gstreamer/plugins/elements/gstaudiosink.c
Wim Taymans 196c67d7ff Changed the GstPadFactory and added the GstPadTemplate.
Original commit message from CVS:
Changed the GstPadFactory and added the GstPadTemplate.

The GstPadFactory is an easy way to define a pad with capabilities.
The GstPadFactory is converted into a GstPadTemplate. The template is
used to create new pads and to expose the possible pads used in an
element to the plugin system.

updated mp3parse, mpg123, cdparanoia to the new API.
Rerun gstreamer-register because the XML definition has changed.
2000-12-13 19:29:35 +00:00

421 lines
12 KiB
C

/* Gnome-Streamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* 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.
*/
#include <sys/types.h>
#include <sys/stat.h>
#include <sys/ioctl.h>
#include <fcntl.h>
#include <sys/soundcard.h>
#include <unistd.h>
//#define DEBUG_ENABLED
#include <gstaudiosink.h>
#include <gst/meta/audioraw.h>
GstElementDetails gst_audiosink_details = {
"Audio Sink (OSS)",
"Sink/Audio",
"Output to a sound card via OSS",
VERSION,
"Erik Walthinsen <omega@cse.ogi.edu>",
"(C) 1999",
};
static void gst_audiosink_class_init (GstAudioSinkClass *klass);
static void gst_audiosink_init (GstAudioSink *audiosink);
static gboolean gst_audiosink_open_audio (GstAudioSink *sink);
static void gst_audiosink_close_audio (GstAudioSink *sink);
static GstElementStateReturn gst_audiosink_change_state (GstElement *element);
static void gst_audiosink_set_arg (GtkObject *object, GtkArg *arg, guint id);
static void gst_audiosink_get_arg (GtkObject *object, GtkArg *arg, guint id);
static void gst_audiosink_chain (GstPad *pad,GstBuffer *buf);
/* AudioSink signals and args */
enum {
SIGNAL_HANDOFF,
LAST_SIGNAL
};
enum {
ARG_0,
ARG_MUTE,
ARG_FORMAT,
ARG_CHANNELS,
ARG_FREQUENCY,
/* FILL ME */
};
static GstPadFactory audiosink_sink_factory = {
"sink",
GST_PAD_FACTORY_SINK,
GST_PAD_FACTORY_ALWAYS,
"audio/raw",
"format", GST_PROPS_INT (AFMT_S16_LE),
"depth", GST_PROPS_LIST (
GST_PROPS_INT (8),
GST_PROPS_INT (16)
),
"rate", GST_PROPS_INT_RANGE (8000, 48000),
"channels", GST_PROPS_INT_RANGE (1, 2),
NULL
};
#define GST_TYPE_AUDIOSINK_FORMATS (gst_audiosink_formats_get_type())
static GtkType
gst_audiosink_formats_get_type(void) {
static GtkType audiosink_formats_type = 0;
static GtkEnumValue audiosink_formats[] = {
{8, "8", "8 Bits"},
{16, "16", "16 Bits"},
{0, NULL, NULL},
};
if (!audiosink_formats_type) {
audiosink_formats_type = gtk_type_register_enum("GstAudiosinkFormats", audiosink_formats);
}
return audiosink_formats_type;
}
#define GST_TYPE_AUDIOSINK_CHANNELS (gst_audiosink_channels_get_type())
static GtkType
gst_audiosink_channels_get_type(void) {
static GtkType audiosink_channels_type = 0;
static GtkEnumValue audiosink_channels[] = {
{1, "1", "Mono"},
{2, "2", "Stereo"},
{0, NULL, NULL},
};
if (!audiosink_channels_type) {
audiosink_channels_type = gtk_type_register_enum("GstAudiosinkChannels", audiosink_channels);
}
return audiosink_channels_type;
}
static GstSinkClass *parent_class = NULL;
static guint gst_audiosink_signals[LAST_SIGNAL] = { 0 };
static GstPadTemplate *gst_audiosink_sink_template;
GtkType
gst_audiosink_get_type (void)
{
static GtkType audiosink_type = 0;
if (!audiosink_type) {
static const GtkTypeInfo audiosink_info = {
"GstAudioSink",
sizeof(GstAudioSink),
sizeof(GstAudioSinkClass),
(GtkClassInitFunc)gst_audiosink_class_init,
(GtkObjectInitFunc)gst_audiosink_init,
(GtkArgSetFunc)NULL,
(GtkArgGetFunc)NULL,
(GtkClassInitFunc)NULL,
};
audiosink_type = gtk_type_unique (GST_TYPE_SINK, &audiosink_info);
}
return audiosink_type;
}
static void
gst_audiosink_class_init (GstAudioSinkClass *klass)
{
GtkObjectClass *gtkobject_class;
