gstreamer/sys/oss/gstosssink.c
Andy Wingo d1d21600f8 sys/oss/gstossmixer.*: Refactored to be more like alsamixer.
Original commit message from CVS:
2005-08-23  Andy Wingo  <wingo@pobox.com>

* sys/oss/gstossmixer.h:
* sys/oss/gstossmixer.c: Refactored to be more like alsamixer.

* sys/oss/gstossmixertrack.h:
* sys/oss/gstossmixertrack.c: Split out from gstossmixer.[ch],
like gstalsamixer.

* sys/oss/gstosssrc.c:
* sys/oss/gstosssink.c: Where before we used a gstosselement
object as a helper library, now just call functions from
gstosshelper.

* sys/oss/gstosshelper.h:
* sys/oss/gstosshelper.c: Made a real library. Removed
propertyprobe for now, should add it back later.

* sys/oss/gstosselement.h:
* sys/oss/gstosselement.c: Removed, we don't have a shared base
class.

* sys/oss/gstosshelper.c (gst_oss_helper_probe_caps): Search
higher-to-lower, makes 16 bit appear earlier in the caps, which
makes it preferred.
2005-08-23 13:26:21 +00:00

410 lines
9.9 KiB
C

/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000,2005 Wim Taymans <wim@fluendo.com>
*
* gstosssink.c:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <sys/ioctl.h>
#include <fcntl.h>
#include <errno.h>
#include <unistd.h>
#include <string.h>
#include <sys/soundcard.h>
#include "gstosssink.h"
/* elementfactory information */
static GstElementDetails gst_oss_sink_details =
GST_ELEMENT_DETAILS ("Audio Sink (OSS)",
"Sink/Audio",
"Output to a sound card via OSS",
"Erik Walthinsen <omega@cse.ogi.edu>, "
"Wim Taymans <wim.taymans@chello.be>");
static void gst_oss_sink_base_init (gpointer g_class);
static void gst_oss_sink_class_init (GstOssSinkClass * klass);
static void gst_oss_sink_init (GstOssSink * osssink);
static void gst_oss_sink_dispose (GObject * object);
static GstCaps *gst_oss_sink_getcaps (GstBaseSink * bsink);
static gboolean gst_oss_sink_open (GstAudioSink * asink);
static gboolean gst_oss_sink_close (GstAudioSink * asink);
static gboolean gst_oss_sink_prepare (GstAudioSink * asink,
GstRingBufferSpec * spec);
static gboolean gst_oss_sink_unprepare (GstAudioSink * asink);
static guint gst_oss_sink_write (GstAudioSink * asink, gpointer data,
guint length);
static guint gst_oss_sink_delay (GstAudioSink * asink);
static void gst_oss_sink_reset (GstAudioSink * asink);
/* OssSink signals and args */
enum
{
LAST_SIGNAL
};
static GstStaticPadTemplate osssink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
"audio/x-raw-int, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 8, "
"depth = (int) 8, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
);
static GstElementClass *parent_class = NULL;
/* static guint gst_oss_sink_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_oss_sink_get_type (void)
{
static GType osssink_type = 0;
if (!osssink_type) {
static const GTypeInfo osssink_info = {
sizeof (GstOssSinkClass),
gst_oss_sink_base_init,
NULL,
(GClassInitFunc) gst_oss_sink_class_init,
NULL,
NULL,
sizeof (GstOssSink),
0,
(GInstanceInitFunc) gst_oss_sink_init,
};
osssink_type =
g_type_register_static (GST_TYPE_AUDIO_SINK, "GstOssSink",
&osssink_info, 0);
}
return osssink_type;
}
static void
gst_oss_sink_dispose (GObject * object)
{
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_oss_sink_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &gst_oss_sink_details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&osssink_sink_factory));
}
static void
gst_oss_sink_class_init (GstOssSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
GstBaseAudioSinkClass *gstbaseaudiosink_class;
GstAudioSinkClass *gstaudiosink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
gstaudiosink_class = (GstAudioSinkClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_BASE_AUDIO_SINK);
gobject_class->dispose = gst_oss_sink_dispose;
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_sink_getcaps);
gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_oss_sink_open);
gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_oss_sink_close);
gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_sink_prepare);
gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_sink_unprepare);
gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_oss_sink_write);
gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_oss_sink_delay);
gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_oss_sink_reset);
}
static void
gst_oss_sink_init (GstOssSink * osssink)
{
GST_DEBUG ("initializing osssink");
