seal 0.1.0

data/mpg123/src/output/aix.c

@@ -0,0 +1,300 @@
+/*
+	aix: Driver for IBM RS/6000 with AIX Ultimedia Services
+
+	copyright ?-2006 by the mpg123 project - free software under the terms of the LGPL 2.1
+	see COPYING and AUTHORS files in distribution or http://mpg123.org
+	initially written by Juergen Schoew and Tomas Oegren
+*/
+
+#include <stdio.h>
+#include <unistd.h>
+#include <stdlib.h>
+
+#include <errno.h>
+#include <fcntl.h>
+#include <sys/audio.h>
+#include <stropts.h>
+#include <sys/types.h>
+#include <sys/file.h>
+#include <sys/stat.h>
+#include <sys/param.h>
+
+#include "mpg123app.h"
+#include "debug.h"
+
+/* use AUDIO_BSIZE to set the msec for audio buffering in Ultimedia library
+ */
+/* #define AUDIO_BSIZE AUDIO_IGNORE */
+#define AUDIO_BSIZE 200
+
+static int rate_best_match(audio_output_t *ao)
+{
+	static long valid [ ] = {  5510,  6620,  8000,  9600, 11025, 16000, 18900,
+							22050, 27420, 32000, 33075, 37800, 44100, 48000, 0 };
+	int  i = 0;
+	long best = 8000;
+
+	if(!ai || ao->fn < 0 || ao->rate < 0) {
+		return -1;
+	} 
+	
+	while (valid [i]) {
+		if (abs(valid[i] - ao->rate) < abs(best - ao->rate))
+		{
+			best = valid [i];
+		}
+		i = i + 1;
+	}
+	
+	ao->rate = best;
+	return best;
+}
+
+static int reset_parameters(audio_output_t *ao)
+{
+	audio_control  acontrol;
+	audio_change   achange;
+	audio_init     ainit;
+	int ret;
+
+	memset ( & achange, '\0', sizeof (achange));
+	memset ( & acontrol, '\0', sizeof (acontrol));
+	
+	achange.balance        = 0x3fff0000;
+	achange.balance_delay  = 0;
+	achange.volume         = (long) (0x7fff << 16);
+	achange.volume_delay   = 0;
+	achange.input          = AUDIO_IGNORE;
+	if (ao->flags == -1) achange.output = INTERNAL_SPEAKER;
+	else achange.output      = 0;
+	
+	if(ao->flags & AUDIO_OUT_INTERNAL_SPEAKER)
+	achange.output     |= INTERNAL_SPEAKER;
+	if(ao->flags & AUDIO_OUT_HEADPHONES)
+	achange.output     |= EXTERNAL_SPEAKER;
+	if(ao->flags & AUDIO_OUT_LINE_OUT)
+	achange.output     |= OUTPUT_1;
+	if(ao->flags == 0)
+	achange.output      = AUDIO_IGNORE;
+
+	achange.treble         = AUDIO_IGNORE;
+	achange.bass          = AUDIO_IGNORE;
+	achange.pitch          = AUDIO_IGNORE;
+	achange.monitor        = AUDIO_IGNORE;
+	achange.dev_info       = (char *) NULL;
+	
+	acontrol.ioctl_request = AUDIO_CHANGE;
+	acontrol.position      = 0;
+	acontrol.request_info  = (char *) & achange;
+	
+	ret = ioctl (ao->fn, AUDIO_CONTROL, & acontrol);
+	if (ret < 0)
+	return ret;
+	
+	/* Init Device for new values */
+	if (ao->rate >0) {
+		memset ( & ainit, '\0', sizeof (ainit));
+		ainit.srate                 = rate_best_match(ai);
+		if (ao->channels > 0)
+		ainit.channels          = ao->channels;
+		else
+		ainit.channels           = 1;
+		switch (ao->format) {
+			default :
+				ainit.mode             = PCM;
+				ainit.bits_per_sample  = 8;
+				ainit.flags            = BIG_ENDIAN | TWOS_COMPLEMENT;
+			break;
+			case MPG123_ENC_SIGNED_16:
