gstreamer/ext/shout2/gstshout2.c

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2006> Tim-Philipp Müller <tim centricular net>
*
* 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 "gstshout2.h"
#include <stdlib.h>
#include <string.h>
#include "gst/gst-i18n-plugin.h"
GST_DEBUG_CATEGORY_STATIC (shout2_debug);
#define GST_CAT_DEFAULT shout2_debug
static const GstElementDetails shout2send_details =
better/unified long descriptions Original commit message from CVS: * ext/aalib/gstaasink.c: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttimeoverlay.c: * ext/cdio/gstcdiocddasrc.c: * ext/dv/gstdvdec.c: * ext/esd/esdmon.c: * ext/esd/esdsink.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init): * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init): * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init): * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init): * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/pixbufscale.c: * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init): * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init): * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokeenc.c: * ext/libcaca/gstcacasink.c: * ext/libmng/gstmngdec.c: * ext/libmng/gstmngenc.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/mikmod/gstmikmod.c: * ext/raw1394/gstdv1394src.c: * ext/shout2/gstshout2.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_base_init): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_base_init): * gst/avi/gstavimux.c: (gst_avimux_base_init): * gst/cutter/gstcutter.c: * gst/debug/breakmydata.c: * gst/debug/efence.c: * gst/debug/gstnavigationtest.c: * gst/debug/negotiation.c: * gst/debug/progressreport.c: * gst/debug/testplugin.c: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstquark.c: * gst/effectv/gstrev.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/law/alaw-decode.c: (gst_alawdec_base_init): * gst/law/alaw-encode.c: (gst_alawenc_base_init): * gst/law/mulaw-decode.c: (gst_mulawdec_base_init): * gst/law/mulaw-encode.c: (gst_mulawenc_base_init): * gst/level/gstlevel.c: * gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init): * gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init): * gst/median/gstmedian.c: * gst/monoscope/gstmonoscope.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/oldcore/gstmd5sink.c: * gst/oldcore/gstmultifilesrc.c: * gst/oldcore/gstpipefilter.c: * gst/oldcore/gstshaper.c: * gst/oldcore/gststatistics.c: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtsp/gstrtpdec.c: * gst/smpte/gstsmpte.c: * gst/videobox/gstvideobox.c: * gst/videofilter/gstgamma.c: (gst_gamma_base_init): * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideotemplate.c: (gst_videotemplate_base_init): * gst/videomixer/videomixer.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init): better/unified long descriptions Fixed #336602 Some cleanups to auparse, don't send multiple newsegments.
2006-03-30 15:37:05 +00:00
GST_ELEMENT_DETAILS ("Icecast network sink",
"Sink/Network",
"Sends data to an icecast server",
"Wim Taymans <wim.taymans@chello.be>\n"
"Pedro Corte-Real <typo@netcabo.pt>\n"
"Zaheer Abbas Merali <zaheerabbas at merali dot org>");
enum
{
SIGNAL_CONNECTION_PROBLEM, /* 0.11 FIXME: remove this */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_IP, /* the ip of the server */
ARG_PORT, /* the encoder port number on the server */
ARG_PASSWORD, /* the encoder password on the server */
ARG_PUBLIC, /* is this stream public? */
ARG_STREAMNAME, /* Name of the stream */
ARG_DESCRIPTION, /* Description of the stream */
ARG_GENRE, /* Genre of the stream */
ARG_PROTOCOL, /* Protocol to connect with */
