mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-27 18:50:48 +00:00
6cc1c73d2b
Original commit message from CVS: don't mix tabs and spaces
485 lines
13 KiB
C
485 lines
13 KiB
C
/* GStreamer
|
|
* Copyright (C) <2001> Richard Boulton <richard-gst@tartarus.org>
|
|
*
|
|
* Based on example.c:
|
|
* 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.
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
#include "config.h"
|
|
#endif
|
|
#include "esdsink.h"
|
|
#include <esd.h>
|
|
#include <unistd.h>
|
|
#include <errno.h>
|
|
|
|
GST_DEBUG_CATEGORY_EXTERN (esd_debug);
|
|
#define GST_CAT_DEFAULT esd_debug
|
|
|
|
/* elementfactory information */
|
|
static GstElementDetails esdsink_details = {
|
|
"Esound audio sink",
|
|
"Sink/Audio",
|
|
"Plays audio to an esound server",
|
|
"Richard Boulton <richard-gst@tartarus.org>",
|
|
};
|
|
|
|
/* Signals and args */
|
|
enum
|
|
{
|
|
/* FILL ME */
|
|
LAST_SIGNAL
|
|
};
|
|
|
|
enum
|
|
{
|
|
ARG_0,
|
|
ARG_MUTE,
|
|
ARG_HOST,
|
|
ARG_SYNC,
|
|
ARG_FALLBACK,
|
|
};
|
|
|
|
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-raw-int, "
|
|
"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", "
|
|
"signed = (boolean) TRUE, "
|
|
"width = (int) 16, "
|
|
"depth = (int) 16, " "rate = 44100, " "channels = 2")
|
|
);
|
|
|
|
static void gst_esdsink_base_init (gpointer g_class);
|
|
static void gst_esdsink_class_init (gpointer g_class, gpointer class_data);
|
|
static void gst_esdsink_init (GTypeInstance * instance, gpointer g_class);
|
|
|
|
static gboolean gst_esdsink_open_audio (GstEsdsink * sink);
|
|
static void gst_esdsink_close_audio (GstEsdsink * sink);
|
|
static GstElementStateReturn gst_esdsink_change_state (GstElement * element);
|
|
|
|
static GstClockTime gst_esdsink_get_time (GstClock * clock, gpointer data);
|
|
static GstClock *gst_esdsink_get_clock (GstElement * element);
|
|
static void gst_esdsink_set_clock (GstElement * element, GstClock * clock);
|
|
static void gst_esdsink_chain (GstPad * pad, GstData * _data);
|
|
|
|
static void gst_esdsink_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec);
|
|
static void gst_esdsink_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec);
|
|
|
|
static GstElementClass *parent_class = NULL;
|
|
|
|
/*static guint gst_esdsink_signals[LAST_SIGNAL] = { 0 }; */
|
|
|
|
GType
|
|
gst_esdsink_get_type (void)
|
|
{
|
|
static GType esdsink_type = 0;
|
|
|
|
if (!esdsink_type) {
|
|
static const GTypeInfo esdsink_info = {
|
|
sizeof (GstEsdsinkClass),
|
|
gst_esdsink_base_init,
|
|
NULL,
|
|
gst_esdsink_class_init,
|
|
NULL,
|
|
NULL,
|
|
sizeof (GstEsdsink),
|
|
0,
|
|
gst_esdsink_init,
|
|
};
|
|
|
|
esdsink_type =
|
|
g_type_register_static (GST_TYPE_ELEMENT, "GstEsdsink", &esdsink_info,
|
|
0);
|
|
}
|
|
return esdsink_type;
|
|
}
|
|
|
|
static void
|
|
gst_esdsink_base_init (gpointer g_class)
|
|
{
|
|
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
|
|
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&sink_factory));
|
|
gst_element_class_set_details (element_class, &esdsink_details);
|
|
}
|
|
|
|
static void
|
|
gst_esdsink_class_init (gpointer g_class, gpointer class_data)
|
|
{
|
|
GObjectClass *gobject_class = G_OBJECT_CLASS (g_class);
|
|
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
|
|
