--- /dev/null
+/*
+** Copyright (C) 1999-2011 Erik de Castro Lopo <erikd@mega-nerd.com>
+**
+** All rights reserved.
+**
+** Redistribution and use in source and binary forms, with or without
+** modification, are permitted provided that the following conditions are
+** met:
+**
+** * Redistributions of source code must retain the above copyright
+** notice, this list of conditions and the following disclaimer.
+** * Redistributions in binary form must reproduce the above copyright
+** notice, this list of conditions and the following disclaimer in
+** the documentation and/or other materials provided with the
+** distribution.
+** * Neither the author nor the names of any contributors may be used
+** to endorse or promote products derived from this software without
+** specific prior written permission.
+**
+** THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+** "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED
+** TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+** PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR
+** CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+** EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+** PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+** OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+** WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+** OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+** ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include "sfconfig.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <errno.h>
+
+#if HAVE_UNISTD_H
+#include <unistd.h>
+#endif
+
+#include <sndfile.h>
+
+#include "common.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 defined (__linux__) || defined (__FreeBSD_kernel__) || defined (__FreeBSD__)
+ #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 (HAVE_SNDIO_H)
+ #include <sndio.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
+
+#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, unsigned srate, int realtime) ;
+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 ;
+ } ;
+
+ if ((alsa_dev = alsa_open (sfinfo.channels, (unsigned) sfinfo.samplerate, SF_FALSE)) == NULL)
+ continue ;
+
+ 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_drain (alsa_dev) ;
+ snd_pcm_close (alsa_dev) ;
+
+ sf_close (sndfile) ;
+ } ;
+
+ return ;
+} /* alsa_play */
+
+static snd_pcm_t *
+alsa_open (int channels, unsigned samplerate, int realtime)
+{ const char * device = "default" ;
+ snd_pcm_t *alsa_dev = NULL ;
+ 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 ;
+
+ if (realtime)
+ { alsa_period_size = 256 ;
+ alsa_buffer_frames = 3 * alsa_period_size ;
+ }
+ else
+ { alsa_period_size = 1024 ;
+ alsa_buffer_frames = 4 * 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)) ;
+ goto catch_error ;
+ } ;
+
+ 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)) ;
+ goto catch_error ;
+ } ;
+
+ if ((err = snd_pcm_hw_params_any (alsa_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_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_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_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_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_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_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_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_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_dev, sw_params) ;
