C++ GST_BUFFER_TIMESTAMP函数代码示例

static GstFlowReturn
audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
{
  int outsize;
  int outsamples;
  ResampleState *r;

  r = audioresample->resample;

  outsize = resample_get_output_size (r);
  GST_LOG_OBJECT (audioresample, "audioresample can give me %d bytes", outsize);

  /* protect against mem corruption */
  if (outsize > GST_BUFFER_SIZE (outbuf)) {
    GST_WARNING_OBJECT (audioresample,
        "overriding audioresample's outsize %d with outbuffer's size %d",
        outsize, GST_BUFFER_SIZE (outbuf));
    outsize = GST_BUFFER_SIZE (outbuf);
  }
  /* catch possibly wrong size differences */
  if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) {
    GST_WARNING_OBJECT (audioresample,
        "audioresample's outsize %d too far from outbuffer's size %d",
        outsize, GST_BUFFER_SIZE (outbuf));
  }

  outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize);
  outsamples = outsize / r->sample_size;
  GST_LOG_OBJECT (audioresample, "resample gave me %d bytes or %d samples",
      outsize, outsamples);

  GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
  GST_BUFFER_TIMESTAMP (outbuf) = audioresample->next_ts;

  if (audioresample->ts_offset != -1) {
    audioresample->offset += outsamples;
    audioresample->ts_offset += outsamples;
    audioresample->next_ts =
        gst_util_uint64_scale_int (audioresample->ts_offset, GST_SECOND,
        audioresample->o_rate);
    GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset;

    /* we calculate DURATION as the difference between "next" timestamp
     * and current timestamp so we ensure a contiguous stream, instead of
     * having rounding errors. */
    GST_BUFFER_DURATION (outbuf) = audioresample->next_ts -
        GST_BUFFER_TIMESTAMP (outbuf);
  } else {
    /* no valid offset know, we can still sortof calculate the duration though */
    GST_BUFFER_DURATION (outbuf) =
        gst_util_uint64_scale_int (outsamples, GST_SECOND,
        audioresample->o_rate);
  }

  /* check for possible mem corruption */
  if (outsize > GST_BUFFER_SIZE (outbuf)) {
    /* this is an error that when it happens, would need fixing in the
     * resample library; we told it we wanted only GST_BUFFER_SIZE (outbuf),
     * and it gave us more ! */
    GST_WARNING_OBJECT (audioresample,
        "audioresample, you memory corrupting bastard. "
        "you gave me outsize %d while my buffer was size %d",
        outsize, GST_BUFFER_SIZE (outbuf));
    return GST_FLOW_ERROR;
  }
  /* catch possibly wrong size differences */
  if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) {
    GST_WARNING_OBJECT (audioresample,
        "audioresample's written outsize %d too far from outbuffer's size %d",
        outsize, GST_BUFFER_SIZE (outbuf));
  }
  GST_BUFFER_SIZE (outbuf) = outsize;

  if (G_UNLIKELY (audioresample->need_discont)) {
    GST_DEBUG_OBJECT (audioresample,
        "marking this buffer with the DISCONT flag");
    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
    audioresample->need_discont = FALSE;
  }

  GST_LOG_OBJECT (audioresample, "transformed to buffer of %d bytes, ts %"
      GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
      G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
      outsize, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
      GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
      GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));


  return GST_FLOW_OK;
}

static GstFlowReturn
gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
    GstBuffer * inbuf, GstBuffer * outbuf)
{
  GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
  GstClockTime timestamp, expected_timestamp;
  gint channels = GST_AUDIO_FILTER_CAST (self)->format.channels;
  gint rate = GST_AUDIO_FILTER_CAST (self)->format.rate;
  gint width = GST_AUDIO_FILTER_CAST (self)->format.width / 8;
  guint input_samples = (GST_BUFFER_SIZE (inbuf) / width) / channels;
  guint output_samples = (GST_BUFFER_SIZE (outbuf) / width) / channels;
  guint generated_samples;
  guint64 output_offset;
  gint64 diff = 0;
  GstClockTime stream_time;

  timestamp = GST_BUFFER_TIMESTAMP (outbuf);

  if (!GST_CLOCK_TIME_IS_VALID (timestamp)
      && !GST_CLOCK_TIME_IS_VALID (self->start_ts)) {
    GST_ERROR_OBJECT (self, "Invalid timestamp");
    return GST_FLOW_ERROR;
  }

  stream_time =
      gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);

  GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,
      GST_TIME_ARGS (timestamp));

  if (GST_CLOCK_TIME_IS_VALID (stream_time))
    gst_object_sync_values (G_OBJECT (self), stream_time);

  g_return_val_if_fail (self->kernel != NULL, GST_FLOW_ERROR);
  g_return_val_if_fail (channels != 0, GST_FLOW_ERROR);

  if (GST_CLOCK_TIME_IS_VALID (self->start_ts))
    expected_timestamp =
        self->start_ts + gst_util_uint64_scale_int (self->nsamples_in,
        GST_SECOND, rate);
  else
    expected_timestamp = GST_CLOCK_TIME_NONE;

  /* Reset the residue if already existing on discont buffers */
  if (GST_BUFFER_IS_DISCONT (inbuf)
      || (GST_CLOCK_TIME_IS_VALID (expected_timestamp)
          && (ABS (GST_CLOCK_DIFF (timestamp,
                      expected_timestamp) > 5 * GST_MSECOND)))) {
    GST_DEBUG_OBJECT (self, "Discontinuity detected - flushing");
    if (GST_CLOCK_TIME_IS_VALID (expected_timestamp))
      gst_audio_fx_base_fir_filter_push_residue (self);
    self->buffer_fill = 0;
    g_free (self->buffer);
    self->buffer = NULL;
    self->start_ts = timestamp;
    self->start_off = GST_BUFFER_OFFSET (inbuf);
    self->nsamples_out = 0;
    self->nsamples_in = 0;
  } else if (!GST_CLOCK_TIME_IS_VALID (self->start_ts)) {
    self->start_ts = timestamp;
    self->start_off = GST_BUFFER_OFFSET (inbuf);
  }

  self->nsamples_in += input_samples;

  generated_samples =
      self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf),
      input_samples);

  g_assert (generated_samples <= output_samples);
  self->nsamples_out += generated_samples;
  if (generated_samples == 0)
    return GST_BASE_TRANSFORM_FLOW_DROPPED;

  /* Calculate the number of samples we can push out now without outputting
   * latency zeros in the beginning */
  diff = ((gint64) self->nsamples_out) - ((gint64) self->latency);
  if (diff < 0) {
    return GST_BASE_TRANSFORM_FLOW_DROPPED;
  } else if (diff < generated_samples) {
    gint64 tmp = diff;
    diff = generated_samples - diff;
    generated_samples = tmp;
    GST_BUFFER_DATA (outbuf) += diff * width * channels;
  }
  GST_BUFFER_SIZE (outbuf) = generated_samples * width * channels;

  output_offset = self->nsamples_out - self->latency - generated_samples;
  GST_BUFFER_TIMESTAMP (outbuf) =
      self->start_ts + gst_util_uint64_scale_int (output_offset, GST_SECOND,
      rate);
  GST_BUFFER_DURATION (outbuf) =
      gst_util_uint64_scale_int (output_samples, GST_SECOND, rate);
  if (self->start_off != GST_BUFFER_OFFSET_NONE) {
    GST_BUFFER_OFFSET (outbuf) = self->start_off + output_offset;
    GST_BUFFER_OFFSET_END (outbuf) =
        GST_BUFFER_OFFSET (outbuf) + generated_samples;
  } else {
    GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
    GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
//.........这里部分代码省略.........

static GstFlowReturn
dvdspu_handle_vid_buffer (GstDVDSpu * dvdspu, GstBuffer * buf)
{
  GstClockTime new_ts;
  GstFlowReturn ret;
  gboolean using_ref = FALSE;

  DVD_SPU_LOCK (dvdspu);

  if (buf == NULL) {
    GstClockTime next_ts = dvdspu->video_seg.position;

    next_ts += gst_util_uint64_scale_int (GST_SECOND,
        dvdspu->spu_state.info.fps_d, dvdspu->spu_state.info.fps_n);

