gst/rtp/: Add DV pay and depayloaders. Fixes #532423.

Original commit message from CVS:
Based on patch by: Olivier Crete <tester at tester dot ca>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpdvdepay.c: (gst_rtp_dv_depay_base_init),
(gst_rtp_dv_depay_class_init), (gst_rtp_dv_depay_init),
(parse_encode), (gst_rtp_dv_depay_setcaps),
(calculate_difblock_location), (gst_rtp_dv_depay_process),
(gst_rtp_dv_depay_reset), (gst_rtp_dv_depay_change_state),
(gst_rtp_dv_depay_plugin_init):
* gst/rtp/gstrtpdvdepay.h:
* gst/rtp/gstrtpdvpay.c: (gst_dv_pay_mode_get_type),
(gst_rtp_dv_pay_base_init), (gst_rtp_dv_pay_class_init),
(gst_rtp_dv_pay_init), (gst_dv_pay_set_property),
(gst_dv_pay_get_property), (gst_rtp_dv_pay_setcaps),
(gst_dv_pay_negotiate), (include_dif),
(gst_rtp_dv_pay_handle_buffer), (gst_rtp_dv_pay_plugin_init):
* gst/rtp/gstrtpdvpay.h:
Add DV pay and depayloaders. Fixes #532423.

1 files changed, 386 insertions, 0 deletions

diff --git a/gst/rtp/gstrtpdvpay.c b/gst/rtp/gstrtpdvpay.c
new file mode 100644
index 00000000..65cfaf98
--- /dev/null
+++ b/gst/rtp/gstrtpdvpay.c
@@ -0,0 +1,386 @@
+/* Farsight
+ * Copyright (C) 2006 Marcel Moreaux <marcelm@spacelabs.nl>
+ *           (C) 2008 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 <stdlib.h>
+#include <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+#include "gstrtpdvpay.h"
+
+GST_DEBUG_CATEGORY (rtpdvpay_debug);
+#define GST_CAT_DEFAULT (rtpdvpay_debug)
+
+/* Elementfactory information */
+static GstElementDetails gst_rtp_dv_pay_details = {
+  "RTP DV Payloader",
+  "Codec/Payloader/Network",
+  "Payloads DV into RTP packets (RFC 3189)",
+  "Marcel Moreaux <marcelm@spacelabs.nl>, Wim Taymans <wim.taymans@gmail.com>"
+};
+
+#define DEFAULT_MODE GST_DV_PAY_MODE_VIDEO
+enum
+{
+  PROP_0,
+  PROP_MODE
+};
+
+/* takes both system and non-system streams */
+static GstStaticPadTemplate gst_rtp_dv_pay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("video/x-dv")
+    );
+
+static GstStaticPadTemplate gst_rtp_dv_pay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) { \"video\", \"audio\" } ,"
+        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+        "encoding-name = (string) \"DV\", "
+        "clock-rate = (int) 90000,"
+        "encode = (string) { \"SD-VCR/525-60\", \"SD-VCR/625-50\", \"HD-VCR/1125-60\","
+        "\"HD-VCR/1250-50\", \"SDL-VCR/525-60\", \"SDL-VCR/625-50\","
+        "\"306M/525-60\", \"306M/625-50\", \"314M-25/525-60\","
+        "\"314M-25/625-50\", \"314M-50/525-60\", \"314M-50/625-50\" }"
+        /* optional parameters can't go in the template
+         * "audio = (string) { \"bundled\", \"none\" }"
+         */
+    )
+    );
+
+static gboolean gst_rtp_dv_pay_setcaps (GstBaseRTPPayload * payload,
+    GstCaps * caps);
+static GstFlowReturn gst_rtp_dv_pay_handle_buffer (GstBaseRTPPayload * payload,
+    GstBuffer * buffer);
+
+#define GST_TYPE_DV_PAY_MODE (gst_dv_pay_mode_get_type())
+static GType
+gst_dv_pay_mode_get_type (void)
+{
+  static GType dv_pay_mode_type = 0;
+  static const GEnumValue dv_pay_modes[] = {
+    {GST_DV_PAY_MODE_VIDEO, "Video only", "video"},
+    {GST_DV_PAY_MODE_BUNDLED, "Video and Audio bundled", "bundled"},
+    {GST_DV_PAY_MODE_AUDIO, "Audio only", "audio"},
+    {0, NULL, NULL},
+  };
+
+  if (!dv_pay_mode_type) {
+    dv_pay_mode_type = g_enum_register_static ("GstDVPayMode", dv_pay_modes);
+  }
+  return dv_pay_mode_type;
+}
+
+
+static void gst_dv_pay_set_property (GObject * object,
+    guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_dv_pay_get_property (GObject * object,
+    guint prop_id, GValue * value, GParamSpec * pspec);
+
+GST_BOILERPLATE (GstRTPDVPay, gst_rtp_dv_pay, GstBaseRTPPayload,
+    GST_TYPE_BASE_RTP_PAYLOAD)
+
+     static void gst_rtp_dv_pay_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_dv_pay_sink_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_dv_pay_src_template));
+  gst_element_class_set_details (element_class, &gst_rtp_dv_pay_details);
+}
+
+static void
+gst_rtp_dv_pay_class_init (GstRTPDVPayClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+
+  gobject_class->set_property = gst_dv_pay_set_property;
+  gobject_class->get_property = gst_dv_pay_get_property;