+
+ if ((err = snd_pcm_sw_params_set_start_threshold (alsa_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_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_dev) ;
+
+catch_error :
+
+ if (err < 0 && alsa_dev != NULL)
+ { snd_pcm_close (alsa_dev) ;
+ return NULL ;
+ } ;
+
+ 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 ;
+
+ 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 */
+
+#endif /* HAVE_ALSA_ASOUNDLIB_H */
+
+/*------------------------------------------------------------------------------
+** Linux/OSS functions for playing a sound.
+*/
+
+#if defined (__linux__) || defined (__FreeBSD_kernel__) || defined (__FreeBSD__)
+
+static int opensoundsys_open_device (int channels, int srate) ;
+
+static int
+opensoundsys_play (int argc, char *argv [])
+{ static short buffer [BUFFER_LEN] ;
+ SNDFILE *sndfile ;
+ SF_INFO sfinfo ;
+ int k, audio_device, readcount, writecount, 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 = opensoundsys_open_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] ;
+ writecount = write (audio_device, buffer, readcount * sizeof (short)) ;
+ } ;
+ }
+ else
+ { while ((readcount = sf_read_short (sndfile, buffer, BUFFER_LEN)))
+ writecount = write (audio_device, buffer, readcount * sizeof (short)) ;
+ } ;
+
+ if (ioctl (audio_device, SNDCTL_DSP_POST, 0) == -1)
+ perror ("ioctl (SNDCTL_DSP_POST) ") ;
+
+ if (ioctl (audio_device, SNDCTL_DSP_SYNC, 0) == -1)
+ perror ("ioctl (SNDCTL_DSP_SYNC) ") ;
+
+ close (audio_device) ;
+
+ sf_close (sndfile) ;
+ } ;
+
+ return writecount ;
+} /* opensoundsys_play */
+
+static int
+opensoundsys_open_device (int channels, int srate)
+{ int fd, stereo, fmt ;
+
+ if ((fd = open ("/dev/dsp", O_WRONLY, 0)) == -1 &&
+ (fd = open ("/dev/sound/dsp", O_WRONLY, 0)) == -1)
+ { perror ("opensoundsys_open_device : open ") ;
+ exit (1) ;
+ } ;
+
+ stereo = 0 ;
+ if (ioctl (fd, SNDCTL_DSP_STEREO, &stereo) == -1)
+ { /* Fatal error */
+ perror ("opensoundsys_open_device : stereo ") ;
+ exit (1) ;
+ } ;
+
+ if (ioctl (fd, SNDCTL_DSP_RESET, 0))
+ { perror ("opensoundsys_open_device : reset ") ;
+ exit (1) ;
+ } ;
+
+ fmt = CPU_IS_BIG_ENDIAN ? AFMT_S16_BE : AFMT_S16_LE ;
+ if (ioctl (fd, SNDCTL_DSP_SETFMT, &fmt) != 0)
+ { perror ("opensoundsys_open_device : set format ") ;
+ exit (1) ;
+ } ;
+
+ if (ioctl (fd, SNDCTL_DSP_CHANNELS, &channels) != 0)
+ { perror ("opensoundsys_open_device : channels ") ;
+ exit (1) ;
+ } ;
+
+ if (ioctl (fd, SNDCTL_DSP_SPEED, &srate) != 0)
+ { perror ("opensoundsys_open_device : sample rate ") ;
+ exit (1) ;
+ } ;
+
+ if (ioctl (fd, SNDCTL_DSP_SYNC, 0) != 0)
+ { perror ("opensoundsys_open_device : sync ") ;
+ exit (1) ;
+ } ;
+
+ return fd ;
+} /* opensoundsys_open_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 ;
+
+ audio_data = (MacOSXAudioData*) client_data ;
+
+ size = data_out->mBuffers [0].mDataByteSize ;
+ sample_count = size / sizeof (float) ;
+
+ buffer = (float*) data_out->mBuffers [0].mData ;
+
+ if (audio_data->fake_stereo != 0)
+ { read_count = sf_read_float (audio_data->sndfile, buffer, sample_count / 2) ;
+
+ for (k = read_count - 1 ; k >= 0 ; k--)
+ { buffer [2 * k ] = buffer [k] ;
+ buffer [2 * k + 1] = buffer [k] ;
+ } ;
+ read_count *= 2 ;
+ }
+ else
+ read_count = sf_read_float (audio_data->sndfile, buffer, sample_count) ;