    /* NULL buffer was passed - use the reference frame and update the timestamp,
     * or else there's nothing to draw, and just return GST_FLOW_OK */
    if (dvdspu->ref_frame == NULL) {
      dvdspu->video_seg.position = next_ts;
      goto no_ref_frame;
    }

    buf = gst_buffer_copy (dvdspu->ref_frame);

#if 0
    g_print ("Duping frame %" GST_TIME_FORMAT " with new TS %" GST_TIME_FORMAT
        "\n", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
        GST_TIME_ARGS (next_ts));
#endif

    GST_BUFFER_TIMESTAMP (buf) = next_ts;
    using_ref = TRUE;
  }

  if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
    dvdspu->video_seg.position = GST_BUFFER_TIMESTAMP (buf);
  }

  new_ts = gst_segment_to_running_time (&dvdspu->video_seg, GST_FORMAT_TIME,
      dvdspu->video_seg.position);

#if 0
  g_print ("TS %" GST_TIME_FORMAT " running: %" GST_TIME_FORMAT "\n",
      GST_TIME_ARGS (dvdspu->video_seg.position), GST_TIME_ARGS (new_ts));
#endif

  gst_dvd_spu_advance_spu (dvdspu, new_ts);

  /* If we have an active SPU command set, we store a copy of the frame in case
   * we hit a still and need to draw on it. Otherwise, a reference is
   * sufficient in case we later encounter a still */
  if ((dvdspu->spu_state.flags & SPU_STATE_FORCED_DSP) ||
      ((dvdspu->spu_state.flags & SPU_STATE_FORCED_ONLY) == 0 &&
          (dvdspu->spu_state.flags & SPU_STATE_DISPLAY))) {
    if (using_ref == FALSE) {
      GstBuffer *copy;

      /* Take a copy in case we hit a still frame and need the pristine 
       * frame around */
      copy = gst_buffer_copy (buf);
      gst_buffer_replace (&dvdspu->ref_frame, copy);
      gst_buffer_unref (copy);
    }

    /* Render the SPU overlay onto the buffer */
    buf = gst_buffer_make_writable (buf);

    gstspu_render (dvdspu, buf);
  } else {
    if (using_ref == FALSE) {
      /* Not going to draw anything on this frame, just store a reference
       * in case we hit a still frame and need it */
      gst_buffer_replace (&dvdspu->ref_frame, buf);
    }
  }

  if (dvdspu->spu_state.flags & SPU_STATE_STILL_FRAME) {
    GST_DEBUG_OBJECT (dvdspu, "Outputting buffer with TS %" GST_TIME_FORMAT
        "from chain while in still",
        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
  }

  DVD_SPU_UNLOCK (dvdspu);

  /* just push out the incoming buffer without touching it */
  ret = gst_pad_push (dvdspu->srcpad, buf);

  return ret;

no_ref_frame:

  DVD_SPU_UNLOCK (dvdspu);

  return GST_FLOW_OK;
}

void test_reuse()
{
  GstElement *audioresample;
  GstEvent *newseg;
  GstBuffer *inbuffer;
  GstCaps *caps;
  xmlfile = "test_reuse";
std_log(LOG_FILENAME_LINE, "Test Started test_reuse");
  audioresample = setup_audioresample (1, 9343, 48000, 16, FALSE);
  caps = gst_pad_get_negotiated_caps (mysrcpad);
  fail_unless (gst_caps_is_fixed (caps));

  fail_unless (gst_element_set_state (audioresample,
          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
      "could not set to playing");

  newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
  fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);

  inbuffer = gst_buffer_new_and_alloc (9343 * 4);
  memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
  GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
  GST_BUFFER_OFFSET (inbuffer) = 0;
  gst_buffer_set_caps (inbuffer, caps);

  /* pushing gives away my reference ... */
  fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);

  /* ... but it ends up being collected on the global buffer list */
  fail_unless_equals_int (g_list_length (buffers), 1);

  /* now reset and try again ... */
  fail_unless (gst_element_set_state (audioresample,
          GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");

  fail_unless (gst_element_set_state (audioresample,
          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
      "could not set to playing");

  newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
  fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);

  inbuffer = gst_buffer_new_and_alloc (9343 * 4);
  memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
  GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
  GST_BUFFER_OFFSET (inbuffer) = 0;
  gst_buffer_set_caps (inbuffer, caps);

  fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);

