gstreamer/ext/esd/esdsink.c
Josep Torra Valles c4e7ebfe35 Fix a bunch of problems discovered by the Forte compiler, mostly type mixups and pointer arithmetics with void pointe...
Original commit message from CVS:
Patch by: Josep Torra Valles  <josep at fluendo com>
* ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform):
* ext/esd/esdsink.c: (gst_esdsink_write):
* ext/flac/gstflacdec.c: (gst_flac_dec_length),
(gst_flac_dec_read_seekable), (gst_flac_dec_chain),
(gst_flac_dec_send_newsegment):
* ext/flac/gstflacenc.c: (gst_flac_enc_seek_callback),
(gst_flac_enc_tell_callback):
* ext/jpeg/smokecodec.c: (find_best_size), (smokecodec_encode),
(smokecodec_parse_header), (smokecodec_decode):
* gst/avi/gstavimux.c: (gst_avi_mux_write_avix_index):
* gst/debug/efence.c: (gst_fenced_buffer_alloc):
* gst/goom/Makefile.am:
* gst/goom/gstgoom.c:
* gst/icydemux/gsticydemux.c: (gst_icydemux_typefind_or_forward):
* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
* gst/udp/gstudpsink.c:
* gst/udp/gstudpsrc.c:
* gst/wavparse/gstwavparse.c: (gst_wavparse_change_state):
* sys/sunaudio/gstsunaudiomixertrack.h:
Fix a bunch of problems discovered by the Forte compiler, mostly type
mixups and pointer arithmetics with void pointers. Fixes #362603.
2006-10-16 18:22:47 +00:00

472 lines
12 KiB
C

/* GStreamer
* Copyright (C) <2005> Arwed v. Merkatz <v.merkatz@gmx.net>
*
* Roughly based on the gstreamer 0.8 esdsink plugin:
* Copyright (C) <2001> Richard Boulton <richard-gst@tartarus.org>
*
* esdsink.c: an EsounD audio sink
*
* 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.
*/
/**
* SECTION:element-esdsink
* @see_also: #GstAlsaSink, #GstAutoAudioSink
*
* <refsect2>
* <para>
* This element outputs sound to an already-running Enlightened Sound Daemon
* (ESound Daemon, esd). Note that a sound daemon will never be auto-spawned
* through this element (regardless of the system configuration), since this
* is actively prevented by the element. If you must use esd, you need to
* make sure it is started automatically with your session or otherwise.
* </para>
* <para>
* TODO: insert some comments about how sucky esd is and that all the cool
* kids use pulseaudio or whatever these days.
* </para>
* <para>
* Simple example pipeline that plays an Ogg/Vorbis file via esd:
* <programlisting>
* gst-launch -v filesrc location=foo.ogg ! decodebin ! audioconvert ! audioresample ! esdsink
* </programlisting>
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "esdsink.h"
#include <esd.h>
#include <unistd.h>
#include <errno.h>
#include <gst/gst-i18n-plugin.h>
/* wtay: from my esd.h (debian unstable libesd0-dev 0.2.36-3) */
#ifndef ESD_MAX_WRITE_SIZE
#define ESD_MAX_WRITE_SIZE (21 * 4096)
#endif
GST_DEBUG_CATEGORY_EXTERN (esd_debug);
#define GST_CAT_DEFAULT esd_debug
/* elementfactory information */
static const GstElementDetails esdsink_details =
GST_ELEMENT_DETAILS ("Esound audio sink",
"Sink/Audio",
"Plays audio to an esound server",
"Arwed von Merkatz <v.merkatz@gmx.net>");
enum
{
PROP_0,
PROP_HOST
};
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) TRUE, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 2 ]; "
"audio/x-raw-int, "
"signed = (boolean) { true, false }, "
"width = (int) 8, "
"depth = (int) 8, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
);
static void gst_esdsink_finalize (GObject * object);
static GstCaps *gst_esdsink_getcaps (GstBaseSink * bsink);
static gboolean gst_esdsink_open (GstAudioSink * asink);
static gboolean gst_esdsink_close (GstAudioSink * asink);
static gboolean gst_esdsink_prepare (GstAudioSink * asink,
GstRingBufferSpec * spec);
static gboolean gst_esdsink_unprepare (GstAudioSink * asink);
static guint gst_esdsink_write (GstAudioSink * asink, gpointer data,
guint length);
static guint gst_esdsink_delay (GstAudioSink * asink);
static void gst_esdsink_reset (GstAudioSink * asink);
