/* Sound_audio.c * * Copyright (C) 1992-2002 Paul Boersma * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or (at * your option) any later version. * * This program is distributed in the hope that it will be useful, but * WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. * See the GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ /* * pb 2002/05/28 * pb 2002/07/16 GPL */ #include #ifdef linux #define DEV_AUDIO "/dev/dsp" #else #define DEV_AUDIO "/dev/audio" #endif /* * In the new Mac SDK/ CodeWarrior, the following must be included before and ...??? */ #include "Sound.h" #include "Resources.h" #if defined (sgi) #include #include /* sginap (): nap while waiting for a sound to finish playing. */ #elif defined (linux) #include #include #include /* ioctl */ #include /* open write close read */ #elif defined (sun) #include #include #include /* ioctl open write close */ #if defined (sun4) #include #include #else #include #endif #elif defined (HPUX) #include #include #include #include #include #elif defined (macintosh) #include "macport_on.h" #include #include #include #ifndef __MACH__ #include #endif #include #include "macport_off.h" #elif defined (_WIN32) #include #include #else #include #endif static int ulaw2linear [] = { -32124, -31100, -30076, -29052, -28028, -27004, -25980, -24956, -23932, -22908, -21884, -20860, -19836, -18812, -17788, -16764, -15996, -15484, -14972, -14460, -13948, -13436, -12924, -12412, -11900, -11388, -10876, -10364, -9852, -9340, -8828, -8316, -7932, -7676, -7420, -7164, -6908, -6652, -6396, -6140, -5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092, -3900, -3772, -3644, -3516, -3388, -3260, -3132, -3004, -2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980, -1884, -1820, -1756, -1692, -1628, -1564, -1500, -1436, -1372, -1308, -1244, -1180, -1116, -1052, -988, -924, -876, -844, -812, -780, -748, -716, -684, -652, -620, -588, -556, -524, -492, -460, -428, -396, -372, -356, -340, -324, -308, -292, -276, -260, -244, -228, -212, -196, -180, -164, -148, -132, -120, -112, -104, -96, -88, -80, -72, -64, -56, -48, -40, -32, -24, -16, -8, 0, 32124, 31100, 30076, 29052, 28028, 27004, 25980, 24956, 23932, 22908, 21884, 20860, 19836, 18812, 17788, 16764, 15996, 15484, 14972, 14460, 13948, 13436, 12924, 12412, 11900, 11388, 10876, 10364, 9852, 9340, 8828, 8316, 7932, 7676, 7420, 7164, 6908, 6652, 6396, 6140, 5884, 5628, 5372, 5116, 4860, 4604, 4348, 4092, 3900, 3772, 3644, 3516, 3388, 3260, 3132, 3004, 2876, 2748, 2620, 2492, 2364, 2236, 2108, 1980, 1884, 1820, 1756, 1692, 1628, 1564, 1500, 1436, 1372, 1308, 1244, 1180, 1116, 1052, 988, 924, 876, 844, 812, 780, 748, 716, 684, 652, 620, 588, 556, 524, 492, 460, 428, 396, 372, 356, 340, 324, 308, 292, 276, 260, 244, 228, 212, 196, 180, 164, 148, 132, 120, 112, 104, 96, 88, 80, 72, 64, 56, 48, 40, 32, 24, 16, 8, 0 }; #ifdef macintosh #ifndef __MWERKS__ typedef SndListPtr *SndListHandle; #endif #endif Sound Sound_recordFixedTime (int inputSource, double gain, double balance, double sampleRate, double duration) { Sound me = NULL; long numberOfSamples, i; short *buffer = NULL; int mulaw = FALSE; int can16bit = TRUE; /* Declare system-dependent data structures. */ #if defined (sgi) ALconfig config; ALport port; #elif defined (macintosh) SPB spb; long refNum, soundFeatures; (void) gain; (void) balance; can16bit = Gestalt (gestaltSoundAttr, & soundFeatures) == 0 && (soundFeatures & (1 << gestalt16BitAudioSupport)) != 0; #elif defined (_WIN32) HWAVEIN hWaveIn = 0; WAVEFORMATEX waveFormat; WAVEHDR