/* GStreamer * Copyright (C) <2005> Wim Taymans * * 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 #include #include "gstrtpmpapay.h" /* elementfactory information */ static const GstElementDetails gst_rtp_mpapay_details = GST_ELEMENT_DETAILS ("RTP packet payloader", "Codec/Payloader/Network", "Payload MPEG audio as RTP packets (RFC 2038)", "Wim Taymans "); static GstStaticPadTemplate gst_rtp_mpa_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/mpeg") ); static GstStaticPadTemplate gst_rtp_mpa_pay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_MPA_STRING ", " "clock-rate = (int) 90000; " "application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) 90000, " "encoding-name = (string) \"MPA\"") ); static void gst_rtp_mpa_pay_class_init (GstRtpMPAPayClass * klass); static void gst_rtp_mpa_pay_base_init (GstRtpMPAPayClass * klass); static void gst_rtp_mpa_pay_init (GstRtpMPAPay * rtpmpapay); static void gst_rtp_mpa_pay_finalize (GObject * object); static gboolean gst_rtp_mpa_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps); static GstFlowReturn gst_rtp_mpa_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer); static GstBaseRTPPayloadClass *parent_class = NULL; static GType gst_rtp_mpa_pay_get_type (void) { static GType rtpmpapay_type = 0; if (!rtpmpapay_type) { static const GTypeInfo rtpmpapay_info = { sizeof (GstRtpMPAPayClass), (GBaseInitFunc) gst_rtp_mpa_pay_base_init, NULL, (GClassInitFunc) gst_rtp_mpa_pay_class_init, NULL, NULL, sizeof (GstRtpMPAPay), 0, (GInstanceInitFunc) gst_rtp_mpa_pay_init, }; rtpmpapay_type = g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpMPAPay", &rtpmpapay_info, 0); } return rtpmpapay_type; } static void gst_rtp_mpa_pay_base_init (GstRtpMPAPayClass * klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_mpa_pay_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_mpa_pay_sink_template)); gst_element_class_set_details (element_class, &gst_rtp_mpapay_details); } static void gst_rtp_mpa_pay_class_init (GstRtpMPAPayClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseRTPPayloadClass *gstbasertppayload_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; parent_class = g_type_class_peek_parent (klass); gobject_class->finalize = gst_rtp_mpa_pay_finalize; gstbasertppayload_class->set_caps = gst_rtp_mpa_pay_setcaps; gstbasertppayload_class->handle_buffer = gst_rtp_mpa_pay_handle_buffer; } static void gst_rtp_mpa_pay_init (GstRtpMPAPay * rtpmpapay) { rtpmpapay->adapter = gst_adapter_new (); } static void gst_rtp_mpa_pay_finalize (GObject * object) { GstRtpMPAPay *rtpmpapay; rtpmpapay = GST_RTP_MPA_PAY (object); g_object_unref (rtpmpapay->adapter); rtpmpapay->adapter = NULL; G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean gst_rtp_mpa_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) { gboolean res; gst_basertppayload_set_options (payload, "audio", TRUE, "MPA", 90000); res = gst_basertppayload_set_outcaps (payload, NULL); return res; } static GstFlowReturn gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay) { guint avail; GstBuffer *outbuf; GstFlowReturn ret; guint16 frag_offset; /* the data available in the adapter is either smaller * than the MTU or bigger. In the case it is smaller, the complete * adapter contents can be put in one packet. In the case the * adapter has more than one MTU, we need to split the MPA data * over multiple packets. The frag_offset in each packet header * needs to be updated with the position in the MPA frame. */ avail = gst_adapter_available (rtpmpapay->adapter); ret = GST_FLOW_OK; frag_offset = 0; while (avail > 0) { guint towrite; guint8 *payload; guint payload_len; guint packet_len; /* this will be the total lenght of the packet */ packet_len = gst_rtp_buffer_calc_packet_len (4 + avail, 0, 0); /* fill one MTU or all available bytes */ towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpmpapay)); /* this is the payload length */ payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0); /* create buffer to hold the payload */ outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); payload_len -= 4; gst_rtp_buffer_set_payload_type (outbuf, GST_RTP_PAYLOAD_MPA); /* * 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 | * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ */ payload = gst_rtp_buffer_get_payload (outbuf); payload[0] = 0; payload[1] = 0; payload[2] = frag_offset >> 8; payload[3] = frag_offset & 0xff; gst_adapter_copy (rtpmpapay->adapter, &payload[4], 0, payload_len); gst_adapter_flush (rtpmpapay->adapter, payload_len); avail -= payload_len; frag_offset += payload_len; if (avail == 0) gst_rtp_buffer_set_marker (outbuf, TRUE); GST_BUFFER_TIMESTAMP (outbuf) = rtpmpapay->first_ts; GST_BUFFER_DURATION (outbuf) = rtpmpapay->duration; ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmpapay), outbuf); } return ret; } static GstFlowReturn gst_rtp_mpa_pay_handle_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstRtpMPAPay *rtpmpapay; GstFlowReturn ret; guint size, avail; guint packet_len; GstClockTime duration; rtpmpapay = GST_RTP_MPA_PAY (basepayload); size = GST_BUFFER_SIZE (buffer); duration = GST_BUFFER_DURATION (buffer); avail = gst_adapter_available (rtpmpapay->adapter); if (avail == 0) { rtpmpapay->first_ts = GST_BUFFER_TIMESTAMP (buffer); rtpmpapay->duration = 0; } /* get packet length of previous data and this new data, * payload length includes a 4 byte header */ packet_len = gst_rtp_buffer_calc_packet_len (4 + avail + size, 0, 0); /* if this buffer is going to overflow the packet, flush what we * have. */ if (gst_basertppayload_is_filled (basepayload, packet_len, rtpmpapay->duration + duration)) { ret = gst_rtp_mpa_pay_flush (rtpmpapay); rtpmpapay->first_ts = GST_BUFFER_TIMESTAMP (buffer); rtpmpapay->duration = 0; } else { ret = GST_FLOW_OK; } gst_adapter_push (rtpmpapay->adapter, buffer); rtpmpapay->duration += duration; return ret; } gboolean gst_rtp_mpa_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmpapay", GST_RANK_NONE, GST_TYPE_RTP_MPA_PAY); }