gstreamer/gst/rtp/gstrtpvorbisdepay.c

/* 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 <gst/rtp/gstrtpbuffer.h>

#include <string.h>
#include "gstrtpvorbisdepay.h"

GST_DEBUG_CATEGORY_STATIC (rtpvorbisdepay_debug);
#define GST_CAT_DEFAULT (rtpvorbisdepay_debug)

/* elementfactory information */
static const GstElementDetails gst_rtp_vorbis_depay_details =
GST_ELEMENT_DETAILS ("RTP packet parser",
    "Codec/Depayloader/Network",
    "Extracts Vorbis Audio from RTP packets (draft-01 of RFC XXXX)",
    "Wim Taymans <wim@fluendo.com>");

/* RtpVorbisDepay signals and args */
enum
{
  /* FILL ME */
  LAST_SIGNAL
};

enum
{
  ARG_0,
};

static GstStaticPadTemplate gst_rtp_vorbis_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
    GST_PAD_SINK,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("application/x-rtp, "
        "media = (string) \"audio\", "
        "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_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
    GST_PAD_SRC,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("audio/x-vorbis")
    );

GST_BOILERPLATE (GstRtpVorbisDepay, gst_rtp_vorbis_depay, GstBaseRTPDepayload,
    GST_TYPE_BASE_RTP_DEPAYLOAD);

static gboolean gst_rtp_vorbis_depay_setcaps (GstBaseRTPDepayload * depayload,
    GstCaps * caps);
static GstBuffer *gst_rtp_vorbis_depay_process (GstBaseRTPDepayload * depayload,
    GstBuffer * buf);

static void gst_rtp_vorbis_depay_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec);
static void gst_rtp_vorbis_depay_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec);
static void gst_rtp_vorbis_depay_finalize (GObject * object);

static GstStateChangeReturn gst_rtp_vorbis_depay_change_state (GstElement *
    element, GstStateChange transition);


static void
gst_rtp_vorbis_depay_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_depay_sink_template));
  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&gst_rtp_vorbis_depay_src_template));

  gst_element_class_set_details (element_class, &gst_rtp_vorbis_depay_details);
}

static void
gst_rtp_vorbis_depay_class_init (GstRtpVorbisDepayClass * klass)
{
  GObjectClass *gobject_class;
  GstElementClass *gstelement_class;
  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;

  gobject_class = (GObjectClass *) klass;
  gstelement_class = (GstElementClass *) klass;
  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;

  gobject_class->set_property = gst_rtp_vorbis_depay_set_property;
  gobject_class->get_property = gst_rtp_vorbis_depay_get_property;
  gobject_class->finalize = gst_rtp_vorbis_depay_finalize;

  gstelement_class->change_state = gst_rtp_vorbis_depay_change_state;

  gstbasertpdepayload_class->process = gst_rtp_vorbis_depay_process;
  gstbasertpdepayload_class->set_caps = gst_rtp_vorbis_depay_setcaps;

  GST_DEBUG_CATEGORY_INIT (rtpvorbisdepay_debug, "rtpvorbisdepay", 0,
      "Vorbis RTP Depayloader");
}

static void
gst_rtp_vorbis_depay_init (GstRtpVorbisDepay * rtpvorbisdepay,
    GstRtpVorbisDepayClass * klass)
{
  rtpvorbisdepay->adapter = gst_adapter_new ();
}
static void
gst_rtp_vorbis_depay_finalize (GObject * object)
{
  GstRtpVorbisDepay *rtpvorbisdepay = GST_RTP_VORBIS_DEPAY (object);

  g_object_unref (rtpvorbisdepay->adapter);

  G_OBJECT_CLASS (parent_class)->finalize (object);
}

static gboolean
gst_rtp_vorbis_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
  GstStructure *structure;
  GstRtpVorbisDepay *rtpvorbisdepay;
  GstCaps *srccaps;
  gint clock_rate;

  rtpvorbisdepay = GST_RTP_VORBIS_DEPAY (depayload);

  structure = gst_caps_get_structure (caps, 0);

  if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
    goto no_rate;

