hg/stx-libbasic2: comparison SoundStream.st

equal deleted inserted replaced

-:9f874a72681c
+:17cec979acdf
 documentation
 "
 Preliminary (unfinished) interface to an audio device.
 Currently works with
-LINUXs
+	LINUXs
-SUNs (and other) /dev/audio driver
+	SUNs (and other) /dev/audio driver
-IRIX (indy).
+	IRIX (indy).
-PORTAUDIO (osx).
+	PORTAUDIO (osx).
 On iris, the default setup is for 8 bit mono
 so I can play the standard sound files I have here.
 It needs much more work, for stereo, different sampling rates etc.
 This is an experimental class - its interface & implementation may change in the future.
 [author:]
-Claus Gittinger
+	Claus Gittinger
 "
 ! !
 !SoundStream class methodsFor:'initialization'!
 !SoundStream class methodsFor:'instance creation'!
 determineConcreteClass
 self allSubclassesDo:[:each |
-each isAbstract ifFalse:[
+	each isAbstract ifFalse:[
-each isSupported ifTrue:[
+	    each isSupported ifTrue:[
-^ each
+		^ each
-]
+	    ]
-]
+	]
 ].
 ^ nil
 "
 self determineConcreteClass
 "
 !
 new
 self == SoundStream ifTrue:[
-ConcreteClass isNil ifTrue:[
+	ConcreteClass isNil ifTrue:[
-ConcreteClass := self determineConcreteClass.
+	    ConcreteClass := self determineConcreteClass.
-ConcreteClass isNil ifTrue:[
+	    ConcreteClass isNil ifTrue:[
-OpenError raiseErrorString:'missing sound support'.
+		OpenError raiseErrorString:'missing sound support'.
-].
+	    ].
-].
+	].
-^ ConcreteClass new.
+	^ ConcreteClass new.
 ].
 ^ self basicNew initialize
 "
 SoundStream new
 |newStream|
 newStream := self new.
 (newStream supportedAudioFormats includesAny:#(S16 U16)) ifFalse:[
-OpenError raiseRequestErrorString:'sound system does not support 16bit audio'
+	OpenError raiseRequestErrorString:'sound system does not support 16bit audio'
 ].
 newStream bitsPerSample:16.
 newStream openForWriting isNil ifTrue:[^nil].
 ^ newStream
 |newStream|
 newStream := self new.
 (newStream supportedAudioFormats includesAny:#(S16 U16)) ifFalse:[
-OpenError raiseRequestErrorString:'sound system does not support 16bit audio'
+	OpenError raiseRequestErrorString:'sound system does not support 16bit audio'
 ].
 newStream bitsPerSample:16.
 newStream numberOfChannels:2.
 newStream openForWriting isNil ifTrue:[^nil].
 ^ newStream
 writingFloat
 |newStream|
 newStream := self new.
 (newStream supportedAudioFormats includes:#F32) ifFalse:[
-OpenError raiseRequestErrorString:'sound system does not support float audio'
+	OpenError raiseRequestErrorString:'sound system does not support float audio'
 ].
 newStream setAudioFormat:#F32.
 newStream numberOfChannels:1.
 newStream openForWriting isNil ifTrue:[^nil].
 ^ newStream
 writingFloatStereo
 |newStream|
 newStream := self new.
 (newStream supportedAudioFormats includes:#F32) ifFalse:[
-OpenError raiseRequestErrorString:'sound system does not support float audio'
+	OpenError raiseRequestErrorString:'sound system does not support float audio'
 ].
 newStream setAudioFormat:#F32.
 newStream numberOfChannels:2.
 newStream openForWriting isNil ifTrue:[^nil].
 ^ newStream
 openAttributes := attributeSpec.
 %{
 #ifdef SUPPORT_JACKAUDIO
 jack_options_t options = JackNullOption;
 jack_client_t *client;
+jack_port_t *output_port;
 struct jackStreamData* streamData;
 int nChannels, sampleRate, bytesPerSample;
 #   define FRAMES_PER_BUFFER 128
 ok = false;
 if (__isSmallInteger(__INST(numberOfChannels))) {
-nChannels = __intVal(__INST(numberOfChannels));
+	nChannels = __intVal(__INST(numberOfChannels));
 } else {
-nChannels = 1;
+	nChannels = 1;
 }
 if (__INST(audioFormat) == @symbol(F32)) {
-bytesPerSample = 4;
+	bytesPerSample = 4;
 } else {
-fprintf(stderr, "SoundStream [warning]: only f32 supported with jack\n");
+	fprintf(stderr, "SoundStream [warning]: only f32 supported with jack\n");
-errorStringOrNil = __MKSTRING("audioFormat must be f32 with jack");
+	errorStringOrNil = __MKSTRING("audioFormat must be f32 with jack");
-goto out;
+	goto out;
-}
-if (__isSmallInteger(__INST(sampleRate))) {
-sampleRate = __intVal(__INST(sampleRate));
-} else {
-fprintf(stderr, "SoundStream [warning]: using default sampleRate 8000\n");
-sampleRate = 8000;
 }
 streamData = (struct jackStreamData*)malloc(sizeof(struct jackStreamData));
