gst/rtp/: Small cleanups.

Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_change_state):
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init):
Small cleanups.
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_base_init),
(gst_rtp_vorbis_depay_class_init), (gst_rtp_vorbis_depay_init),
(gst_rtp_vorbis_depay_finalize), (gst_rtp_vorbis_depay_setcaps),
(gst_rtp_vorbis_depay_process),
(gst_rtp_vorbis_depay_set_property),
(gst_rtp_vorbis_depay_get_property),
(gst_rtp_vorbis_depay_change_state),
(gst_rtp_vorbis_depay_plugin_init):
* gst/rtp/gstrtpvorbisdepay.h:
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_base_init),
(gst_rtp_vorbis_pay_class_init), (gst_rtp_vorbis_pay_init),
(gst_rtp_vorbis_pay_setcaps), (gst_rtp_vorbis_pay_init_packet),
(gst_rtp_vorbis_pay_flush_packet),
(gst_rtp_vorbis_pay_append_buffer),
(gst_rtp_vorbis_pay_handle_buffer),
(gst_rtp_vorbis_pay_plugin_init):
* gst/rtp/gstrtpvorbispay.h:
Add experimental vorbis pay and depayloaders.

1 files changed, 333 insertions, 0 deletions

diff --git a/gst/rtp/gstrtpvorbispay.c b/gst/rtp/gstrtpvorbispay.c
new file mode 100644
index 00000000..30336425
--- /dev/null
+++ b/gst/rtp/gstrtpvorbispay.c
@@ -0,0 +1,333 @@
+/* GStreamer
+ * Copyright (C) <2006> Wim Taymans <wim@fluendo.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 <string.h>
+
+#include <gst/rtp/gstrtpbuffer.h>
+
+#include "gstrtpvorbispay.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpvorbispay_debug);
+#define GST_CAT_DEFAULT (rtpvorbispay_debug)
+
+/* references:
+ * http://svn.xiph.org/trunk/vorbis/doc/draft-ietf-avt-rtp-vorbis-01.txt
+ */
+
+/* elementfactory information */
+static const GstElementDetails gst_rtp_vorbispay_details =
+GST_ELEMENT_DETAILS ("RTP packet parser",
+    "Codec/Payloader/Network",
+    "Payload-encode Vorbis audio into RTP packets (draft-01 RFC XXXX)",
+    "Wim Taymans <wim@fluendo.com>");
+
+static GstStaticPadTemplate gst_rtp_vorbis_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) [ 96, 127 ], "
+        "clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"vorbis\""
+        /* All required parameters
+         *
+         * "encoding-params = (string) <num channels>"
+         * "delivery-method = (string) { inline, in_band, out_band/<specific_name> } "
+         * "configuration = (string) ANY"
+         */
+        /* All optional parameters
+         *
+         * "configuration-uri ="
+         */
+    )
+    );
+
+static GstStaticPadTemplate gst_rtp_vorbis_pay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-vorbis")
+    );
+
+GST_BOILERPLATE (GstRtpVorbisPay, gst_rtp_vorbis_pay, GstBaseRTPPayload,
+    GST_TYPE_BASE_RTP_PAYLOAD);
+
+static gboolean gst_rtp_vorbis_pay_setcaps (GstBaseRTPPayload * basepayload,
+    GstCaps * caps);
+static GstFlowReturn gst_rtp_vorbis_pay_handle_buffer (GstBaseRTPPayload * pad,
+    GstBuffer * buffer);
+
+static void
+gst_rtp_vorbis_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_vorbis_pay_src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_vorbis_pay_sink_template));
+
+  gst_element_class_set_details (element_class, &gst_rtp_vorbispay_details);
+}
+
+static void
+gst_rtp_vorbis_pay_class_init (GstRtpVorbisPayClass * 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);
+
+  gstbasertppayload_class->set_caps = gst_rtp_vorbis_pay_setcaps;