ARG_MOUNT, /* mountpoint of stream (icecast only) */
ARG_URL /* Url of stream (I'm guessing) */
};
#define DEFAULT_IP "127.0.0.1"
#define DEFAULT_PORT 8000
#define DEFAULT_PASSWORD "hackme"
#define DEFAULT_STREAMNAME ""
#define DEFAULT_DESCRIPTION ""
#define DEFAULT_GENRE ""
#define DEFAULT_MOUNT ""
#define DEFAULT_URL ""
#define DEFAULT_PROTOCOL SHOUT2SEND_PROTOCOL_HTTP
static GstElementClass *parent_class = NULL;
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/ogg; "
"audio/mpeg, mpegversion = (int) 1, layer = (int) [ 1, 3 ]")
);
static void gst_shout2send_class_init (GstShout2sendClass * klass);
static void gst_shout2send_base_init (GstShout2sendClass * klass);
static void gst_shout2send_init (GstShout2send * shout2send);
static gboolean gst_shout2send_event (GstBaseSink * sink, GstEvent * event);
static GstFlowReturn gst_shout2send_render (GstBaseSink * sink,
GstBuffer * buffer);
static gboolean gst_shout2send_start (GstBaseSink * basesink);
static gboolean gst_shout2send_stop (GstBaseSink * basesink);
static void gst_shout2send_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_shout2send_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_shout2send_setcaps (GstPad * pad, GstCaps * caps);
static guint gst_shout2send_signals[LAST_SIGNAL] = { 0 };
#define GST_TYPE_SHOUT_PROTOCOL (gst_shout2send_protocol_get_type())
static GType
gst_shout2send_protocol_get_type (void)
{
static GType shout2send_protocol_type = 0;
static const GEnumValue shout2send_protocol[] = {
{SHOUT2SEND_PROTOCOL_XAUDIOCAST,
"Xaudiocast Protocol (icecast 1.3.x)", "xaudiocast"},
{SHOUT2SEND_PROTOCOL_ICY, "Icy Protocol (ShoutCast)", "icy"},
{SHOUT2SEND_PROTOCOL_HTTP, "Http Protocol (icecast 2.x)", "http"},
{0, NULL, NULL},
};
if (!shout2send_protocol_type) {
shout2send_protocol_type =
g_enum_register_static ("GstShout2SendProtocol", shout2send_protocol);
}
return shout2send_protocol_type;
}
GType
gst_shout2send_get_type (void)
{
static GType shout2send_type = 0;
if (!shout2send_type) {
static const GTypeInfo shout2send_info = {
sizeof (GstShout2sendClass),
(GBaseInitFunc) gst_shout2send_base_init,
NULL,
(GClassInitFunc) gst_shout2send_class_init,
NULL,
NULL,
sizeof (GstShout2send),
0,
(GInstanceInitFunc) gst_shout2send_init,
};
static const GInterfaceInfo tag_setter_info = {
NULL,
NULL,
NULL
};
shout2send_type =
g_type_register_static (GST_TYPE_BASE_SINK, "GstShout2send",
&shout2send_info, 0);
g_type_add_interface_static (shout2send_type, GST_TYPE_TAG_SETTER,
&tag_setter_info);
}
return shout2send_type;
}
static void
gst_shout2send_base_init (GstShout2sendClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_details (element_class, &shout2send_details);
GST_DEBUG_CATEGORY_INIT (shout2_debug, "shout2", 0, "shout2send element");
}
static void
gst_shout2send_class_init (GstShout2sendClass * klass)
{
GObjectClass *gobject_class;
GstBaseSinkClass *gstbasesink_class;
gobject_class = (GObjectClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) Original commit message from CVS: * ext/aalib/gstaasink.c: (gst_aasink_class_init): * ext/esd/esdsink.c: (gst_esdsink_class_init): * ext/flac/gstflactag.c: (gst_flac_tag_class_init): * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_class_init): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_class_init): * ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init): * ext/libcaca/gstcacasink.c: (gst_cacasink_class_init): * ext/libmng/gstmngdec.c: (gst_mngdec_class_init): * ext/libmng/gstmngenc.c: (gst_mngenc_class_init): * ext/libpng/gstpngdec.c: (gst_pngdec_class_init): * ext/libpng/gstpngenc.c: (gst_pngenc_class_init): * ext/mikmod/gstmikmod.c: (gst_mikmod_class_init): * ext/shout2/gstshout2.c: (gst_shout2send_class_init): * ext/speex/gstspeexenc.c: (gst_speexenc_class_init): * gst/alpha/gstalpha.c: (gst_alpha_class_init): * gst/avi/gstavimux.c: (gst_avimux_class_init): * gst/debug/efence.c: (gst_efence_class_init): * gst/debug/negotiation.c: (gst_negotiation_class_init): * gst/flx/gstflxdec.c: (gst_flxdec_class_init): * gst/goom/gstgoom.c: (gst_goom_class_init): * gst/id3demux/gstid3demux.c: (gst_id3demux_class_init): * gst/interleave/deinterleave.c: (deinterleave_class_init): * gst/interleave/interleave.c: (interleave_class_init): * gst/law/alaw-decode.c: (gst_alawdec_class_init): * gst/law/alaw-encode.c: (gst_alawenc_class_init): * gst/law/mulaw-encode.c: (gst_mulawenc_class_init): * gst/median/gstmedian.c: (gst_median_class_init): * gst/monoscope/gstmonoscope.c: (gst_monoscope_class_init): * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init): * gst/rtp/gstasteriskh263.c: (gst_asteriskh263_class_init): * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_class_init): * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_class_init): * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init): * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init): * gst/rtp/gstrtpdepay.c: (gst_rtp_depay_class_init): * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_class_init): * gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_class_init): * gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_class_init): * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_class_init): * gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init): * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init): * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init): * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_class_init): * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init): * gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_class_init): * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_class_init): * gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_class_init): * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_class_init): * gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_class_init): * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init): * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_class_init): * gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init): * gst/smpte/gstsmpte.c: (gst_smpte_class_init): * gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init): * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init): * gst/udp/gstudpsink.c: (gst_udpsink_class_init): * gst/videomixer/videomixer.c: (gst_videomixer_class_init): * gst/wavenc/gstwavenc.c: (gst_wavenc_class_init): * sys/oss/gstossdmabuffer.c: (gst_ossdmabuffer_class_init): * sys/oss/gstosssink.c: (gst_oss_sink_class_init): * sys/osxaudio/gstosxaudioelement.c: (gst_osxaudioelement_class_init): * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_class_init): * sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_class_init): * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_class_init): Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
2006-04-08 21:21:45 +00:00
parent_class = g_type_class_peek_parent (klass);
gobject_class->set_property = gst_shout2send_set_property;
gobject_class->get_property = gst_shout2send_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_IP,
g_param_spec_string ("ip", "ip", "ip", DEFAULT_IP, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_PORT,
g_param_spec_int ("port", "port", "port", 1, G_MAXUSHORT, DEFAULT_PORT,
G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_PASSWORD,