  /* ... it also ends up being collected on the global buffer list. If we
   * now have more than 2 buffers, then audioresample probably didn't clean
   * up its internal buffer properly and tried to push the remaining samples
   * when it got the second NEWSEGMENT event */
  fail_unless_equals_int (g_list_length (buffers), 2);

  cleanup_audioresample (audioresample);
  gst_caps_unref (caps);
  
  std_log(LOG_FILENAME_LINE, "Test Successful");
  create_xml(0);
}

//.........这里部分代码省略.........

    /* now open the mux format */
    if (avformat_write_header (ffmpegmux->context, NULL) < 0) {
      GST_ELEMENT_ERROR (ffmpegmux, LIBRARY, SETTINGS, (NULL),
          ("Failed to write file header - check codec settings"));
      return GST_FLOW_ERROR;
    }

    /* we're now opened */
    ffmpegmux->opened = TRUE;

    /* flush the header so it will be used as streamheader */
    avio_flush (ffmpegmux->context->pb);
  }

  /* take the one with earliest timestamp,
   * and push it forward */
  best_pad = NULL;
  best_time = GST_CLOCK_TIME_NONE;
  for (collected = ffmpegmux->collect->data; collected;
      collected = g_slist_next (collected)) {
    GstFFMpegMuxPad *collect_pad = (GstFFMpegMuxPad *) collected->data;
    GstBuffer *buffer = gst_collect_pads_peek (ffmpegmux->collect,
        (GstCollectData *) collect_pad);

    /* if there's no buffer, just continue */
    if (buffer == NULL) {
      continue;
    }

    /* if we have no buffer yet, just use the first one */
    if (best_pad == NULL) {
      best_pad = collect_pad;
      best_time = GST_BUFFER_TIMESTAMP (buffer);
      goto next_pad;
    }

    /* if we do have one, only use this one if it's older */
    if (GST_BUFFER_TIMESTAMP (buffer) < best_time) {
      best_time = GST_BUFFER_TIMESTAMP (buffer);
      best_pad = collect_pad;
    }

  next_pad:
    gst_buffer_unref (buffer);

    /* Mux buffers with invalid timestamp first */
    if (!GST_CLOCK_TIME_IS_VALID (best_time))
      break;
  }

  /* now handle the buffer, or signal EOS if we have
   * no buffers left */
  if (best_pad != NULL) {
    GstBuffer *buf;
    AVPacket pkt;
    gboolean need_free = FALSE;
    GstMapInfo map;

    /* push out current buffer */
    buf =
        gst_collect_pads_pop (ffmpegmux->collect, (GstCollectData *) best_pad);

    ffmpegmux->context->streams[best_pad->padnum]->codec->frame_number++;

    /* set time */

static GstFlowReturn
gst_bml_transform_transform_mono_to_stereo (GstBaseTransform * base,
    GstBuffer * inbuf, GstBuffer * outbuf)
{
  GstMapInfo infoi, infoo;
  GstBMLTransform *bml_transform = GST_BML_TRANSFORM (base);
  GstBMLTransformClass *klass = GST_BML_TRANSFORM_GET_CLASS (bml_transform);
  GstBML *bml = GST_BML (bml_transform);
  GstBMLClass *bml_class = GST_BML_CLASS (klass);
  BMLData *datai, *datao, *seg_datai, *seg_datao;
  gpointer bm = bml->bm;
  guint todo, seg_size, samples_per_buffer;
  gboolean has_data;
  guint mode = 3;               /*WM_READWRITE */

  bml->running_time =
      gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME,
      GST_BUFFER_TIMESTAMP (inbuf));

  if (GST_BUFFER_FLAG_IS_SET (outbuf, GST_BUFFER_FLAG_DISCONT)) {
    bml->subtick_count = (!bml->reverse) ? bml->subticks_per_tick : 1;
  }

  if (bml->subtick_count >= bml->subticks_per_tick) {
    bml (gstbml_reset_triggers (bml, bml_class));
    bml (gstbml_sync_values (bml, bml_class, GST_BUFFER_TIMESTAMP (outbuf)));
    bml (tick (bm));
    bml->subtick_count = 1;
  } else {
    bml->subtick_count++;
  }