+  gstbasertppayload_class->handle_buffer = gst_rtp_vorbis_pay_handle_buffer;
+
+  GST_DEBUG_CATEGORY_INIT (rtpvorbispay_debug, "rtpvorbispay", 0,
+      "Vorbis RTP Payloader");
+}
+
+static void
+gst_rtp_vorbis_pay_init (GstRtpVorbisPay * rtpvorbispay,
+    GstRtpVorbisPayClass * klass)
+{
+}
+
+static gboolean
+gst_rtp_vorbis_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
+{
+  GstRtpVorbisPay *rtpvorbispay;
+
+  rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
+
+  gst_basertppayload_set_options (basepayload, "audio", TRUE, "vorbis", 8000);
+  gst_basertppayload_set_outcaps (basepayload,
+      "encoding-params", G_TYPE_STRING, "1",
+      /* don't set the defaults 
+       */
+      NULL);
+
+  return TRUE;
+}
+
+static void
+gst_rtp_vorbis_pay_init_packet (GstRtpVorbisPay * rtpvorbispay)
+{
+  guint payload_len;
+
+  if (rtpvorbispay->packet)
+    gst_buffer_unref (rtpvorbispay->packet);
+
+  GST_DEBUG_OBJECT (rtpvorbispay, "starting new packet");
+
+  /* new packet allocate max packet size */
+  rtpvorbispay->packet =
+      gst_rtp_buffer_new_allocate_len (GST_BASE_RTP_PAYLOAD_MTU
+      (rtpvorbispay), 0, 0);
+  rtpvorbispay->payload_pos = 4;
+  payload_len = gst_rtp_buffer_get_payload_len (rtpvorbispay->packet);
+  rtpvorbispay->payload_left = payload_len - 4;
+  rtpvorbispay->payload_duration = 0;
+  rtpvorbispay->payload_ident = 0;
+  rtpvorbispay->payload_F = 0;
+  rtpvorbispay->payload_VDT = 0;
+  rtpvorbispay->payload_pkts = 0;
+}
+
+static GstFlowReturn
+gst_rtp_vorbis_pay_flush_packet (GstRtpVorbisPay * rtpvorbispay)
+{
+  GstFlowReturn ret;
+  guint8 *payload;
+  guint hlen;
+
+  /* check for empty packet */
+  if (!rtpvorbispay || rtpvorbispay->payload_pos <= 4)
+    return GST_FLOW_OK;
+
+  GST_DEBUG_OBJECT (rtpvorbispay, "flushing packet");
+
+  /* fix header */
+  payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet);
+  /*
+   *  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
+   * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+   * |                     Ident                     | F |VDT|# pkts.|
+   * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+   *
+   * F: Fragment type (0=none, 1=start, 2=cont, 3=end)
+   * VDT: Vorbis data type (0=vorbis, 1=config, 2=comment, 3=reserved)
+   * pkts: number of packets.
+   */
+  payload[0] = (rtpvorbispay->payload_ident >> 16) & 0xff;
+  payload[1] = (rtpvorbispay->payload_ident >> 8) & 0xff;
+  payload[2] = (rtpvorbispay->payload_ident) & 0xff;
+  payload[3] = (rtpvorbispay->payload_F & 0x3) << 6 |
+      (rtpvorbispay->payload_VDT & 0x3) << 4 |
+      (rtpvorbispay->payload_pkts & 0xf);
+
+  /* shrink the buffer size to the last written byte */
+  hlen = gst_rtp_buffer_calc_header_len (0);
+  GST_BUFFER_SIZE (rtpvorbispay->packet) = hlen + rtpvorbispay->payload_pos;
+
+  /* push, this gives away our ref to the packet, so clear it. */
+  ret =
+      gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpvorbispay),
+      rtpvorbispay->packet);
+  rtpvorbispay->packet = NULL;
+
+  /* prepare new packet */
+  gst_rtp_vorbis_pay_init_packet (rtpvorbispay);
+
+  return ret;
+}
+
+static GstFlowReturn
+gst_rtp_vorbis_pay_append_buffer (GstRtpVorbisPay * rtpvorbispay,
+    GstBuffer * buffer)
+{
+  GstFlowReturn res;
+  guint size;
+  GstClockTime duration;
+  guint plen;
+  guint8 *ppos, *payload, *data;
+  gboolean fragmented;
+