+				ainit.mode             = PCM;
+				ainit.bits_per_sample  = 16;
+				ainit.flags            = BIG_ENDIAN | TWOS_COMPLEMENT;
+			break;
+			case MPG123_ENC_SIGNED_8:
+				ainit.mode             = PCM;
+				ainit.bits_per_sample  = 8;
+				ainit.flags            = BIG_ENDIAN | TWOS_COMPLEMENT;
+			break;
+			case MPG123_ENC_UNSIGNED_16:
+				ainit.mode             = PCM;
+				ainit.bits_per_sample  = 16;
+				ainit.flags            = BIG_ENDIAN | TWOS_COMPLEMENT | SIGNED;
+			break;
+			case MPG123_ENC_UNSIGNED_8:
+				ainit.mode             = PCM;
+				ainit.bits_per_sample  = 8;
+				ainit.flags            = BIG_ENDIAN | TWOS_COMPLEMENT | SIGNED;
+			break;
+			case MPG123_ENC_ULAW_8:
+				ainit.mode             = MU_LAW;
+				ainit.bits_per_sample  = 8;
+				ainit.flags            = BIG_ENDIAN | TWOS_COMPLEMENT;
+			break;
+			case MPG123_ENC_ALAW_8:
+				ainit.mode             = A_LAW;
+				ainit.bits_per_sample  = 8;
+				ainit.flags            = BIG_ENDIAN | TWOS_COMPLEMENT;
+			break;
+		}
+		ainit.operation            = PLAY;
+		ainit.bsize                = AUDIO_BSIZE;
+		
+		ret = ioctl (ao->fn, AUDIO_INIT, & ainit);
+		if (ret < 0) {
+			error("Can't set new audio parameters!");
+			return ret;
+		}
+	}
+	
+	acontrol.ioctl_request   = AUDIO_START;
+	acontrol.request_info    = NULL;
+	acontrol.position        = 0;
+	
+	ret = ioctl (ao->fn, AUDIO_CONTROL, & acontrol);
+	if (ret < 0) {
+	error("Can't reset audio!");
+	return ret;
+	}
+	return 0;
+}
+
+static int open_aix(audio_output_t *ao)
+{
+	audio_init ainit;
+	int ret;
+
+	if(!ao->device) {
+		if(getenv("AUDIODEV")) {
+			ao->device = getenv("AUDIODEV");
+			ao->fn = open(ao->device,O_WRONLY);
+		} else {
+			ao->device = "/dev/paud0/1";                   /* paud0 for PCI */
+			ao->fn = open(ao->device,O_WRONLY);
+			if ((ao->fn == -1) & (errno == ENOENT)) {
+				ao->device = "/dev/baud0/1";                 /* baud0 for MCA */
+				ao->fn = open(ao->device,O_WRONLY);
+			}  
+		}
+	} else ao->fn = open(ao->device,O_WRONLY);
+	
+	if(ao->fn < 0) {
+		error("Can't open audio device!");
+		return ao->fn;
+	}
+	
+	/* Init to default values */
+	memset ( & ainit, '\0', sizeof (ainit));
+	ainit.srate            = 44100;
+	ainit.channels         = 2;
+	ainit.mode             = PCM;
+	ainit.bits_per_sample  = 16;
+	ainit.flags            = BIG_ENDIAN | TWOS_COMPLEMENT;
+	ainit.operation        = PLAY;
+	ainit.bsize            = AUDIO_BSIZE;
+	
+	ret = ioctl (ao->fn, AUDIO_INIT, & ainit);
+	if (ret < 0) return ret;
+	
+	reset_parameters(ai);
+	return ao->fn;
+}
+
+
+
+static int get_formats_aix(audio_output_t *ao)
+{
+	/* ULTIMEDIA DOCUMENTATION SAYS:
+	The Ultimedia Audio Adapter supports fourteen sample rates you can use to
+	capture and playback audio data. The rates are (in kHz): 5.51, 6.62, 8.0,
+	9.6, 11.025, 16.0, 18.9, 22.050, 27.42, 32.0, 33.075, 37.8, 44.1, and 48.0.
+	These rates are supported for mono and stereo PCM (8- and 16-bit), mu-law,
+	and A-law. 
