Ejemplo n.º 1
0
static void
gst_rtp_qcelp_depay_init (GstRtpQCELPDepay * rtpqcelpdepay,
    GstRtpQCELPDepayClass * klass)
{
  GstBaseRTPDepayload *depayload;

  depayload = GST_BASE_RTP_DEPAYLOAD (rtpqcelpdepay);
}
static gboolean
gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps)
{
  GstBaseRTPDepayload *filter;
  GstBaseRTPDepayloadClass *bclass;
  GstBaseRTPDepayloadPrivate *priv;
  gboolean res;
  GstStructure *caps_struct;
  const GValue *value;

  filter = GST_BASE_RTP_DEPAYLOAD (gst_pad_get_parent (pad));
  priv = filter->priv;

  bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);

  GST_DEBUG_OBJECT (filter, "Set caps");

  caps_struct = gst_caps_get_structure (caps, 0);

  /* get other values for newsegment */
  value = gst_structure_get_value (caps_struct, "npt-start");
  if (value && G_VALUE_HOLDS_UINT64 (value))
    priv->npt_start = g_value_get_uint64 (value);
  else
    priv->npt_start = 0;
  GST_DEBUG_OBJECT (filter, "NPT start %" G_GUINT64_FORMAT, priv->npt_start);

  value = gst_structure_get_value (caps_struct, "npt-stop");
  if (value && G_VALUE_HOLDS_UINT64 (value))
    priv->npt_stop = g_value_get_uint64 (value);
  else
    priv->npt_stop = -1;

  GST_DEBUG_OBJECT (filter, "NPT stop %" G_GUINT64_FORMAT, priv->npt_stop);

  value = gst_structure_get_value (caps_struct, "play-speed");
  if (value && G_VALUE_HOLDS_DOUBLE (value))
    priv->play_speed = g_value_get_double (value);
  else
    priv->play_speed = 1.0;

  value = gst_structure_get_value (caps_struct, "play-scale");
  if (value && G_VALUE_HOLDS_DOUBLE (value))
    priv->play_scale = g_value_get_double (value);
  else
    priv->play_scale = 1.0;

  if (bclass->set_caps)
    res = bclass->set_caps (filter, caps);
  else
    res = TRUE;

  priv->negotiated = res;

  gst_object_unref (filter);

  return res;
}
static void
gst_base_rtp_depayload_finalize (GObject * object)
{
  GstBaseRTPDepayload *filter = GST_BASE_RTP_DEPAYLOAD (object);

  g_queue_free (filter->queue);

  G_OBJECT_CLASS (parent_class)->finalize (object);
}
Ejemplo n.º 4
0
static void
gst_rtp_amr_depay_init (GstRtpAMRDepay * rtpamrdepay,
    GstRtpAMRDepayClass * klass)
{
  GstBaseRTPDepayload *depayload;

  depayload = GST_BASE_RTP_DEPAYLOAD (rtpamrdepay);

  gst_pad_use_fixed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
}
Ejemplo n.º 5
0
static void
gst_dtmf_src_generate_tone (GstRtpDTMFDepay * rtpdtmfdepay,
    GstRTPDTMFPayload payload, GstBuffer * buffer)
{
  gint16 *p;
  gint tone_size;
  double i = 0;
  double amplitude, f1, f2;
  double volume_factor;
  DTMF_KEY key = DTMF_KEYS[payload.event];
  guint32 clock_rate = 8000 /* default */ ;
  GstBaseRTPDepayload *depayload = GST_BASE_RTP_DEPAYLOAD (rtpdtmfdepay);
  gint volume;

  clock_rate = depayload->clock_rate;

  /* Create a buffer for the tone */
  tone_size = (payload.duration * SAMPLE_SIZE * CHANNELS) / 8;
  GST_BUFFER_SIZE (buffer) = tone_size;
  GST_BUFFER_MALLOCDATA (buffer) = g_malloc (tone_size);
  GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
  GST_BUFFER_DURATION (buffer) = payload.duration * GST_SECOND / clock_rate;
  volume = payload.volume;

  p = (gint16 *) GST_BUFFER_MALLOCDATA (buffer);

  volume_factor = pow (10, (-volume) / 20);

