xemacs-beta: src/sgiplay.c comparison

comparison src/sgiplay.c @ 394:7d59cb494b73 r21-2-12

Import from CVS: tag r21-2-12

author	cvs
date	Mon, 13 Aug 2007 11:11:37 +0200
parents	cc15677e0335
children	74fd4e045ea6

comparison

equal deleted inserted replaced

-:2e030b8965b1
+:7d59cb494b73
 /* Synched up with: Not in FSF. */
 #include <config.h>
 #include "lisp.h"
-#include <audio.h>
+#include <string.h>
 #include <sys/file.h>
 #include <sys/types.h>
 #include <sys/stat.h>
 #include <fcntl.h>
-#include <string.h>
+#include <unistd.h>
+#include <audio.h>
 #include <netinet/in.h>		/* for ntohl() etc. */
 /* Configuration options */
 /* ability to parse Sun/NeXT (.au or .snd) audio file headers.  The
 /* are we looking at an NeXT/Sun audio header? */
 #define LOOKING_AT_SND_HEADER_P(address) \
 (!strncmp(".snd", (char *)(address), 4))
 static Lisp_Object
-close_sound_file (closure)
+close_sound_file (Lisp_Object closure)
-Lisp_Object closure;
 {
 close (XINT (closure));
 return Qnil;
 }
 void
-play_sound_file (sound_file, volume)
+play_sound_file (char *sound_file, int volume)
-char * sound_file;
-int volume;
 {
 int count = specpdl_depth ();
 int input_fd;
 unsigned char buffer[CHUNKSIZE];
 int bytes_read;
 AL_LEFT_SPEAKER_GAIN, 0,
 AL_RIGHT_SPEAKER_GAIN, 0,
 };
 static Lisp_Object
-restore_audio_port (closure)
+restore_audio_port (Lisp_Object closure)
-Lisp_Object closure;
 {
 Lisp_Object * contents = XVECTOR_DATA (closure);
 saved_device_state[1] = XINT (contents[0]);
 saved_device_state[3] = XINT (contents[1]);
 saved_device_state[5] = XINT (contents[2]);
 ALsetparams (AL_DEFAULT_DEVICE, saved_device_state, 6);
 return Qnil;
 }
 void
-play_sound_data (data, length, volume)
+play_sound_data (unsigned char *data, int length, int volume)
-unsigned char * data;
-int length;
-int volume;
 {
 int count = specpdl_depth ();
 AudioContext ac;
 ac = audio_initialize (data, length, volume);
 drain_audio_port (ac);
 unbind_to (count, Qnil);
 }
 static AudioContext
-audio_initialize (data, length, volume)
+audio_initialize (unsigned char *data, int length, int volume)
-unsigned char * data;
-int length;
-int volume;
 {
 Lisp_Object audio_port_state[3];
 static AudioContextRec desc;
 AudioContext ac;
 desc = * ac;
 return ac;
 }
 static void
-play_internal (data, length, ac)
+play_internal (unsigned char *data, int length, AudioContext ac)
-unsigned char * data;
-int length;
-AudioContext ac;
 {
 unsigned char * limit;
 if (ac == (AudioContext) 0)
 return;
 data = chunklimit;
 }
 }
 static void
-drain_audio_port (ac)
+drain_audio_port (AudioContext ac)
-AudioContext ac;
 {
 while (ALgetfilled (ac->ac_port) > 0)
 sginap(1);
 }
 #if USE_MULAW_DECODE_TABLE
 #include "libst.h"
 #else /* not USE_MULAW_DECODE_TABLE */
 static int
-st_ulaw_to_linear (u)
+st_ulaw_to_linear (int u)
-int u;
 {
 static CONST short table[] = {0,132,396,924,1980,4092,8316,16764};
 int u1 = ~u;
 short exponent = (u1 >> 4) & 0x07;
 int mantissa = u1 & 0x0f;
 return u1 & 0x80 ? -unsigned_result : unsigned_result;
 }
 #endif /* not USE_MULAW_DECODE_TABLE */
