mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-28 12:41:05 +00:00
1762dfbf98
Original commit message from CVS: initial checkin
291 lines
8.6 KiB
C
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;
|
|
}
|