|
|
parent_class = g_type_class_peek_parent (g_class);
|
|
|
|
g_object_class_install_property (gobject_class, ARG_MUTE, g_param_spec_boolean ("mute", "mute", "mute", TRUE, G_PARAM_READWRITE)); /* CHECKME */
|
|
g_object_class_install_property (gobject_class, ARG_HOST, g_param_spec_string ("host", "host", "host", NULL, G_PARAM_READWRITE)); /* CHECKME */
|
|
g_object_class_install_property (gobject_class, ARG_SYNC,
|
|
g_param_spec_boolean ("sync", "sync", "Synchronize output to clock",
|
|
TRUE, G_PARAM_READWRITE));
|
|
#if 0
|
|
/* This option is disabled because it is dumb in GStreamer's architecture. */
|
|
g_object_class_install_property (gobject_class, ARG_FALLBACK,
|
|
g_param_spec_boolean ("fallback", "fallback",
|
|
"Fall back to using OSS if Esound daemon is not present", FALSE,
|
|
G_PARAM_READWRITE));
|
|
#endif
|
|
|
|
gobject_class->set_property = gst_esdsink_set_property;
|
|
gobject_class->get_property = gst_esdsink_get_property;
|
|
|
|
gstelement_class->change_state = gst_esdsink_change_state;
|
|
gstelement_class->set_clock = gst_esdsink_set_clock;
|
|
gstelement_class->get_clock = gst_esdsink_get_clock;
|
|
}
|
|
|
|
static void
|
|
gst_esdsink_init (GTypeInstance * instance, gpointer g_class)
|
|
{
|
|
GstEsdsink *esdsink = GST_ESDSINK (instance);
|
|
|
|
esdsink->sinkpad =
|
|
gst_pad_new_from_template (gst_element_class_get_pad_template
|
|
(GST_ELEMENT_GET_CLASS (instance), "sink"), "sink");
|
|
gst_element_add_pad (GST_ELEMENT (esdsink), esdsink->sinkpad);
|
|
gst_pad_set_chain_function (esdsink->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_esdsink_chain));
|
|
|
|
GST_FLAG_SET (esdsink, GST_ELEMENT_EVENT_AWARE);
|
|
|
|
esdsink->mute = FALSE;
|
|
esdsink->fd = -1;
|
|
/* FIXME: get default from somewhere better than just putting them inline. */
|
|
/*esdsink->negotiated = FALSE; */
|
|
/* we have static caps on our template, so it always is negotiated */
|
|
esdsink->negotiated = TRUE;
|
|
esdsink->format = 16;
|
|
esdsink->depth = 16;
|
|
esdsink->channels = 2;
|
|
esdsink->frequency = 44100;
|
|
esdsink->bytes_per_sample = esdsink->channels * (esdsink->depth / 8);
|
|
esdsink->host = getenv ("ESPEAKER");
|
|
esdsink->provided_clock =
|
|
gst_audio_clock_new ("esdclock", gst_esdsink_get_time, esdsink);
|
|
gst_object_set_parent (GST_OBJECT (esdsink->provided_clock),
|
|
GST_OBJECT (esdsink));
|
|
esdsink->sync = TRUE;
|
|
esdsink->fallback = FALSE;
|
|
}
|
|
|
|
#ifdef unused
|
|
static GstPadLinkReturn
|
|
gst_esdsink_link (GstPad * pad, const GstCaps * caps)
|
|
{
|
|
GstEsdsink *esdsink;
|
|
GstStructure *structure;
|
|
|
|
esdsink = GST_ESDSINK (gst_pad_get_parent (pad));
|
|
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
gst_structure_get_int (structure, "depth", &esdsink->depth);
|
|
gst_structure_get_int (structure, "channels", &esdsink->channels);
|
|
gst_structure_get_int (structure, "rate", &esdsink->frequency);
|
|
|
|
esdsink->bytes_per_sample = esdsink->channels * (esdsink->depth / 8);
|
|
|
|
gst_esdsink_close_audio (esdsink);
|
|
if (gst_esdsink_open_audio (esdsink)) {
|
|
esdsink->negotiated = TRUE;
|
|
return GST_PAD_LINK_OK;
|
|
}
|
|
/* FIXME: is it supposed to be correct to have closed audio when caps nego
|
|
failed? */
|
|
|
|
GST_DEBUG ("esd link function could not negotiate, returning delayed");