waveHeader; MMRESULT err; (void) inputSource; (void) gain; (void) balance; #else int fd = -1; /* Other systems use stream I/O with a file descriptor. */ #if defined (sun) struct audio_info info; #elif defined (HPUX) struct audio_describe info; struct audio_gains gains; struct audio_limits limits; #elif defined (linux) #define min(a,b) a > b ? b : a int dev_mask; int fd_mixer; int val; #endif #endif /* Check representation of shorts. */ if (sizeof (short) != 2) return Melder_errorp ("(Sound_record:) Cannot record a sound on this machine."); /* Check sample rate. */ #if defined (sgi) if (sampleRate != 8000 && sampleRate != 9800 && sampleRate != 11025 && sampleRate != 16000 && sampleRate != 22050 && sampleRate != 32000 && sampleRate != 44100 && sampleRate != 48000 && sampleRate != 0) #elif defined (macintosh) if (sampleRate != 11025 && sampleRate != 22050 && sampleRate != 44100) #elif defined (sun) || defined (linux) if (sampleRate != 8000 && sampleRate != 11025 && sampleRate != 16000 && sampleRate != 22050 && sampleRate != 32000 && sampleRate != 44100 && sampleRate != 48000) #elif defined (HPUX) if (sampleRate != 8000 && sampleRate != 11025 && sampleRate != 16000 && sampleRate != 22050 && sampleRate != 32000 && sampleRate != 44100 && sampleRate != 48000 && sampleRate != 5512) #elif defined (_WIN32) if (sampleRate != 8000 && sampleRate != 11025 && sampleRate != 16000 && sampleRate != 22050 && sampleRate != 32000 && sampleRate != 44100 && sampleRate != 48000) #endif return Melder_errorp ("(Sound_record:) Audio hardware does not support sample rate %.8g.", sampleRate); /* Create a buffer for recording, and the resulting sound. */ numberOfSamples = floor (sampleRate * duration + 0.5); if (numberOfSamples < 1) return Melder_errorp ("(Sound_record:) Duration too short."); if (! (buffer = NUMsvector (1, numberOfSamples))) return NULL; if (! (me = Sound_createSimple (numberOfSamples / sampleRate, sampleRate))) { NUMsvector_free (buffer, 1); return NULL; } Melder_assert (my nx == numberOfSamples); /* * Open phase 1. * On some systems, the info is filled in before the audio port is opened. * On other systems, the info is filled in after the port is opened. */ #if defined (sgi) config = ALnewconfig (); if (! config) return Melder_errorp ("(Sound_record:) Do not know how to record a sound on this machine."); /* We do not open the port yet, because the info is an argument to opening the port. */ #elif defined (macintosh) if (SPBOpenDevice (NULL, siWritePermission, & refNum) != noErr) return Melder_errorp ("(Sound_record:) Cannot open audio input device."); #elif defined (_WIN32) #else /* We must open the port now, because we use an ioctl to set the info to an open port. */ fd = open (DEV_AUDIO, O_RDONLY); if (fd == -1) { if (errno == EBUSY) Melder_error ("(Sound_record:) Audio device in use by another program."); else #ifdef linux Melder_error ("(Sound_record:) Cannot open audio device.\n" "Consult /usr/doc/HOWTO/Sound-HOWTO."); #else Melder_error ("(Sound_record:) Cannot open audio device."); #endif goto error; } /* The device immediately started recording into its buffer, but probably at the wrong rate etc. */ /* Pause and flush this rubbish. */ #if defined (sun) AUDIO_INITINFO (& info); info. record. pause = 1; ioctl (fd, AUDIO_SETINFO, & info); /* Pause! */ ioctl (fd, I_FLUSH, FLUSHR); /* Discard buffers! */ #elif defined (HPUX) ioctl (fd, AUDIO_RESET, RESET_RX_BUF | RESET_RX_OVF); ioctl (fd, AUDIO_PAUSE, AUDIO_RECEIVE); #elif defined (linux) ioctl (fd, SNDCTL_DSP_RESET, NULL); #endif #endif /* Set the input source; the default is the microphone. */ #if defined (sgi) if (inputSource == 0) { /* Do not change input source: e.g., use the one set by the Audio Control Panel. */ } else { long params [2]; params [0] = AL_INPUT_SOURCE; params [1] = inputSource == 2 ? AL_INPUT_LINE : inputSource == 3 ? AL_INPUT_DIGITAL : AL_INPUT_MIC; ALsetparams (AL_DEFAULT_DEVICE, params, 2); } #elif defined (macintosh) { short macInputSource = inputSource; OSErr err = SPBSetDeviceInfo (refNum, siInputSource, & macInputSource); if (err != noErr) { static int notified = FALSE; if (! notified) Melder_warning ("(Sound_record:) " "Cannot set input source to %d. Error #%d.", inputSource, err); notified = TRUE; } } #elif defined (sun) AUDIO_INITINFO (& info); info. record. port = inputSource == 2 ? AUDIO_LINE_IN : AUDIO_MICROPHONE; /* No digital. */ if (ioctl (fd, AUDIO_SETINFO, & info) == -1) { Melder_error ("(Sound_record:) Cannot set port."); goto error; } #elif defined (HPUX) if (ioctl (fd, AUDIO_SET_INPUT, inputSource == 2 ? AUDIO_IN_LINE : AUDIO_IN_MIKE) == -1) { Melder_error ("(Sound_record:) Cannot set input source."); goto error; } #elif defined (linux) fd_mixer = open("/dev/mixer", O_WRONLY); if (fd_mixer == -1) { Melder_error ("(Sound_record:) Cannot open /dev/mixer."); goto error; } if (ioctl(fd_mixer, SOUND_MIXER_READ_RECMASK, &dev_mask) == -1) { Melder_error ("(Sound_record:) Cannot access /dev/mixer."); goto error; } /* printf("%d %d %d\n", dev_mask, (dev_mask&SOUND_MASK_LINE), (dev_mask&SOUND_MASK_MIC));*/ if (inputSource == 2) { /* AUDIO_LINE_IN */ if (dev_mask&SOUND_MASK_LINE) { dev_mask = SOUND_MASK_LINE; } else { Melder_error ("(Sound_record:) Can't set LINE as recording device"); goto error; } } else { /* AUDIO_MICROPHONE */ if (dev_mask&SOUND_MASK_MIC) { dev_mask = SOUND_MASK_MIC; } else { Melder_error ("(Sound_record:) Can't set MIC as recording device"); goto error; } } /* printf("%d\n", dev_mask);*/ if (ioctl(fd_mixer, SOUND_MIXER_WRITE_RECSRC, &dev_mask) == -1) { Melder_error ("(Sound_record:) Can't set recording device in mixer"); } if (ioctl(fd_mixer, SOUND_MIXER_READ_RECSRC, &dev_mask) == -1) { Melder_error ("(Sound_record:) Can't read recording device from mixer"); } else { /* printf("%x\n",dev_mask);*/ if (dev_mask&SOUND_MASK_MIC) { inputSource = 1; } else if (dev_mask&SOUND_MASK_LINE) { inputSource = 2; } } #endif /* Set gain and balance. */ #if defined (sgi) || defined (HPUX) || defined (macintosh) /* Taken from Audio Control Panel. */ #elif defined (sun) AUDIO_INITINFO (& info); info. record. gain = gain <= 0.0 ? 0 : gain >= 1.0 ? 255 : floor (gain * 255 + 0.5); info. record. balance = balance <= 0.0 ? 0 : balance >= 1.0 ? 64 : floor (balance * 64 + 0.5); if (ioctl (fd, AUDIO_SETINFO, & info) == -1) { Melder_error ("(Sound_record:) Cannot set gain and balance."); goto error; } #elif defined (linux) val = (gain <= 0.0 ? 0 : gain >= 1.0 ? 100 : floor (gain * 100 + 0.5)); balance = balance <= 0 ? 0 : balance >= 1 ? 