+
+ /* Fill the remainder with zeroes. */
+ if (read_count < sample_count)
+ { if (audio_data->fake_stereo == 0)
+ memset (&(buffer [read_count]), 0, (sample_count - read_count) * sizeof (float)) ;
+ /* Tell the main application to terminate. */
+ audio_data->done_playing = SF_TRUE ;
+ } ;
+
+ return noErr ;
+} /* macosx_audio_out_callback */
+
+static void
+macosx_play (int argc, char *argv [])
+{ MacOSXAudioData audio_data ;
+ OSStatus err ;
+ UInt32 count, buffer_size ;
+ int k ;
+
+ audio_data.fake_stereo = 0 ;
+ audio_data.device = kAudioDeviceUnknown ;
+
+ /* get the default output device for the HAL */
+ count = sizeof (AudioDeviceID) ;
+ if ((err = AudioHardwareGetProperty (kAudioHardwarePropertyDefaultOutputDevice,
+ &count, (void *) &(audio_data.device))) != noErr)
+ { printf ("AudioHardwareGetProperty (kAudioDevicePropertyDefaultOutputDevice) failed.\n") ;
+ return ;
+ } ;
+
+ /* get the buffersize that the default device uses for IO */
+ count = sizeof (UInt32) ;
+ if ((err = AudioDeviceGetProperty (audio_data.device, 0, false, kAudioDevicePropertyBufferSize,
+ &count, &buffer_size)) != noErr)
+ { printf ("AudioDeviceGetProperty (kAudioDevicePropertyBufferSize) failed.\n") ;
+ return ;
+ } ;
+
+ /* get a description of the data format used by the default device */
+ count = sizeof (AudioStreamBasicDescription) ;
+ if ((err = AudioDeviceGetProperty (audio_data.device, 0, false, kAudioDevicePropertyStreamFormat,
+ &count, &(audio_data.format))) != noErr)
+ { printf ("AudioDeviceGetProperty (kAudioDevicePropertyStreamFormat) failed.\n") ;
+ return ;
+ } ;
+
+ /* Base setup completed. Now play files. */
+ for (k = 1 ; k < argc ; k++)
+ { printf ("Playing %s\n", argv [k]) ;
+ if (! (audio_data.sndfile = sf_open (argv [k], SFM_READ, &(audio_data.sfinfo))))
+ { puts (sf_strerror (NULL)) ;
+ continue ;
+ } ;
+
+ if (audio_data.sfinfo.channels < 1 || audio_data.sfinfo.channels > 2)
+ { printf ("Error : channels = %d.\n", audio_data.sfinfo.channels) ;
+ continue ;
+ } ;
+
+ audio_data.format.mSampleRate = audio_data.sfinfo.samplerate ;
+
+ if (audio_data.sfinfo.channels == 1)
+ { audio_data.format.mChannelsPerFrame = 2 ;
+ audio_data.fake_stereo = 1 ;
+ }
+ else
+ audio_data.format.mChannelsPerFrame = audio_data.sfinfo.channels ;
+
+ if ((err = AudioDeviceSetProperty (audio_data.device, NULL, 0, false, kAudioDevicePropertyStreamFormat,
+ sizeof (AudioStreamBasicDescription), &(audio_data.format))) != noErr)
+ { printf ("AudioDeviceSetProperty (kAudioDevicePropertyStreamFormat) failed.\n") ;
+ return ;
+ } ;
+
+ /* we want linear pcm */
+ if (audio_data.format.mFormatID != kAudioFormatLinearPCM)
+ return ;
+
+ /* Fire off the device. */
+ if ((err = AudioDeviceAddIOProc (audio_data.device, macosx_audio_out_callback,
+ (void *) &audio_data)) != noErr)
+ { printf ("AudioDeviceAddIOProc failed.\n") ;
+ return ;
+ } ;
+
+ err = AudioDeviceStart (audio_data.device, macosx_audio_out_callback) ;
+ if (err != noErr)
+ return ;
+
+ audio_data.done_playing = SF_FALSE ;
+
+ while (audio_data.done_playing == SF_FALSE)
+ usleep (10 * 1000) ; /* 10 000 milliseconds. */
+
+ if ((err = AudioDeviceStop (audio_data.device, macosx_audio_out_callback)) != noErr)
+ { printf ("AudioDeviceStop failed.\n") ;
+ return ;
+ } ;
+
+ err = AudioDeviceRemoveIOProc (audio_data.device, macosx_audio_out_callback) ;
+ if (err != noErr)