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);
GST_BOILERPLATE (GstEsdSink, gst_esdsink, GstAudioSink, GST_TYPE_AUDIO_SINK);
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 (GstEsdSinkClass * klass)
{
GObjectClass *gobject_class;
GstBaseSinkClass *gstbasesink_class;
GstBaseAudioSinkClass *gstbaseaudiosink_class;
GstAudioSinkClass *gstaudiosink_class;
gobject_class = (GObjectClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
gstaudiosink_class = (GstAudioSinkClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_esdsink_finalize;
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_esdsink_getcaps);
gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_esdsink_open);
gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_esdsink_close);
gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_esdsink_prepare);
gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_esdsink_unprepare);
gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_esdsink_write);
gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_esdsink_delay);
gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_esdsink_reset);
gobject_class->set_property = gst_esdsink_set_property;
gobject_class->get_property = gst_esdsink_get_property;
/* default value is filled in the _init method */
g_object_class_install_property (gobject_class, PROP_HOST,
g_param_spec_string ("host", "Host",
"The host running the esound daemon", NULL, G_PARAM_READWRITE));
}
static void
gst_esdsink_init (GstEsdSink * esdsink, GstEsdSinkClass * klass)
{
esdsink->fd = -1;
esdsink->ctrl_fd = -1;
esdsink->host = g_strdup (g_getenv ("ESPEAKER"));
}
static void
gst_esdsink_finalize (GObject * object)
{
GstEsdSink *esdsink = GST_ESDSINK (object);
gst_caps_replace (&esdsink->cur_caps, NULL);
g_free (esdsink->host);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstCaps *
gst_esdsink_getcaps (GstBaseSink * bsink)
{
GstEsdSink *esdsink;
esdsink = GST_ESDSINK (bsink);
/* no fd, we're done with the template caps */
if (esdsink->ctrl_fd < 0 || esdsink->cur_caps == NULL) {
GST_LOG_OBJECT (esdsink, "getcaps called, returning template caps");
return NULL;
}
GST_LOG_OBJECT (esdsink, "returning %" GST_PTR_FORMAT, esdsink->cur_caps);
return gst_caps_ref (esdsink->cur_caps);
}
static gboolean
gst_esdsink_open (GstAudioSink * asink)
{
esd_server_info_t *server_info;
GstPadTemplate *pad_template;
GstEsdSink *esdsink;
gchar *saved_env;
gint i;
esdsink = GST_ESDSINK (asink);
GST_DEBUG_OBJECT (esdsink, "open");
/* ensure libesd doesn't auto-spawn a sound daemon if none is running yet */
saved_env = g_strdup (g_getenv ("ESD_NO_SPAWN"));
g_setenv ("ESD_NO_SPAWN", "1", TRUE);
/* now try to connect to any existing/running sound daemons */
esdsink->ctrl_fd = esd_open_sound (esdsink->host);
/* and restore the previous state */
if (saved_env != NULL) {
g_setenv ("ESD_NO_SPAWN", saved_env, TRUE);
} else {
g_unsetenv ("ESD_NO_SPAWN");
}
g_free (saved_env);
if (esdsink->ctrl_fd < 0)
goto couldnt_connect;
/* get server info */
server_info = esd_get_server_info (esdsink->ctrl_fd);
if (!server_info)
goto no_server_info;
GST_INFO_OBJECT (esdsink, "got server info rate: %i", server_info->rate);
pad_template = gst_static_pad_template_get (&sink_factory);
esdsink->cur_caps = gst_caps_copy (gst_pad_template_get_caps (pad_template));
for (i = 0; i < esdsink->cur_caps->structs->len; i++) {
GstStructure *s;
s = gst_caps_get_structure (esdsink->cur_caps, i);
gst_structure_set (s, "rate", G_TYPE_INT, server_info->rate, NULL);
}
esd_free_server_info (server_info);
GST_INFO_OBJECT (esdsink, "server caps: %" GST_PTR_FORMAT, esdsink->cur_caps);
return TRUE;
/* ERRORS */
couldnt_connect:
{
GST_ELEMENT_ERROR (esdsink, RESOURCE, OPEN_WRITE,
(_("Could not establish connection to sound server")),
("can't open connection to esound server"));
return FALSE;
}
no_server_info:
{
GST_ELEMENT_ERROR (esdsink, RESOURCE, OPEN_WRITE,
