))
+ win32_close (audio_out) ;
+#else
+ #warning "*** Playing sound not yet supported on this platform."
+ #warning "*** Please feel free to submit a patch."
+ printf ("Error : Playing sound not yet supported on this platform.\n") ;
+ return ;
+#endif
+
+ return ;
+} /* audio_close */
+
+#else /* (HAVE_SNDFILE == 0) */
+
+/* Do not have libsndfile installed so just return. */
+
+AUDIO_OUT *
+audio_open (int channels, int samplerate)
+{
+ (void) channels ;
+ (void) samplerate ;
+
+ return NULL ;
+} /* audio_open */
+
+void
+audio_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
+{
+ (void) callback ;
+ (void) audio_out ;
+ (void) callback_data ;
+
+ return ;
+} /* audio_play */
+
+void
+audio_close (AUDIO_OUT *audio_out)
+{
+ audio_out = audio_out ;
+
+ return ;
+} /* audio_close */
+
+#endif /* HAVE_SNDFILE */
+
projects / audio-libsamplerate.git / blobdiff