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authorZeeshan Ali <zeenix@gmail.com>2003-07-13 00:20:44 +0000
committerZeeshan Ali <zeenix@gmail.com>2003-07-13 00:20:44 +0000
commit333468b5d448062b40c428e5c3b95dcb8159807c (patch)
treeddaa5d09eae9309ef88260439ef4379d26ed9af8 /gst/rtp/gstrtpgsmpay.c
parent8c4aec99981a02e6fe6fdf61d4156746131a442d (diff)
GSM -> RTP and viceversa
Original commit message from CVS: GSM -> RTP and viceversa
Diffstat (limited to 'gst/rtp/gstrtpgsmpay.c')
-rw-r--r--gst/rtp/gstrtpgsmpay.c310
1 files changed, 310 insertions, 0 deletions
diff --git a/gst/rtp/gstrtpgsmpay.c b/gst/rtp/gstrtpgsmpay.c
new file mode 100644
index 00000000..2fff85c6
--- /dev/null
+++ b/gst/rtp/gstrtpgsmpay.c
@@ -0,0 +1,310 @@
+/* GStreamer
+ * 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 <string.h>
+#include "gstrtpgsmenc.h"
+
+/* elementfactory information */
+static GstElementDetails gst_rtpgsmenc_details = {
+ "RTP GSM Audio Encoder",
+ "RtpGSMEnc",
+ "LGPL",
+ "Encodes GSM audio into an RTP packet",
+ VERSION,
+ "Zeeshan Ali <zak147@yahoo.com>",
+ "(C) 2003",
+};
+
+/* RtpGSMEnc signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ /* FILL ME */
+ ARG_0,
+};
+
+GST_PAD_TEMPLATE_FACTORY (sink_factory,
+ "sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_CAPS_NEW (
+ "gsm_gsm",
+ "audio/x-gsm",
+ "rate", GST_PROPS_INT_RANGE (1000, 48000)
+ )
+);
+
+GST_PAD_TEMPLATE_FACTORY (src_factory,
+ "src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_CAPS_NEW (
+ "rtp",
+ "application/x-rtp",
+ NULL)
+);
+
+static void gst_rtpgsmenc_class_init (GstRtpGSMEncClass * klass);
+static void gst_rtpgsmenc_init (GstRtpGSMEnc * rtpgsmenc);
+static void gst_rtpgsmenc_chain (GstPad * pad, GstBuffer * buf);
+static void gst_rtpgsmenc_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtpgsmenc_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static GstPadLinkReturn gst_rtpgsmenc_sinkconnect (GstPad * pad, GstCaps * caps);
+static GstElementStateReturn gst_rtpgsmenc_change_state (GstElement * element);
+
+static GstElementClass *parent_class = NULL;
+
+static GType gst_rtpgsmenc_get_type (void)
+{
+ static GType rtpgsmenc_type = 0;
+
+ if (!rtpgsmenc_type) {
+ static const GTypeInfo rtpgsmenc_info = {
+ sizeof (GstRtpGSMEncClass),
+ NULL,
+ NULL,
+ (GClassInitFunc) gst_rtpgsmenc_class_init,
+ NULL,
+ NULL,
+ sizeof (GstRtpGSMEnc),
+ 0,
+ (GInstanceInitFunc) gst_rtpgsmenc_init,
+ };
+
+ rtpgsmenc_type = g_type_register_static (GST_TYPE_ELEMENT, "GstRtpGSMEnc", &rtpgsmenc_info, 0);
+ }
+ return rtpgsmenc_type;
+}
+
+static void
+gst_rtpgsmenc_class_init (GstRtpGSMEncClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+
+ parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+
+ gobject_class->set_property = gst_rtpgsmenc_set_property;
