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/* Play sound using the SGI audio library
written by Simon Leinen <simon@lia.di.epfl.ch>
Copyright (C) 1992 Free Software Foundation, Inc.
This file is part of XEmacs.
XEmacs 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, or (at your option) any
later version.
XEmacs 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 XEmacs; see the file COPYING. If not, write to the Free
Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. */
/* Synched up with: Not in FSF. */
#include <config.h>
#include "lisp.h"
#include <audio.h>
#include <sys/file.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <string.h>
#include <netinet/in.h> /* for ntohl() etc. */
/* Configuration options */
/* ability to parse Sun/NeXT (.au or .snd) audio file headers. The
.snd format supports all sampling rates and sample widths that are
commonly used, as well as stereo. It is also easy to parse. */
#ifndef HAVE_SND_FILES
#define HAVE_SND_FILES 1
#endif
/* support for eight-but mu-law encoding. This is a useful compaction
technique, and most sounds from the Sun universe are in this
format. */
#ifndef HAVE_MULAW_8
#define HAVE_MULAW_8 1
#endif
/* if your machine is very slow, you have to use a table lookup to
convert mulaw samples to linear. This makes Emacs bigger so try to
avoid it. */
#ifndef USE_MULAW_DECODE_TABLE
#define USE_MULAW_DECODE_TABLE 0
#endif
/* support for linear encoding -- useful if you want better quality.
This enables 8, 16 and 24 bit wide samples. */
#ifndef HAVE_LINEAR
#define HAVE_LINEAR 1
#endif
/* support for 32 bit wide samples. If you notice the difference
between 32 and 24 bit samples, you must have very good ears. Since
the SGI audio library only supports 24 bit samples, each sample has
to be shifted right by 8 bits anyway. So you should probably just
convert all your 32 bit audio files to 24 bit. */
#ifndef HAVE_LINEAR_32
#define HAVE_LINEAR_32 0
#endif
/* support for stereo sound. Imagine the cool applications of this:
finally you don't just hear a beep -- you also know immediately
*where* something went wrong! Unfortunately the programming
interface only takes a single volume argument so far. */
#ifndef HAVE_STEREO
#define HAVE_STEREO 1
#endif
/* the play routine can be interrupted between chunks, so we choose a
small chunksize to keep the system responsive (2000 samples
correspond to a quarter of a second for .au files. If you
HAVE_STEREO, the chunksize should probably be even. */
#define CHUNKSIZE 8000
/* the format assumed for header-less audio data. The following
assumes ".au" format (8000 samples/sec mono 8-bit mulaw). */
#define DEFAULT_SAMPLING_RATE 8000
#define DEFAULT_CHANNEL_COUNT 1
#define DEFAULT_FORMAT AFmulaw8
/* Exports */
/* all compilers on machines that have the SGI audio library
understand prototypes, right? */
extern void play_sound_file (char *, int);
extern void play_sound_data (unsigned char *, int, int);
/* Data structures */
/* an AudioContext describes everything we want to know about how a
particular sound snippet should be played. It is split into three
parts (device, port and buffer) for implementation reasons. The
device part corresponds to the state of the output device and must
be reverted after playing the samples. The port part corresponds