  /* don't process data in passthrough-mode */
  if (gst_base_transform_is_passthrough (base)) {
    // we would actually need to convert mono to stereo here
    // but this is not even called
    GST_WARNING_OBJECT (bml_transform, "m2s in passthrough mode");
    //return GST_FLOW_OK;
  }

  if (!gst_buffer_map (inbuf, &infoi, GST_MAP_READ)) {
    GST_WARNING_OBJECT (base, "unable to map input buffer for read");
    return GST_FLOW_ERROR;
  }
  datai = (BMLData *) infoi.data;
  samples_per_buffer = infoi.size / sizeof (BMLData);
  if (!gst_buffer_map (outbuf, &infoo, GST_MAP_READ | GST_MAP_WRITE)) {
    GST_WARNING_OBJECT (base, "unable to map output buffer for read & write");
    return GST_FLOW_ERROR;
  }
  datao = (BMLData *) infoo.data;

  // some buzzmachines expect a cleared buffer
  //for(i=0;i<samples_per_buffer*2;i++) datao[i]=0.0f;
  memset (datao, 0, samples_per_buffer * 2 * sizeof (BMLData));

  /* if buffer has only silence process with different mode */
  if (GST_BUFFER_FLAG_IS_SET (outbuf, GST_BUFFER_FLAG_GAP)) {
    mode = 2;                   /* WM_WRITE */
  } else {
    gfloat fc = 32768.0;
    orc_scalarmultiply_f32_ns (datai, datai, fc, samples_per_buffer);
  }

  GST_DEBUG_OBJECT (bml_transform, "  calling work_m2s(%d,%d)",
      samples_per_buffer, mode);
  todo = samples_per_buffer;
  seg_datai = datai;
  seg_datao = datao;
  has_data = FALSE;
  while (todo) {
    // 256 is MachineInterface.h::MAX_BUFFER_LENGTH
    seg_size = (todo > 256) ? 256 : todo;
    has_data |= bml (work_m2s (bm, seg_datai, seg_datao, (int) seg_size, mode));
    seg_datai = &seg_datai[seg_size];
    seg_datao = &seg_datao[seg_size * 2];
    todo -= seg_size;
  }
  if (gstbml_fix_data ((GstElement *) bml_transform, &infoo, has_data)) {
    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
  } else {
    GST_BUFFER_FLAG_UNSET (outbuf, GST_BUFFER_FLAG_GAP);
  }

  gst_buffer_unmap (inbuf, &infoi);
  gst_buffer_unmap (outbuf, &infoo);
  return GST_FLOW_OK;
}

GstPadProbeReturn GstEnginePipeline::HandoffCallback(GstPad*,
                                                     GstPadProbeInfo* info,
                                                     gpointer self) {
  GstEnginePipeline* instance = reinterpret_cast<GstEnginePipeline*>(self);
  GstBuffer* buf = gst_pad_probe_info_get_buffer(info);

  QList<BufferConsumer*> consumers;
  {
    QMutexLocker l(&instance->buffer_consumers_mutex_);
    consumers = instance->buffer_consumers_;
  }

  for (BufferConsumer* consumer : consumers) {
    gst_buffer_ref(buf);
    consumer->ConsumeBuffer(buf, instance->id());
  }

  // Calculate the end time of this buffer so we can stop playback if it's
  // after the end time of this song.
  if (instance->end_offset_nanosec_ > 0) {
    quint64 start_time = GST_BUFFER_TIMESTAMP(buf) - instance->segment_start_;
    quint64 duration = GST_BUFFER_DURATION(buf);
    quint64 end_time = start_time + duration;

    if (end_time > instance->end_offset_nanosec_) {
      if (instance->has_next_valid_url()) {
        if (instance->next_url_ == instance->url_ &&
            instance->next_beginning_offset_nanosec_ ==
                instance->end_offset_nanosec_) {
          // The "next" song is actually the next segment of this file - so
          // cheat and keep on playing, but just tell the Engine we've moved on.
          instance->end_offset_nanosec_ = instance->next_end_offset_nanosec_;
          instance->next_url_ = QUrl();
          instance->next_beginning_offset_nanosec_ = 0;
          instance->next_end_offset_nanosec_ = 0;