GstElementClass *gstelement_class;
gtkobject_class = (GtkObjectClass*)klass;
gstelement_class = (GstElementClass*)klass;
parent_class = gtk_type_class(GST_TYPE_FILTER);
gtk_object_add_arg_type ("GstAudioSink::mute", GTK_TYPE_BOOL,
GTK_ARG_READWRITE, ARG_MUTE);
gtk_object_add_arg_type ("GstAudioSink::format", GST_TYPE_AUDIOSINK_FORMATS,
GTK_ARG_READWRITE, ARG_FORMAT);
gtk_object_add_arg_type ("GstAudioSink::channels", GST_TYPE_AUDIOSINK_CHANNELS,
GTK_ARG_READWRITE, ARG_CHANNELS);
gtk_object_add_arg_type ("GstAudioSink::frequency", GTK_TYPE_INT,
GTK_ARG_READWRITE, ARG_FREQUENCY);
gtkobject_class->set_arg = gst_audiosink_set_arg;
gtkobject_class->get_arg = gst_audiosink_get_arg;
gst_audiosink_signals[SIGNAL_HANDOFF] =
gtk_signal_new("handoff",GTK_RUN_LAST,gtkobject_class->type,
GTK_SIGNAL_OFFSET(GstAudioSinkClass,handoff),
gtk_marshal_NONE__NONE,GTK_TYPE_NONE,0);
gtk_object_class_add_signals(gtkobject_class,gst_audiosink_signals,
LAST_SIGNAL);
gstelement_class->change_state = gst_audiosink_change_state;
}
static void
gst_audiosink_init (GstAudioSink *audiosink)
{
audiosink->sinkpad = gst_pad_new_from_template (gst_audiosink_sink_template, "sink");
gst_element_add_pad (GST_ELEMENT (audiosink), audiosink->sinkpad);
gst_pad_set_chain_function (audiosink->sinkpad, gst_audiosink_chain);
audiosink->fd = -1;
audiosink->clock = gst_clock_get_system();
gst_clock_register (audiosink->clock, GST_OBJECT (audiosink));
GST_FLAG_SET (audiosink, GST_ELEMENT_THREAD_SUGGESTED);
}
static void
gst_audiosink_sync_parms (GstAudioSink *audiosink)
{
audio_buf_info ospace;
int frag;
g_return_if_fail (audiosink != NULL);
g_return_if_fail (GST_IS_AUDIOSINK (audiosink));
if (audiosink->fd == -1) return;
ioctl (audiosink->fd,SNDCTL_DSP_RESET, 0);
ioctl (audiosink->fd, SNDCTL_DSP_SETFMT, &audiosink->format);
ioctl (audiosink->fd, SNDCTL_DSP_CHANNELS, &audiosink->channels);
ioctl (audiosink->fd, SNDCTL_DSP_SPEED, &audiosink->frequency);
ioctl (audiosink->fd, SNDCTL_DSP_GETBLKSIZE, &frag);
ioctl (audiosink->fd, SNDCTL_DSP_GETOSPACE, &ospace);
g_print("audiosink: setting sound card to %dKHz %d bit %s (%d bytes buffer, %d fragment)\n",
audiosink->frequency, audiosink->format,
(audiosink->channels == 2) ? "stereo" : "mono", ospace.bytes, frag);
}
static void
gst_audiosink_chain (GstPad *pad, GstBuffer *buf)
{
GstAudioSink *audiosink;
MetaAudioRaw *meta;
gboolean in_flush;
audio_buf_info ospace;
g_return_if_fail (pad != NULL);
g_return_if_fail (GST_IS_PAD (pad));
g_return_if_fail (buf != NULL);
/* this has to be an audio buffer */
// g_return_if_fail(((GstMeta *)buf->meta)->type !=
//gst_audiosink_type_audio);
audiosink = GST_AUDIOSINK (pad->parent);
// g_return_if_fail(GST_FLAG_IS_SET(audiosink,GST_STATE_RUNNING));
if ((in_flush = GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLUSH))) {
DEBUG ("audiosink: flush\n");
ioctl (audiosink->fd, SNDCTL_DSP_RESET, 0);
}
meta = (MetaAudioRaw *)gst_buffer_get_first_meta (buf);
if (meta != NULL) {
if ((meta->format != audiosink->format) ||
(meta->channels != audiosink->channels) ||
(meta->frequency != audiosink->frequency))
{
audiosink->format = meta->format;
audiosink->channels = meta->channels;
audiosink->frequency = meta->frequency;
gst_audiosink_sync_parms (audiosink);
g_print("audiosink: sound device set to format %d, %d channels, %dHz\n",
audiosink->format, audiosink->channels, audiosink->frequency);
}
}
gtk_signal_emit (GTK_OBJECT (audiosink), gst_audiosink_signals[SIGNAL_HANDOFF],
audiosink);
if (GST_BUFFER_DATA (buf) != NULL) {
gst_trace_add_entry(NULL, 0, buf, "audiosink: writing to soundcard");
//g_print("audiosink: writing to soundcard\n");
if (audiosink->fd > 2) {
if (!audiosink->mute) {
gst_clock_wait (audiosink->clock, GST_BUFFER_TIMESTAMP (buf), GST_OBJECT (audiosink));