|
|
return GST_PAD_LINK_REFUSED;
|
|
}
|
|
#endif
|
|
|
|
static GstClockTime
|
|
gst_esdsink_get_time (GstClock * clock, gpointer data)
|
|
{
|
|
GstEsdsink *esdsink = GST_ESDSINK (data);
|
|
GstClockTime res;
|
|
|
|
res = (esdsink->handled * GST_SECOND) / esdsink->frequency;
|
|
//- GST_SECOND * 2;
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstClock *
|
|
gst_esdsink_get_clock (GstElement * element)
|
|
{
|
|
GstEsdsink *esdsink;
|
|
|
|
esdsink = GST_ESDSINK (element);
|
|
|
|
return GST_CLOCK (esdsink->provided_clock);
|
|
}
|
|
|
|
static void
|
|
gst_esdsink_set_clock (GstElement * element, GstClock * clock)
|
|
{
|
|
GstEsdsink *esdsink;
|
|
|
|
esdsink = GST_ESDSINK (element);
|
|
|
|
esdsink->clock = clock;
|
|
}
|
|
|
|
static void
|
|
gst_esdsink_chain (GstPad * pad, GstData * _data)
|
|
{
|
|
GstBuffer *buf = GST_BUFFER (_data);
|
|
GstEsdsink *esdsink;
|
|
|
|
esdsink = GST_ESDSINK (gst_pad_get_parent (pad));
|
|
|
|
if (!esdsink->negotiated) {
|
|
GST_ELEMENT_ERROR (esdsink, CORE, NEGOTIATION, (NULL),
|
|
("element wasn't negotiated before chain function"));
|
|
goto done;
|
|
}
|
|
|
|
if (GST_IS_EVENT (buf)) {
|
|
GstEvent *event = GST_EVENT (buf);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_EOS:
|
|
gst_audio_clock_set_active (GST_AUDIO_CLOCK (esdsink->provided_clock),
|
|
FALSE);
|
|
gst_pad_event_default (pad, event);
|
|
return;
|
|
default:
|
|
gst_pad_event_default (pad, event);
|
|
return;
|
|
}
|
|
gst_event_unref (event);
|
|
return;
|
|
}
|
|
|
|
if (GST_BUFFER_DATA (buf) != NULL) {
|
|
if (!esdsink->mute && esdsink->fd >= 0) {
|
|
guchar *data = GST_BUFFER_DATA (buf);
|
|
gint size = GST_BUFFER_SIZE (buf);
|
|
gint to_write = 0;
|
|
|
|
to_write = size;
|
|
|
|
GST_LOG ("fd=%d data=%p size=%d",
|
|
esdsink->fd, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
|
|
while (to_write > 0) {
|
|
int done;
|
|
|
|
done = write (esdsink->fd, data, to_write);
|
|
|
|
if (done < 0) {
|
|
if (errno == EINTR) {
|
|
goto done;
|
|
}
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
to_write -= done;
|
|
data += done;
|
|
esdsink->handled += done / esdsink->bytes_per_sample;
|
|
}
|
|
|
|
}
|
|
}
|
|
|
|
gst_audio_clock_update_time ((GstAudioClock *) esdsink->provided_clock,
|
|
gst_esdsink_get_time (esdsink->provided_clock, esdsink));
|
|
|
|
done:
|
|
gst_buffer_unref (buf);
|
|
}
|
|
|
|
static void
|
|
gst_esdsink_set_property (GObject * object, guint prop_id, const GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstEsdsink *esdsink;
|
|
|
|
/* it's not null if we got it, but it might not be ours */
|
|
g_return_if_fail (GST_IS_ESDSINK (object));
|
|
esdsink = GST_ESDSINK (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_MUTE:
|
|
esdsink->mute = g_value_get_boolean (value);
|
|
break;
|
|
case ARG_HOST:
|
|
g_free (esdsink->host);
|
|
if (g_value_get_string (value) == NULL)
|
|
esdsink->host = NULL;
|
|
else
|
|
esdsink->host = g_strdup (g_value_get_string (value));
|
|
break;
|
|
case ARG_SYNC:
|
|
esdsink->sync = g_value_get_boolean (value);
|
|
break;
|
|
case ARG_FALLBACK:
|
|
esdsink->fallback = g_value_get_boolean (value);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_esdsink_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstEsdsink *esdsink;