+	*/
+	
+	long rate;
+	
+	rate = ao->rate;
+	rate_best_match(ai);
+	if (ao->rate == rate)
+		return (MPG123_ENC_SIGNED_16|MPG123_ENC_UNSIGNED_16|
+			MPG123_ENC_UNSIGNED_8|MPG123_ENC_SIGNED_8|
+			MPG123_ENC_ULAW_8|MPG123_ENC_ALAW_8);
+	else
+		return 0;
+}
+
+static int write_aix(audio_output_t *ao,unsigned char *buf,int len)
+{
+	return write(ao->fn,buf,len);
+}
+
+static int close_aix(audio_output_t *ao)
+{
+	audio_control acontrol;
+	audio_buffer  abuffer;
+	int           ret,i;
+	
+	/* Don't close the audio-device until it's played all its contents */
+	memset ( & acontrol, '\0', sizeof ( acontrol ) );
+	acontrol.request_info = &abuffer;
+	acontrol.position = 0;
+	i=50;   /* Don't do this forever on a bad day :-) */
+	
+	while (i-- > 0) {
+		if ((ioctl(ao->fn, AUDIO_BUFFER, &acontrol))< 0) {
+			error1("buffer read failed: %d", errno);
+			break;
+		} else {
+			if (abuffer.flags <= 0) break;
+		}
+		usleep(200000); /* sleep 0.2 sec */
+	}
+	
+	memset ( & acontrol, '\0', sizeof ( acontrol ) );
+	acontrol.ioctl_request = AUDIO_STOP;
+	acontrol.request_info  = NULL;
+	acontrol.position      = 0;
+	
+	ret = ioctl ( ao->fn, AUDIO_CONTROL, & acontrol );
+	if (ret < 0) error("Can't close audio!");
+	
+	ret = close (ao->fn);
+	if (ret < 0) error("Can't close audio!");
+	
+	return 0;
+}
+
+static void flush_aix(audio_output_t *ao)
+{
+}
+
+static int init_aix(audio_output_t* ao)
+{
+	if (ao==NULL) return -1;
+
+	/* Set callbacks */
+	ao->open = open_aix;
+	ao->flush = flush_aix;
+	ao->write = write_aix;
+	ao->get_formats = get_formats_aix;
+	ao->close = close_aix;
+
+	/* Success */
+	return 0;
+}
+
+
+
+/* 
+	Module information data structure
+*/
+mpg123_module_t mpg123_output_module_info = {
+	/* api_version */	MPG123_MODULE_API_VERSION,
+	/* name */			"AIX",						
+	/* description */	"Output audio on IBM RS/6000 with AIX Ultimedia Services.",
+	/* revision */		"$Rev: 932 $",						
+	/* handle */		NULL,
+	
+	/* init_output */	init_aix,						
+};
+

data/mpg123/src/output/alib.c

@@ -0,0 +1,209 @@
+/*
+	alib: audio output for HP-UX using alib
+
+	copyright ?-2006 by the mpg123 project - free software under the terms of the LGPL 2.1
+	see COPYING and AUTHORS files in distribution or http://mpg123.org
+	initially written by Erwan Ducroquet
+	based on source code from HP (Audio SDK)
+*/
+
+/*
+ *
+ * for mpg123 :
+ * hpux:
+ *  $(MAKE) \
+ *  CC=cc \
+ *  LDFLAGS=-L/opt/audio/lib \
+ *  AUDIO_LIB=-lAlib \
+ *  OBJECTS=decode.o dct64.o \
+ *  CFLAGS=-Ae +O3 -DREAL_IS_FLOAT -D_HPUX_SOURCE -DHPUX -I/opt/audio/include \
+ *  mpg123
+ */
+
+
+/*
+ * For the user :
+ * If you launch mpg123 on a XTerm with sound capabilities, it's OK
+ * Else, you have to set the environment variable "AUDIO" to the name of
+ * an HP Xterm with sound card.