to an ALport; we want to allocate a minimal number of these since
there are only four of them system-wide, but on the other hand we
can't use the same port for mono and stereo. The buffer part
corresponds to the sound data itself. */
typedef struct _AudioContextRec * AudioContext;
typedef struct
{
long device;
int left_speaker_gain;
int right_speaker_gain;
long output_rate;
}
AudioDeviceRec, * AudioDevice;
/* supported sound data formats */
typedef enum
{
AFunknown,
#if HAVE_MULAW_8
AFmulaw8,
#endif
#if HAVE_LINEAR
AFlinear8,
AFlinear16,
AFlinear24,
#if HAVE_LINEAR_32
AFlinear32,
#endif
#endif
AFillegal
}
AudioFormat;
typedef struct
{
ALport port;
AudioFormat format;
unsigned nchan;
unsigned queue_size;
}
AudioPortRec, * AudioPort;
typedef struct
{
void * data;
unsigned long size;
void (* write_chunk_function) (void *, void *, AudioContext);
}
AudioBufferRec, * AudioBuffer;
typedef struct _AudioContextRec
{
AudioDeviceRec device;
AudioPortRec port;
AudioBufferRec buffer;
}
AudioContextRec;
#define ac_device device.device
#define ac_left_speaker_gain device.left_speaker_gain
#define ac_right_speaker_gain device.right_speaker_gain
#define ac_output_rate device.output_rate
#define ac_port port.port
#define ac_format port.format
#define ac_nchan port.nchan
#define ac_queue_size port.queue_size
#define ac_data buffer.data
#define ac_size buffer.size
#define ac_write_chunk_function buffer.write_chunk_function
/* Forward declarations */
static Lisp_Object close_sound_file (Lisp_Object);
static AudioContext audio_initialize (unsigned char *, int, int);
static void play_internal (unsigned char *, int, AudioContext);
static void drain_audio_port (AudioContext);
static void write_mulaw_8_chunk (void *, void *, AudioContext);
static void write_linear_chunk (void *, void *, AudioContext);
static void write_linear_32_chunk (void *, void *, AudioContext);
static Lisp_Object restore_audio_port (Lisp_Object);
static AudioContext initialize_audio_port (AudioContext);
static int open_audio_port (AudioContext, AudioContext);
static void adjust_audio_volume (AudioDevice);
static void get_current_volumes (AudioDevice);
static int set_channels (ALconfig, unsigned);
static int set_output_format (ALconfig, AudioFormat);
static int parse_snd_header (void*, long, AudioContext);
/* 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)
Lisp_Object closure;
{
close (XINT (closure));
return Qnil;
}
void
play_sound_file (sound_file, volume)
char * sound_file;
int volume;
{
int count = specpdl_depth ();
int input_fd;
unsigned char buffer[CHUNKSIZE];
int bytes_read;
AudioContext ac = (AudioContext) 0;
input_fd = open (sound_file, O_RDONLY);
if (input_fd == -1)
/* no error message -- this can't happen
because Fplay_sound_file has checked the
file for us. */
return;
record_unwind_protect (close_sound_file, make_number (input_fd));
while ((bytes_read = read (input_fd, buffer, CHUNKSIZE)) > 0)
{
if (ac == (AudioContext) 0)
{
ac = audio_initialize (buffer, bytes_read, volume);
if (ac == 0)
return;
}
else
{
ac->ac_data = buffer;
ac->ac_size = bytes_read;
}
play_internal (buffer, bytes_read, ac);
}
drain_audio_port (ac);
unbind_to (count, Qnil);
}
static long
saved_device_state[] = {
AL_OUTPUT_RATE, 0,
AL_LEFT_SPEAKER_GAIN, 0,
AL_RIGHT_SPEAKER_GAIN, 0,
};
static Lisp_Object
restore_audio_port (closure)
Lisp_Object closure;
