gstrtpsbcdepay.c 8.34 KB
Newer Older
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27
/*
 * GStreamer RTP SBC depayloader
 *
 * Copyright (C) 2012  Collabora Ltd.
 *   @author: Arun Raghavan <arun.raghavan@collabora.co.uk>
 *
 * 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 <gst/rtp/gstrtpbuffer.h>
28
#include <gst/audio/audio.h>
29
#include "gstrtpsbcdepay.h"
30
#include "gstrtputils.h"
31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63

GST_DEBUG_CATEGORY_STATIC (rtpsbcdepay_debug);
#define GST_CAT_DEFAULT (rtpsbcdepay_debug)

static GstStaticPadTemplate gst_rtp_sbc_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("audio/x-sbc, "
        "rate = (int) { 16000, 32000, 44100, 48000 }, "
        "channels = (int) [ 1, 2 ], "
        "mode = (string) { mono, dual, stereo, joint }, "
        "blocks = (int) { 4, 8, 12, 16 }, "
        "subbands = (int) { 4, 8 }, "
        "allocation-method = (string) { snr, loudness }, "
        "bitpool = (int) [ 2, 64 ]")
    );

static GstStaticPadTemplate gst_rtp_sbc_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("application/x-rtp, "
        "media = (string) audio,"
        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
        "clock-rate = (int) { 16000, 32000, 44100, 48000 },"
        "encoding-name = (string) SBC")
    );

#define gst_rtp_sbc_depay_parent_class parent_class
G_DEFINE_TYPE (GstRtpSbcDepay, gst_rtp_sbc_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);

static void gst_rtp_sbc_depay_finalize (GObject * object);

static gboolean gst_rtp_sbc_depay_setcaps (GstRTPBaseDepayload * base,
    GstCaps * caps);
static GstBuffer *gst_rtp_sbc_depay_process (GstRTPBaseDepayload * base,
64
    GstRTPBuffer * rtp);
65 66 67 68 69 70 71 72 73 74 75 76

static void
gst_rtp_sbc_depay_class_init (GstRtpSbcDepayClass * klass)
{
  GstRTPBaseDepayloadClass *gstbasertpdepayload_class =
      GST_RTP_BASE_DEPAYLOAD_CLASS (klass);
  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);

  gobject_class->finalize = gst_rtp_sbc_depay_finalize;

  gstbasertpdepayload_class->set_caps = gst_rtp_sbc_depay_setcaps;
77
  gstbasertpdepayload_class->process_rtp_packet = gst_rtp_sbc_depay_process;
78

79 80 81 82
  gst_element_class_add_static_pad_template (element_class,
      &gst_rtp_sbc_depay_src_template);
  gst_element_class_add_static_pad_template (element_class,
      &gst_rtp_sbc_depay_sink_template);
83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141

  GST_DEBUG_CATEGORY_INIT (rtpsbcdepay_debug, "rtpsbcdepay", 0,
      "SBC Audio RTP Depayloader");

  gst_element_class_set_static_metadata (element_class,
      "RTP SBC audio depayloader",
      "Codec/Depayloader/Network/RTP",
      "Extracts SBC audio from RTP packets",
      "Arun Raghavan <arun.raghavan@collabora.co.uk>");
}

static void
gst_rtp_sbc_depay_init (GstRtpSbcDepay * rtpsbcdepay)
{
  rtpsbcdepay->adapter = gst_adapter_new ();
}

static void
gst_rtp_sbc_depay_finalize (GObject * object)
{
  GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (object);

  gst_object_unref (depay->adapter);

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

/* FIXME: This duplicates similar functionality rtpsbcpay, but there isn't a
 * simple way to consolidate the two. This is best done by moving the function
 * to the codec-utils library in gst-plugins-base when these elements move to
 * GStreamer. */
static int
gst_rtp_sbc_depay_get_params (GstRtpSbcDepay * depay, const guint8 * data,
    gint size, int *framelen, int *samples)
{
  int blocks, channel_mode, channels, subbands, bitpool;
  int length;

  if (size < 3) {
    /* Not enough data for the header */
    return -1;
  }

  /* Sanity check */
  if (data[0] != 0x9c) {
    GST_WARNING_OBJECT (depay, "Bad packet: couldn't find syncword");
    return -2;
  }

  blocks = (data[1] >> 4) & 0x3;
  blocks = (blocks + 1) * 4;
  channel_mode = (data[1] >> 2) & 0x3;
  channels = channel_mode ? 2 : 1;
  subbands = (data[1] & 0x1);
  subbands = (subbands + 1) * 4;
  bitpool = data[2];

  length = 4 + ((4 * subbands * channels) / 8);