g_param_spec_string ("password", "password", "password", DEFAULT_PASSWORD,
G_PARAM_READWRITE));
/* metadata */
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_STREAMNAME,
g_param_spec_string ("streamname", "streamname", "name of the stream",
DEFAULT_STREAMNAME, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DESCRIPTION,
g_param_spec_string ("description", "description", "description",
DEFAULT_DESCRIPTION, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_GENRE,
g_param_spec_string ("genre", "genre", "genre", DEFAULT_GENRE,
G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_PROTOCOL,
g_param_spec_enum ("protocol", "protocol", "Connection Protocol to use",
GST_TYPE_SHOUT_PROTOCOL, DEFAULT_PROTOCOL, G_PARAM_READWRITE));
/* icecast only */
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MOUNT,
g_param_spec_string ("mount", "mount", "mount", DEFAULT_MOUNT,
G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_URL,
g_param_spec_string ("url", "url", "url", DEFAULT_URL,
G_PARAM_READWRITE));
/* signals */
gst_shout2send_signals[SIGNAL_CONNECTION_PROBLEM] =
g_signal_new ("connection-problem", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_CLEANUP, G_STRUCT_OFFSET (GstShout2sendClass,
connection_problem), NULL, NULL, g_cclosure_marshal_VOID__INT,
G_TYPE_NONE, 1, G_TYPE_INT);
gstbasesink_class->start = GST_DEBUG_FUNCPTR (gst_shout2send_start);
gstbasesink_class->stop = GST_DEBUG_FUNCPTR (gst_shout2send_stop);
gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_shout2send_render);
gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_shout2send_event);
}
static void
gst_shout2send_init (GstShout2send * shout2send)
{
gst_base_sink_set_sync (GST_BASE_SINK (shout2send), FALSE);
gst_pad_set_setcaps_function (GST_BASE_SINK_PAD (shout2send),
GST_DEBUG_FUNCPTR (gst_shout2send_setcaps));
shout2send->ip = g_strdup (DEFAULT_IP);
shout2send->port = DEFAULT_PORT;
shout2send->password = g_strdup (DEFAULT_PASSWORD);
shout2send->streamname = g_strdup (DEFAULT_STREAMNAME);
shout2send->description = g_strdup (DEFAULT_DESCRIPTION);
shout2send->genre = g_strdup (DEFAULT_GENRE);
shout2send->mount = g_strdup (DEFAULT_MOUNT);
shout2send->url = g_strdup (DEFAULT_URL);
shout2send->protocol = DEFAULT_PROTOCOL;
shout2send->tags = gst_tag_list_new ();
shout2send->conn = NULL;
shout2send->audio_format = SHOUT_FORMAT_VORBIS;
shout2send->connected = FALSE;
shout2send->songmetadata = NULL;
}
static void
set_shout_metadata (const GstTagList * list, const gchar * tag,
gpointer user_data)
{
char **shout_metadata = (char **) user_data;
gchar *value, *temp;
GST_DEBUG ("tag: %s being added", tag);
if (strcmp (tag, GST_TAG_ARTIST) == 0) {
if (gst_tag_get_type (tag) == G_TYPE_STRING) {
if (!gst_tag_list_get_string (list, tag, &value)) {
GST_DEBUG ("Error reading \"%s\" tag value", tag);
return;
}
/* shout_metadata should be NULL if title is after artist in list */
if (*shout_metadata == NULL) {
*shout_metadata = g_strdup (value);
} else {
temp = g_strdup_printf ("%s - %s", value, *shout_metadata);
g_free (*shout_metadata);
*shout_metadata = temp;
}
}
} else if (strcmp (tag, GST_TAG_TITLE) == 0) {
if (gst_tag_get_type (tag) == G_TYPE_STRING) {
if (!gst_tag_list_get_string (list, tag, &value)) {
GST_DEBUG ("Error reading \"%s\" tag value", tag);
return;
}
/* shout_metadata should be NULL if title is before artist in list */
if (*shout_metadata == NULL) {
*shout_metadata = g_strdup (value);
} else {
temp = g_strdup_printf ("%s - %s", *shout_metadata, value);
g_free (*shout_metadata);
*shout_metadata = temp;
}
}
}
GST_LOG ("shout metadata is now: %s", *shout_metadata);
}
#if 0
static void