|
|
|
|
esdsink = GST_ESDSINK (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_MUTE:
|
|
g_value_set_boolean (value, esdsink->mute);
|
|
break;
|
|
case ARG_HOST:
|
|
g_value_set_string (value, esdsink->host);
|
|
break;
|
|
case ARG_SYNC:
|
|
g_value_set_boolean (value, esdsink->sync);
|
|
break;
|
|
case ARG_FALLBACK:
|
|
g_value_set_boolean (value, esdsink->fallback);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
gboolean
|
|
gst_esdsink_factory_init (GstPlugin * plugin)
|
|
{
|
|
if (!gst_element_register (plugin, "esdsink", GST_RANK_NONE,
|
|
GST_TYPE_ESDSINK))
|
|
return FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_esdsink_open_audio (GstEsdsink * sink)
|
|
{
|
|
/* Name used by esound for this connection. */
|
|
const char *connname = "GStreamer";
|
|
|
|
/* Bitmap describing audio format. */
|
|
esd_format_t esdformat = ESD_STREAM | ESD_PLAY;
|
|
|
|
g_return_val_if_fail (sink->fd == -1, FALSE);
|
|
|
|
if (sink->depth == 16)
|
|
esdformat |= ESD_BITS16;
|
|
else if (sink->depth == 8)
|
|
esdformat |= ESD_BITS8;
|
|
else {
|
|
GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL),
|
|
("invalid bit depth (%d)", sink->depth));
|
|
return FALSE;
|
|
}
|
|
|
|
if (sink->channels == 2)
|
|
esdformat |= ESD_STEREO;
|
|
else if (sink->channels == 1)
|
|
esdformat |= ESD_MONO;
|
|
else {
|
|
GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL),
|
|
("invalid number of channels (%d)", sink->channels));
|
|
return FALSE;
|
|
}
|
|
|
|
GST_INFO ("attempting to open connection to esound server");
|
|
if (sink->fallback) {
|
|
sink->fd =
|
|
esd_play_stream_fallback (esdformat, sink->frequency, sink->host,
|
|
connname);
|
|
} else {
|
|
sink->fd =
|
|
esd_play_stream (esdformat, sink->frequency, sink->host, connname);
|
|
}
|
|
if (sink->fd < 0) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, (NULL),
|
|
("can't open connection to esound server"));
|
|
return FALSE;
|
|
}
|
|
GST_INFO ("successfully opened connection to esound server");
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_esdsink_close_audio (GstEsdsink * sink)
|
|
{
|
|
if (sink->fd < 0)
|
|
return;
|
|
|
|
close (sink->fd);
|
|
sink->fd = -1;
|
|
|
|
GST_INFO ("esdsink: closed sound device");
|
|
}
|
|
|
|
static GstElementStateReturn
|
|
gst_esdsink_change_state (GstElement * element)
|
|
{
|
|
GstEsdsink *esdsink;
|
|
|
|
esdsink = GST_ESDSINK (element);
|
|
|
|
switch (GST_STATE_TRANSITION (element)) {
|
|
case GST_STATE_NULL_TO_READY:
|
|
if (!gst_esdsink_open_audio (GST_ESDSINK (element))) {
|
|
return GST_STATE_FAILURE;
|
|
}
|
|
break;
|
|
case GST_STATE_READY_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_PAUSED_TO_PLAYING:
|
|
gst_audio_clock_set_active (GST_AUDIO_CLOCK (esdsink->provided_clock),
|
|
TRUE);
|
|
break;
|
|
case GST_STATE_PLAYING_TO_PAUSED:
|
|
gst_audio_clock_set_active (GST_AUDIO_CLOCK (esdsink->provided_clock),
|
|
FALSE);
|
|
esdsink->resync = TRUE;
|
|
break;
|
|
case GST_STATE_PAUSED_TO_READY:
|
|
break;
|
|
case GST_STATE_READY_TO_NULL:
|
|
gst_esdsink_close_audio (GST_ESDSINK (element));
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (GST_ELEMENT_CLASS (parent_class)->change_state)
|
|
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
|
|
|
|
return GST_STATE_SUCCESS;
|
|
}
|