/* ** Copyright (C) 1999-2009 Erik de Castro Lopo ** ** 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 #include #include #include #if HAVE_UNISTD_H #include #endif #if HAVE_ALSA_ASOUNDLIB_H #define ALSA_PCM_NEW_HW_PARAMS_API #define ALSA_PCM_NEW_SW_PARAMS_API #include #include #endif #if defined (__linux__) #include #include #include #elif (defined (__MACH__) && defined (__APPLE__)) #include #include #elif (defined (sun) && defined (unix)) #include #include #include #elif (OS_IS_WIN32 == 1) #include #include #endif #include #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__) static int linux_open_dsp_device (int channels, int srate) ; static void linux_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 = linux_open_dsp_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 ; } /* linux_play */ static int linux_open_dsp_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 ("linux_open_dsp_device : open ") ; exit (1) ; } ; stereo = 0 ; if (ioctl (fd, SNDCTL_DSP_STEREO, &stereo) == -1) { /* Fatal error */ perror ("linux_open_dsp_device : stereo ") ; exit (1) ; } ; if (ioctl (fd, SNDCTL_DSP_RESET, 0)) { perror ("linux_open_dsp_device : reset ") ; exit (1) ; } ; fmt = CPU_IS_BIG_ENDIAN ? AFMT_S16_BE : AFMT_S16_LE ; if (ioctl (fd, SNDCTL_DSP_SETFMT, &fmt) != 0) { perror ("linux_open_dsp_device : set format ") ; exit (1) ; } ; if (ioctl (fd, SNDCTL_DSP_CHANNELS, &channels) != 0) { perror ("linux_open_dsp_device : channels ") ; exit (1) ; } ; if (ioctl (fd, SNDCTL_DSP_SPEED, &srate) != 0) { perror ("linux_open_dsp_device : sample rate ") ; exit (1) ; } ; if (ioctl (fd, SNDCTL_DSP_SYNC, 0) != 0) { perror ("linux_open_dsp_device : sync ") ; exit (1) ; } ; return fd ; } /* linux_open_dsp_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 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 */ /*------------------------------------------------------------------------------ ** 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 \n\n", argv [0]) ; #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") ; /* This is the officially blessed by microsoft way but I can't get ** it to link. ** Sleep (15) ; ** Instead, use this: */ 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 linux_play (argc, argv) ; #elif (defined (__MACH__) && defined (__APPLE__)) macosx_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 */