 if (streamData == NULL) {
-fprintf(stderr, "SoundStream [warning]: failed to allocate jackStream\n");
+	fprintf(stderr, "SoundStream [warning]: failed to allocate jackStream\n");
-errorStringOrNil = __MKSTRING("failed to allocate jackStream");
+	errorStringOrNil = __MKSTRING("failed to allocate jackStream");
-goto out;
+	goto out;
 }
 if ((client = jack_client_open ("stx", JackNullOption, NULL)) == 0) {
-fprintf(stderr, "SoundStream [warning]: client_open\n");
+	fprintf(stderr, "SoundStream [warning]: JACK server not running?\n");
-free(streamData);
+	free(streamData);
-errorStringOrNil = __MKSTRING("client_open");
+	errorStringOrNil = __MKSTRING("JACK server not running");
-goto out;
+	goto out;
 }
 jack_set_process_callback (client, jack_callback, 0);
+output_port = jack_port_register (client, "stx_out", JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
+sampleRate = jack_get_sample_rate (client);
+__INST(sampleRate) = __MKSMALLINT(sampleRate);
+// if (__isSmallInteger(__INST(sampleRate))) {
+//     sampleRate = __intVal(__INST(sampleRate));
+// } else {
+//     fprintf(stderr, "SoundStream [warning]: using default sampleRate 8000\n");
+//     sampleRate = 8000;
+// }
 streamData->jack_client = client;
+streamData->jack_output_port = output_port;
 INITLOCK(streamData->lock);
 streamData->readOffset = 0;
 streamData->bytesPerSample = bytesPerSample;
 streamData->nChannels = nChannels;
 streamData->lastBuffer = NULL;
 streamData->freeList = NULL;
 streamData->hasFinished = 0;
 {
-OBJ t;
+	OBJ t;
-t = __MKEXTERNALADDRESS(streamData); __INST(handle1) = t; __STORE(self, t);
+	t = __MKEXTERNALADDRESS(streamData); __INST(handle1) = t; __STORE(self, t);
 }
 __INST(binary) = true;
 //    paErr = Pa_SetStreamFinishedCallback( stream, &paStreamFinished );
 //    if( paErr != paNoError ) {
 out:;
 #endif /* SUPPORT_JACKAUDIO */
 %}.
 ok == false ifTrue:[
-lastErrorString := errorStringOrNil.
+	lastErrorString := errorStringOrNil.
-lastErrorNumber := error ? -1.
+	lastErrorNumber := error ? -1.
-self openError:error.
+	self openError:error.
-"normally not reached"
+	"normally not reached"
-^ nil.
+	^ nil.
 ].
 self registerForFinalization.
 !
 reopenStream
 !SoundStream::JackAudio methodsFor:'queries'!
 supportedAudioFormats
 "return a collection of supported audio formats.
 possibly returned symbols are:
-U8        unsigned 8bit samples
+	U8        unsigned 8bit samples
-S8        signed 8bit samples
+	S8        signed 8bit samples
-U16       unsigned 16bit samples in native format
+	U16       unsigned 16bit samples in native format
-U16_LE    unsigned 16bit big endian samples
+	U16_LE    unsigned 16bit big endian samples
-U16_BE    unsigned 16bit big endian samples
+	U16_BE    unsigned 16bit big endian samples
-S16       signed 16bit little endian samples in native format
+	S16       signed 16bit little endian samples in native format
-S16_LE    signed 16bit little endian samples
+	S16_LE    signed 16bit little endian samples
-S16_BE    signed 16bit big endian samples
+	S16_BE    signed 16bit big endian samples
-S24       signed 24bit little endian samples in native format
+	S24       signed 24bit little endian samples in native format
-S24_LE    signed 24bit little endian samples
+	S24_LE    signed 24bit little endian samples
-S24_BE    signed 24bit big endian samples
+	S24_BE    signed 24bit big endian samples
-S32       signed 32bit little endian samples in native format
+	S32       signed 32bit little endian samples in native format
-S32_LE    signed 32bit little endian samples
+	S32_LE    signed 32bit little endian samples
-S32_BE    signed 32bit big endian samples
+	S32_BE    signed 32bit big endian samples
-F32       float samples
+	F32       float samples
-MPEG      audio mpeg encoded
+	MPEG      audio mpeg encoded
-MU_LAW    u-law encoded 8bit samples
+	MU_LAW    u-law encoded 8bit samples
-A_LAW     a-law encoded 8bit samples
+	A_LAW     a-law encoded 8bit samples
-IMA_ADPCM adpcm encoded
+	IMA_ADPCM adpcm encoded
 the set of returned symbols depends on the underlying sound hardware.
 "
 ^ #(F32)

changeset 5425	17cec979acdf
parent 5424	9f874a72681c
child 5426	c9d3c51e1b53