  /* caps seem good, configure element */
  depayload->clock_rate = clock_rate;

  /* set caps on pad and on header */
  srccaps = gst_caps_new_simple ("audio/x-vorbis", NULL);
  gst_pad_set_caps (depayload->srcpad, srccaps);
  gst_caps_unref (srccaps);

  return TRUE;

no_rate:
  {
    GST_ERROR_OBJECT (rtpvorbisdepay, "no clock-rate specified");
    return FALSE;
  }
}

static GstBuffer *
gst_rtp_vorbis_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
  GstRtpVorbisDepay *rtpvorbisdepay;
  GstBuffer *outbuf;
  GstFlowReturn ret;
  gint payload_len;
  guint8 *payload, *to_free = NULL;
  guint32 timestamp;
  guint32 header, ident;
  guint8 F, VDT, packets;
  gboolean free_payload;

  rtpvorbisdepay = GST_RTP_VORBIS_DEPAY (depayload);

  if (!gst_rtp_buffer_validate (buf))
    goto bad_packet;

  payload_len = gst_rtp_buffer_get_payload_len (buf);

  GST_DEBUG_OBJECT (depayload, "got RTP packet of size %d", payload_len);

  /* we need at least 4 bytes for the packet header */
  if (G_UNLIKELY (payload_len < 4))
    goto packet_short;

  payload = gst_rtp_buffer_get_payload (buf);
  free_payload = FALSE;

  header = GST_READ_UINT32_BE (payload);
  /*
   *  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.
   */
  VDT = (header & 0x30) >> 4;
  if (G_UNLIKELY (VDT == 3))
    goto ignore_reserved;

  ident = (header >> 8) & 0xffffff;
  F = (header & 0xc0) >> 6;
  packets = (header & 0xf);

  if (VDT == 0) {
    /* FIXME, if we have a raw payload, we need the codebook for the ident */
  }

  /* skip header */
  payload += 4;
  payload_len -= 4;

  GST_DEBUG_OBJECT (depayload, "ident: %u, F: %d, VDT: %d, packets: %d", ident,
      F, VDT, packets);

  /* fragmented packets, assemble */
  if (F != 0) {
    GstBuffer *vdata;
    guint headerskip;

    if (F == 1) {
      /* if we start a packet, clear adapter and start assembling. */
      gst_adapter_clear (rtpvorbisdepay->adapter);
      GST_DEBUG_OBJECT (depayload, "start assemble");
      rtpvorbisdepay->assembling = TRUE;
    }

    if (!rtpvorbisdepay->assembling)
      goto no_output;

    /* first assembled packet, reuse 2 bytes to store the length */
    headerskip = (F == 1 ? 4 : 6);
    /* skip header and length. */
    vdata = gst_rtp_buffer_get_payload_subbuffer (buf, headerskip, -1);

    GST_DEBUG_OBJECT (depayload, "assemble vorbis packet");
    gst_adapter_push (rtpvorbisdepay->adapter, vdata);

    /* packet is not complete, we are done */
    if (F != 3)
      goto no_output;

    /* construct assembled buffer */
    payload_len = gst_adapter_available (rtpvorbisdepay->adapter);
    payload = gst_adapter_take (rtpvorbisdepay->adapter, payload_len);
    /* fix the length */
    payload[0] = ((payload_len - 2) >> 8) & 0xff;
    payload[1] = (payload_len - 2) & 0xff;
    to_free = payload;
  }

  GST_DEBUG_OBJECT (depayload, "assemble done");

  /* we not assembling anymore now */
  rtpvorbisdepay->assembling = FALSE;
  gst_adapter_clear (rtpvorbisdepay->adapter);