+ gobject_class->get_property = gst_rtpgsmenc_get_property;
+
+ gstelement_class->change_state = gst_rtpgsmenc_change_state;
+}
+
+static void
+gst_rtpgsmenc_init (GstRtpGSMEnc * rtpgsmenc)
+{
+ rtpgsmenc->sinkpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (sink_factory), "sink");
+ rtpgsmenc->srcpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (src_factory), "src");
+ gst_element_add_pad (GST_ELEMENT (rtpgsmenc), rtpgsmenc->sinkpad);
+ gst_element_add_pad (GST_ELEMENT (rtpgsmenc), rtpgsmenc->srcpad);
+ gst_pad_set_chain_function (rtpgsmenc->sinkpad, gst_rtpgsmenc_chain);
+ gst_pad_set_link_function (rtpgsmenc->sinkpad, gst_rtpgsmenc_sinkconnect);
+
+ rtpgsmenc->frequency = 8000;
+
+ rtpgsmenc->next_time = 0;
+ rtpgsmenc->time_interval = 0;
+
+ rtpgsmenc->seq = 0;
+ rtpgsmenc->ssrc = random ();
+}
+
+static GstPadLinkReturn
+gst_rtpgsmenc_sinkconnect (GstPad * pad, GstCaps * caps)
+{
+ GstRtpGSMEnc *rtpgsmenc;
+
+ rtpgsmenc = GST_RTP_GSM_ENC (gst_pad_get_parent (pad));
+
+ gst_caps_get_int (caps, "rate", &rtpgsmenc->frequency);
+
+ /* Pre-calculate what we can */
+ rtpgsmenc->time_interval = GST_SECOND / (2 * rtpgsmenc->frequency);
+
+ return GST_PAD_LINK_OK;
+}
+
+
+void
+gst_rtpgsmenc_htons (GstBuffer *buf)
+{
+ gint16 *i, *len;
+
+ /* FIXME: is this code correct or even sane at all? */
+ i = (gint16 *) GST_BUFFER_DATA(buf);
+ len = i + GST_BUFFER_SIZE (buf) / sizeof (gint16 *);
+
+ for (; i<len; i++) {
+ *i = g_htons (*i);
+ }
+}
+
+static void
+gst_rtpgsmenc_chain (GstPad * pad, GstBuffer * buf)
+{
+ GstRtpGSMEnc *rtpgsmenc;
+ GstBuffer *outbuf;
+ Rtp_Packet packet;
+
+ g_return_if_fail (pad != NULL);
+ g_return_if_fail (GST_IS_PAD (pad));
+ g_return_if_fail (buf != NULL);
+
+ rtpgsmenc = GST_RTP_GSM_ENC (GST_OBJECT_PARENT (pad));
+
+ g_return_if_fail (rtpgsmenc != NULL);
+ g_return_if_fail (GST_IS_RTP_GSM_ENC (rtpgsmenc));
+
+ if (GST_IS_EVENT (buf)) {
+ GstEvent *event = GST_EVENT (buf);
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_DISCONTINUOUS:
+ GST_DEBUG (GST_CAT_EVENT, "discont");
+ rtpgsmenc->next_time = 0;
+ gst_pad_event_default (pad, event);
+ return;
+ default:
+ gst_pad_event_default (pad, event);
+ return;
+ }
+ }
+
+ /* We only need the header */
+ packet = rtp_packet_new_allocate (0, 0, 0);
+
+ rtp_packet_set_csrc_count (packet, 0);
+ rtp_packet_set_extension (packet, 0);
+ rtp_packet_set_padding (packet, 0);
+ rtp_packet_set_version (packet, RTP_VERSION);
+ rtp_packet_set_marker (packet, 0);
+ rtp_packet_set_ssrc (packet, g_htonl (rtpgsmenc->ssrc));
+ rtp_packet_set_seq (packet, g_htons (rtpgsmenc->seq));
+ rtp_packet_set_timestamp (packet, g_htonl ((guint32) rtpgsmenc->next_time / GST_SECOND));
+ rtp_packet_set_payload_type (packet, (guint8) PAYLOAD_GSM);
+
+ /* FIXME: According to RFC 1890, this is required, right? */
