/* melder_audio.cpp * * Copyright (C) 1992-2011 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 2003/08/22 used getenv ("AUDIODEV") on Sun (thanks to Michel Scheffers) * pb 2003/10/22 fake mono for Linux drivers that do not support mono * pb 2003/12/06 use sys/soundcard.h instead of linux/soundcard.h for FreeBSD compatibility * pb 2004/05/07 removed motif_mac_setNullEventWaitingTime (we nowadays use 1 clock tick everywhere anyway) * pb 2004/08/10 fake mono for Linux drivers etc, also if not asynchronous * pb 2005/02/13 added O_NDELAY when opening /dev/dsp on Linux (suggestion by Rafael Laboissiere) * pb 2005/03/31 undid previous change (four complaints that sound stopped playing) * pb 2005/05/19 redid previous change (with fctrl fix suggested by Rafael Laboissiere) * pb 2005/10/13 edition for OpenBSD * pb 2006/10/28 erased MacOS 9 stuff * pb 2006/12/16 Macintosh uses CoreAudio (via PortAudio) * pb 2007/01/03 best sample rate can be over 64 kHz * pb 2007/05/13 null pointer test for deviceInfo (thanks to Stefan de Konink) * pb 2007/08/12 wchar_t * Stefan de Konink 2007/12/02 big-endian Linux * pb 2007/12/04 enums * pb 2008/06/01 removed SPEXLAB audio server * pb 2008/06/10 made PortAudio and foreground playing optional * pb 2008/07/03 DirectSound * pb 2010/05/09 GTK * pb 2011/02/11 better message * pb 2011/04/05 C++ */ #include "melder.h" #include "Gui.h" #include "Preferences.h" #include "NUM.h" #include #define my me -> #if defined (macintosh) #include #include #elif defined (_WIN32) #include #include #elif defined (linux) #include #include #include #include #include #include #include #if defined (__OpenBSD__) || defined (__NetBSD__) #include #else #include #endif #include #endif #include "../external/portaudio/portaudio.h" static struct { enum kMelder_asynchronicityLevel maximumAsynchronicity; bool useInternalSpeaker, inputUsesPortAudio, outputUsesPortAudio, outputUsesBlocking; double silenceBefore, silenceAfter; } preferences; void Melder_audio_prefs (void) { Preferences_addEnum (L"Audio.maximumAsynchronicity", & preferences. maximumAsynchronicity, kMelder_asynchronicityLevel, DEFAULT); Preferences_addBool (L"Audio.useInternalSpeaker", & preferences. useInternalSpeaker, true); Preferences_addBool (L"Audio.outputUsesPortAudio", & preferences. outputUsesPortAudio, kMelderAudio_outputUsesPortAudio_DEFAULT); Preferences_addBool (L"Audio.outputUsesBlocking", & preferences. outputUsesBlocking, false); Preferences_addDouble (L"Audio.silenceBefore", & preferences. silenceBefore, 0.0); Preferences_addDouble (L"Audio.silenceAfter", & preferences. silenceAfter, 0.5); Preferences_addBool (L"Audio.inputUsesPortAudio", & preferences.inputUsesPortAudio, kMelderAudio_inputUsesPortAudio_DEFAULT); } void MelderAudio_setOutputMaximumAsynchronicity (enum kMelder_asynchronicityLevel maximumAsynchronicity) { MelderAudio_stopPlaying (MelderAudio_IMPLICIT); preferences. maximumAsynchronicity = maximumAsynchronicity; } enum kMelder_asynchronicityLevel MelderAudio_getOutputMaximumAsynchronicity (void) { return preferences. maximumAsynchronicity; } void MelderAudio_setInputUsesPortAudio (bool inputUsesPortAudio) { preferences. inputUsesPortAudio = inputUsesPortAudio; } bool MelderAudio_getInputUsesPortAudio (void) { return preferences. inputUsesPortAudio; } void MelderAudio_setOutputUsesPortAudio (bool outputUsesPortAudio) { MelderAudio_stopPlaying (MelderAudio_IMPLICIT); preferences. outputUsesPortAudio = outputUsesPortAudio; } bool MelderAudio_getOutputUsesPortAudio (void) { return preferences. outputUsesPortAudio; } void MelderAudio_setOutputUsesBlocking (bool outputUsesBlocking) { MelderAudio_stopPlaying (MelderAudio_IMPLICIT); preferences. outputUsesBlocking = outputUsesBlocking; } bool MelderAudio_getOutputUsesBlocking (void) { return preferences. outputUsesBlocking; } void MelderAudio_setUseInternalSpeaker (bool useInternalSpeaker) { MelderAudio_stopPlaying (MelderAudio_IMPLICIT); preferences. useInternalSpeaker = useInternalSpeaker; } bool