1 : balance; if (balance >= 0.5) { val = (int)(((int)(val*balance/(1-balance)) << 8) | val); } else { val = (int)(val | ((int)(val*(1-balance)/balance) << 8)); } val = (int)((min(2-2*balance,1))*val) | ((int)((min(2*balance,1))*val) << 8); if (inputSource == 1) { /* MIC */ if (ioctl(fd_mixer, MIXER_WRITE(SOUND_MIXER_MIC), &val) == -1) { Melder_error ("(Sound_record:) Cannot set gain and balance."); goto error; } } else { /* LINE */ if (ioctl(fd_mixer, MIXER_WRITE(SOUND_MIXER_LINE), &val) == -1) { Melder_error ("(Sound_record:) Cannot set gain and balance."); goto error; } } close(fd_mixer); #endif /* Set the sample rate. */ #if defined (sgi) if (sampleRate == 0.0) { /* Do not change sample rate. Get it from audio panel. */ long params [2]; params [0] = AL_INPUT_RATE; ALgetparams (AL_DEFAULT_DEVICE, params, 2); switch (params [1]) { case AL_RATE_8000: sampleRate = 8000; break; case 9800: sampleRate = 9800; break; case AL_RATE_11025: sampleRate = 11025; break; case AL_RATE_16000: sampleRate = 16000; break; case AL_RATE_22050: sampleRate = 22050; break; case AL_RATE_32000: sampleRate = 32000; break; case AL_RATE_44100: sampleRate = 44100; break; case AL_RATE_48000: sampleRate = 48000; break; default: return Melder_errorp ("(Sound_recordFixedTime:) Wrong sampling frequency."); } } else { /* Set sampling frequency. */ long params [2]; params [0] = AL_INPUT_RATE; params [1] = sampleRate; ALsetparams (AL_DEFAULT_DEVICE, params, 2); } #elif defined (macintosh) { unsigned long sampleRate_uf = sampleRate * 65536.0; if (SPBSetDeviceInfo (refNum, siSampleRate, & sampleRate_uf) != noErr) { Melder_error ("(Sound_record:) Cannot set sample rate to %f.", sampleRate); goto error; } } #elif defined (sun) AUDIO_INITINFO (& info); info. record. sample_rate = sampleRate; if (ioctl (fd, AUDIO_SETINFO, & info) == -1) { Melder_error ("(Sound_record:) Cannot set sample rate to %f.", sampleRate); goto error; } #elif defined (HPUX) if (ioctl (fd, AUDIO_SET_SAMPLE_RATE, (int) sampleRate) == -1) { Melder_error ("(Sound_record:) Cannot set sample rate to %d.", (int) sampleRate); goto error; } #elif defined (linux) if (ioctl (fd, SNDCTL_DSP_SPEED, &sampleRate) == -1) { Melder_error ("(Sound_record:) Cannot set sample rate to %d.", (int) sampleRate); goto error; } #elif defined (_WIN32) waveFormat. nSamplesPerSec = sampleRate; #endif /* Set the number of channels to 1 (mono). */ #if defined (sgi) ALsetchannels (config, AL_MONO); #elif defined (macintosh) { short numberOfChannels = 1; if (SPBSetDeviceInfo (refNum, siNumberChannels, & numberOfChannels) != noErr) { Melder_error ("(Sound_record:) Cannot set to mono."); goto error; } } #elif defined (sun) AUDIO_INITINFO (& info); info. record. channels = 1; if (ioctl (fd, AUDIO_SETINFO, & info) == -1) { Melder_error ("(Sound_record:) Cannot set to mono."); goto error; } #elif defined (HPUX) if (ioctl (fd, AUDIO_SET_CHANNELS, 1) == -1) { Melder_error ("(Sound_record:) Cannot set to mono."); goto error; } #elif defined (linux) val = 1; if (ioctl (fd, SNDCTL_DSP_CHANNELS, &val) == -1) { Melder_error ("(Sound_record:) Cannot set to mono."); goto error; } #elif defined (_WIN32) waveFormat. nChannels = 1; #endif /* Set the encoding to 16-bit linear (or to 8-bit linear, if 16-bit is not available). */ #if defined (sgi) ALsetwidth (config, AL_SAMPLE_16); #elif defined (macintosh) { OSType compressionType = 'NONE'; short sampleSize = can16bit ? 16 : 8; if (SPBSetDeviceInfo (refNum, siCompressionType, & compressionType) != noErr) { Melder_error ("(Sound_record:) Cannot set to linear."); goto error; } if (SPBSetDeviceInfo (refNum, siSampleSize, & sampleSize) != noErr) { Melder_error ("(Sound_record:) Cannot set to %d-bit.", sampleSize); goto error; } } #elif defined (sun) AUDIO_INITINFO (& info); info. record. precision = 16; info. record. encoding = AUDIO_ENCODING_LINEAR; if (ioctl (fd, AUDIO_SETINFO, & info) == -1) { Melder_error ("(Sound_record:) Cannot set to 16-bit linear."); goto error; } #elif defined (HPUX) if (ioctl (fd, AUDIO_SET_DATA_FORMAT, AUDIO_FORMAT_LINEAR16BIT) == -1) { Melder_error ("(Sound_record:) Cannot set 16-bit linear."); goto error; } #elif defined (linux) val = AFMT_S16_LE; if (ioctl (fd, SNDCTL_DSP_SETFMT, &val) == -1) { Melder_error ("(Sound_record:) Cannot set 16-bit linear."); goto error; } #elif defined (_WIN32) waveFormat. wFormatTag = WAVE_FORMAT_PCM; waveFormat. wBitsPerSample = 16; waveFormat. nBlockAlign = waveFormat. nChannels * waveFormat. wBitsPerSample / 8; waveFormat. nAvgBytesPerSec = waveFormat. nBlockAlign * waveFormat. nSamplesPerSec; #endif /* * Open phase 2. * This starts recording now. */ #if defined (sgi) port = ALopenport ("Sound_record", "r", config); if (! port) { return Melder_errorp ("(Sound_recordFixedTime:) Cannot open audio port."); goto error; } #elif defined (macintosh) spb. inRefNum = refNum; spb. bufferLength = spb. count = numberOfSamples * (can16bit ? 2 : 1); spb. milliseconds = 0; spb. bufferPtr = (void *) & buffer [1]; spb. completionRoutine = NULL; spb. interruptRoutine = NULL; spb. userLong = 0; spb. error = noErr; spb. unused1 = 0; if (SPBRecord (& spb, false) != noErr) { Melder_errorp ("(Sound_recordFixedTime:) Cannot create audio buffer."); goto error; } #elif defined (sun) if (info. record. encoding == AUDIO_ENCODING_ULAW) { mulaw = TRUE; sampleRate = info. record. sample_rate; numberOfSamples = floor (sampleRate * duration + 0.5); Melder_casual ("Recording with mulaw encoding at sample rate %f.", sampleRate); forget (me); me = Sound_createSimple (numberOfSamples / sampleRate, sampleRate); } AUDIO_INITINFO (& info); info. record. pause = 0; /* Resume. */ ioctl (fd, AUDIO_SETINFO, & info); #elif defined (HPUX) ioctl (fd, AUDIO_GET_LIMITS, & limits); ioctl (fd, AUDIO_SET_RXBUFSIZE, limits. max_receive_buffer_size); ioctl (fd, AUDIO_RESUME, AUDIO_RECEIVE); #elif defined (_WIN32) waveFormat. cbSize = 0; err = waveInOpen (& hWaveIn, WAVE_MAPPER, & waveFormat, 0, 0, CALLBACK_NULL); if (err != MMSYSERR_NOERROR) { Melder_error ("(Sound_record:) Error %d while opening.", err); goto error; } #endif /* Read the sound into the buffer. */ #if defined (sgi) ALreadsamps (port, & buffer [1], numberOfSamples); #elif defined (macintosh) #elif defined (_WIN32) waveHeader. dwFlags = 0; waveHeader. lpData = (char *) & buffer [1]; waveHeader. dwBufferLength = numberOfSamples * 2; waveHeader. dwLoops = 0; waveHeader. lpNext = NULL; waveHeader. reserved = 0; err = waveInPrepareHeader (hWaveIn, & waveHeader, sizeof (WAVEHDR)); if (err != MMSYSERR_NOERROR) { Melder_error ("(Sound_record:) Error %d while preparing header.", err); goto error; } err = waveInAddBuffer (hWaveIn, & waveHeader, sizeof (WAVEHDR)); if (err != MMSYSERR_NOERROR) { Melder_error ("(Sound_record:) Error %d while listening.", err); goto error; } err = waveInStart (hWaveIn); if (err != MMSYSERR_NOERROR) { Melder_error ("(Sound_record:) Error %d while starting.", err); goto error; } while (! (waveHeader. dwFlags & WHDR_DONE)) { (void) clock (); } err = waveInUnprepareHeader (hWaveIn, & waveHeader, sizeof (WAVEHDR)); if (err != MMSYSERR_NOERROR) { Melder_error ("(Sound_record:) Error %d while unpreparing header.", err); goto error; } #else if (mulaw) read (fd, & ((char *) buffer) [1], numberOfSamples); else { long bytesLeft = 2 * numberOfSamples, dbytes, bytesRead = 0; while (bytesLeft) { dbytes = read (fd, & ((char *) buffer) [2 + bytesRead], bytesLeft > 4000 ? 