  /*
   * For each sample point we calculate 'x' as the
   * the amplitude value.
   */
  for (i = 0; i < (tone_size / (SAMPLE_SIZE / 8)); i++) {
    /*
     * We add the fundamental frequencies together.
     */
    f1 = sin (2 * M_PI * key.low_frequency * (rtpdtmfdepay->sample /
            clock_rate));
    f2 = sin (2 * M_PI * key.high_frequency * (rtpdtmfdepay->sample /
            clock_rate));

    amplitude = (f1 + f2) / 2;

    /* Adjust the volume */
    amplitude *= volume_factor;

    /* Make the [-1:1] interval into a [-32767:32767] interval */
    amplitude *= 32767;

    /* Store it in the data buffer */
    *(p++) = (gint16) amplitude;

    (rtpdtmfdepay->sample)++;
  }
}
Ejemplo n.º 6
0
static void
gst_rtp_ilbc_depay_init (GstRTPiLBCDepay * rtpilbcdepay,
    GstRTPiLBCDepayClass * klass)
{
  GstBaseRTPDepayload *depayload;

  depayload = GST_BASE_RTP_DEPAYLOAD (rtpilbcdepay);

  /* Set default mode */
  rtpilbcdepay->mode = DEFAULT_MODE;
}
static void
gst_base_rtp_depayload_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec)
{
  GstBaseRTPDepayload *filter;

  filter = GST_BASE_RTP_DEPAYLOAD (object);

  switch (prop_id) {
    case PROP_QUEUE_DELAY:
      g_value_set_uint (value, filter->queue_delay);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}
static GstStateChangeReturn
gst_base_rtp_depayload_change_state (GstElement * element,
    GstStateChange transition)
{
  GstBaseRTPDepayload *filter;
  GstBaseRTPDepayloadPrivate *priv;
  GstStateChangeReturn ret;

  filter = GST_BASE_RTP_DEPAYLOAD (element);
  priv = filter->priv;

  switch (transition) {
    case GST_STATE_CHANGE_NULL_TO_READY:
      break;
    case GST_STATE_CHANGE_READY_TO_PAUSED:
      filter->need_newsegment = TRUE;
      priv->npt_start = 0;
      priv->npt_stop = -1;
      priv->play_speed = 1.0;
      priv->play_scale = 1.0;
      priv->next_seqnum = -1;
      priv->negotiated = FALSE;
      break;
    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
      break;
    default:
      break;
  }

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

  switch (transition) {
    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
      break;
    case GST_STATE_CHANGE_PAUSED_TO_READY:
      break;
    case GST_STATE_CHANGE_READY_TO_NULL:
      break;
    default:
      break;
  }
  return ret;
}
Ejemplo n.º 9
0
static void
flush_packets (GstRtpQCELPDepay * depay)
{
  guint i, size;

  GST_DEBUG_OBJECT (depay, "flushing packets");

  size = depay->packets->len;

  for (i = 0; i < size; i++) {
    GstBuffer *outbuf;

    outbuf = g_ptr_array_index (depay->packets, i);
    g_ptr_array_index (depay->packets, i) = NULL;

    gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (depay), outbuf);
  }

  /* and reset interleaving state */
  depay->interleaved = FALSE;
  depay->bundling = 0;
}
static gboolean
gst_base_rtp_depayload_handle_sink_event (GstPad * pad, GstEvent * event)
{
  GstBaseRTPDepayload *filter;
  gboolean res = TRUE;

  filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_FLUSH_STOP:
      res = gst_pad_push_event (filter->srcpad, event);

      gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
      filter->need_newsegment = TRUE;
      filter->priv->next_seqnum = -1;
      break;
    case GST_EVENT_NEWSEGMENT:
    {
      gboolean update;
      gdouble rate;
      GstFormat fmt;
      gint64 start, stop, position;

      gst_event_parse_new_segment (event, &update, &rate, &fmt, &start, &stop,
          &position);

      gst_segment_set_newsegment (&filter->segment, update, rate, fmt,
          start, stop, position);