{
Lisp_Object * contents = (vector_data (XVECTOR (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)
unsigned char * data;
int length;
int volume;
{
int count = specpdl_depth ();
AudioContext ac;
ac = audio_initialize (data, length, volume);
if (ac == (AudioContext) 0)
return;
play_internal (data, length, ac);
drain_audio_port (ac);
unbind_to (count, Qnil);
}
static AudioContext
audio_initialize (data, length, volume)
unsigned char * data;
int length;
int volume;
{
Lisp_Object audio_port_state[3];
static AudioContextRec desc;
AudioContext ac;
desc.ac_right_speaker_gain
= desc.ac_left_speaker_gain
= volume * 256 / 100;
desc.ac_device = AL_DEFAULT_DEVICE;
#if HAVE_SND_FILES
if (LOOKING_AT_SND_HEADER_P (data))
{
if (parse_snd_header (data, length, & desc)==-1)
report_file_error ("decoding .snd header", Qnil);
}
else
#endif
{
desc.ac_data = data;
desc.ac_size = length;
desc.ac_output_rate = DEFAULT_SAMPLING_RATE;
desc.ac_nchan = DEFAULT_CHANNEL_COUNT;
desc.ac_format = DEFAULT_FORMAT;
desc.ac_write_chunk_function = write_mulaw_8_chunk;
}
/* Make sure that the audio port is reset to
its initial characteristics after exit */
ALgetparams (desc.ac_device, saved_device_state,
sizeof (saved_device_state) / sizeof (long));
audio_port_state[0] = make_number (saved_device_state[1]);
audio_port_state[1] = make_number (saved_device_state[3]);
audio_port_state[2] = make_number (saved_device_state[5]);
record_unwind_protect (restore_audio_port,
Fvector (3, &audio_port_state[0]));
ac = initialize_audio_port (& desc);
desc = * ac;
return ac;
}
static void
play_internal (data, length, ac)
unsigned char * data;
int length;
AudioContext ac;
{
unsigned char * limit;
if (ac == (AudioContext) 0)
return;
data = ac->ac_data;
limit = data + ac->ac_size;
while (data < limit)
{
unsigned char * chunklimit = data + CHUNKSIZE;
if (chunklimit > limit)
chunklimit = limit;
QUIT;
(* ac->ac_write_chunk_function) (data, chunklimit, ac);
data = chunklimit;
}
}
static void
drain_audio_port (ac)
AudioContext ac;
{
while (ALgetfilled (ac->ac_port) > 0)
sginap(1);
}
/* Methods to write a "chunk" from a buffer containing audio data to
an audio port. This may involve some conversion if the output
device doesn't directly support the format the audio data is in. */
#if HAVE_MULAW_8
#if USE_MULAW_DECODE_TABLE
#include "libst.h"
#else /* not USE_MULAW_DECODE_TABLE */
static int
st_ulaw_to_linear (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;
int unsigned_result = table[exponent]+(mantissa << (exponent+3));
return u1 & 0x80 ? -unsigned_result : unsigned_result;
}
#endif /* not USE_MULAW_DECODE_TABLE */
static void
write_mulaw_8_chunk (buffer, chunklimit, 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;
obuf = alloca (n_samples * sizeof (short));
bufp = &obuf[0];
while (data < limit)
*bufp++ = st_ulaw_to_linear (*data++);
ALwritesamps (ac->ac_port, obuf, n_samples);
}
#endif /* HAVE_MULAW_8 */
#if HAVE_LINEAR
static void
write_linear_chunk (data, limit, ac)
void * data;
void * limit;
AudioContext ac;
{
unsigned n_samples;
switch (ac->ac_format)
{
case AFlinear16: n_samples = (short *) limit - (short *) data; break;
case AFlinear8: n_samples = (char *) limit - (char *) data; break;
default: n_samples = (long *) limit - (long *) data; break;
}
ALwritesamps (ac->ac_port, data, (long) n_samples);