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+ gst_rtpgsmenc_htons (buf);
+#endif
+
+ outbuf = gst_buffer_new ();
+ GST_BUFFER_SIZE (outbuf) = rtp_packet_get_packet_len (packet) + GST_BUFFER_SIZE (buf);
+ GST_BUFFER_DATA (outbuf) = g_malloc (GST_BUFFER_SIZE (outbuf));
+ GST_BUFFER_TIMESTAMP (outbuf) = rtpgsmenc->next_time;
+
+ memcpy (GST_BUFFER_DATA (outbuf), packet->data, rtp_packet_get_packet_len (packet));
+ memcpy (GST_BUFFER_DATA (outbuf) + rtp_packet_get_packet_len(packet), GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
+
+ GST_DEBUG (0,"gst_rtpgsmenc_chain: pushing buffer of size %d", GST_BUFFER_SIZE(outbuf));
+ gst_pad_push (rtpgsmenc->srcpad, outbuf);
+
+ ++rtpgsmenc->seq;
+ rtpgsmenc->next_time += rtpgsmenc->time_interval * GST_BUFFER_SIZE (buf);
+
+ rtp_packet_free (packet);
+ gst_buffer_unref (buf);
+}
+
+static void
+gst_rtpgsmenc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec)
+{
+ GstRtpGSMEnc *rtpgsmenc;
+
+ /* it's not null if we got it, but it might not be ours */
+ g_return_if_fail (GST_IS_RTP_GSM_ENC (object));
+ rtpgsmenc = GST_RTP_GSM_ENC (object);
+
+ switch (prop_id) {
+ default:
+ break;
+ }
+}
+
+static void
+gst_rtpgsmenc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec)
+{
+ GstRtpGSMEnc *rtpgsmenc;
+
+ /* it's not null if we got it, but it might not be ours */
+ g_return_if_fail (GST_IS_RTP_GSM_ENC (object));
+ rtpgsmenc = GST_RTP_GSM_ENC (object);
+
+ switch (prop_id) {
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static GstElementStateReturn
+gst_rtpgsmenc_change_state (GstElement * element)
+{
+ GstRtpGSMEnc *rtpgsmenc;
+
+ g_return_val_if_fail (GST_IS_RTP_GSM_ENC (element), GST_STATE_FAILURE);
+
+ rtpgsmenc = GST_RTP_GSM_ENC (element);
+
+ GST_DEBUG (0, "state pending %d\n", GST_STATE_PENDING (element));
+
+ /* if going down into NULL state, close the file if it's open */
+ switch (GST_STATE_TRANSITION (element)) {
+ case GST_STATE_NULL_TO_READY:
+ break;
+
+ case GST_STATE_READY_TO_NULL:
+ break;
+
+ default:
+ break;
+ }
+
+ /* if we haven't failed already, give the parent class a chance to ;-) */
+ if (GST_ELEMENT_CLASS (parent_class)->change_state)
+ return GST_ELEMENT_CLASS (parent_class)->change_state (element);
+
+ return GST_STATE_SUCCESS;
+}
+
+gboolean
+gst_rtpgsmenc_plugin_init (GModule * module, GstPlugin * plugin)
+{
+ GstElementFactory *rtpgsmenc;
+
+ rtpgsmenc = gst_element_factory_new ("rtpgsmenc", GST_TYPE_RTP_GSM_ENC, &gst_rtpgsmenc_details);
+ g_return_val_if_fail (rtpgsmenc != NULL, FALSE);
+
+ gst_element_factory_add_pad_template (rtpgsmenc, GST_PAD_TEMPLATE_GET (sink_factory));
+ gst_element_factory_add_pad_template (rtpgsmenc, GST_PAD_TEMPLATE_GET (src_factory));
+
+ gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (rtpgsmenc));
+
+ return TRUE;
+}