      /* don't pass the event downstream, we generate our own segment including
       * the NTP time and other things we receive in caps */
      gst_event_unref (event);
      break;
    }
    case GST_EVENT_CUSTOM_DOWNSTREAM:
    {
      GstBaseRTPDepayloadClass *bclass;

      bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);

      if (gst_event_has_name (event, "GstRTPPacketLost")) {
        /* we get this event from the jitterbuffer when it considers a packet as
         * being lost. We send it to our packet_lost vmethod. The default
         * implementation will make time progress by pushing out a NEWSEGMENT
         * update event. Subclasses can override and to one of the following:
         *  - Adjust timestamp/duration to something more accurate before
         *    calling the parent (default) packet_lost method.
         *  - do some more advanced error concealing on the already received
         *    (fragmented) packets.
         *  - ignore the packet lost.
         */
        if (bclass->packet_lost)
          res = bclass->packet_lost (filter, event);
      }
      gst_event_unref (event);
      break;
    }
    default:
      /* pass other events forward */
      res = gst_pad_push_event (filter->srcpad, event);
      break;
  }
  return res;
}
static GstFlowReturn
gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in)
{
  GstBaseRTPDepayload *filter;
  GstBaseRTPDepayloadPrivate *priv;
  GstBaseRTPDepayloadClass *bclass;
  GstFlowReturn ret = GST_FLOW_OK;
  GstBuffer *out_buf;
  GstClockTime timestamp;
  guint16 seqnum;
  guint32 rtptime;
  gboolean reset_seq, discont;
  gint gap;

  filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
  priv = filter->priv;

  /* we must have a setcaps first */
  if (G_UNLIKELY (!priv->negotiated))
    goto not_negotiated;

  /* we must validate, it's possible that this element is plugged right after a
   * network receiver and we don't want to operate on invalid data */
  if (G_UNLIKELY (!gst_rtp_buffer_validate (in)))
    goto invalid_buffer;

  priv->discont = GST_BUFFER_IS_DISCONT (in);

  timestamp = GST_BUFFER_TIMESTAMP (in);
  /* convert to running_time and save the timestamp, this is the timestamp
   * we put on outgoing buffers. */
  timestamp = gst_segment_to_running_time (&filter->segment, GST_FORMAT_TIME,
      timestamp);
  priv->timestamp = timestamp;
  priv->duration = GST_BUFFER_DURATION (in);

  seqnum = gst_rtp_buffer_get_seq (in);
  rtptime = gst_rtp_buffer_get_timestamp (in);
  reset_seq = TRUE;
  discont = FALSE;

  GST_LOG_OBJECT (filter, "discont %d, seqnum %u, rtptime %u, timestamp %"
      GST_TIME_FORMAT, priv->discont, seqnum, rtptime,
      GST_TIME_ARGS (timestamp));

  /* Check seqnum. This is a very simple check that makes sure that the seqnums
   * are striclty increasing, dropping anything that is out of the ordinary. We
   * can only do this when the next_seqnum is known. */
  if (G_LIKELY (priv->next_seqnum != -1)) {
    gap = gst_rtp_buffer_compare_seqnum (seqnum, priv->next_seqnum);

    /* if we have no gap, all is fine */
    if (G_UNLIKELY (gap != 0)) {
      GST_LOG_OBJECT (filter, "got packet %u, expected %u, gap %d", seqnum,
          priv->next_seqnum, gap);
      if (gap < 0) {
        /* seqnum > next_seqnum, we are missing some packets, this is always a
         * DISCONT. */
        GST_LOG_OBJECT (filter, "%d missing packets", gap);
        discont = TRUE;
      } else {
        /* seqnum < next_seqnum, we have seen this packet before or the sender
         * could be restarted. If the packet is not too old, we throw it away as
         * a duplicate, otherwise we mark discont and continue. 100 misordered
         * packets is a good threshold. See also RFC 4737. */
        if (gap < 100)
          goto dropping;

        GST_LOG_OBJECT (filter,
            "%d > 100, packet too old, sender likely restarted", gap);
        discont = TRUE;
      }
    }
  }
  priv->next_seqnum = (seqnum + 1) & 0xffff;

  if (G_UNLIKELY (discont && !priv->discont)) {
    GST_LOG_OBJECT (filter, "mark DISCONT on input buffer");
    /* we detected a seqnum discont but the buffer was not flagged with a discont,
     * set the discont flag so that the subclass can throw away old data. */
    priv->discont = TRUE;
    GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT);
  }

  bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);

  if (G_UNLIKELY (bclass->process == NULL))
    goto no_process;