}
#if HAVE_LINEAR_32
static void
write_linear_32_chunk (buffer, chunklimit, ac)
void * buffer;
void * chunklimit;
AudioContext ac;
{
long * data = (long *) buffer;
long * limit = (long *) chunklimit;
long * obuf, * bufp;
long n_samples = limit-data;
obuf = alloca (n_samples * sizeof (long));
bufp = &obuf[0];
while (data < limit)
*bufp++ = *data++ >> 8;
ALwritesamps (ac->ac_port, obuf, n_samples);
}
#endif /* HAVE_LINEAR_32 */
#endif /* HAVE_LINEAR */
static AudioContext
initialize_audio_port (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 },
{ (void *) 0, (unsigned long) 0 } };
#if HAVE_STEREO
static AudioContextRec stereo_port_state
= { { 0, 0, 0, 0 },
{ (ALport) 0, AFunknown, 2, 0 },
{ (void *) 0, (unsigned long) 0 } };
static AudioContext return_ac;
switch (desc->ac_nchan)
{
case 1: return_ac = & mono_port_state; break;
case 2: return_ac = & stereo_port_state; break;
default: return (AudioContext) 0;
}
#else /* not HAVE_STEREO */
static AudioContext return_ac = & mono_port_state;
#endif /* not HAVE_STEREO */
return_ac->device = desc->device;
return_ac->buffer = desc->buffer;
return_ac->ac_format = desc->ac_format;
return_ac->ac_queue_size = desc->ac_queue_size;
if (return_ac->ac_port==(ALport) 0)
{
if ((open_audio_port (return_ac, desc))==-1)
{
report_file_error ("Open audio port", Qnil);
return (AudioContext) 0;
}
}
else
{
ALconfig config = ALgetconfig (return_ac->ac_port);
int changed = 0;
long params[2];
params[0] = AL_OUTPUT_RATE;
ALgetparams (return_ac->ac_device, params, 2);
return_ac->ac_output_rate = params[1];
if (return_ac->ac_output_rate != desc->ac_output_rate)
{
return_ac->ac_output_rate = params[1] = desc->ac_output_rate;
ALsetparams (return_ac->ac_device, params, 2);
}
if ((changed = set_output_format (config, return_ac->ac_format))==-1)
return (AudioContext) 0;
return_ac->ac_format = desc->ac_format;
if (changed)
ALsetconfig (return_ac->ac_port, config);
}
return_ac->ac_write_chunk_function = desc->ac_write_chunk_function;
get_current_volumes (& return_ac->device);
if (return_ac->ac_left_speaker_gain != desc->ac_left_speaker_gain
|| return_ac->ac_right_speaker_gain != desc->ac_right_speaker_gain)
adjust_audio_volume (& desc->device);
return return_ac;
}
static int
open_audio_port (return_ac, desc)
AudioContext return_ac;
AudioContext desc;
{
ALconfig config = ALnewconfig();
long params[2];
adjust_audio_volume (& desc->device);
return_ac->ac_left_speaker_gain = desc->ac_left_speaker_gain;
return_ac->ac_right_speaker_gain = desc->ac_right_speaker_gain;
params[0] = AL_OUTPUT_RATE;
params[1] = desc->ac_output_rate;
ALsetparams (desc->ac_device, params, 2);
return_ac->ac_output_rate = desc->ac_output_rate;
if (set_channels (config, desc->ac_nchan)==-1)
return -1;
return_ac->ac_nchan = desc->ac_nchan;
if (set_output_format (config, desc->ac_format)==-1)
return -1;
return_ac->ac_format = desc->ac_format;
ALsetqueuesize (config, (long) CHUNKSIZE);
return_ac->ac_port = ALopenport("XEmacs audio output", "w", config);
ALfreeconfig (config);
if (return_ac->ac_port==0)
{
report_file_error ("Opening audio output port", Qnil);
return -1;
}
return 0;
}
static int
set_channels (config, nchan)
ALconfig config;
unsigned nchan;
{
switch (nchan)
{
case 1: ALsetchannels (config, AL_MONO); break;
#if HAVE_STEREO
case 2: ALsetchannels (config, AL_STEREO); break;
