gstreamer/gst/elements/gstaudiosink.c
Wim Taymans 9bae9d4b91 More Docs updates.
Original commit message from CVS:
More Docs updates.
Added plugin documentation. I fear we need a gstdoc implementation
that loads plugins and does introspection on them. I think we should
automatically create the docs for the pads and mime types the plugins
provide. Does anyone have enough perl knowledge to add these features? I
allready changed the C code to output the pad definitions but my perl
knowledge is too limited, for now, to implement the rest of the needed
functionality...
2000-10-25 19:09:53 +00:00

390 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 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);
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 */
};
#define GST_TYPE_AUDIOSINK_FORMATS (gst_audiosink_formats_get_type())
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())
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 void gst_audiosink_class_init(GstAudioSinkClass *klass);
static void gst_audiosink_init(GstAudioSink *audiosink);
static GstSinkClass *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_SINK,&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);
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("sink",GST_PAD_SINK);
gst_element_add_pad(GST_ELEMENT(audiosink),audiosink->sinkpad);
gst_pad_set_type_id(audiosink->sinkpad,gst_audiosink_type_audio);
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));
//audiosink->clocktime = 0LL;
GST_FLAG_SET(audiosink, GST_ELEMENT_THREAD_SUGGESTED);
}
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);
}
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;
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) {
if (!gst_audiosink_type_audio)
gst_audiosink_type_audio = gst_type_find_by_mime("audio/raw");
gst_type_add_sink(gst_audiosink_type_audio, factory);
return TRUE;
}