/* GStreamer * Copyright (C) <2005> Wim Taymans <wim.taymans@gmail.com> * * 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 <gst/rtp/gstrtpbuffer.h> #include <string.h> #include "gstrtpmpadepay.h" GST_DEBUG_CATEGORY_STATIC (rtpmpadepay_debug); #define GST_CAT_DEFAULT (rtpmpadepay_debug) /* elementfactory information */ static const GstElementDetails gst_rtp_mpadepay_details = GST_ELEMENT_DETAILS ("RTP MPEG audio depayloader", "Codec/Depayloader/Network", "Extracts MPEG audio from RTP packets (RFC 2038)", "Wim Taymans <wim.taymans@gmail.com>"); static GstStaticPadTemplate gst_rtp_mpa_depay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 1") ); static GstStaticPadTemplate gst_rtp_mpa_depay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) 90000, " "encoding-name = (string) \"MPA\";" "application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_MPA_STRING ", " "clock-rate = (int) 90000") ); GST_BOILERPLATE (GstRtpMPADepay, gst_rtp_mpa_depay, GstBaseRTPDepayload, GST_TYPE_BASE_RTP_DEPAYLOAD); static gboolean gst_rtp_mpa_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps); static GstBuffer *gst_rtp_mpa_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf); static void gst_rtp_mpa_depay_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_mpa_depay_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_mpa_depay_sink_template)); gst_element_class_set_details (element_class, &gst_rtp_mpadepay_details); } static void gst_rtp_mpa_depay_class_init (GstRtpMPADepayClass * klass) { GstBaseRTPDepayloadClass *gstbasertpdepayload_class; gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass; parent_class = g_type_class_peek_parent (klass); gstbasertpdepayload_class->set_caps = gst_rtp_mpa_depay_setcaps; gstbasertpdepayload_class->process = gst_rtp_mpa_depay_process; GST_DEBUG_CATEGORY_INIT (rtpmpadepay_debug, "rtpmpadepay", 0, "MPEG Audio RTP Depayloader"); } static void gst_rtp_mpa_depay_init (GstRtpMPADepay * rtpmpadepay, GstRtpMPADepayClass * klass) { /* needed because of GST_BOILERPLATE */ } static gboolean gst_rtp_mpa_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) { GstStructure *structure; GstCaps *outcaps; gint clock_rate; gboolean res; structure = gst_caps_get_structure (caps, 0); if (!gst_structure_get_int (structure, "clock-rate", &clock_rate)) clock_rate = 90000; depayload->clock_rate = clock_rate; outcaps = gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 1, NULL); res = gst_pad_set_caps (depayload->srcpad, outcaps); gst_caps_unref (outcaps); return res; } static GstBuffer * gst_rtp_mpa_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) { GstRtpMPADepay *rtpmpadepay; GstBuffer *outbuf; rtpmpadepay = GST_RTP_MPA_DEPAY (depayload); { gint payload_len; guint8 *payload; guint16 frag_offset; gboolean marker; payload_len = gst_rtp_buffer_get_payload_len (buf); if (payload_len <= 4) goto empty_packet; payload = gst_rtp_buffer_get_payload (buf); /* strip off header * * 0 1 2 3 * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * | MBZ | Frag_offset | * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ */ frag_offset = (payload[2] << 8) | payload[3]; /* subbuffer skipping the 4 header bytes */ outbuf = gst_rtp_buffer_get_payload_subbuffer (buf, 4, -1); marker = gst_rtp_buffer_get_marker (buf); if (marker) { /* mark start of talkspurt with discont */ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); } GST_DEBUG_OBJECT (rtpmpadepay, "gst_rtp_mpa_depay_chain: pushing buffer of size %d", GST_BUFFER_SIZE (outbuf)); /* FIXME, we can push half mpeg frames when they are split over multiple * RTP packets */ return outbuf; } return NULL; /* ERRORS */ empty_packet: { GST_ELEMENT_WARNING (rtpmpadepay, STREAM, DECODE, ("Empty Payload."), (NULL)); return NULL; } } gboolean gst_rtp_mpa_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmpadepay", GST_RANK_MARGINAL, GST_TYPE_RTP_MPA_DEPAY); }