+  res = GST_FLOW_OK;
+
+  if (rtpvorbispay->payload_left < 2)
+    return res;
+
+  size = GST_BUFFER_SIZE (buffer);
+  /* skip packets that are too big */
+  if (size > 0xffff)
+    return res;
+
+  data = GST_BUFFER_DATA (buffer);
+  duration = GST_BUFFER_DURATION (buffer);
+  payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet);
+  ppos = payload + rtpvorbispay->payload_pos;
+  fragmented = FALSE;
+
+  while (size) {
+    plen = MIN (rtpvorbispay->payload_left - 2, size);
+
+    GST_DEBUG_OBJECT (rtpvorbispay, "append %u bytes", plen);
+
+    ppos[0] = (plen >> 8) & 0xff;
+    ppos[1] = (plen & 0xff);
+    memcpy (&ppos[2], data, plen);
+
+    size -= plen;
+    data += plen;
+
+    rtpvorbispay->payload_pos += plen + 2;
+    rtpvorbispay->payload_left -= plen + 2;
+
+    if (fragmented) {
+      if (size == 0)
+        /* last fragment, set F to 0x3. */
+        rtpvorbispay->payload_F = 0x3;
+      else
+        /* fragment continues, set F to 0x2. */
+        rtpvorbispay->payload_F = 0x2;
+    } else {
+      if (size == 0) {
+        /* unfragmented packet, update stats for next packet */
+        rtpvorbispay->payload_pkts++;
+        if (duration != GST_CLOCK_TIME_NONE)
+          rtpvorbispay->payload_duration += duration;
+      } else {
+        /* fragmented packet starts, set F to 0x1, mark ourselves as
+         * fragmented. */
+        rtpvorbispay->payload_F = 0x1;
+        fragmented = TRUE;
+      }
+    }
+    if (fragmented) {
+      /* fragmented packets are always flushed and have ptks of 0 */
+      rtpvorbispay->payload_pkts = 0;
+      res = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay);
+      /* get new pointers */
+      payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet);
+      ppos = payload + rtpvorbispay->payload_pos;
+    }
+  }
+
+  return res;
+}
+
+static GstFlowReturn
+gst_rtp_vorbis_pay_handle_buffer (GstBaseRTPPayload * basepayload,
+    GstBuffer * buffer)
+{
+  GstRtpVorbisPay *rtpvorbispay;
+  GstFlowReturn ret;
+  guint size, newsize;
+  guint packet_len;
+  GstClockTime duration, newduration;
+  gboolean flush;
+
+  rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
+
+  size = GST_BUFFER_SIZE (buffer);
+  duration = GST_BUFFER_DURATION (buffer);
+
+  GST_DEBUG_OBJECT (rtpvorbispay, "size %u, duration %" GST_TIME_FORMAT,
+      size, GST_TIME_ARGS (duration));
+
+  if (!rtpvorbispay->packet)
+    gst_rtp_vorbis_pay_init_packet (rtpvorbispay);
+
+  /* size increases with packet length and 2 bytes size eader. */
+  newduration = rtpvorbispay->payload_duration;
+  if (duration != GST_CLOCK_TIME_NONE)
+    newduration += duration;
+
+  newsize = rtpvorbispay->payload_pos + 2 + size;
+  packet_len = gst_rtp_buffer_calc_packet_len (newsize, 0, 0);
+
+  /* check buffer filled against length and max latency */
+  flush = gst_basertppayload_is_filled (basepayload, packet_len, newduration);
+  /* we can store up to 15 vorbis packets in one RTP packet. */
+  flush |= (rtpvorbispay->payload_pkts == 15);
+
+  if (flush)
+    ret = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay);
+
+  /* put buffer in packet */
+  ret = gst_rtp_vorbis_pay_append_buffer (rtpvorbispay, buffer);
+
+  return ret;
+}
+
+gboolean
+gst_rtp_vorbis_pay_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpvorbispay",
+      GST_RANK_NONE, GST_TYPE_RTP_VORBIS_PAY);
+}