  /* payload now points to a length with that many vorbis data bytes.
   * Iterate over the packets and send them out.
   *
   *  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
   * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   * |             length            |          vorbis data         ..
   * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   * ..                        vorbis data                           |
   * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   * |            length             |   next vorbis packet data    ..
   * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   * ..                        vorbis data                           |
   * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+*
   */
  timestamp = gst_rtp_buffer_get_timestamp (buf);

  while (payload_len > 2) {
    guint16 length;

    length = GST_READ_UINT16_BE (payload);
    payload += 2;
    payload_len -= 2;

    GST_DEBUG_OBJECT (depayload, "read length %u, avail: %d", length,
        payload_len);

    /* skip packet if something odd happens */
    if (G_UNLIKELY (length > payload_len))
      goto length_short;

    /* create buffer for packet */
    if (G_UNLIKELY (to_free)) {
      outbuf = gst_buffer_new ();
      GST_BUFFER_DATA (outbuf) = payload;
      GST_BUFFER_MALLOCDATA (outbuf) = to_free;
      GST_BUFFER_SIZE (outbuf) = length;
      to_free = NULL;
    } else {
      outbuf = gst_buffer_new_and_alloc (length);
      memcpy (GST_BUFFER_DATA (outbuf), payload, length);
    }

    payload += length;
    payload_len -= length;

    if (timestamp != -1)
      /* push with timestamp of the last packet, which is the same timestamp that
       * should apply to the first assembled packet. */
      ret = gst_base_rtp_depayload_push_ts (depayload, timestamp, outbuf);
    else
      ret = gst_base_rtp_depayload_push (depayload, outbuf);

    if (ret != GST_FLOW_OK)
      break;

    /* make sure we don't set a timestamp on next buffers */
    timestamp = -1;
  }

  g_free (to_free);

  return NULL;

no_output:
  {
    return NULL;
  }
  /* ERORRS */
bad_packet:
  {
    GST_ELEMENT_WARNING (rtpvorbisdepay, STREAM, DECODE,
        ("Packet did not validate"), (NULL));
    return NULL;
  }
packet_short:
  {
    GST_ELEMENT_WARNING (rtpvorbisdepay, STREAM, DECODE,
        ("Packet was too short (%d < 4)", payload_len), (NULL));
    return NULL;
  }
ignore_reserved:
  {
    GST_WARNING_OBJECT (rtpvorbisdepay, "reserved VDT ignored");
    return NULL;
  }
length_short:
  {
    GST_ELEMENT_WARNING (rtpvorbisdepay, STREAM, DECODE,
        ("Packet contains invalid data"), (NULL));
    return NULL;
  }
}

static void
gst_rtp_vorbis_depay_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec)
{
  GstRtpVorbisDepay *rtpvorbisdepay;

  rtpvorbisdepay = GST_RTP_VORBIS_DEPAY (object);

  switch (prop_id) {
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
gst_rtp_vorbis_depay_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec)
{
  GstRtpVorbisDepay *rtpvorbisdepay;

  rtpvorbisdepay = GST_RTP_VORBIS_DEPAY (object);

  switch (prop_id) {
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static GstStateChangeReturn
gst_rtp_vorbis_depay_change_state (GstElement * element,
    GstStateChange transition)
{
  GstRtpVorbisDepay *rtpvorbisdepay;
  GstStateChangeReturn ret;

  rtpvorbisdepay = GST_RTP_VORBIS_DEPAY (element);

  switch (transition) {
    case GST_STATE_CHANGE_NULL_TO_READY:
      break;
    case GST_STATE_CHANGE_READY_TO_PAUSED:
      break;
    default:
      break;
  }

  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);

  switch (transition) {
    case GST_STATE_CHANGE_READY_TO_NULL:
      break;
    default:
      break;
  }
  return ret;
}

gboolean
gst_rtp_vorbis_depay_plugin_init (GstPlugin * plugin)
{
  return gst_element_register (plugin, "rtpvorbisdepay",
      GST_RANK_MARGINAL, GST_TYPE_RTP_VORBIS_DEPAY);
}