Mercurial > hg > sv-dependency-builds
diff src/libsamplerate-0.1.9/examples/audio_out.c @ 126:4a7071416412
Current libsamplerate source
author | Chris Cannam <cannam@all-day-breakfast.com> |
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date | Tue, 18 Oct 2016 13:24:45 +0100 |
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--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/src/libsamplerate-0.1.9/examples/audio_out.c Tue Oct 18 13:24:45 2016 +0100 @@ -0,0 +1,1097 @@ +/* +** 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 */ +