gstrtpg722depay.c 7.3 KB
Newer Older
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
/* GStreamer
 * Copyright (C) <2007> 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
Tim-Philipp Müller's avatar
Tim-Philipp Müller committed
16 17
 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
 * Boston, MA 02110-1301, USA.
18 19 20 21 22 23 24 25 26 27 28 29 30
 */

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

#include <string.h>
#include <stdlib.h>

#include <gst/audio/audio.h>

#include "gstrtpg722depay.h"
#include "gstrtpchannels.h"
31
#include "gstrtputils.h"
32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48

GST_DEBUG_CATEGORY_STATIC (rtpg722depay_debug);
#define GST_CAT_DEFAULT (rtpg722depay_debug)

static GstStaticPadTemplate gst_rtp_g722_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
    GST_PAD_SRC,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("audio/G722, "
        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
    );

static GstStaticPadTemplate gst_rtp_g722_depay_sink_template =
    GST_STATIC_PAD_TEMPLATE ("sink",
    GST_PAD_SINK,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("application/x-rtp, "
49
        "media = (string) \"audio\", " "clock-rate = (int) 8000, "
50 51 52 53 54 55 56 57 58 59 60 61 62
        /* "channels = (int) [1, MAX]"  */
        /* "channel-order = (string) ANY" */
        "encoding-name = (string) \"G722\";"
        "application/x-rtp, "
        "media = (string) \"audio\", "
        "payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", "
        "clock-rate = (int) [ 1, MAX ]"
        /* "channels = (int) [1, MAX]" */
        /* "emphasis = (string) ANY" */
        /* "channel-order = (string) ANY" */
    )
    );

Wim Taymans's avatar
Wim Taymans committed
63 64
#define gst_rtp_g722_depay_parent_class parent_class
G_DEFINE_TYPE (GstRtpG722Depay, gst_rtp_g722_depay,
Wim Taymans's avatar
Wim Taymans committed
65
    GST_TYPE_RTP_BASE_DEPAYLOAD);
66

Wim Taymans's avatar
Wim Taymans committed
67
static gboolean gst_rtp_g722_depay_setcaps (GstRTPBaseDepayload * depayload,
68
    GstCaps * caps);
Wim Taymans's avatar
Wim Taymans committed
69
static GstBuffer *gst_rtp_g722_depay_process (GstRTPBaseDepayload * depayload,
70
    GstRTPBuffer * rtp);
71 72

static void
Wim Taymans's avatar
Wim Taymans committed
73
gst_rtp_g722_depay_class_init (GstRtpG722DepayClass * klass)
74
{
Wim Taymans's avatar
Wim Taymans committed
75
  GstElementClass *gstelement_class;
Wim Taymans's avatar
Wim Taymans committed
76
  GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
Wim Taymans's avatar
Wim Taymans committed
77 78 79

  GST_DEBUG_CATEGORY_INIT (rtpg722depay_debug, "rtpg722depay", 0,
      "G722 RTP Depayloader");
80

Wim Taymans's avatar
Wim Taymans committed
81
  gstelement_class = (GstElementClass *) klass;
Wim Taymans's avatar
Wim Taymans committed
82
  gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
Wim Taymans's avatar
Wim Taymans committed
83 84

  gst_element_class_add_pad_template (gstelement_class,
85
      gst_static_pad_template_get (&gst_rtp_g722_depay_src_template));
Wim Taymans's avatar
Wim Taymans committed
86
  gst_element_class_add_pad_template (gstelement_class,
87 88
      gst_static_pad_template_get (&gst_rtp_g722_depay_sink_template));

89
  gst_element_class_set_static_metadata (gstelement_class,
Wim Taymans's avatar
Wim Taymans committed
90
      "RTP audio depayloader", "Codec/Depayloader/Network/RTP",
91 92 93
      "Extracts G722 audio from RTP packets",
      "Wim Taymans <wim.taymans@gmail.com>");