(_("Failed to query sound server capabilities")),
("couldn't get server info!"));
return FALSE;
}
}
static gboolean
gst_esdsink_close (GstAudioSink * asink)
{
GstEsdSink *esdsink = GST_ESDSINK (asink);
GST_DEBUG_OBJECT (esdsink, "close");
gst_caps_replace (&esdsink->cur_caps, NULL);
esd_close (esdsink->ctrl_fd);
esdsink->ctrl_fd = -1;
return TRUE;
}
static gboolean
gst_esdsink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
{
GstEsdSink *esdsink = GST_ESDSINK (asink);
esd_format_t esdformat;
/* Name used by esound for this connection. */
const char connname[] = "GStreamer";
GST_DEBUG_OBJECT (esdsink, "prepare");
/* Bitmap describing audio format. */
esdformat = ESD_STREAM | ESD_PLAY;
switch (spec->depth) {
case 8:
esdformat |= ESD_BITS8;
break;
case 16:
esdformat |= ESD_BITS16;
break;
default:
goto unsupported_depth;
}
switch (spec->channels) {
case 1:
esdformat |= ESD_MONO;
break;
case 2:
esdformat |= ESD_STEREO;
break;
default:
goto unsupported_channels;
}
GST_INFO_OBJECT (esdsink,
"attempting to open data connection to esound server");
esdsink->fd =
esd_play_stream (esdformat, spec->rate, esdsink->host, connname);
if ((esdsink->fd < 0) || (esdsink->ctrl_fd < 0))
goto cannot_open;
esdsink->rate = spec->rate;
spec->segsize = ESD_BUF_SIZE;
spec->segtotal = (ESD_MAX_WRITE_SIZE / spec->segsize);
/* FIXME: this is wrong for signed ints (and the
* audioringbuffers should do it for us anyway) */
spec->silence_sample[0] = 0;
spec->silence_sample[1] = 0;
spec->silence_sample[2] = 0;
spec->silence_sample[3] = 0;
GST_INFO_OBJECT (esdsink, "successfully opened connection to esound server");
return TRUE;
/* ERRORS */
unsupported_depth:
{
GST_ELEMENT_ERROR (esdsink, STREAM, WRONG_TYPE, (NULL),
("can't handle sample depth of %d, only 8 or 16 supported",
spec->depth));
return FALSE;
}
unsupported_channels:
{
GST_ELEMENT_ERROR (esdsink, STREAM, WRONG_TYPE, (NULL),
("can't handle %d channels, only 1 or 2 supported", spec->channels));
return FALSE;
}
cannot_open:
{
GST_ELEMENT_ERROR (esdsink, RESOURCE, OPEN_WRITE,
(_("Could not establish connection to sound server")),
("can't open connection to esound server"));
return FALSE;
}
}
static gboolean
gst_esdsink_unprepare (GstAudioSink * asink)
{
GstEsdSink *esdsink = GST_ESDSINK (asink);
if ((esdsink->fd < 0) && (esdsink->ctrl_fd < 0))
return TRUE;
close (esdsink->fd);
esdsink->fd = -1;
GST_INFO_OBJECT (esdsink, "closed sound device");
return TRUE;
}
static guint
gst_esdsink_write (GstAudioSink * asink, gpointer data, guint length)
{
GstEsdSink *esdsink = GST_ESDSINK (asink);
gint to_write = 0;
to_write = length;
while (to_write > 0) {
int done;
done = write (esdsink->fd, data, to_write);
if (done < 0)
goto write_error;
to_write -= done;
data = (char *) data + done;
}
return length;
/* ERRORS */
write_error:
{
GST_ELEMENT_ERROR (esdsink, RESOURCE, WRITE,
("Failed to write data to the esound daemon"), GST_ERROR_SYSTEM);
return 0;
}
}
static guint
gst_esdsink_delay (GstAudioSink * asink)
{
GstEsdSink *esdsink = GST_ESDSINK (asink);
guint latency;
latency = esd_get_latency (esdsink->ctrl_fd);
/* latency is measured in samples at a rate of 44100, this
* cannot overflow. */
latency = latency * G_GINT64_CONSTANT (44100) / esdsink->rate;
GST_DEBUG_OBJECT (asink, "got latency: %u", latency);
return latency;
}
static void
gst_esdsink_reset (GstAudioSink * asink)
{
GST_DEBUG_OBJECT (asink, "reset called");
}
static void
gst_esdsink_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstEsdSink *esdsink = GST_ESDSINK (object);
switch (prop_id) {
case PROP_HOST:
g_free (esdsink->host);
esdsink->host = g_value_dup_string (value);
break;
default:
break;
}
}
static void
gst_esdsink_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstEsdSink *esdsink = GST_ESDSINK (object);
switch (prop_id) {
case PROP_HOST:
g_value_set_string (value, esdsink->host);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}