 static void
-write_mulaw_8_chunk (buffer, chunklimit, ac)
+write_mulaw_8_chunk (void *buffer, void *chunklimit, AudioContext ac)
-void * buffer;
-void * chunklimit;
-AudioContext ac;
 {
 unsigned char * data = (unsigned char *) buffer;
 unsigned char * limit = (unsigned char *) chunklimit;
 short * obuf, * bufp;
 long n_samples = limit - data;
 }
 #endif /* HAVE_MULAW_8 */
 #if HAVE_LINEAR
 static void
-write_linear_chunk (data, limit, ac)
+write_linear_chunk (void *data, void *limit, AudioContext ac)
-void * data;
-void * limit;
-AudioContext ac;
 {
 unsigned n_samples;
 switch (ac->ac_format)
 {
 ALwritesamps (ac->ac_port, data, (long) n_samples);
 }
 #if HAVE_LINEAR_32
 static void
-write_linear_32_chunk (buffer, chunklimit, ac)
+write_linear_32_chunk (void *buffer, void *chunklimit, AudioContext ac)
-void * buffer;
-void * chunklimit;
-AudioContext ac;
 {
 long * data = (long *) buffer;
 long * limit = (long *) chunklimit;
 long * obuf, * bufp;
 long n_samples = limit-data;
 }
 #endif /* HAVE_LINEAR_32 */
 #endif /* HAVE_LINEAR */
 static AudioContext
-initialize_audio_port (desc)
+initialize_audio_port (AudioContext desc)
-AudioContext desc;
 {
 /* we can't use the same port for mono and stereo */
 static AudioContextRec mono_port_state
 = { { 0, 0, 0, 0 },
 	{ (ALport) 0, AFunknown, 1, 0 },
 adjust_audio_volume (& desc->device);
 return return_ac;
 }
 static int
-open_audio_port (return_ac, desc)
+open_audio_port (AudioContext return_ac, AudioContext desc)
-AudioContext return_ac;
-AudioContext desc;
 {
 ALconfig config = ALnewconfig();
 long params[2];
 adjust_audio_volume (& desc->device);
 }
 return 0;
 }
 static int
-set_channels (config, nchan)
+set_channels (ALconfig config, unsigned int nchan)
-ALconfig config;
-unsigned nchan;
 {
 switch (nchan)
 {
 case 1: ALsetchannels (config, AL_MONO); break;
 #if HAVE_STEREO
 }
 return 0;
 }
 static int
-set_output_format (config, format)
+set_output_format (ALconfig config, AudioFormat format)
-ALconfig config;
-AudioFormat format;
 {
 long samplesize;
 long old_samplesize;
 switch (format)
 ALsetwidth (config, samplesize);
 return 1;
 }
 static void
-adjust_audio_volume (device)
+adjust_audio_volume (AudioDevice device)
-AudioDevice device;
 {
 long params[4];
 params[0] = AL_LEFT_SPEAKER_GAIN;
 params[1] = device->left_speaker_gain;
 params[2] = AL_RIGHT_SPEAKER_GAIN;
 params[3] = device->right_speaker_gain;
 ALsetparams (device->device, params, 4);
 }
 static void
-get_current_volumes (device)
+get_current_volumes (AudioDevice device)
-AudioDevice device;
 {
 long params[4];
 params[0] = AL_LEFT_SPEAKER_GAIN;
 params[2] = AL_RIGHT_SPEAKER_GAIN;
 ALgetparams (device->device, params, 4);
 SND_FORMAT_DSP_COMMANDS_SAMPLES
 }
 SNDFormatCode;
 static int
-parse_snd_header (header, length, desc)
+parse_snd_header (void *header, long length, AudioContext desc)
-void * header;
-long length;
-AudioContext desc;
 {
 #define hp ((SNDSoundStruct *) (header))
 long limit;
 #if HAVE_LINEAR

Mercurial > hg > xemacs-beta

comparison src/sgiplay.c @ 394:7d59cb494b73 r21-2-12