ioctl (audiosink->fd, SNDCTL_DSP_GETOSPACE, &ospace);
DEBUG ("audiosink: (%d bytes buffer) %d %p %d\n", ospace.bytes,
audiosink->fd, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
write (audiosink->fd, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
//write(STDOUT_FILENO,GST_BUFFER_DATA(buf),GST_BUFFER_SIZE(buf));
}
}
}
gst_buffer_unref (buf);
}
static void
gst_audiosink_set_arg (GtkObject *object, GtkArg *arg, guint id)
{
GstAudioSink *audiosink;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail (GST_IS_AUDIOSINK (object));
audiosink = GST_AUDIOSINK (object);
switch(id) {
case ARG_MUTE:
audiosink->mute = GTK_VALUE_BOOL (*arg);
break;
case ARG_FORMAT:
audiosink->format = GTK_VALUE_ENUM (*arg);
gst_audiosink_sync_parms (audiosink);
break;
case ARG_CHANNELS:
audiosink->channels = GTK_VALUE_ENUM (*arg);
gst_audiosink_sync_parms (audiosink);
break;
case ARG_FREQUENCY:
audiosink->frequency = GTK_VALUE_INT (*arg);
gst_audiosink_sync_parms (audiosink);
break;
default:
break;
}
}
static void
gst_audiosink_get_arg (GtkObject *object, GtkArg *arg, guint id)
{
GstAudioSink *audiosink;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail (GST_IS_AUDIOSINK (object));
audiosink = GST_AUDIOSINK (object);
switch(id) {
case ARG_MUTE:
GTK_VALUE_BOOL (*arg) = audiosink->mute;
break;
case ARG_FORMAT:
GTK_VALUE_ENUM (*arg) = audiosink->format;
break;
case ARG_CHANNELS:
GTK_VALUE_ENUM (*arg) = audiosink->channels;
break;
case ARG_FREQUENCY:
GTK_VALUE_INT (*arg) = audiosink->frequency;
break;
default:
break;
}
}
static gboolean
gst_audiosink_open_audio (GstAudioSink *sink)
{
g_return_val_if_fail (sink->fd == -1, FALSE);
g_print ("audiosink: attempting to open sound device\n");
/* first try to open the sound card */
sink->fd = open("/dev/dsp", O_WRONLY);
/* if we have it, set the default parameters and go have fun */
if (sink->fd > 0) {
/* set card state */
sink->format = AFMT_S16_LE;
sink->channels = 2; /* stereo */
sink->frequency = 44100;
gst_audiosink_sync_parms (sink);
ioctl(sink->fd, SNDCTL_DSP_GETCAPS, &sink->caps);
g_print("audiosink: Capabilities\n");
if (sink->caps & DSP_CAP_DUPLEX) g_print("audiosink: Full duplex\n");
if (sink->caps & DSP_CAP_REALTIME) g_print("audiosink: Realtime\n");
if (sink->caps & DSP_CAP_BATCH) g_print("audiosink: Batch\n");
if (sink->caps & DSP_CAP_COPROC) g_print("audiosink: Has coprocessor\n");
if (sink->caps & DSP_CAP_TRIGGER) g_print("audiosink: Trigger\n");
if (sink->caps & DSP_CAP_MMAP) g_print("audiosink: Direct access\n");
g_print("audiosink: opened audio with fd=%d\n", sink->fd);
GST_FLAG_SET (sink, GST_AUDIOSINK_OPEN);
return TRUE;
}
return FALSE;
}
static void
gst_audiosink_close_audio (GstAudioSink *sink)
{
if (sink->fd < 0) return;
close(sink->fd);
sink->fd = -1;
GST_FLAG_UNSET (sink, GST_AUDIOSINK_OPEN);
g_print("audiosink: closed sound device\n");
}
static GstElementStateReturn
gst_audiosink_change_state (GstElement *element)
{
g_return_val_if_fail (GST_IS_AUDIOSINK (element), FALSE);
/* if going down into NULL state, close the file if it's open */
if (GST_STATE_PENDING (element) == GST_STATE_NULL) {
if (GST_FLAG_IS_SET (element, GST_AUDIOSINK_OPEN))
gst_audiosink_close_audio (GST_AUDIOSINK (element));
/* otherwise (READY or higher) we need to open the sound card */
} else {
if (!GST_FLAG_IS_SET (element, GST_AUDIOSINK_OPEN)) {
if (!gst_audiosink_open_audio (GST_AUDIOSINK (element)))
return GST_STATE_FAILURE;
}
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}
gboolean
gst_audiosink_factory_init (GstElementFactory *factory)
{
gst_audiosink_sink_template = gst_padtemplate_new (&audiosink_sink_factory);
gst_elementfactory_add_padtemplate (factory, gst_audiosink_sink_template);
return TRUE;
}