+
+  gstbasertppayload_class->set_caps = gst_rtp_dv_pay_setcaps;
+  gstbasertppayload_class->handle_buffer = gst_rtp_dv_pay_handle_buffer;
+
+  g_object_class_install_property (gobject_class, PROP_MODE,
+      g_param_spec_enum ("mode", "Mode",
+          "The payload mode of payloading",
+          GST_TYPE_DV_PAY_MODE, DEFAULT_MODE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  GST_DEBUG_CATEGORY_INIT (rtpdvpay_debug, "rtpdvpay", 0, "DV RTP Payloader");
+}
+
+static void
+gst_rtp_dv_pay_init (GstRTPDVPay * rtpdvpay, GstRTPDVPayClass * klass)
+{
+}
+
+static void
+gst_dv_pay_set_property (GObject * object,
+    guint prop_id, const GValue * value, GParamSpec * pspec)
+{
+  GstRTPDVPay *rtpdvpay = GST_RTP_DV_PAY (object);
+
+  switch (prop_id) {
+    case PROP_MODE:
+      rtpdvpay->mode = g_value_get_enum (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_dv_pay_get_property (GObject * object,
+    guint prop_id, GValue * value, GParamSpec * pspec)
+{
+  GstRTPDVPay *rtpdvpay = GST_RTP_DV_PAY (object);
+
+  switch (prop_id) {
+    case PROP_MODE:
+      g_value_set_enum (value, rtpdvpay->mode);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static gboolean
+gst_rtp_dv_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
+{
+  GstRTPDVPay *rtpdvpay;
+
+  rtpdvpay = GST_RTP_DV_PAY (payload);
+
+  /* We don't do anything here, but we could check if it's a system stream and if
+   * it's not, default to sending the video only. We will negotiate downstream
+   * caps when we get to see the first frame. */
+
+  return TRUE;
+}
+
+static gboolean
+gst_dv_pay_negotiate (GstRTPDVPay * rtpdvpay, guint8 * data, guint size)
+{
+  gchar *encode, *media;
+  gboolean audio_bundled;
+
+  if ((data[3] & 0x80) == 0) {  /* DSF flag */
+    /* it's an NTSC format */
+    if ((data[80 * 5 + 48 + 3] & 0x4) && (data[80 * 5 + 48] == 0x60)) { /* 4:2:2 sampling */
+      /* NTSC 50Mbps */
+      encode = "314M-25/525-60";
+    } else {                    /* 4:1:1 sampling */
+      /* NTSC 25Mbps */
+      encode = "SD-VCR/525-60";
+    }
+  } else {
+    /* it's a PAL format */
+    if ((data[80 * 5 + 48 + 3] & 0x4) && (data[80 * 5 + 48] == 0x60)) { /* 4:2:2 sampling */
+      /* PAL 50Mbps */
+      encode = "314M-50/625-50";
+    } else if ((data[5] & 0x07) == 0) { /* APT flag */
+      /* PAL 25Mbps 4:2:0 */
+      encode = "SD-VCR/625-50";
+    } else
+      /* PAL 25Mbps 4:1:1 */
+      encode = "314M-25/625-50";
+  }
+
+  media = "video";
+  audio_bundled = FALSE;
+
+  switch (rtpdvpay->mode) {
+    case GST_DV_PAY_MODE_AUDIO:
+      media = "audio";
+      break;
+    case GST_DV_PAY_MODE_BUNDLED:
+      audio_bundled = TRUE;
+      break;
+    default:
+      break;
+  }
+  gst_basertppayload_set_options (GST_BASE_RTP_PAYLOAD (rtpdvpay), media, TRUE,
+      "DV", 90000);
+
+  if (audio_bundled) {
+    gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpdvpay),
+        "encode", G_TYPE_STRING, encode,
+        "audio", G_TYPE_STRING, "bundled", NULL);
+  } else {
+    gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpdvpay),
+        "encode", G_TYPE_STRING, encode, NULL);
+  }
+  return TRUE;
+}
+
+static gboolean
+include_dif (GstRTPDVPay * rtpdvpay, guint8 * data)
+{
+  gint block_type;
+  gboolean res;
+
+  block_type = data[0] >> 5;
+
+  switch (block_type) {
+    case 0:                    /* Header block */
+    case 1:                    /* Subcode block */
+    case 2:                    /* VAUX block */
+      /* always include these blocks */
+      res = TRUE;
+      break;
+    case 3:                    /* Audio block */
+      /* never include audio if we are doing video only */
+      if (rtpdvpay->mode == GST_DV_PAY_MODE_VIDEO)
+        res = FALSE;
+      else
+        res = TRUE;
+      break;
+    case 4:                    /* Video block */
+      /* never include video if we are doing audio only */
+      if (rtpdvpay->mode == GST_DV_PAY_MODE_AUDIO)
+        res = FALSE;
+      else
+        res = TRUE;
+      break;
+    default:                   /* Something bogus, just ignore */
+      res = FALSE;
+      break;
+  }
+  return res;
+}
+
+/* Get a DV frame, chop it up in pieces, and push the pieces to the RTP layer.