MelderAudio_getUseInternalSpeaker (void) { return preferences. useInternalSpeaker; } void MelderAudio_setOutputSilenceBefore (double silenceBefore) { MelderAudio_stopPlaying (MelderAudio_IMPLICIT); preferences. silenceBefore = silenceBefore; } double MelderAudio_getOutputSilenceBefore (void) { return preferences. silenceBefore; } void MelderAudio_setOutputSilenceAfter (double silenceAfter) { MelderAudio_stopPlaying (MelderAudio_IMPLICIT); preferences. silenceAfter = silenceAfter; } double MelderAudio_getOutputSilenceAfter (void) { return preferences. silenceAfter; } long MelderAudio_getOutputBestSampleRate (long fsamp) { #if defined (macintosh) return fsamp; #elif defined (_WIN32) return fsamp == 8000 || fsamp == 11025 || fsamp == 16000 || fsamp == 22050 || fsamp == 32000 || fsamp == 44100 || fsamp == 48000 || fsamp == 96000 ? fsamp : 44100; #elif defined (linux) return fsamp == 44100 || fsamp == 48000 || fsamp == 96000 ? fsamp : 44100; #else return 44100; #endif } bool MelderAudio_isPlaying; static double theStartingTime = 0.0; static struct MelderPlay { const int16_t *buffer; long sampleRate, numberOfSamples, samplesLeft, samplesSent, samplesPlayed; unsigned int asynchronicity; int numberOfChannels; bool explicitStop, fakeMono; bool (*callback) (void *closure, long samplesPlayed); void *closure; #if motif XtWorkProcId workProcId_motif; #elif gtk gint workProcId_gtk; #endif bool usePortAudio, blocking, supports_paComplete; PaStream *stream; double paStartingTime; #if defined (macintosh) SndChannelPtr soundChannel; #elif defined (linux) int audio_fd, val, err; #elif defined (_WIN32) HWAVEOUT hWaveOut; WAVEHDR waveHeader; MMRESULT status; #endif } thePlay; long MelderAudio_getSamplesPlayed (void) { return thePlay. samplesPlayed; } bool MelderAudio_stopWasExplicit (void) { return thePlay. explicitStop; } /* * The flush () procedure will always have to be invoked after normal play, i.e. in the following cases: * 1. After synchronous play (asynchronicity = 0, 1, or 2). * 2. After interruption of asynchronicity 2 by the ESCAPE key. * 3. After asynchronous play, by the workProc. * 4. After interruption of asynchronicity 3 by MelderAudio_stopPlaying (). */ static bool flush (void) { struct MelderPlay *me = & thePlay; if (my usePortAudio) { if (my stream != NULL) Pa_CloseStream (my stream), my stream = NULL; } else { #if defined (macintosh) if (my asynchronicity == kMelder_asynchronicityLevel_ASYNCHRONOUS) { /* Other asynchronicities are handled within Melder_play16 (). */ SndDisposeChannel (my soundChannel, 1), my soundChannel = NULL; } #elif defined (linux) /* * As on Sun. */ if (my audio_fd) { ioctl (my audio_fd, SNDCTL_DSP_RESET, (my val = 0, & my val)); close (my audio_fd), my audio_fd = 0; } #elif defined (_WIN32) /* * FIX: Do not reset the sound card if played to the end: * the last 20 milliseconds may be truncated! * This used to happen on Barbertje's Dell PC, not with SoundBlaster. */ if (my samplesPlayed != my numberOfSamples || Melder_debug == 2) waveOutReset (my hWaveOut); my status = waveOutUnprepareHeader (my hWaveOut, & my waveHeader, sizeof (WAVEHDR)); if (/* Melder_debug == 3 && */ my status == WAVERR_STILLPLAYING) { waveOutReset (my hWaveOut); waveOutUnprepareHeader (my hWaveOut, & my waveHeader, sizeof (WAVEHDR)); } waveOutClose (my hWaveOut), my hWaveOut = 0; #endif } if (my fakeMono) { NUMvector_free ((short *) my buffer, 0); my buffer = NULL; } MelderAudio_isPlaying = false; if (my samplesPlayed >= my numberOfSamples) my samplesPlayed = my numberOfSamples; if (my samplesPlayed <= 0) my samplesPlayed = 1; /* * Call the callback for the last time, which is recognizable by the value of MelderAudio_isPlaying. * In this way, the caller of Melder_play16 can be notified. * The caller can examine the actual number of samples played by testing MelderAudio_getSamplesPlayed (). */ if (my callback) my callback (my closure, my samplesPlayed); my callback = 0; my closure = 0; return true; /* Remove workProc if called from