4000 : bytesLeft); if (dbytes <= 0) break; bytesLeft -= dbytes; bytesRead += dbytes; }; } #endif /* Copy the buffered data to the sound object, and discard the buffer. */ if (mulaw) for (i = 1; i <= numberOfSamples; i ++) my z [1] [i] = ulaw2linear [((unsigned char *) buffer) [i]] * (1.0 / 32768); else if (can16bit) for (i = 1; i <= numberOfSamples; i ++) my z [1] [i] = buffer [i] * (1.0 / 32768); else for (i = 1; i <= numberOfSamples; i ++) my z [1] [i] = ((int) ((unsigned char *) buffer) [i + 1] - 128) * (1.0 / 128); NUMsvector_free (buffer, 1); /* Close the audio device. */ #if defined (sgi) ALcloseport (port); ALfreeconfig (config); #elif defined (macintosh) SPBCloseDevice (refNum); #elif defined (_WIN32) err = waveInClose (hWaveIn); if (err != MMSYSERR_NOERROR) { Melder_error ("(Sound_record:) Error %d while closing.", err); goto error; } #else close (fd); #endif /* Hand the resulting sound to the caller. */ return me; error: forget (me); NUMsvector_free (buffer, 1); #if defined (sgi) if (port) ALcloseport (port); if (config) ALfreeconfig (config); #elif defined (macintosh) SPBCloseDevice (refNum); #elif defined (_WIN32) if (hWaveIn != 0) waveInClose (hWaveIn); #else if (fd != -1) close (fd); #endif return Melder_errorp ("Sound not recorded."); } /********** PLAYING A SOUND **********/ static struct SoundPlay { long numberOfSamples, i1, i2, zeroPadding; double tmin, tmax, dt, t1; int (*callback) (void *closure, int phase, double tmin, double tmax, double t); void *closure; short *buffer; } thePlayingSound; static int melderPlayCallback (void *closure, long samplesPlayed) { struct SoundPlay *me = (struct SoundPlay *) closure; int phase = 2; double t = samplesPlayed <= my zeroPadding ? my tmin : samplesPlayed >= my zeroPadding + my numberOfSamples ? my tmax : my t1 + (my i1 - 1.5 + samplesPlayed - my zeroPadding) * my dt; if (! Melder_isPlaying) { NUMsvector_free (my buffer, my i1 - my zeroPadding), my buffer = 0; phase = 3; } if (my callback) return my callback (my closure, phase, my tmin, my tmax, t); return 1; } void Sound_playPart (Sound me, double tmin, double tmax, int (*callback) (void *closure, int phase, double tmin, double tmax, double t), void *closure) { long ifsamp = floor (1.0 / my dx + 0.5), bestSampleRate = Melder_getBestSampleRate (ifsamp); if (ifsamp == bestSampleRate) { struct SoundPlay *thee = (struct SoundPlay *) & thePlayingSound; float *from = my z [1]; short *to; long i, i1, i2; Melder_stopPlaying (Melder_IMPLICIT); if ((thy numberOfSamples = Matrix_getWindowSamplesX (me, tmin, tmax, & i1, & i2)) < 1) return; thy tmin = tmin; thy tmax = tmax; thy dt = my dx; thy t1 = my x1; thy callback = callback; thy closure = closure; thy zeroPadding = (long) (ifsamp * Melder_getZeroPadding ()); to = NUMsvector (i1 - thy zeroPadding, i2 + thy zeroPadding); if (! to) { Melder_flushError (NULL); return; } thy i1 = i1; thy i2 = i2; thy buffer = to; for (i = i1; i <= i2; i ++) { long value = floor (from [i] * 32768.0 + 0.5); to [i] = value < -32768 ? -32768 : value > 32767 ? 32767 : value; } if (thy callback) thy callback (thy closure, 1, tmin, tmax, tmin); if (! Melder_play16 (thy buffer + i1 - thy zeroPadding, ifsamp, thy zeroPadding + thy numberOfSamples + thy zeroPadding, 1, melderPlayCallback, thee)) Melder_flushError (NULL); } else { Sound resampled = Sound_resample (me, bestSampleRate, 1); Sound_playPart (resampled, tmin, tmax, callback, closure); /* Recursively. */ forget (resampled); } } void Sound_play (Sound me, int (*playCallback) (void *playClosure, int phase, double tmin, double tmax, double t), void *playClosure) { Sound_playPart (me, my xmin, my xmax, playCallback, playClosure); } /* End of file Sound_audio.c */