gst_shout2send_set_metadata (GstShout2send * shout2send)
{
const GstTagList *user_tags;
GstTagList *copy;
char *tempmetadata;
shout_metadata_t *pmetadata;
g_return_if_fail (shout2send != NULL);
user_tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (shout2send));
if ((shout2send->tags == NULL) && (user_tags == NULL)) {
return;
}
copy = gst_tag_list_merge (user_tags, shout2send->tags,
gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (shout2send)));
/* lets get the artist and song tags */
tempmetadata = NULL;
gst_tag_list_foreach ((GstTagList *) copy, set_shout_metadata,
(gpointer) & tempmetadata);
if (tempmetadata) {
pmetadata = shout_metadata_new ();
shout_metadata_add (pmetadata, "song", tempmetadata);
shout_set_metadata (shout2send->conn, pmetadata);
shout_metadata_free (pmetadata);
}
gst_tag_list_free (copy);
}
#endif
static gboolean
gst_shout2send_event (GstBaseSink * sink, GstEvent * event)
{
GstShout2send *shout2send;
gboolean ret = TRUE;
shout2send = GST_SHOUT2SEND (sink);
GST_LOG_OBJECT (shout2send, "got %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_TAG:{
/* vorbis audio doesnt need metadata setting on the icecast level, only mp3 */
if (shout2send->tags && shout2send->audio_format == SHOUT_FORMAT_MP3) {
GstTagList *list;
gst_event_parse_tag (event, &list);
GST_DEBUG_OBJECT (shout2send, "tags=%" GST_PTR_FORMAT, list);
gst_tag_list_insert (shout2send->tags,
list,
gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (shout2send)));
/* lets get the artist and song tags */
gst_tag_list_foreach ((GstTagList *) shout2send->tags,
set_shout_metadata, &shout2send->songmetadata);
if (shout2send->songmetadata && shout2send->connected) {
shout_metadata_t *pmetadata;
GST_DEBUG_OBJECT (shout2send, "metadata now: %s",
shout2send->songmetadata);
pmetadata = shout_metadata_new ();
shout_metadata_add (pmetadata, "song", shout2send->songmetadata);
shout_set_metadata (shout2send->conn, pmetadata);
shout_metadata_free (pmetadata);
}
}
break;
}
default:{
GST_LOG_OBJECT (shout2send, "let base class handle event");
if (GST_BASE_SINK_CLASS (parent_class)->event) {
event = gst_event_ref (event);
ret = GST_BASE_SINK_CLASS (parent_class)->event (sink, event);
}
break;
}
}
return ret;
}
static gboolean
gst_shout2send_start (GstBaseSink * basesink)
{
GstShout2send *sink = GST_SHOUT2SEND (basesink);
const gchar *cur_prop;
gshort proto = 3;
gchar *version_string;
GST_DEBUG_OBJECT (sink, "starting");
sink->conn = shout_new ();
switch (sink->protocol) {
case SHOUT2SEND_PROTOCOL_XAUDIOCAST:
proto = SHOUT_PROTOCOL_XAUDIOCAST;
break;
case SHOUT2SEND_PROTOCOL_ICY:
proto = SHOUT_PROTOCOL_ICY;
break;
case SHOUT2SEND_PROTOCOL_HTTP:
proto = SHOUT_PROTOCOL_HTTP;
break;
}
cur_prop = "protocol";
GST_DEBUG_OBJECT (sink, "setting protocol: %d", sink->protocol);
if (shout_set_protocol (sink->conn, proto) != SHOUTERR_SUCCESS)
goto set_failed;
/* --- FIXME: shout requires an ip, and fails if it is given a host. */
/* may want to put convert_to_ip(shout2send->ip) here */
cur_prop = "ip";
GST_DEBUG_OBJECT (sink, "setting ip: %s", sink->ip);
if (shout_set_host (sink->conn, sink->ip) != SHOUTERR_SUCCESS)
goto set_failed;
cur_prop = "port";
GST_DEBUG_OBJECT (sink, "setting port: %u", sink->port);
if (shout_set_port (sink->conn, sink->port) != SHOUTERR_SUCCESS)
goto set_failed;
cur_prop = "password";
GST_DEBUG_OBJECT (sink, "setting password: %s", sink->password);
if (shout_set_password (sink->conn, sink->password) != SHOUTERR_SUCCESS)
goto set_failed;
cur_prop = "streamname";
GST_DEBUG_OBJECT (sink, "setting %s: %s", cur_prop, sink->streamname);
if (shout_set_name (sink->conn, sink->streamname) != SHOUTERR_SUCCESS)
goto set_failed;
cur_prop = "description";