+ */
+
+/**************************************************************************/
+
+#include "mpg123app.h"
+
+#include <fcntl.h>
+
+#include <sys/socket.h>
+#include <netdb.h>
+#include <netinet/in.h>
+#include <netinet/tcp.h>
+#include <errno.h>
+
+
+#include <Alib.h>   /* /opt/audio/include */
+#include <CUlib.h>  /* /opt/audio/include */
+
+#include "debug.h"
+
+/**************************************************************************/
+
+
+/* FIXME: These globals should be moved into a structure */
+static Audio *audioServer = (Audio *) NULL;
+static struct protoent *tcpProtocolEntry;
+static ATransID xid;
+
+static void printAudioError(Audio *audio,char *message,int errorCode) {
+	char    errorbuff[132];
+	AGetErrorText(audio, errorCode, errorbuff, 131);
+	error2("%s: %s", message, errorbuff);
+}
+static long myHandler(Audio *audio,AErrorEvent *err_event) {
+	printAudioError( audio, "Audio error", err_event->error_code ); 
+	/* we cannot just do random exists, that messes terminal up
+	 need proper error propagation in that case for future, setting intflag or such */
+	/* exit(1); */
+}
+
+/**************************************************************************/
+
+/*
+ * Set the fn element of ai
+ * Use ao->rate and ao->channels
+ * Doesn't set any volume
+ */
+
+/* return on error leaves stuff dirty here... */
+static int open_alib(audio_output_t *ao)
+{
+	AudioAttributes Attribs;
+	AudioAttrMask   AttribsMask;
+	AGainEntry      gainEntry[4];
+	SSPlayParams    playParams;
+	SStream	  audioStream;
+	AErrorHandler   prevHandler;
+	char		  server[1];
+	int		  i;
+	long            status;
+	
+	if (audioServer) {
+		error("openAudio: audio already open");
+		return -1;
+	}
+	
+	prevHandler = ASetErrorHandler(myHandler);
+	
+	server[0] = '\0';
+	audioServer = AOpenAudio( server, NULL );
+	if (audioServer==NULL) {
+		error("Error: could not open audio\n");
+		return -1;
+	}
+	
+	ao->fn = socket( AF_INET, SOCK_STREAM, 0 );
+	if(ao->fn<0) {
+		error("Socket creation failed");
+		return -1;
+	}
+	
+	Attribs.type = ATSampled;
+	Attribs.attr.sampled_attr.sampling_rate = ao->rate;
+	Attribs.attr.sampled_attr.channels	  = ao->channels;
+	Attribs.attr.sampled_attr.data_format	  = ADFLin16;
+	AttribsMask = ASSamplingRateMask | ASChannelsMask  | ASDataFormatMask;
+	
+	gainEntry[0].gain = AUnityGain;
+	gainEntry[0].u.o.out_ch  = AOCTMono;
+	gainEntry[0].u.o.out_dst = AODTDefaultOutput;
+	
+	playParams.gain_matrix.type = AGMTOutput;  /* gain matrix */
+	playParams.gain_matrix.num_entries = 1;
+	playParams.gain_matrix.gain_entries = gainEntry;
+	playParams.play_volume = AUnityGain;       /* play volume */
+	playParams.priority = APriorityNormal;     /* normal priority */
+	playParams.event_mask = 0;                 /* don't solicit any events */
+	
+	xid=APlaySStream(audioServer,AttribsMask,&Attribs,
+	&playParams,&audioStream,NULL);
+	
+	status=connect(ao->fn,
+	(struct sockaddr *) &audioStream.tcp_sockaddr,
+	sizeof(struct sockaddr_in) );
+	if (status<0) {
+		error("Connect failed");
+		return -1;
+	}
+	
+	i=-1;
+	tcpProtocolEntry=getprotobyname("tcp");
+	setsockopt(ao->fn,tcpProtocolEntry->p_proto,TCP_NODELAY,&i,sizeof(i));
+	
+	return ao->fn;
+}
+
+/**************************************************************************/
+
+static int close_alib(audio_output_t *ao)
+{
+	close(ao->fn);
+	ASetCloseDownMode( audioServer, AKeepTransactions, NULL );
+	ACloseAudio( audioServer, NULL );
+	audioServer = (Audio *) NULL;
+	return 0;
+}
+
+/**************************************************************************/
+
+/*
+ * very simple
+ * deserv to be inline
+ */
+
+static int write_alib(audio_output_t *ao,unsigned char *buf,int len)
+{
+	return write(ao->fn,buf,len*2);
+}
+
+/**************************************************************************/
+
+static int get_formats_alib(audio_output_t *ao)
+{
+	return MPG123_ENC_SIGNED_16;
+}
+
+static void flush_alib(audio_output_t *ao)
+{
+}
+
+
+static int init_alib(audio_output_t* ao)
+{
+	if (ao==NULL) return -1;
+
+	/* Set callbacks */
+	ao->open = open_alib;
+	ao->flush = flush_alib;
+	ao->write = write_alib;
+	ao->get_formats = get_formats_alib;
+	ao->close = close_alib;
+
+	/* Success */
+	return 0;
+}
+
+
+
+/* 
+	Module information data structure
+*/
+mpg123_module_t mpg123_output_module_info = {
+	/* api_version */	MPG123_MODULE_API_VERSION,
+	/* name */			"alib",						
+	/* description */	"Output audio HP-UX using alib.",
+	/* revision */		"$Rev:$",						
+	/* handle */		NULL,
+	
+	/* init_output */	init_alib,						
+};
+