/* GStreamer * * 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 "gstrtph264pay.h" GST_DEBUG_CATEGORY_STATIC (rtph264pay_debug); #define GST_CAT_DEFAULT (rtph264pay_debug) /* references: * * RFC 3984 */ /* elementfactory information */ static const GstElementDetails gst_rtp_h264pay_details = GST_ELEMENT_DETAILS ("RTP packet payloader", "Codec/Payloader/Network", "Payload-encode H264 video into RTP packets (RFC 3984)", "Laurent Glayal "); static GstStaticPadTemplate gst_rtp_h264_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("video/x-h264") ); static GstStaticPadTemplate gst_rtp_h264_pay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"video\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) 90000, " "encoding-name = (string) \"H264\"") ); static void gst_rtp_h264_pay_finalize (GObject * object); static void gst_rtp_h264_pay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_h264_pay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstStateChangeReturn gst_rtp_h264_pay_change_state (GstElement * element, GstStateChange transition); static gboolean gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps); static GstFlowReturn gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * pad, GstBuffer * buffer); GST_BOILERPLATE (GstRtpH264Pay, gst_rtp_h264_pay, GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD); static void gst_rtp_h264_pay_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_h264_pay_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_h264_pay_sink_template)); gst_element_class_set_details (element_class, &gst_rtp_h264pay_details); } static void gst_rtp_h264_pay_class_init (GstRtpH264PayClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseRTPPayloadClass *gstbasertppayload_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; gobject_class->finalize = gst_rtp_h264_pay_finalize; gobject_class->set_property = gst_rtp_h264_pay_set_property; gobject_class->get_property = gst_rtp_h264_pay_get_property; gstelement_class->change_state = gst_rtp_h264_pay_change_state; gstbasertppayload_class->set_caps = gst_rtp_h264_pay_setcaps; gstbasertppayload_class->handle_buffer = gst_rtp_h264_pay_handle_buffer; GST_DEBUG_CATEGORY_INIT (rtph264pay_debug, "rtph264pay", 0, "H264 RTP Payloader"); } static void gst_rtp_h264_pay_init (GstRtpH264Pay * rtph264pay, GstRtpH264PayClass * klass) { } static void gst_rtp_h264_pay_finalize (GObject * object) { GstRtpH264Pay *rtph264pay; rtph264pay = GST_RTP_H264_PAY (object); G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps) { GstRtpH264Pay *rtph264pay; rtph264pay = GST_RTP_H264_PAY (basepayload); gst_basertppayload_set_options (basepayload, "video", TRUE, "H264", 90000); gst_basertppayload_set_outcaps (basepayload, NULL); return TRUE; } static GstFlowReturn gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstRtpH264Pay *rtph264pay; GstFlowReturn ret; guint size, idxdata; GstBuffer *outbuf; guint8 *payload, *data, *pdata; guint8 nalType; GstClockTime timestamp; guint packet_len, payload_len, mtu; rtph264pay = GST_RTP_H264_PAY (basepayload); mtu = GST_BASE_RTP_PAYLOAD_MTU (rtph264pay); size = GST_BUFFER_SIZE (buffer); data = GST_BUFFER_DATA (buffer); timestamp = GST_BUFFER_TIMESTAMP (buffer); GST_DEBUG_OBJECT (basepayload, "got %d bytes", size); /* H264 stream analysis */ pdata = data; idxdata = size; while (idxdata > 5 && (pdata[0] != 0x00 || pdata[1] != 0x00 || pdata[2] != 0x1 || (pdata[3] & 0x1f) < 1 || (pdata[3] & 0x1f) > 23) ) { pdata++; idxdata--; GST_DEBUG_OBJECT (basepayload, "idxdata=%d", idxdata); } if (idxdata < 5) { GST_DEBUG_OBJECT (basepayload, "Returning GST_FLOW_OK without creating RTP packet"); return GST_FLOW_OK; } pdata += 3; idxdata -= 3; nalType = pdata[0] & 0x1f; GST_DEBUG_OBJECT (basepayload, "Processing Buffer with NAL TYPE=%d", nalType); packet_len = gst_rtp_buffer_calc_packet_len (idxdata, 0, 0); if (packet_len < mtu) { GST_DEBUG_OBJECT (basepayload, "NAL Unit fit in one packet datasize=%d mtu=%d", idxdata, mtu); /* will fit in one packet */ outbuf = gst_rtp_buffer_new_allocate (idxdata, 0, 0); GST_BUFFER_TIMESTAMP (outbuf) = timestamp; gst_rtp_buffer_set_marker (outbuf, 1); payload = gst_rtp_buffer_get_payload (outbuf); GST_DEBUG_OBJECT (basepayload, "Copying %d bytes to outbuf", idxdata); memcpy (payload, pdata, idxdata); gst_buffer_unref (buffer); ret = gst_basertppayload_push (basepayload, outbuf); return ret; } else { /* Fragmentation Units FU-A */ guint8 nalHeader; guint limitedSize; int ii = 0, start = 1, end = 0, first = 0; GST_DEBUG_OBJECT (basepayload, "NAL Unit DOES NOT fit in one packet datasize=%d mtu=%d", idxdata, mtu); nalHeader = *pdata; pdata++; idxdata--; ret = GST_FLOW_OK; GST_DEBUG_OBJECT (basepayload, "Using FU-A fragmentation for data size=%d", idxdata); payload_len = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0); /* We keep 2 bytes for FU indicator and FU Header */ while (end == 0) { limitedSize = idxdata < payload_len ? idxdata : payload_len; GST_DEBUG_OBJECT (basepayload, "Inside FU-A fragmentation limitedSize=%d iteration=%d", limitedSize, ii); outbuf = gst_rtp_buffer_new_allocate (limitedSize + 2, 0, 0); GST_BUFFER_TIMESTAMP (outbuf) = timestamp; gst_rtp_buffer_set_marker (outbuf, end); payload = gst_rtp_buffer_get_payload (outbuf); if (limitedSize == idxdata) { GST_DEBUG_OBJECT (basepayload, "end idxdata=%d iteration=%d", idxdata, ii); end = 1; } /* FU indicator */ payload[0] = (nalHeader & 0x60) | 28; /* FU Header */ payload[1] = (start << 7) | (end << 6) | (nalHeader & 0x1f); memcpy (&payload[2], pdata + first, limitedSize); GST_DEBUG_OBJECT (basepayload, "recorded %d payload bytes into packet iteration=%d", limitedSize + 2, ii); ret = gst_basertppayload_push (basepayload, outbuf); if (ret != GST_FLOW_OK) break; idxdata -= limitedSize; first += limitedSize; ii++; start = 0; } gst_buffer_unref (buffer); return ret; } GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT, (NULL), ("Should not be there !!")); gst_buffer_unref (buffer); return GST_FLOW_ERROR; } static void gst_rtp_h264_pay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRtpH264Pay *rtph264pay; rtph264pay = GST_RTP_H264_PAY (object); switch (prop_id) { default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtp_h264_pay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRtpH264Pay *rtph264pay; rtph264pay = GST_RTP_H264_PAY (object); switch (prop_id) { default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstStateChangeReturn gst_rtp_h264_pay_change_state (GstElement * element, GstStateChange transition) { GstRtpH264Pay *rtph264pay; GstStateChangeReturn ret; rtph264pay = GST_RTP_H264_PAY (element); switch (transition) { default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { default: break; } return ret; } gboolean gst_rtp_h264_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtph264pay", GST_RANK_NONE, GST_TYPE_RTP_H264_PAY); }