Wim Taymans's avatar
Wim Taymans committed
94
  gstrtpbasedepayload_class->set_caps = gst_rtp_g722_depay_setcaps;
95
  gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_g722_depay_process;
96 97 98
}

static void
Wim Taymans's avatar
Wim Taymans committed
99
gst_rtp_g722_depay_init (GstRtpG722Depay * rtpg722depay)
100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119
{
}

static gint
gst_rtp_g722_depay_parse_int (GstStructure * structure, const gchar * field,
    gint def)
{
  const gchar *str;
  gint res;

  if ((str = gst_structure_get_string (structure, field)))
    return atoi (str);

  if (gst_structure_get_int (structure, field, &res))
    return res;

  return def;
}

static gboolean
Wim Taymans's avatar
Wim Taymans committed
120
gst_rtp_g722_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
121 122 123 124 125 126 127
{
  GstStructure *structure;
  GstRtpG722Depay *rtpg722depay;
  gint clock_rate, payload, samplerate;
  gint channels;
  GstCaps *srccaps;
  gboolean res;
128
#if 0
129 130
  const gchar *channel_order;
  const GstRTPChannelOrder *order;
131
#endif
132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181

  rtpg722depay = GST_RTP_G722_DEPAY (depayload);

  structure = gst_caps_get_structure (caps, 0);

  payload = 96;
  gst_structure_get_int (structure, "payload", &payload);
  switch (payload) {
    case GST_RTP_PAYLOAD_G722:
      channels = 1;
      clock_rate = 8000;
      samplerate = 16000;
      break;
    default:
      /* no fixed mapping, we need clock-rate */
      channels = 0;
      clock_rate = 0;
      samplerate = 0;
      break;
  }

  /* caps can overwrite defaults */
  clock_rate =
      gst_rtp_g722_depay_parse_int (structure, "clock-rate", clock_rate);
  if (clock_rate == 0)
    goto no_clockrate;

  if (clock_rate == 8000)
    samplerate = 16000;

  if (samplerate == 0)
    samplerate = clock_rate;

  channels =
      gst_rtp_g722_depay_parse_int (structure, "encoding-params", channels);
  if (channels == 0) {
    channels = gst_rtp_g722_depay_parse_int (structure, "channels", channels);
    if (channels == 0) {
      /* channels defaults to 1 otherwise */
      channels = 1;
    }
  }

  depayload->clock_rate = clock_rate;
  rtpg722depay->rate = samplerate;
  rtpg722depay->channels = channels;

  srccaps = gst_caps_new_simple ("audio/G722",
      "rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL);

182 183
  /* FIXME: Do something with the channel order */
#if 0
184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201
  /* add channel positions */
  channel_order = gst_structure_get_string (structure, "channel-order");

  order = gst_rtp_channels_get_by_order (channels, channel_order);
  if (order) {
    gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0),
        order->pos);
  } else {
    GstAudioChannelPosition *pos;

    GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
        (NULL), ("Unknown channel order '%s' for %d channels",
            GST_STR_NULL (channel_order), channels));
    /* create default NONE layout */
    pos = gst_rtp_channels_create_default (channels);
    gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos);
    g_free (pos);
  }
202
#endif
203 204 205 206 207 208 209 210 211 212 213 214 215 216 217

  res = gst_pad_set_caps (depayload->srcpad, srccaps);
  gst_caps_unref (srccaps);

  return res;

  /* ERRORS */
no_clockrate:
  {
    GST_ERROR_OBJECT (depayload, "no clock-rate specified");
    return FALSE;
  }
}

static GstBuffer *
218
gst_rtp_g722_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
219 220 221 222 223 224 225 226
{
  GstRtpG722Depay *rtpg722depay;
  GstBuffer *outbuf;
  gint payload_len;
  gboolean marker;

  rtpg722depay = GST_RTP_G722_DEPAY (depayload);

227
  payload_len = gst_rtp_buffer_get_payload_len (rtp);
228 229 230 231 232 233

  if (payload_len <= 0)
    goto empty_packet;

  GST_DEBUG_OBJECT (rtpg722depay, "got payload of %d bytes", payload_len);

234 235
  outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
  marker = gst_rtp_buffer_get_marker (rtp);
236

237
  if (marker && outbuf) {
238 239
    /* mark talk spurt with RESYNC */
    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
240 241
  }

242 243 244 245 246
  if (outbuf) {
    gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpg722depay), outbuf,
        g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
  }

247 248 249 250 251 252 253 254 255 256 257 258 259 260 261
  return outbuf;

  /* ERRORS */
empty_packet:
  {
    GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
        ("Empty Payload."), (NULL));
    return NULL;
  }
}

gboolean
gst_rtp_g722_depay_plugin_init (GstPlugin * plugin)
{
  return gst_element_register (plugin, "rtpg722depay",
262
      GST_RANK_SECONDARY, GST_TYPE_RTP_G722_DEPAY);
263
}