gstreamer/plugins/elements/gstaudiosink.c
Richard Boulton 2211a01d91 Various small fixes.
Original commit message from CVS:
Various small fixes.
2001-01-03 18:55:28 +00:00

429 lines
12 KiB
C

/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
*
* gstaudiosink.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.
*/
#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,
GST_PAD_FACTORY_CAPS (
"audiosink_sink",
"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 GstElementClass *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_ELEMENT, &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_ELEMENT);
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();
audiosink->format = 16;
audiosink->channels = 2;
audiosink->frequency = 44100;
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))) {
GST_DEBUG (0,"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 >= 0) {
if (!audiosink->mute) {
gst_clock_wait (audiosink->clock, GST_BUFFER_TIMESTAMP (buf), GST_OBJECT (audiosink));
ioctl (audiosink->fd, SNDCTL_DSP_GETOSPACE, &ospace);
GST_DEBUG (0,"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;
}