+ */
+static GstFlowReturn
+gst_rtp_dv_pay_handle_buffer (GstBaseRTPPayload * basepayload,
+    GstBuffer * buffer)
+{
+  GstRTPDVPay *rtpdvpay;
+  guint max_payload_size;
+  GstBuffer *outbuf;
+  GstFlowReturn ret = GST_FLOW_OK;
+  gint hdrlen;
+  guint size;
+  guint8 *data;
+  guint8 *dest;
+  guint filled;
+
+  rtpdvpay = GST_RTP_DV_PAY (basepayload);
+
+  hdrlen = gst_rtp_buffer_calc_header_len (0);
+  /* DV frames are made up from a bunch of DIF blocks. DIF blocks are 80 bytes
+   * each, and we should put an integral number of them in each RTP packet.
+   * Therefore, we round the available room down to the nearest multiple of 80.
+   *
+   * The available room is just the packet MTU, minus the RTP header length. */
+  max_payload_size = ((GST_BASE_RTP_PAYLOAD_MTU (rtpdvpay) - hdrlen) / 80) * 80;
+
+  /* The length of the buffer to transmit. */
+  size = GST_BUFFER_SIZE (buffer);
+  data = GST_BUFFER_DATA (buffer);
+
+  GST_DEBUG_OBJECT (rtpdvpay,
+      "DV RTP payloader got buffer of %u bytes, splitting in %u byte "
+      "payload fragments, at time %" GST_TIME_FORMAT, size, max_payload_size,
+      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
+
+  if (!rtpdvpay->negotiated) {
+    gst_dv_pay_negotiate (rtpdvpay, data, size);
+    /* if we have not yet scanned the stream for its type, do so now */
+    rtpdvpay->negotiated = TRUE;
+  }
+
+  outbuf = NULL;
+  dest = NULL;
+  filled = 0;
+
+  /* while we have a complete DIF chunks left */
+  while (size >= 80) {
+    /* Allocate a new buffer, set the timestamp */
+    if (outbuf == NULL) {
+      outbuf = gst_rtp_buffer_new_allocate (max_payload_size, 0, 0);
+      GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buffer);
+      dest = gst_rtp_buffer_get_payload (outbuf);
+      filled = 0;
+    }
+
+    /* inspect the DIF chunk, if we don't need to include it, skip to the next one. */
+    if (include_dif (rtpdvpay, data)) {
+      /* copy data in packet */
+      memcpy (dest, data, 80);
+
+      dest += 80;
+      filled += 80;
+    }
+
+    /* go to next dif chunk */
+    size -= 80;
+    data += 80;
+
+    /* push out the buffer if the next one would exceed the max packet size or
+     * when we are pushing the last packet */
+    if (filled + 80 > max_payload_size || size < 80) {
+      if (size < 160) {
+        guint hlen;
+
+        /* set marker */
+        gst_rtp_buffer_set_marker (outbuf, TRUE);
+
+        /* shrink buffer to last packet */
+        hlen = gst_rtp_buffer_get_header_len (outbuf);
+        gst_rtp_buffer_set_packet_len (outbuf, hlen + filled);
+      }
+      /* Push out the created piece, and check for errors. */
+      ret = gst_basertppayload_push (basepayload, outbuf);
+      if (ret != GST_FLOW_OK)
+        break;
+
+      outbuf = NULL;
+    }
+  }
+  gst_buffer_unref (buffer);
+
+  return ret;
+}
+
+gboolean
+gst_rtp_dv_pay_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpdvpay",
+      GST_RANK_NONE, GST_TYPE_RTP_DV_PAY);
+}