GST_DEBUG_OBJECT (sink, "setting %s: %s", cur_prop, sink->description);
if (shout_set_description (sink->conn, sink->description) != SHOUTERR_SUCCESS)
goto set_failed;
cur_prop = "genre";
GST_DEBUG_OBJECT (sink, "setting %s: %s", cur_prop, sink->genre);
if (shout_set_genre (sink->conn, sink->genre) != SHOUTERR_SUCCESS)
goto set_failed;
cur_prop = "mount";
GST_DEBUG_OBJECT (sink, "setting %s: %s", cur_prop, sink->mount);
if (shout_set_mount (sink->conn, sink->mount) != SHOUTERR_SUCCESS)
goto set_failed;
cur_prop = "user";
GST_DEBUG_OBJECT (sink, "setting %s: %s", cur_prop, "source");
if (shout_set_user (sink->conn, "source") != SHOUTERR_SUCCESS)
goto set_failed;
version_string = gst_version_string ();
cur_prop = "agent";
GST_DEBUG_OBJECT (sink, "setting %s: %s", cur_prop, version_string);
if (shout_set_agent (sink->conn, version_string) != SHOUTERR_SUCCESS) {
g_free (version_string);
goto set_failed;
}
g_free (version_string);
return TRUE;
/* ERROR */
set_failed:
{
GST_ELEMENT_ERROR (sink, LIBRARY, SETTINGS, (NULL),
("Error setting %s: %s", cur_prop, shout_get_error (sink->conn)));
return FALSE;
}
}
static gboolean
gst_shout2send_connect (GstShout2send * sink)
{
GST_DEBUG_OBJECT (sink, "Connection format is: %s",
(sink->audio_format == SHOUT_FORMAT_VORBIS) ? "vorbis" :
((sink->audio_format == SHOUT_FORMAT_MP3) ? "mp3" : "unknown"));
if (shout_set_format (sink->conn, sink->audio_format) != SHOUTERR_SUCCESS)
goto could_not_set_format;
if (shout_open (sink->conn) != SHOUTERR_SUCCESS)
goto could_not_connect;
GST_DEBUG_OBJECT (sink, "connected to server");
sink->connected = TRUE;
/* let's set metadata */
if (sink->songmetadata) {
shout_metadata_t *pmetadata;
GST_DEBUG_OBJECT (sink, "shout metadata now: %s", sink->songmetadata);
pmetadata = shout_metadata_new ();
shout_metadata_add (pmetadata, "song", sink->songmetadata);
shout_set_metadata (sink->conn, pmetadata);
shout_metadata_free (pmetadata);
}
return TRUE;
/* ERRORS */
could_not_set_format:
{
GST_ELEMENT_ERROR (sink, LIBRARY, SETTINGS, (NULL),
("Error setting connection format: %s", shout_get_error (sink->conn)));
return FALSE;
}
could_not_connect:
{
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
(_("Could not connect to server")),
("shout_open() failed: err=%s", shout_get_error (sink->conn)));
g_signal_emit (sink, gst_shout2send_signals[SIGNAL_CONNECTION_PROBLEM], 0,
shout_get_errno (sink->conn));
return FALSE;
}
}
static gboolean
gst_shout2send_stop (GstBaseSink * basesink)
{
GstShout2send *sink = GST_SHOUT2SEND (basesink);
if (sink->conn) {
if (sink->connected)
shout_close (sink->conn);
shout_free (sink->conn);
sink->conn = NULL;
}
if (sink->songmetadata) {
g_free (sink->songmetadata);
sink->songmetadata = NULL;
}
sink->connected = FALSE;
return TRUE;
}
static GstFlowReturn
gst_shout2send_render (GstBaseSink * basesink, GstBuffer * buf)
{
GstShout2send *sink;
glong ret;
sink = GST_SHOUT2SEND (basesink);
/* presumably we connect here because we need to know the format before
* we can set up the connection, which we don't know yet in _start() */
if (!sink->connected) {
if (!gst_shout2send_connect (sink))
return GST_FLOW_ERROR;
}
/* FIXME: do we want to do syncing here at all? (tpm) */
/* GST_LOG_OBJECT (sink, "using libshout to sync"); */
shout_sync (sink->conn);
GST_LOG_OBJECT (sink, "sending %u bytes of data", GST_BUFFER_SIZE (buf));
ret = shout_send (sink->conn, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
if (ret != SHOUTERR_SUCCESS)
goto send_error;
return GST_FLOW_OK;
/* ERRORS */
send_error:
{
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
("shout_send() failed: %s", shout_get_error (sink->conn)));