workProc. */ } bool MelderAudio_stopPlaying (bool explicitStop) { //Melder_casual ("stop playing!"); struct MelderPlay *me = & thePlay; my explicitStop = explicitStop; if (! MelderAudio_isPlaying || my asynchronicity < kMelder_asynchronicityLevel_ASYNCHRONOUS) return false; (void) flush (); #if motif XtRemoveWorkProc (thePlay. workProcId_motif); #elif gtk gtk_idle_remove (thePlay. workProcId_gtk); #endif return true; } static bool workProc (void *closure) { struct MelderPlay *me = & thePlay; //static long n = 0; //n ++; //Melder_casual("workProc %ld", n); if (my usePortAudio) { if (my blocking) { if (my samplesLeft > 0) { int dsamples = my samplesLeft > 2000 ? 2000 : my samplesLeft; Pa_WriteStream (my stream, (void *) & my buffer [my samplesSent * my numberOfChannels], dsamples); my samplesLeft -= dsamples; my samplesSent += dsamples; my samplesPlayed = (Melder_clock () - theStartingTime - Pa_GetStreamInfo (my stream) -> outputLatency) * my sampleRate; if (my callback && ! my callback (my closure, my samplesPlayed)) return flush (); } else /*if (my samplesPlayed >= my numberOfSamples)*/ { Pa_StopStream (my stream); my samplesPlayed = my numberOfSamples; return flush (); } } else { /* * Not all hostApis support paComplete or wait till all buffers have been played in Pa_StopStream. * Once pa_win_ds implements this, we can simply do: * if (Pa_IsStreamActive (my stream)) { * if (my callback && ! my callback (my closure, my samplesPlayed)) * return flush (); * } else { * my samplesPlayed = my numberOfSamples; * return flush (); * } * But then we also have to use paComplete in the stream callback. */ double timeElapsed = Melder_clock () - theStartingTime - Pa_GetStreamInfo (my stream) -> outputLatency; my samplesPlayed = timeElapsed * my sampleRate; if (my supports_paComplete && Pa_IsStreamActive (my stream)) { if (my callback && ! my callback (my closure, my samplesPlayed)) { Pa_AbortStream (my stream); return flush (); } } else if (my samplesPlayed < my numberOfSamples + my sampleRate / 20) { // allow the latency estimate to be 50 ms off. if (my callback && ! my callback (my closure, my samplesPlayed)) { Pa_AbortStream (my stream); return flush (); } } else { Pa_AbortStream (my stream); my samplesPlayed = my numberOfSamples; return flush (); } Pa_Sleep (10); } } else { #if defined (macintosh) SCStatus status; SndChannelStatus (my soundChannel, sizeof (SCStatus), & status); if (status. scChannelBusy) { my samplesPlayed = (Melder_clock () - theStartingTime) * my sampleRate; if (my callback && ! my callback (my closure, my samplesPlayed)) return flush (); } else { my samplesPlayed = my numberOfSamples; return flush (); } #elif defined (linux) if (my samplesLeft > 0) { int dsamples = my samplesLeft > 500 ? 500 : my samplesLeft; write (my audio_fd, (char *) & my buffer [my samplesSent * my numberOfChannels], 2 * dsamples * my numberOfChannels); my samplesLeft -= dsamples; my samplesSent += dsamples; my samplesPlayed = (Melder_clock () - theStartingTime) * my sampleRate; if (my callback && ! my callback (my closure, my samplesPlayed)) return flush (); } else /*if (my samplesPlayed >= my numberOfSamples)*/ { close (my audio_fd), my audio_fd = 0; my samplesPlayed = my numberOfSamples; return flush (); /*} else { my samplesPlayed = (Melder_clock () - theStartingTime) * my sampleRate; if (my callback && ! my callback (my closure, my samplesPlayed)) return flush ();*/ } #elif defined (_WIN32) if (my waveHeader. dwFlags & WHDR_DONE) { my samplesPlayed = my numberOfSamples; return flush (); } else { static long previousTime = 0; unsigned long currentTime = clock (); if (Melder_debug == 1) { my samplesPlayed = (Melder_clock () - theStartingTime) * my sampleRate; } else { MMTIME mmtime; mmtime. wType = TIME_BYTES; waveOutGetPosition (my hWaveOut, & mmtime, sizeof (MMTIME)); my samplesPlayed = mmtime. u.cb / (2 * my numberOfChannels); } if (/* Melder_debug != 4 || */ currentTime - previousTime > CLOCKS_PER_SEC / 100) { previousTime = currentTime; if (my