  /* let's send it out to processing */
  out_buf = bclass->process (filter, in);
  if (out_buf) {
    /* we pass rtptime as backward compatibility, in reality, the incomming
     * buffer timestamp is always applied to the outgoing packet. */
    ret = gst_base_rtp_depayload_push_ts (filter, rtptime, out_buf);
  }
  gst_buffer_unref (in);

  return ret;

  /* ERRORS */
not_negotiated:
  {
    /* this is not fatal but should be filtered earlier */
    GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION, (NULL),
        ("Not RTP format was negotiated"));
    gst_buffer_unref (in);
    return GST_FLOW_NOT_NEGOTIATED;
  }
invalid_buffer:
  {
    /* this is not fatal but should be filtered earlier */
    GST_ELEMENT_WARNING (filter, STREAM, DECODE, (NULL),
        ("Received invalid RTP payload, dropping"));
    gst_buffer_unref (in);
    return GST_FLOW_OK;
  }
dropping:
  {
    GST_WARNING_OBJECT (filter, "%d <= 100, dropping old packet", gap);
    gst_buffer_unref (in);
    return GST_FLOW_OK;
  }
no_process:
  {
    /* this is not fatal but should be filtered earlier */
    GST_ELEMENT_ERROR (filter, STREAM, NOT_IMPLEMENTED, (NULL),
        ("The subclass does not have a process method"));
    gst_buffer_unref (in);
    return GST_FLOW_ERROR;
  }
}
Ejemplo n.º 12
0
static gboolean
gst_rtp_celt_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
  GstStructure *structure;
  GstRtpCELTDepay *rtpceltdepay;
  gint clock_rate, nb_channels, frame_size;
  GstBuffer *buf;
  guint8 *data;
  const gchar *params;
  GstCaps *srccaps;
  gboolean res;

  rtpceltdepay = GST_RTP_CELT_DEPAY (depayload);

  structure = gst_caps_get_structure (caps, 0);

  if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
    goto no_clockrate;
  depayload->clock_rate = clock_rate;

  if (!(params = gst_structure_get_string (structure, "encoding-params")))
    nb_channels = 1;
  else {
    nb_channels = atoi (params);
  }

  if (!(params = gst_structure_get_string (structure, "frame-size")))
    frame_size = 480;
  else {
    frame_size = atoi (params);
  }

  /* construct minimal header and comment packet for the decoder */
  buf = gst_buffer_new_and_alloc (60);
  data = GST_BUFFER_DATA (buf);
  memcpy (data, "CELT    ", 8);
  data += 8;
  memcpy (data, "1.1.12", 7);
  data += 20;
  GST_WRITE_UINT32_LE (data, 0x80000006);       /* version */
  data += 4;
  GST_WRITE_UINT32_LE (data, 56);       /* header_size */
  data += 4;
  GST_WRITE_UINT32_LE (data, clock_rate);       /* rate */
  data += 4;
  GST_WRITE_UINT32_LE (data, nb_channels);      /* channels */
  data += 4;
  GST_WRITE_UINT32_LE (data, frame_size);       /* frame-size */
  data += 4;
  GST_WRITE_UINT32_LE (data, -1);       /* overlap */
  data += 4;
  GST_WRITE_UINT32_LE (data, -1);       /* bytes_per_packet */
  data += 4;
  GST_WRITE_UINT32_LE (data, 0);        /* extra headers */

  srccaps = gst_caps_new_simple ("audio/x-celt", NULL);
  res = gst_pad_set_caps (depayload->srcpad, srccaps);
  gst_caps_unref (srccaps);

  gst_buffer_set_caps (buf, GST_PAD_CAPS (depayload->srcpad));
  gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpceltdepay), buf);

  buf = gst_buffer_new_and_alloc (sizeof (gst_rtp_celt_comment));
  memcpy (GST_BUFFER_DATA (buf), gst_rtp_celt_comment,
      sizeof (gst_rtp_celt_comment));

  gst_buffer_set_caps (buf, GST_PAD_CAPS (depayload->srcpad));
  gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpceltdepay), buf);

  return res;

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