142
  if (channel_mode == 0 || channel_mode == 1) {
143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183
    /* Mono || Dual channel */
    length += ((blocks * channels * bitpool)
        + 4 /* round up */ ) / 8;
  } else {
    /* Stereo || Joint stereo */
    gboolean joint = (channel_mode == 3);

    length += ((joint * subbands) + (blocks * bitpool)
        + 4 /* round up */ ) / 8;
  }

  *framelen = length;
  *samples = blocks * subbands;

  return 0;
}

static gboolean
gst_rtp_sbc_depay_setcaps (GstRTPBaseDepayload * base, GstCaps * caps)
{
  GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (base);
  GstStructure *structure;
  GstCaps *outcaps, *oldcaps;

  structure = gst_caps_get_structure (caps, 0);

  if (!gst_structure_get_int (structure, "clock-rate", &depay->rate))
    goto bad_caps;

  outcaps = gst_caps_new_simple ("audio/x-sbc", "rate", G_TYPE_INT,
      depay->rate, NULL);

  gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (base), outcaps);

  oldcaps = gst_pad_get_current_caps (GST_RTP_BASE_DEPAYLOAD_SINKPAD (base));
  if (oldcaps && !gst_caps_can_intersect (oldcaps, caps)) {
    /* Caps have changed, flush old data */
    gst_adapter_clear (depay->adapter);
  }

  gst_caps_unref (outcaps);
184
  if (oldcaps)
Tim-Philipp Müller's avatar
Tim-Philipp Müller committed
185
    gst_caps_unref (oldcaps);
186 187 188 189

  return TRUE;

bad_caps:
190
  GST_WARNING_OBJECT (depay, "Can't support the caps we got: %"
191 192 193 194 195
      GST_PTR_FORMAT, caps);
  return FALSE;
}

static GstBuffer *
196
gst_rtp_sbc_depay_process (GstRTPBaseDepayload * base, GstRTPBuffer * rtp)
197 198 199 200 201 202 203 204 205
{
  GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (base);
  GstBuffer *data = NULL;

  gboolean fragment, start, last;
  guint8 nframes;
  guint8 *payload;
  guint payload_len;

206
  GST_LOG_OBJECT (depay, "Got %" G_GSIZE_FORMAT " bytes",
207
      gst_buffer_get_size (rtp->buffer));
208

209
  if (gst_rtp_buffer_get_marker (rtp)) {
210 211 212 213 214
    /* Marker isn't supposed to be set */
    GST_WARNING_OBJECT (depay, "Marker bit was set");
    goto bad_packet;
  }

215 216
  payload = gst_rtp_buffer_get_payload (rtp);
  payload_len = gst_rtp_buffer_get_payload_len (rtp);
217 218 219 220 221 222 223 224 225

  fragment = payload[0] & 0x80;
  start = payload[0] & 0x40;
  last = payload[0] & 0x20;
  nframes = payload[0] & 0x0f;

  payload += 1;
  payload_len -= 1;

226
  data = gst_rtp_buffer_get_payload_subbuffer (rtp, 1, -1);
227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247

  if (fragment) {
    /* Got a packet with a fragment */
    GST_LOG_OBJECT (depay, "Got fragment");

    if (start && gst_adapter_available (depay->adapter)) {
      GST_WARNING_OBJECT (depay, "Missing last fragment");
      gst_adapter_clear (depay->adapter);

    } else if (!start && !gst_adapter_available (depay->adapter)) {
      GST_WARNING_OBJECT (depay, "Missing start fragment");
      gst_buffer_unref (data);
      data = NULL;
      goto out;
    }

    gst_adapter_push (depay->adapter, data);

    if (last) {
      data = gst_adapter_take_buffer (depay->adapter,
          gst_adapter_available (depay->adapter));
248
      gst_rtp_drop_non_audio_meta (depay, data);
249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285
    } else
      data = NULL;

  } else {
    /* !fragment */
    gint framelen, samples;

    GST_LOG_OBJECT (depay, "Got %d frames", nframes);

    if (gst_rtp_sbc_depay_get_params (depay, payload,
            payload_len, &framelen, &samples) < 0) {
      gst_adapter_clear (depay->adapter);
      goto bad_packet;
    }

    GST_LOG_OBJECT (depay, "Got payload of %d", payload_len);

    if (nframes * framelen > (gint) payload_len) {
      GST_WARNING_OBJECT (depay, "Short packet");
      goto bad_packet;
    } else if (nframes * framelen < (gint) payload_len) {
      GST_WARNING_OBJECT (depay, "Junk at end of packet");
    }
  }

out:
  return data;

bad_packet:
  GST_ELEMENT_WARNING (depay, STREAM, DECODE,
      ("Received invalid RTP payload, dropping"), (NULL));
  goto out;
}

gboolean
gst_rtp_sbc_depay_plugin_init (GstPlugin * plugin)
{
286
  return gst_element_register (plugin, "rtpsbcdepay", GST_RANK_SECONDARY,
287 288
      GST_TYPE_RTP_SBC_DEPAY);
}