+ { printf ("AudioDeviceRemoveIOProc failed.\n") ;
+ return ;
+ } ;
+
+ sf_close (audio_data.sndfile) ;
+ } ;
+
+ return ;
+} /* macosx_play */
+
+#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 (OS_IS_WIN32 == 1)
+
+#define WIN32_BUFFER_LEN (1<<15)
+
+typedef struct
+{ HWAVEOUT hwave ;
+ WAVEHDR whdr [2] ;
+
+ CRITICAL_SECTION mutex ; /* to control access to BuffersInUSe */
+ HANDLE Event ; /* signal that a buffer is free */
+
+ short buffer [WIN32_BUFFER_LEN / sizeof (short)] ;
+ int current, bufferlen ;
+ int BuffersInUse ;
+
+ SNDFILE *sndfile ;
+ SF_INFO sfinfo ;
+
+ sf_count_t remaining ;
+} Win32_Audio_Data ;
+
+
+static void
+win32_play_data (Win32_Audio_Data *audio_data)
+{ int thisread, readcount ;
+
+ /* fill a buffer if there is more data and we can read it sucessfully */
+ readcount = (audio_data->remaining > audio_data->bufferlen) ? audio_data->bufferlen : (int) audio_data->remaining ;
+
+ thisread = (int) sf_read_short (audio_data->sndfile, (short *) (audio_data->whdr [audio_data->current].lpData), readcount) ;
+
+ audio_data->remaining -= thisread ;
+
+ if (thisread > 0)
+ { /* Fix buffer length if this is only a partial block. */
+ if (thisread < audio_data->bufferlen)
+ audio_data->whdr [audio_data->current].dwBufferLength = thisread * sizeof (short) ;
+
+ /* Queue the WAVEHDR */
+ waveOutWrite (audio_data->hwave, (LPWAVEHDR) &(audio_data->whdr [audio_data->current]), sizeof (WAVEHDR)) ;
+
+ /* count another buffer in use */
+ EnterCriticalSection (&audio_data->mutex) ;
+ audio_data->BuffersInUse ++ ;
+ LeaveCriticalSection (&audio_data->mutex) ;
+
+ /* use the other buffer next time */
+ audio_data->current = (audio_data->current + 1) % 2 ;
+ } ;
+
+ return ;
+} /* win32_play_data */
+
+static void CALLBACK
+win32_audio_out_callback (HWAVEOUT hwave, UINT msg, DWORD_PTR data, DWORD param1, DWORD param2)
+{ Win32_Audio_Data *audio_data ;
+
+ /* Prevent compiler warnings. */
+ hwave = hwave ;
+ param1 = param2 ;
+
+ if (data == 0)
+ return ;
+
+ /*
+ ** I consider this technique of passing a pointer via an integer as
+ ** fundamentally broken but thats the way microsoft has defined the
+ ** interface.
+ */
+ audio_data = (Win32_Audio_Data*) data ;
+
+ /* let main loop know a buffer is free */
+ if (msg == MM_WOM_DONE)
+ { EnterCriticalSection (&audio_data->mutex) ;
+ audio_data->BuffersInUse -- ;
+ LeaveCriticalSection (&audio_data->mutex) ;
+ SetEvent (audio_data->Event) ;
+ } ;
+
+ return ;
+} /* win32_audio_out_callback */
+
+static void
+win32_play (int argc, char *argv [])
+{ Win32_Audio_Data audio_data ;
+
+ WAVEFORMATEX wf ;
+ int k, error ;
+
+ audio_data.sndfile = NULL ;
+ audio_data.hwave = 0 ;
+
+ for (k = 1 ; k < argc ; k++)
+ { printf ("Playing %s\n", argv [k]) ;
+
+ if (! (audio_data.sndfile = sf_open (argv [k], SFM_READ, &(audio_data.sfinfo))))
+ { puts (sf_strerror (NULL)) ;
+ continue ;
+ } ;
+
+ audio_data.remaining = audio_data.sfinfo.frames * audio_data.sfinfo.channels ;
+ audio_data.current = 0 ;
+
+ InitializeCriticalSection (&audio_data.mutex) ;
+ audio_data.Event = CreateEvent (0, FALSE, FALSE, 0) ;
+
+ wf.nChannels = audio_data.sfinfo.channels ;
+ wf.wFormatTag = WAVE_FORMAT_PCM ;
+ wf.cbSize = 0 ;
+ wf.wBitsPerSample = 16 ;
+
+ wf.nSamplesPerSec = audio_data.sfinfo.samplerate ;
+
+ wf.nBlockAlign = audio_data.sfinfo.channels * sizeof (short) ;
+