osssink->fd = -1;
}
static GstCaps *
gst_oss_sink_getcaps (GstBaseSink * bsink)
{
GstOssSink *osssink;
GstCaps *caps;
osssink = GST_OSSSINK (bsink);
if (osssink->fd == -1) {
caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD
(bsink)));
} else {
caps = gst_oss_helper_probe_caps (osssink->fd);
}
return caps;
}
static gint
ilog2 (gint x)
{
/* well... hacker's delight explains... */
x = x | (x >> 1);
x = x | (x >> 2);
x = x | (x >> 4);
x = x | (x >> 8);
x = x | (x >> 16);
x = x - ((x >> 1) & 0x55555555);
x = (x & 0x33333333) + ((x >> 2) & 0x33333333);
x = (x + (x >> 4)) & 0x0f0f0f0f;
x = x + (x >> 8);
x = x + (x >> 16);
return (x & 0x0000003f) - 1;
}
#define SET_PARAM(_oss, _name, _val) \
G_STMT_START { \
int _tmp = _val; \
if (ioctl(_oss->fd, _name, &_tmp) == -1) { \
perror(G_STRINGIFY (_name)); \
return FALSE; \
} \
GST_DEBUG(G_STRINGIFY (name) " %d", _tmp); \
} G_STMT_END
#define GET_PARAM(oss, name, val) \
G_STMT_START { \
if (ioctl(oss->fd, name, val) == -1) { \
perror(G_STRINGIFY (name)); \
return FALSE; \
} \
} G_STMT_END
static gint
gst_oss_sink_get_format (GstBufferFormat fmt)
{
gint result;
switch (fmt) {
case GST_MU_LAW:
result = AFMT_MU_LAW;
break;
case GST_A_LAW:
result = AFMT_A_LAW;
break;
case GST_IMA_ADPCM:
result = AFMT_IMA_ADPCM;
break;
case GST_U8:
result = AFMT_U8;
break;
case GST_S16_LE:
result = AFMT_S16_LE;
break;
case GST_S16_BE:
result = AFMT_S16_BE;
break;
case GST_S8:
result = AFMT_S8;
break;
case GST_U16_LE:
result = AFMT_U16_LE;
break;
case GST_U16_BE:
result = AFMT_U16_BE;
break;
case GST_MPEG:
result = AFMT_MPEG;
break;
default:
result = 0;
break;
}
return result;
}
static gboolean
gst_oss_sink_open (GstAudioSink * asink)
{
GstOssSink *oss;
int mode;
oss = GST_OSSSINK (asink);
mode = O_WRONLY;
mode |= O_NONBLOCK;
oss->fd = open ("/dev/dsp", mode, 0);
if (oss->fd == -1) {
perror ("/dev/dsp");
return FALSE;
}
return TRUE;
}
static gboolean
gst_oss_sink_close (GstAudioSink * asink)
{
close (GST_OSSSINK (asink)->fd);
return TRUE;
}
static gboolean
gst_oss_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
{
GstOssSink *oss;
struct audio_buf_info info;
int mode;
int tmp;
oss = GST_OSSSINK (asink);
mode = fcntl (oss->fd, F_GETFL);
mode &= ~O_NONBLOCK;
if (fcntl (oss->fd, F_SETFL, mode) == -1) {
perror ("/dev/dsp");
return FALSE;
}
tmp = gst_oss_sink_get_format (spec->format);
if (tmp == 0)
goto wrong_format;
SET_PARAM (oss, SNDCTL_DSP_SETFMT, tmp);
if (spec->channels == 2)
SET_PARAM (oss, SNDCTL_DSP_STEREO, 1);
SET_PARAM (oss, SNDCTL_DSP_CHANNELS, spec->channels);
SET_PARAM (oss, SNDCTL_DSP_SPEED, spec->rate);
tmp = ilog2 (spec->segsize);
tmp = ((spec->segtotal & 0x7fff) << 16) | tmp;
GST_DEBUG ("set segsize: %d, segtotal: %d, value: %08x", spec->segsize,
spec->segtotal, tmp);
SET_PARAM (oss, SNDCTL_DSP_SETFRAGMENT, tmp);
GET_PARAM (oss, SNDCTL_DSP_GETOSPACE, &info);
spec->segsize = info.fragsize;
spec->segtotal = info.fragstotal;
spec->bytes_per_sample = 4;
oss->bytes_per_sample = 4;
memset (spec->silence_sample, 0, spec->bytes_per_sample);
GST_DEBUG ("got segsize: %d, segtotal: %d, value: %08x", spec->segsize,
spec->segtotal, tmp);
return TRUE;
wrong_format:
{
GST_DEBUG ("wrong format %d\n", spec->format);
return FALSE;
}
}
static gboolean
gst_oss_sink_unprepare (GstAudioSink * asink)
{
/* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */
if (!gst_oss_sink_close (asink))
goto couldnt_close;
if (!gst_oss_sink_open (asink))
goto couldnt_reopen;
return TRUE;
couldnt_close:
{
GST_DEBUG ("Could not close the audio device");
return FALSE;
}
couldnt_reopen:
{
GST_DEBUG ("Could not reopen the audio device");
return FALSE;
}
}
static guint
gst_oss_sink_write (GstAudioSink * asink, gpointer data, guint length)
{
return write (GST_OSSSINK (asink)->fd, data, length);
}
static guint
gst_oss_sink_delay (GstAudioSink * asink)
{
GstOssSink *oss;
gint delay = 0;
gint ret;
oss = GST_OSSSINK (asink);
#ifdef SNDCTL_DSP_GETODELAY
ret = ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay);
#else
ret = -1;
#endif
if (ret < 0) {
audio_buf_info info;
ret = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &info);
delay = (ret < 0 ? 0 : (info.fragstotal * info.fragsize) - info.bytes);
}
return delay / oss->bytes_per_sample;
}
static void
gst_oss_sink_reset (GstAudioSink * asink)
{
GstOssSink *oss;
//gint ret;
oss = GST_OSSSINK (asink);
/* deadlocks on my machine... */
//ret = ioctl (oss->fd, SNDCTL_DSP_RESET, 0);
}