g_signal_emit (sink, gst_shout2send_signals[SIGNAL_CONNECTION_PROBLEM], 0,
shout_get_errno (sink->conn));
return GST_FLOW_ERROR;
}
}
static void
gst_shout2send_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstShout2send *shout2send;
shout2send = GST_SHOUT2SEND (object);
switch (prop_id) {
case ARG_IP:
if (shout2send->ip)
g_free (shout2send->ip);
shout2send->ip = g_strdup (g_value_get_string (value));
break;
case ARG_PORT:
shout2send->port = g_value_get_int (value);
break;
case ARG_PASSWORD:
if (shout2send->password)
g_free (shout2send->password);
shout2send->password = g_strdup (g_value_get_string (value));
break;
case ARG_STREAMNAME: /* Name of the stream */
if (shout2send->streamname)
g_free (shout2send->streamname);
shout2send->streamname = g_strdup (g_value_get_string (value));
break;
case ARG_DESCRIPTION: /* Description of the stream */
if (shout2send->description)
g_free (shout2send->description);
shout2send->description = g_strdup (g_value_get_string (value));
break;
case ARG_GENRE: /* Genre of the stream */
if (shout2send->genre)
g_free (shout2send->genre);
shout2send->genre = g_strdup (g_value_get_string (value));
break;
case ARG_PROTOCOL: /* protocol to connect with */
shout2send->protocol = g_value_get_enum (value);
break;
case ARG_MOUNT: /* mountpoint of stream (icecast only) */
if (shout2send->mount)
g_free (shout2send->mount);
shout2send->mount = g_strdup (g_value_get_string (value));
break;
case ARG_URL: /* Url of the stream (I'm guessing) */
if (shout2send->url)
g_free (shout2send->url);
shout2send->url = g_strdup (g_value_get_string (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_shout2send_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstShout2send *shout2send;
shout2send = GST_SHOUT2SEND (object);
switch (prop_id) {
case ARG_IP:
g_value_set_string (value, shout2send->ip);
break;
case ARG_PORT:
g_value_set_int (value, shout2send->port);
break;
case ARG_PASSWORD:
g_value_set_string (value, shout2send->password);
break;
case ARG_STREAMNAME: /* Name of the stream */
g_value_set_string (value, shout2send->streamname);
break;
case ARG_DESCRIPTION: /* Description of the stream */
g_value_set_string (value, shout2send->description);
break;
case ARG_GENRE: /* Genre of the stream */
g_value_set_string (value, shout2send->genre);
break;
case ARG_PROTOCOL: /* protocol to connect with */
g_value_set_enum (value, shout2send->protocol);
break;
case ARG_MOUNT: /* mountpoint of stream (icecast only) */
g_value_set_string (value, shout2send->mount);
break;
case ARG_URL: /* Url of stream (I'm guessing) */
g_value_set_string (value, shout2send->url);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_shout2send_setcaps (GstPad * pad, GstCaps * caps)
{
const gchar *mimetype;
GstShout2send *shout2send;
gboolean ret = TRUE;
shout2send = GST_SHOUT2SEND (GST_OBJECT_PARENT (pad));
mimetype = gst_structure_get_name (gst_caps_get_structure (caps, 0));
GST_DEBUG_OBJECT (shout2send, "mimetype of caps given is: %s", mimetype);
if (!strcmp (mimetype, "audio/mpeg")) {
shout2send->audio_format = SHOUT_FORMAT_MP3;
} else if (!strcmp (mimetype, "application/ogg")) {
shout2send->audio_format = SHOUT_FORMAT_VORBIS;
} else {
ret = FALSE;
}
return ret;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
#ifdef ENABLE_NLS
setlocale (LC_ALL, "");
bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
#endif /* ENABLE_NLS */
return gst_element_register (plugin, "shout2send", GST_RANK_NONE,
GST_TYPE_SHOUT2SEND);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"shout2send",
"Sends data to an icecast server using libshout2",
plugin_init,
VERSION, "LGPL", "libshout2", "http://www.icecast.org/download.html")