+ wf.nAvgBytesPerSec = wf.nBlockAlign * wf.nSamplesPerSec ;
+
+ error = waveOutOpen (&(audio_data.hwave), WAVE_MAPPER, &wf, (DWORD_PTR) win32_audio_out_callback,
+ (DWORD_PTR) &audio_data, CALLBACK_FUNCTION) ;
+ if (error)
+ { puts ("waveOutOpen failed.") ;
+ audio_data.hwave = 0 ;
+ continue ;
+ } ;
+
+ audio_data.whdr [0].lpData = (char*) audio_data.buffer ;
+ audio_data.whdr [1].lpData = ((char*) audio_data.buffer) + sizeof (audio_data.buffer) / 2 ;
+
+ audio_data.whdr [0].dwBufferLength = sizeof (audio_data.buffer) / 2 ;
+ audio_data.whdr [1].dwBufferLength = sizeof (audio_data.buffer) / 2 ;
+
+ audio_data.whdr [0].dwFlags = 0 ;
+ audio_data.whdr [1].dwFlags = 0 ;
+
+ /* length of each audio buffer in samples */
+ audio_data.bufferlen = sizeof (audio_data.buffer) / 2 / sizeof (short) ;
+
+ /* Prepare the WAVEHDRs */
+ if ((error = waveOutPrepareHeader (audio_data.hwave, &(audio_data.whdr [0]), sizeof (WAVEHDR))))
+ { printf ("waveOutPrepareHeader [0] failed : %08X\n", error) ;
+ waveOutClose (audio_data.hwave) ;
+ continue ;
+ } ;
+
+ if ((error = waveOutPrepareHeader (audio_data.hwave, &(audio_data.whdr [1]), sizeof (WAVEHDR))))
+ { printf ("waveOutPrepareHeader [1] failed : %08X\n", error) ;
+ waveOutUnprepareHeader (audio_data.hwave, &(audio_data.whdr [0]), sizeof (WAVEHDR)) ;
+ waveOutClose (audio_data.hwave) ;
+ continue ;
+ } ;
+
+ /* Fill up both buffers with audio data */
+ audio_data.BuffersInUse = 0 ;
+ win32_play_data (&audio_data) ;
+ win32_play_data (&audio_data) ;
+
+ /* loop until both buffers are released */
+ while (audio_data.BuffersInUse > 0)
+ {
+ /* wait for buffer to be released */
+ WaitForSingleObject (audio_data.Event, INFINITE) ;
+
+ /* refill the buffer if there is more data to play */
+ win32_play_data (&audio_data) ;
+ } ;
+
+ waveOutUnprepareHeader (audio_data.hwave, &(audio_data.whdr [0]), sizeof (WAVEHDR)) ;
+ waveOutUnprepareHeader (audio_data.hwave, &(audio_data.whdr [1]), sizeof (WAVEHDR)) ;
+
+ waveOutClose (audio_data.hwave) ;
+ audio_data.hwave = 0 ;
+
+ DeleteCriticalSection (&audio_data.mutex) ;
+
+ sf_close (audio_data.sndfile) ;
+ } ;
+
+} /* win32_play */
+
+#endif /* Win32 */
+
+/*------------------------------------------------------------------------------
+** OpenBDS's sndio.
+*/
+
+#if defined (HAVE_SNDIO_H)
+
+static void
+sndio_play (int argc, char *argv [])
+{ struct sio_hdl *hdl ;
+ struct sio_par par ;
+ short buffer [BUFFER_LEN] ;
+ SNDFILE *sndfile ;
+ SF_INFO sfinfo ;
+ int k, readcount ;
+
+ for (k = 1 ; k < argc ; k++)
+ { 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 ;
+ } ;
+
+ if ((hdl = sio_open (NULL, SIO_PLAY, 0)) == NULL)
+ { fprintf (stderr, "open sndio device failed") ;
+ return ;
+ } ;
+
+ sio_initpar (&par) ;
+ par.rate = sfinfo.samplerate ;
+ par.pchan = sfinfo.channels ;
+ par.bits = 16 ;
+ par.sig = 1 ;
+ par.le = SIO_LE_NATIVE ;
+
+ if (! sio_setpar (hdl, &par) || ! sio_getpar (hdl, &par))
+ { fprintf (stderr, "set sndio params failed") ;
+ return ;
+ } ;
+
+ if (! sio_start (hdl))
+ { fprintf (stderr, "sndio start failed") ;
+ return ;
+ } ;
+
+ while ((readcount = sf_read_short (sndfile, buffer, BUFFER_LEN)))
+ sio_write (hdl, buffer, readcount * sizeof (short)) ;
+
+ sio_close (hdl) ;
+ } ;
+
+ return ;
+} /* sndio_play */
+
+#endif /* sndio */
+
+/*------------------------------------------------------------------------------
+** Solaris.