#endif /* HAVE_STEREO */
default:
report_file_error ("Unsupported channel count",
Fcons (make_number (nchan), Qnil));
return -1;
}
return 0;
}
static int
set_output_format (config, format)
ALconfig config;
AudioFormat format;
{
long samplesize;
long old_samplesize;
switch (format)
{
#if HAVE_MULAW_8
case AFmulaw8:
#endif
#if HAVE_LINEAR
case AFlinear16:
#endif
#if HAVE_MULAW_8 || HAVE_LINEAR
samplesize = AL_SAMPLE_16;
break;
#endif
#if HAVE_LINEAR
case AFlinear8:
samplesize = AL_SAMPLE_8;
break;
case AFlinear24:
#if HAVE_LINEAR_32
case AFlinear32:
samplesize = AL_SAMPLE_24;
break;
#endif
#endif
default:
report_file_error ("Unsupported audio format",
Fcons (make_number (format), Qnil));
return -1;
}
old_samplesize = ALgetwidth (config);
if (old_samplesize==samplesize)
return 0;
ALsetwidth (config, samplesize);
return 1;
}
static void
adjust_audio_volume (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)
AudioDevice device;
{
long params[4];
params[0] = AL_LEFT_SPEAKER_GAIN;
params[2] = AL_RIGHT_SPEAKER_GAIN;
ALgetparams (device->device, params, 4);
device->left_speaker_gain = params[1];
device->right_speaker_gain = params[3];
}
#if HAVE_SND_FILES
/* Parsing .snd (NeXT/Sun) headers */
typedef struct
{
int magic;
int dataLocation;
int dataSize;
int dataFormat;
int samplingRate;
int channelCount;
char info[4];
}
SNDSoundStruct;
#define SOUND_TO_HOST_INT(x) ntohl(x)
typedef enum
{
SND_FORMAT_FORMAT_UNSPECIFIED,
SND_FORMAT_MULAW_8,
SND_FORMAT_LINEAR_8,
SND_FORMAT_LINEAR_16,
SND_FORMAT_LINEAR_24,
SND_FORMAT_LINEAR_32,
SND_FORMAT_FLOAT,
SND_FORMAT_DOUBLE,
SND_FORMAT_INDIRECT,
SND_FORMAT_NESTED,
SND_FORMAT_DSP_CODE,
SND_FORMAT_DSP_DATA_8,
SND_FORMAT_DSP_DATA_16,
SND_FORMAT_DSP_DATA_24,
SND_FORMAT_DSP_DATA_32,
SND_FORMAT_DSP_unknown_15,
SND_FORMAT_DISPLAY,
SND_FORMAT_MULAW_SQUELCH,
SND_FORMAT_EMPHASIZED,
SND_FORMAT_COMPRESSED,
SND_FORMAT_COMPRESSED_EMPHASIZED,
SND_FORMAT_DSP_COMMANDS,
SND_FORMAT_DSP_COMMANDS_SAMPLES
}
SNDFormatCode;
static int
parse_snd_header (header, length, desc)
void * header;
long length;
AudioContext desc;
{
#define hp ((SNDSoundStruct *) (header))
long limit;
#if HAVE_LINEAR
desc->ac_write_chunk_function = write_linear_chunk;
#endif
switch ((SNDFormatCode) SOUND_TO_HOST_INT (hp->dataFormat))
{
#if HAVE_MULAW_8
case SND_FORMAT_MULAW_8:
desc->ac_format = AFmulaw8;
desc->ac_write_chunk_function = write_mulaw_8_chunk;
break;
#endif
#if HAVE_LINEAR
case SND_FORMAT_LINEAR_8:
desc->ac_format = AFlinear8;
break;
case SND_FORMAT_LINEAR_16:
desc->ac_format = AFlinear16;
break;
case SND_FORMAT_LINEAR_24:
desc->ac_format = AFlinear24;
break;
#endif
#if HAVE_LINEAR_32
case SND_FORMAT_LINEAR_32:
desc->ac_format = AFlinear32;
desc->ac_write_chunk_function = write_linear_32_chunk;
break;
#endif
default:
desc->ac_format = AFunknown;
}
desc->ac_output_rate = SOUND_TO_HOST_INT (hp->samplingRate);
desc->ac_nchan = SOUND_TO_HOST_INT (hp->channelCount);
desc->ac_data = (char *) header + SOUND_TO_HOST_INT (hp->dataLocation);
limit = (char *) header + length - (char *) desc->ac_data;
desc->ac_size = SOUND_TO_HOST_INT (hp->dataSize);
if (desc->ac_size > limit) desc->ac_size = limit;
return 0;
#undef hp
}
#endif /* HAVE_SND_FILES */
These are the contents of the former NiCE NeXT User Group NeXTSTEP/OpenStep software archive, currently hosted by Netfuture.ch.