ref: 83a9482e8049c7eb96a305516fe5efca570b0a3a
dir: /examples/audio_out.c/
/* ** 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 #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)) /*------------------------------------------------------------------------------ ** 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 <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. */ 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] = 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 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 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 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 */