gstreamer/gst/elements/gstaudiosink.c
Erik Walthinsen 1762dfbf98 initial checkin
Original commit message from CVS:
initial checkin
2000-01-30 09:03:00 +00:00

291 lines
8.6 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 <fcntl.h>
#include <sys/soundcard.h>
#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 gboolean gst_audiosink_open_audio(GstAudioSink *sink);
static void gst_audiosink_close_audio(GstAudioSink *sink);
static gboolean gst_audiosink_start(GstElement *element,
GstElementState state);
static gboolean gst_audiosink_stop(GstElement *element);
static gboolean gst_audiosink_change_state(GstElement *element,
GstElementState state);
/* AudioSink signals and args */
enum {
HANDOFF,
LAST_SIGNAL
};
enum {
ARG_0,
/* FILL ME */
};
static void gst_audiosink_class_init(GstAudioSinkClass *klass);
static void gst_audiosink_init(GstAudioSink *audiosink);
static GstFilterClass *parent_class = NULL;
static guint gst_audiosink_signals[LAST_SIGNAL] = { 0 };
static guint16 gst_audiosink_type_audio = 0;
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_FILTER,&audiosink_info);
}
if (!gst_audiosink_type_audio)
gst_audiosink_type_audio = gst_type_find_by_mime("audio/raw");
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);
gst_audiosink_signals[HANDOFF] =
gtk_signal_new("handoff",GTK_RUN_LAST,gtkobject_class->type,
GTK_SIGNAL_OFFSET(GstAudioSinkClass,handoff),
gtk_marshal_NONE__POINTER_POINTER,GTK_TYPE_NONE,2,
GTK_TYPE_POINTER,GTK_TYPE_POINTER);
gtk_object_class_add_signals(gtkobject_class,gst_audiosink_signals,
LAST_SIGNAL);
gstelement_class->start = gst_audiosink_start;
gstelement_class->stop = gst_audiosink_stop;
gstelement_class->change_state = gst_audiosink_change_state;
}
static void gst_audiosink_init(GstAudioSink *audiosink) {
audiosink->sinkpad = gst_pad_new("sink",GST_PAD_SINK);
gst_element_add_pad(GST_ELEMENT(audiosink),audiosink->sinkpad);
if (!gst_audiosink_type_audio)
gst_audiosink_type_audio = gst_type_find_by_mime("audio/raw");
gst_pad_set_type_id(audiosink->sinkpad,gst_audiosink_type_audio);
gst_pad_set_chain_function(audiosink->sinkpad,gst_audiosink_chain);
audiosink->fd = -1;
gst_element_set_state(GST_ELEMENT(audiosink),GST_STATE_COMPLETE);
}
void gst_audiosink_sync_parms(GstAudioSink *audiosink) {
audio_buf_info ospace;
g_return_if_fail(audiosink != NULL);
g_return_if_fail(GST_IS_AUDIOSINK(audiosink));
g_return_if_fail(audiosink->fd > 0);
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_GETOSPACE,&ospace);
g_print("setting sound card to %dKHz %d bit %s (%d bytes buffer)\n",
audiosink->frequency,audiosink->format,
(audiosink->channels == 2) ? "stereo" : "mono",ospace.bytes);
}
GstElement *gst_audiosink_new(gchar *name) {
GstElement *audiosink = GST_ELEMENT(gtk_type_new(GST_TYPE_AUDIOSINK));
gst_element_set_name(GST_ELEMENT(audiosink),name);
return audiosink;
}
void gst_audiosink_chain(GstPad *pad,GstBuffer *buf) {
GstAudioSink *audiosink;
MetaAudioRaw *meta;
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));
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("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[HANDOFF],
audiosink);
if (GST_BUFFER_DATA(buf) != NULL) {
gst_trace_add_entry(NULL,0,buf,"audiosink: writing to soundcard");
if (audiosink->fd > 2)
write(audiosink->fd,GST_BUFFER_DATA(buf),GST_BUFFER_SIZE(buf));
}
gst_buffer_unref(buf);
// g_print("a");
}
void gst_audiosink_set_format(GstAudioSink *audiosink,gint format) {
g_return_if_fail(audiosink != NULL);
g_return_if_fail(GST_IS_AUDIOSINK(audiosink));
audiosink->format = format;
gst_audiosink_sync_parms(audiosink);
}
void gst_audiosink_set_channels(GstAudioSink *audiosink,gint channels) {
g_return_if_fail(audiosink != NULL);
g_return_if_fail(GST_IS_AUDIOSINK(audiosink));
audiosink->channels = channels;
gst_audiosink_sync_parms(audiosink);
}
void gst_audiosink_set_frequency(GstAudioSink *audiosink,gint frequency) {
g_return_if_fail(audiosink != NULL);
g_return_if_fail(GST_IS_AUDIOSINK(audiosink));
audiosink->frequency = frequency;
gst_audiosink_sync_parms(audiosink);
}
static gboolean gst_audiosink_open_audio(GstAudioSink *sink) {
g_return_if_fail(sink->fd == -1);
g_print("attempting to open sound device\n");
/* first try to open the sound card */
sink->fd = open("/dev/dsp",O_RDWR);
/* 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);
g_print("opened audio\n");
return TRUE;
}
return FALSE;
}
static void gst_audiosink_close_audio(GstAudioSink *sink) {
if (sink->fd < 0) return;
close(sink->fd);
sink->fd = -1;
g_print("closed sound device\n");
}
static gboolean gst_audiosink_start(GstElement *element,
GstElementState state) {
g_return_if_fail(GST_IS_AUDIOSINK(element));
if (gst_audiosink_open_audio(GST_AUDIOSINK(element)) == TRUE) {
gst_element_set_state(element,GST_STATE_RUNNING | state);
return TRUE;
}
return FALSE;
}
static gboolean gst_audiosink_stop(GstElement *element) {
g_return_if_fail(GST_IS_AUDIOSINK(element));
gst_audiosink_close_audio(GST_AUDIOSINK(element));
gst_element_set_state(element,~GST_STATE_RUNNING);
return TRUE;
}
static gboolean gst_audiosink_change_state(GstElement *element,
GstElementState state) {
g_return_if_fail(GST_IS_AUDIOSINK(element));
switch (state) {
case GST_STATE_RUNNING:
if (!gst_audiosink_open_audio(GST_AUDIOSINK(element)))
return FALSE;
break;
case ~GST_STATE_RUNNING:
gst_audiosink_close_audio(GST_AUDIOSINK(element));
break;
default:
break;
}
if (GST_ELEMENT_CLASS(parent_class)->change_state)
return GST_ELEMENT_CLASS(parent_class)->change_state(element,state);
return TRUE;
}