callback && ! my callback (my closure, my samplesPlayed)) return flush (); } Sleep (10); } #endif } (void) closure; return false; } #if motif static Boolean workProc_motif (XtPointer closure) { return workProc ((void *) closure); } #elif gtk static gint workProc_gtk (gpointer closure) { return ! workProc ((void *) closure); } #endif #if defined (macintosh) # define FloatToUnsigned(f) \ ((unsigned long)(((long)((f) - 2147483648.0)) + 2147483647L + 1)) static void double2real10 (double x, unsigned char *bytes) { int sign, exponent; double fMantissa, fsMantissa; unsigned long highMantissa, lowMantissa; if (x < 0.0) { sign = 0x8000; x *= -1; } else sign = 0; if (x == 0.0) { exponent = 0; highMantissa = 0; lowMantissa = 0; } else { fMantissa = frexp (x, & exponent); if ((exponent > 16384) || ! (fMantissa < 1)) /* Infinity or Not-a-Number. */ { exponent = sign | 0x7FFF; highMantissa = 0; lowMantissa = 0; } /* Infinity. */ else { /* Finite */ exponent += 16382; /* Add bias. */ if (exponent < 0) { /* Denormalized. */ fMantissa = ldexp (fMantissa, exponent); exponent = 0; } exponent |= sign; fMantissa = ldexp (fMantissa, 32); fsMantissa = floor (fMantissa); highMantissa = FloatToUnsigned (fsMantissa); fMantissa = ldexp (fMantissa - fsMantissa, 32); fsMantissa = floor (fMantissa); lowMantissa = FloatToUnsigned (fsMantissa); } } bytes [0] = exponent >> 8; bytes [1] = exponent; bytes [2] = highMantissa >> 24; bytes [3] = highMantissa >> 16; bytes [4] = highMantissa >> 8; bytes [5] = highMantissa; bytes [6] = lowMantissa >> 24; bytes [7] = lowMantissa >> 16; bytes [8] = lowMantissa >> 8; bytes [9] = lowMantissa; } #endif static int thePaStreamCallback (const void *input, void *output, unsigned long frameCount, const PaStreamCallbackTimeInfo* timeInfo, PaStreamCallbackFlags statusFlags, void *userData) { (void) input; (void) timeInfo; (void) userData; struct MelderPlay *me = & thePlay; if (statusFlags & paOutputUnderflow) { if (Melder_debug == 20) Melder_casual ("output underflow"); } if (statusFlags & paOutputOverflow) { if (Melder_debug == 20) Melder_casual ("output overflow"); } if (my samplesLeft > 0) { long dsamples = my samplesLeft > (long) frameCount ? (long) frameCount : my samplesLeft; if (Melder_debug == 20) Melder_casual ("play %ls %ls", Melder_integer (dsamples), Melder_double (Pa_GetStreamCpuLoad (my stream))); memset (output, '\0', 2 * frameCount * my numberOfChannels); memcpy (output, (char *) & my buffer [my samplesSent * my numberOfChannels], 2 * dsamples * my numberOfChannels); my samplesLeft -= dsamples; my samplesSent += dsamples; my samplesPlayed = my samplesSent; } else /*if (my samplesPlayed >= my numberOfSamples)*/ { memset (output, '\0', 2 * frameCount * my numberOfChannels); my samplesPlayed = my numberOfSamples; return my supports_paComplete ? paComplete : paContinue; } return paContinue; } void MelderAudio_play16 (const int16_t *buffer, long sampleRate, long numberOfSamples, int numberOfChannels, bool (*playCallback) (void *playClosure, long numberOfSamplesPlayed), void *playClosure) { struct MelderPlay *me = & thePlay; #ifdef _WIN32 bool wasPlaying = MelderAudio_isPlaying; #endif if (MelderAudio_isPlaying) MelderAudio_stopPlaying (MelderAudio_IMPLICIT); // otherwise, keep "explicitStop" tag my buffer = buffer; my sampleRate = sampleRate; my numberOfSamples = numberOfSamples; my numberOfChannels = numberOfChannels; my callback = playCallback; my closure = playClosure; my asynchronicity = Melder_batch ? kMelder_asynchronicityLevel_SYNCHRONOUS : Melder_backgrounding ? kMelder_asynchronicityLevel_INTERRUPTABLE : kMelder_asynchronicityLevel_ASYNCHRONOUS; if (my asynchronicity > preferences. maximumAsynchronicity) my asynchronicity = preferences. maximumAsynchronicity; my usePortAudio = preferences. outputUsesPortAudio; my blocking = preferences. outputUsesBlocking; my explicitStop = MelderAudio_IMPLICIT; my fakeMono = false; my samplesLeft = numberOfSamples; my samplesSent = 0; my samplesPlayed = 0; MelderAudio_isPlaying = true; if (my usePortAudio) { PaError err; static bool paInitialized = false; if (! paInitialized) { err = Pa_Initialize (); if (err) Melder_fatal ("PortAudio does not initialize: %s", Pa_GetErrorText (err)); paInitialized = true; } my supports_paComplete = Pa_GetHostApiInfo (Pa_GetDefaultHostApi ()) -> type != paDirectSound; PaStreamParameters outputParameters = { 0 }; outputParameters. device = Pa_GetDefaultOutputDevice (); const PaDeviceInfo *deviceInfo = Pa_GetDeviceInfo (outputParameters. device); outputParameters. channelCount = my numberOfChannels; outputParameters. sampleFormat = paInt16; if (deviceInfo != NULL) outputParameters. suggestedLatency = deviceInfo -> defaultLowOutputLatency; outputParameters. hostApiSpecificStreamInfo = NULL; err = Pa_OpenStream (& my stream, NULL, & outputParameters, my sampleRate, paFramesPerBufferUnspecified, paDitherOff, my blocking ? NULL : thePaStreamCallback, me); if (err) Melder_throw ("PortAudio cannot open sound output: ", Pa_GetErrorText (err), "."); theStartingTime = Melder_clock (); if (my blocking) { err = Pa_StartStream (my stream); if (err) Melder_throw ("PortAudio cannot start sound output: ", Pa_GetErrorText (err), "."); if (my asynchronicity == kMelder_asynchronicityLevel_SYNCHRONOUS) { Pa_WriteStream (my stream, buffer, numberOfSamples); Pa_StopStream (my stream); my samplesPlayed = my numberOfSamples; } else if (my asynchronicity <= kMelder_asynchronicityLevel_INTERRUPTABLE) { bool interrupted = false; while (my samplesLeft > 0 && ! interrupted) { long numberOfWriteSamplesAvailableWithoutBlocking = Pa_GetStreamWriteAvailable (my stream); if (Melder_debug == 20) Melder_casual ("Pa_GetStreamWriteAvailable: %ld", numberOfWriteSamplesAvailableWithoutBlocking); long maximumNumberOfSamplesToWrite = 1000; long dsamples = my samplesLeft > maximumNumberOfSamplesToWrite ? maximumNumberOfSamplesToWrite : my samplesLeft; Pa_WriteStream (my stream, (void *) & my buffer [my samplesSent * my numberOfChannels], dsamples); my samplesLeft -= dsamples; my samplesSent += dsamples; my samplesPlayed = (Melder_clock () - theStartingTime - Pa_GetStreamInfo (my stream) -> outputLatency) * my sampleRate; if (my callback && ! my callback (my closure, my samplesPlayed)) interrupted = true; if (my asynchronicity == kMelder_asynchronicityLevel_INTERRUPTABLE && ! interrupted) { #if defined (macintosh) EventRecord event; if (EventAvail (keyDownMask, & event)) { /* * Remove the event, even if it was a different key. * Otherwise, the key will block the future availability of the Escape key. */ FlushEvents (keyDownMask, 0); /* * Catch Escape and Command-period. */ if ((event. message & charCodeMask) == 27 || ((event. modifiers & cmdKey) && (event. message & charCodeMask) == '.')) { my explicitStop = MelderAudio_EXPLICIT; interrupted = true; } } #elif defined (_WIN32) MSG event; if (PeekMessage (& event, 0, 0, 0, PM_REMOVE) && event. message == WM_KEYDOWN) { if (LOWORD (event. wParam) == VK_ESCAPE) { my explicitStop = MelderAudio_EXPLICIT; interrupted = true; } } #endif } if (interrupted) { flush (); return; } } Pa_StopStream (my stream); } else { #if motif my workProcId_motif = GuiAddWorkProc (workProc_motif, NULL); #elif gtk my workProcId_gtk = gtk_idle_add (workProc_gtk, NULL); #endif return; } } else { err = Pa_StartStream (my stream); if (err) Melder_throw ("PortAudio cannot start sound output: ", Pa_GetErrorText (err), "."); my paStartingTime = Pa_GetStreamTime (my stream); if (my asynchronicity <= kMelder_asynchronicityLevel_INTERRUPTABLE) { for (;;) { double timeElapsed = Melder_clock () - theStartingTime - Pa_GetStreamInfo (my stream) -> outputLatency; my samplesPlayed = timeElapsed * my sampleRate; if (my samplesPlayed >= my numberOfSamples + my sampleRate / 20) { my samplesPlayed = my numberOfSamples; break; } bool interrupted = false; if (my asynchronicity != kMelder_asynchronicityLevel_SYNCHRONOUS && my callback && ! my callback (my closure, my samplesPlayed)) interrupted = true; /* * Safe