+*/
+
+#if (defined (sun) && defined (unix)) /* ie Solaris */
+
+static void
+solaris_play (int argc, char *argv [])
+{ static short buffer [BUFFER_LEN] ;
+ audio_info_t audio_info ;
+ SNDFILE *sndfile ;
+ SF_INFO sfinfo ;
+ unsigned long delay_time ;
+ long k, start_count, output_count, write_count, read_count ;
+ int audio_fd, error, done ;
+
+ for (k = 1 ; k < argc ; k++)
+ { 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 ;
+ } ;
+
+ /* open the audio device - write only, non-blocking */
+ if ((audio_fd = open ("/dev/audio", O_WRONLY | O_NONBLOCK)) < 0)
+ { perror ("open (/dev/audio) failed") ;
+ return ;
+ } ;
+
+ /* Retrive standard values. */
+ AUDIO_INITINFO (&audio_info) ;
+
+ audio_info.play.sample_rate = sfinfo.samplerate ;
+ audio_info.play.channels = sfinfo.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 (audio_fd, AUDIO_SETINFO, &audio_info)))
+ { perror ("ioctl (AUDIO_SETINFO) failed") ;
+ return ;
+ } ;
+
+ /* Delay time equal to 1/4 of a buffer in microseconds. */
+ delay_time = (BUFFER_LEN * 1000000) / (audio_info.play.sample_rate * 4) ;
+
+ done = 0 ;
+ while (! done)
+ { read_count = sf_read_short (sndfile, buffer, BUFFER_LEN) ;
+ if (read_count < BUFFER_LEN)
+ { memset (&(buffer [read_count]), 0, (BUFFER_LEN - read_count) * sizeof (short)) ;
+ /* Tell the main application to terminate. */
+ done = SF_TRUE ;
+ } ;
+
+ start_count = 0 ;
+ output_count = BUFFER_LEN * sizeof (short) ;
+
+ while (output_count > 0)
+ { /* write as much data as possible */
+ write_count = write (audio_fd, &(buffer [start_count]), output_count) ;
+ if (write_count > 0)
+ { output_count -= write_count ;
+ start_count += write_count ;
+ }
+ else
+ { /* Give the audio output time to catch up. */
+ usleep (delay_time) ;
+ } ;
+ } ; /* while (outpur_count > 0) */
+ } ; /* while (! done) */
+
+ close (audio_fd) ;
+ } ;
+
+ return ;
+} /* solaris_play */
+
+#endif /* Solaris */
+
+/*==============================================================================
+** Main function.
+*/
+
+int
+main (int argc, char *argv [])
+{
+ if (argc < 2)
+ {
+ printf ("\nUsage : %s <input sound file>\n\n", program_name (argv [0])) ;
+ printf (" Using %s.\n\n", sf_version_string ()) ;
+#if (OS_IS_WIN32 == 1)
+ printf ("This is a Unix style command line application which\n"
+ "should be run in a MSDOS box or Command Shell window.\n\n") ;
+ printf ("Sleeping for 5 seconds before exiting.\n\n") ;
+
+ Sleep (5 * 1000) ;
+#endif
+ return 1 ;
+ } ;
+
+#if defined (__linux__)
+ #if HAVE_ALSA_ASOUNDLIB_H
+ if (access ("/proc/asound/cards", R_OK) == 0)
+ alsa_play (argc, argv) ;
+ else
+ #endif
+ opensoundsys_play (argc, argv) ;
+#elif defined (__FreeBSD_kernel__) || defined (__FreeBSD__)
+ opensoundsys_play (argc, argv) ;
+#elif (defined (__MACH__) && defined (__APPLE__))
+ macosx_play (argc, argv) ;
+#elif defined HAVE_SNDIO_H
+ sndio_play (argc, argv) ;
+#elif (defined (sun) && defined (unix))
+ solaris_play (argc, argv) ;
+#elif (OS_IS_WIN32 == 1)
+ win32_play (argc, argv) ;
+#elif defined (__BEOS__)
+ printf ("This program cannot be compiled on BeOS.\n") ;
+ printf ("Instead, compile the file sfplay_beos.cpp.\n") ;
+ return 1 ;
+#else
+ puts ("*** Playing sound not yet supported on this platform.") ;
+ puts ("*** Please feel free to submit a patch.") ;
+ return 1 ;
+#endif
+
+ return 0 ;
+} /* main */
+
projects / Faustine.git / blobdiff