gst/rtp/: Added H264 payloader. Fixes #423782.

Original commit message from CVS:
Patch by: Laurent Glayal <spglegle at yahoo dot fr>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_base_init),
(gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init),
(gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps),
(gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property),
(gst_rtp_h264_pay_get_property), (gst_rtp_h264_pay_change_state),
(gst_rtp_h264_pay_plugin_init):
* gst/rtp/gstrtph264pay.h:
Added H264 payloader. Fixes #423782.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
Small fixes.

1 files changed, 337 insertions, 0 deletions

diff --git a/gst/rtp/gstrtph264pay.c b/gst/rtp/gstrtph264pay.c
new file mode 100644
index 00000000..5dd7f6fa
--- /dev/null
+++ b/gst/rtp/gstrtph264pay.c
@@ -0,0 +1,337 @@
+/* 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 <string.h>
+
+#include <gst/rtp/gstrtpbuffer.h>
+
+#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 <spglegle@yahoo.fr>");
+
+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 {
+    GST_DEBUG_OBJECT (basepayload,
+        "NAL Unit DOES NOT fit in one packet datasize=%d mtu=%d", idxdata, mtu);
+
+    /* Fragmentation Units FU-A */
+    guint8 nalHeader;
+    guint limitedSize;
+
+    int ii = 0, start = 1, end = 0, first = 0;
+
+    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);
+}