operation: only listen to key-down events. * Do this on the lowest level that will work. */ if (my asynchronicity == kMelder_asynchronicityLevel_INTERRUPTABLE && ! interrupted) { #if defined (macintosh) EventRecord event; if (EventAvail (keyDownMask, & event)) { /* * Remove the event, even if it was a different key. * Otherwise, the key will block the future availability of the Escape key. */ FlushEvents (keyDownMask, 0); /* * Catch Escape and Command-period. */ if ((event. message & charCodeMask) == 27 || ((event. modifiers & cmdKey) && (event. message & charCodeMask) == '.')) { my explicitStop = MelderAudio_EXPLICIT; interrupted = true; } } #elif defined (_WIN32) MSG event; if (PeekMessage (& event, 0, 0, 0, PM_REMOVE) && event. message == WM_KEYDOWN) { if (LOWORD (event. wParam) == VK_ESCAPE) { my explicitStop = MelderAudio_EXPLICIT; interrupted = true; } } #endif } if (interrupted) { flush (); return; } Pa_Sleep (10); } if (my samplesPlayed != my numberOfSamples) { Melder_fatal ("Played %ld instead of %ld samples.", my samplesPlayed, my numberOfSamples); } Pa_AbortStream (my stream); } else /* my asynchronicity == kMelder_asynchronicityLevel_ASYNCHRONOUS */ { #if motif my workProcId_motif = GuiAddWorkProc (workProc_motif, NULL); #elif gtk my workProcId_gtk = gtk_idle_add (workProc_gtk, NULL); #endif return; } } flush (); return; } else { #if defined (macintosh) if (my asynchronicity == kMelder_asynchronicityLevel_SYNCHRONOUS) { static char listResource [500]; SndListPtr data = (SndListPtr) & listResource [0]; ExtSoundHeaderPtr soundHeader; data -> format = 1; data -> numModifiers = 1; data -> modifierPart [0]. modNumber = sampledSynth; data -> modifierPart [0]. modInit = initStereo; data -> numCommands = 1; data -> commandPart [0]. cmd = bufferCmd + dataOffsetFlag; data -> commandPart [0]. param1 = 0; data -> commandPart [0]. param2 = 20; soundHeader = (ExtSoundHeader *) & data -> dataPart; soundHeader -> samplePtr = (char *) my buffer; soundHeader -> numChannels = my numberOfChannels; soundHeader -> sampleRate = (UnsignedFixed) (unsigned long) floor (65536.0 * my sampleRate); soundHeader -> loopStart = 0L; soundHeader -> loopEnd = 0L; soundHeader -> encode = extSH; soundHeader -> baseFrequency = 60; soundHeader -> numFrames = my numberOfSamples; soundHeader -> sampleSize = 16; (void) SndPlay (NULL, (SndListResource **) (Handle) & data, false); my samplesPlayed = my numberOfSamples; } else { /* long soundFeatures; int can16bit = Gestalt (gestaltSoundAttr, & soundFeatures) == 0 && (soundFeatures & (1 << gestalt16BitAudioSupport)) != 0; */ static char hdr [300]; SndCommand cmd; ExtSoundHeaderPtr soundHeader = (ExtSoundHeaderPtr) & hdr; soundHeader -> samplePtr = (char *) my buffer; soundHeader -> numChannels = my numberOfChannels; soundHeader -> sampleRate = (UnsignedFixed) (unsigned long) floor (65536.0 * my sampleRate); double2real10 (my sampleRate, (unsigned char *) & soundHeader -> AIFFSampleRate); soundHeader -> loopStart = 0L; soundHeader -> loopEnd = 0L; soundHeader -> encode = extSH; soundHeader -> baseFrequency = 60; soundHeader -> numFrames = my numberOfSamples; soundHeader -> sampleSize = 16; SndNewChannel (& my soundChannel, sampledSynth, initStereo, NULL /*NewSndCallBackProc (channelCallback)*/); cmd. cmd = bufferCmd; cmd. param1 = 0; cmd. param2 = (long) soundHeader; theStartingTime = Melder_clock (); SndDoCommand (my soundChannel, & cmd, 0); if (my asynchronicity <= kMelder_asynchronicityLevel_INTERRUPTABLE) { SCStatus status; while (SndChannelStatus (my soundChannel, sizeof (SCStatus), & status), status. scChannelBusy) { bool interrupted = false; EventRecord event; my samplesPlayed = (Melder_clock () - theStartingTime) * my sampleRate; if (my callback && ! my callback (my closure, my samplesPlayed)) interrupted = true; /* * Safe operation: only listen to key-down events. * Do this on the lowest level that will work. */ if (my asynchronicity == kMelder_asynchronicityLevel_INTERRUPTABLE && ! interrupted && EventAvail (keyDownMask, & event)) { /* * Remove the event, even if it was a different key. * Otherwise, the key will block the future availability of the Escape key. */ FlushEvents (keyDownMask, 0); /* * Catch Escape and Command-period. */ if ((event. message & charCodeMask) == 27 || ((event. modifiers & cmdKey) && (event. message & charCodeMask) == '.')) my explicitStop = MelderAudio_EXPLICIT, interrupted = true; } if (interrupted) { my samplesPlayed = (Melder_clock () - theStartingTime) * my sampleRate; /*FlushEvents (everyEvent, 0);*/ SndDisposeChannel (my soundChannel, 1), my soundChannel = NULL; flush (); return; } Pa_Sleep (10); } SndDisposeChannel (my soundChannel, 0), my soundChannel = NULL; my samplesPlayed = my numberOfSamples; } else /* my asynchronicity == kMelder_asynchronicityLevel_ASYNCHRONOUS */ { my workProcId_motif = GuiAddWorkProc (workProc_motif, NULL); return; } } flush (); return; #elif defined (linux) try { /* Big-endian version added by Stefan de Konink, Nov 29, 2007 */ #if __BYTE_ORDER == __BIG_ENDIAN int fmt = AFMT_S16_BE; #else int fmt = AFMT_S16_LE; #endif /* O_NDELAY option added by Rafael Laboissiere, May 19, 2005 */ if ((my audio_fd = open ("/dev/dsp", O_WRONLY | (Melder_debug == 16 ? 0 : O_NDELAY))) == -1) { Melder_throw (errno == EBUSY ? "Audio device already in use." : "Cannot open audio device.\nPlease switch on PortAudio in Praat's Sound Playing Preferences."); } fcntl (my audio_fd, F_SETFL, 0); /* Added by Rafael Laboissiere, May 19, 2005 */ if (ioctl (my audio_fd, SNDCTL_DSP_SETFMT, /* Changed SND_DSP_SAMPLESIZE to SNDCTL_DSP_SETFMT; Stefan de Konink, Nov 29, 2007 */ (my val = fmt, & my val)) == -1 || /* Error? */ my val != fmt) /* Has sound card overridden our sample size? */ { Melder_throw ("Cannot set sample size to 16 bit."); } if (ioctl (my audio_fd, SNDCTL_DSP_CHANNELS, (my val = my numberOfChannels, & my val)) == -1 || /* Error? */ my val != my numberOfChannels) /* Has sound card overridden our number of channels? */ { /* * There is one specific case in which we can work around the current failure, * namely when we are trying to play in mono but the driver of the sound card supports stereo only * and notified us of this by overriding our number of channels. */ if (my numberOfChannels == 1 && my val == 2) { my fakeMono = true; int16_t *newBuffer = NUMvector (0, 2 * numberOfSamples - 1); for (long isamp = 0; isamp < numberOfSamples; isamp ++) { newBuffer [isamp + isamp] = newBuffer [isamp + isamp + 1] = buffer [isamp]; } my buffer = (const int16_t *) newBuffer; my numberOfChannels = 2; } else { Melder_throw ("Cannot set number of channels to .", my numberOfChannels, "."); } } if (ioctl (my audio_fd, SNDCTL_DSP_SPEED, (my val = my sampleRate, & my val)) == -1 || // error? my val != my sampleRate) /* Has sound card overridden our sampling frequency? */ { Melder_throw ("Cannot set sampling frequency to ", my sampleRate, " Hz."); } theStartingTime = Melder_clock (); if (my asynchronicity == kMelder_asynchronicityLevel_SYNCHRONOUS) { if (write (my audio_fd, & my buffer [0], 2 * numberOfChannels * numberOfSamples) == -1) Melder_throw ("Cannot write audio output."); close (my audio_fd), my audio_fd = 0; /* Drain. Set to zero in order to notify flush (). */ my samplesPlayed = my numberOfSamples; } else if (my asynchronicity <= kMelder_asynchronicityLevel_INTERRUPTABLE) { bool interrupted = false; while (my samplesLeft && ! interrupted) { int dsamples = my samplesLeft > 500 ? 