gstreamer/gst/rtp/gstrtpgsmdepay.c

/* GStreamer
 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
 * Copyright (C) <2005> Zeeshan Ali <zeenix@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., 51 Franklin St, Fifth Floor,
 * Boston, MA 02110-1301, USA.
 */

#ifdef HAVE_CONFIG_H
#  include "config.h"
#endif

#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpelements.h"
#include "gstrtpgsmdepay.h"
#include "gstrtputils.h"

GST_DEBUG_CATEGORY_STATIC (rtpgsmdepay_debug);
#define GST_CAT_DEFAULT (rtpgsmdepay_debug)

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

static GstStaticPadTemplate gst_rtp_gsm_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
    GST_PAD_SRC,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = 1")
    );

static GstStaticPadTemplate gst_rtp_gsm_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) 8000, " "encoding-name = (string) \"GSM\";"
        "application/x-rtp, "
        "media = (string) \"audio\", "
        "payload = (int) " GST_RTP_PAYLOAD_GSM_STRING ", "
        "clock-rate = (int) 8000")
    );

static GstBuffer *gst_rtp_gsm_depay_process (GstRTPBaseDepayload * _depayload,
    GstRTPBuffer * rtp);
static gboolean gst_rtp_gsm_depay_setcaps (GstRTPBaseDepayload * _depayload,
    GstCaps * caps);

#define gst_rtp_gsm_depay_parent_class parent_class
G_DEFINE_TYPE (GstRTPGSMDepay, gst_rtp_gsm_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpgsmdepay, "rtpgsmdepay",
    GST_RANK_SECONDARY, GST_TYPE_RTP_GSM_DEPAY, rtp_element_init (plugin));

static void
gst_rtp_gsm_depay_class_init (GstRTPGSMDepayClass * klass)
{
  GstElementClass *gstelement_class;
  GstRTPBaseDepayloadClass *gstrtpbase_depayload_class;

  gstelement_class = (GstElementClass *) klass;
  gstrtpbase_depayload_class = (GstRTPBaseDepayloadClass *) klass;

  gst_element_class_add_static_pad_template (gstelement_class,
      &gst_rtp_gsm_depay_src_template);
  gst_element_class_add_static_pad_template (gstelement_class,
      &gst_rtp_gsm_depay_sink_template);

  gst_element_class_set_static_metadata (gstelement_class,
      "RTP GSM depayloader", "Codec/Depayloader/Network/RTP",
      "Extracts GSM audio from RTP packets", "Zeeshan Ali <zeenix@gmail.com>");

  gstrtpbase_depayload_class->process_rtp_packet = gst_rtp_gsm_depay_process;
  gstrtpbase_depayload_class->set_caps = gst_rtp_gsm_depay_setcaps;

  GST_DEBUG_CATEGORY_INIT (rtpgsmdepay_debug, "rtpgsmdepay", 0,
      "GSM Audio RTP Depayloader");
}

static void
gst_rtp_gsm_depay_init (GstRTPGSMDepay * rtpgsmdepay)
{
}

static gboolean
gst_rtp_gsm_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
  GstCaps *srccaps;
  gboolean ret;
  GstStructure *structure;
  gint clock_rate;

  structure = gst_caps_get_structure (caps, 0);

  if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
    clock_rate = 8000;          /* default */
  depayload->clock_rate = clock_rate;

  srccaps = gst_caps_new_simple ("audio/x-gsm",
      "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, clock_rate, NULL);
  ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
  gst_caps_unref (srccaps);

  return ret;
}

static GstBuffer *
gst_rtp_gsm_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
  GstBuffer *outbuf = NULL;
  gboolean marker;

  marker = gst_rtp_buffer_get_marker (rtp);

  GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
      gst_buffer_get_size (rtp->buffer), marker,
      gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp));

  outbuf = gst_rtp_buffer_get_payload_buffer (rtp);

  if (marker && outbuf) {
    /* mark start of talkspurt with RESYNC */
    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
  }

  if (outbuf) {
    gst_rtp_drop_non_audio_meta (depayload, outbuf);
  }

  return outbuf;
}