500 : my samplesLeft; if (write (my audio_fd, (char *) & my buffer [my samplesSent * my numberOfChannels], 2 * dsamples * my numberOfChannels) == -1) Melder_throw ("Cannot write audio output."); my samplesLeft -= dsamples; my samplesSent += dsamples; my samplesPlayed = (Melder_clock () - theStartingTime) * my sampleRate; if (my callback && ! my callback (my closure, my samplesPlayed)) interrupted = true; } if (! interrupted) { /* * Wait for playing to end. */ close (my audio_fd), my audio_fd = 0; // BUG: should do a loop my samplesPlayed = my numberOfSamples; } } else /* my asynchronicity == kMelder_asynchronicityLevel_ASYNCHRONOUS */ { my workProcId_gtk = gtk_idle_add (workProc_gtk, NULL); return; } flush (); return; } catch (MelderError) { struct MelderPlay *me = & thePlay; if (my audio_fd) close (my audio_fd), my audio_fd = 0; MelderAudio_isPlaying = false; Melder_throw ("16-bit audio not played."); } #elif defined (_WIN32) try { WAVEFORMATEX waveFormat; MMRESULT err; waveFormat. wFormatTag = WAVE_FORMAT_PCM; waveFormat. nChannels = my numberOfChannels; waveFormat. nSamplesPerSec = my sampleRate; waveFormat. wBitsPerSample = 16; waveFormat. nBlockAlign = my numberOfChannels * waveFormat. wBitsPerSample / 8; waveFormat. nAvgBytesPerSec = waveFormat. nBlockAlign * waveFormat. nSamplesPerSec; waveFormat. cbSize = 0; err = waveOutOpen (& my hWaveOut, WAVE_MAPPER, & waveFormat, 0, 0, CALLBACK_NULL | WAVE_ALLOWSYNC); if (err != MMSYSERR_NOERROR) { MelderAudio_isPlaying = false; if (err == MMSYSERR_ALLOCATED) Melder_throw ("Previous sound is still playing? Should not occur!\n" "Report bug to the author: ", err, "; wasPlaying: ", wasPlaying, "."); if (err == MMSYSERR_BADDEVICEID) Melder_throw ("Cannot play a sound. Perhaps another program is playing a sound at the same time?"); if (err == MMSYSERR_NODRIVER) Melder_throw ("This computer probably has no sound card."); if (err == MMSYSERR_NOMEM) Melder_throw ("Not enough free memory to play any sound at all."); if (err == WAVERR_BADFORMAT) Melder_throw ("Bad sound format? Should not occur! Report bug to the author!"); Melder_throw ("Unknown error ", err, " while trying to play a sound? Report bug to the author!"); } my waveHeader. dwFlags = 0; my waveHeader. lpData = (char *) my buffer; my waveHeader. dwBufferLength = my numberOfSamples * 2 * my numberOfChannels; my waveHeader. dwLoops = 1; my waveHeader. lpNext = NULL; my waveHeader. reserved = 0; err = waveOutPrepareHeader (my hWaveOut, & my waveHeader, sizeof (WAVEHDR)); //waveOutReset (my hWaveOut); if (err != MMSYSERR_NOERROR) { waveOutClose (my hWaveOut), my hWaveOut = 0; MelderAudio_isPlaying = false; if (err == MMSYSERR_INVALHANDLE) Melder_throw ("No device? Should not occur!"); if (err == MMSYSERR_NODRIVER) Melder_throw ("No driver? Should not occur!"); if (err == MMSYSERR_NOMEM) Melder_throw ("Not enough free memory to play this sound.\n" "Remove some objects, play a shorter sound, or buy more memory."); Melder_throw ("Unknown error ", err, " while preparing header? Should not occur!"); } err = waveOutWrite (my hWaveOut, & my waveHeader, sizeof (WAVEHDR)); if (err != MMSYSERR_NOERROR) { waveOutReset (my hWaveOut); waveOutUnprepareHeader (my hWaveOut, & my waveHeader, sizeof (WAVEHDR)); waveOutClose (my hWaveOut), my hWaveOut = 0; MelderAudio_isPlaying = false; Melder_throw ("Error ", err, " while writing audio output."); // BUG: should flush } theStartingTime = Melder_clock (); if (my asynchronicity == kMelder_asynchronicityLevel_SYNCHRONOUS) { while (! (my waveHeader. dwFlags & WHDR_DONE)) { Sleep (10); } my samplesPlayed = my numberOfSamples; } else if (my asynchronicity <= kMelder_asynchronicityLevel_INTERRUPTABLE) { while (! (my waveHeader. dwFlags & WHDR_DONE)) { MSG event; Sleep (10); my samplesPlayed = (Melder_clock () - theStartingTime) * my sampleRate; if (my callback && ! my callback (my closure, my samplesPlayed)) break; if (my asynchronicity == kMelder_asynchronicityLevel_INTERRUPTABLE && PeekMessage (& event, 0, 0, 0, PM_REMOVE) && event. message == WM_KEYDOWN) { if (LOWORD (event. wParam) == VK_ESCAPE) { my explicitStop = MelderAudio_EXPLICIT; break; } } } } else { my workProcId_motif = GuiAddWorkProc (workProc_motif, NULL); return; } flush (); return; } catch (MelderError) { Melder_throw ("16-bit audio not played."); } #else Melder_throw ("Cannot play a sound on this computer.\n16-bit audio not played."); #endif } } /********** WAITING FOR SOUND INPUT **********/ void Melder_audioTrigger (void) { } /* End of file melder_audio.cpp */