          // GstEngine will try to seek to the start of the new section, but
          // we're already there so ignore it.
          instance->ignore_next_seek_ = true;
          emit instance->EndOfStreamReached(instance->id(), true);
        } else {
          // We have a next song but we can't cheat, so move to it normally.
          instance->TransitionToNext();
        }
      } else {
        // There's no next song
        emit instance->EndOfStreamReached(instance->id(), false);
      }
    }
  }

  if (instance->emit_track_ended_on_time_discontinuity_) {
    if (GST_BUFFER_FLAG_IS_SET(buf, GST_BUFFER_FLAG_DISCONT) ||
        GST_BUFFER_OFFSET(buf) < instance->last_buffer_offset_) {
      qLog(Debug) << "Buffer discontinuity - emitting EOS";
      instance->emit_track_ended_on_time_discontinuity_ = false;
      emit instance->EndOfStreamReached(instance->id(), true);
    }
  }

  instance->last_buffer_offset_ = GST_BUFFER_OFFSET(buf);

  return GST_PAD_PROBE_OK;
}

static GstBuffer *
gst_rtp_qdm2_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
  GstRtpQDM2Depay *rtpqdm2depay;
  GstBuffer *outbuf = NULL;
  guint16 seq;
  GstRTPBuffer rtp = { NULL };

  rtpqdm2depay = GST_RTP_QDM2_DEPAY (depayload);

  {
    gint payload_len;
    guint8 *payload;
    guint avail;
    guint pos = 0;

    gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
    payload_len = gst_rtp_buffer_get_payload_len (&rtp);
    if (payload_len < 3)
      goto bad_packet;

    payload = gst_rtp_buffer_get_payload (&rtp);
    seq = gst_rtp_buffer_get_seq (&rtp);
    if (G_UNLIKELY (seq != rtpqdm2depay->nextseq)) {
      GST_DEBUG ("GAP in sequence number, Resetting data !");
      /* Flush previous data */
      flush_data (rtpqdm2depay);
      /* And store new timestamp */
      rtpqdm2depay->ptimestamp = rtpqdm2depay->timestamp;
      rtpqdm2depay->timestamp = GST_BUFFER_TIMESTAMP (buf);
      /* And that previous data will be pushed at the bottom */
    }
    rtpqdm2depay->nextseq = seq + 1;

    GST_DEBUG ("Payload size %d 0x%x sequence:%d", payload_len, payload_len,
        seq);

    GST_MEMDUMP ("Incoming payload", payload, payload_len);

    while (pos < payload_len) {
      switch (payload[pos]) {
        case 0x80:{
          GST_DEBUG ("Unrecognized 0x80 marker, skipping 12 bytes");
          pos += 12;
        }
          break;
        case 0xff:
          /* HEADERS */
          GST_DEBUG ("Headers");
          /* Store the incoming timestamp */
          rtpqdm2depay->ptimestamp = rtpqdm2depay->timestamp;
          rtpqdm2depay->timestamp = GST_BUFFER_TIMESTAMP (buf);
          /* flush the internal data if needed */
          flush_data (rtpqdm2depay);
          if (G_UNLIKELY (!rtpqdm2depay->configured)) {
            guint8 *ourdata;
            GstBuffer *codecdata;
            GstMapInfo cmap;
            GstCaps *caps;

            /* First bytes are unknown */
            GST_MEMDUMP ("Header", payload + pos, 32);
            ourdata = payload + pos + 10;
            pos += 10;
            rtpqdm2depay->channs = GST_READ_UINT32_BE (payload + pos + 4);
            rtpqdm2depay->samplerate = GST_READ_UINT32_BE (payload + pos + 8);
            rtpqdm2depay->bitrate = GST_READ_UINT32_BE (payload + pos + 12);
            rtpqdm2depay->blocksize = GST_READ_UINT32_BE (payload + pos + 16);
            rtpqdm2depay->framesize = GST_READ_UINT32_BE (payload + pos + 20);
            rtpqdm2depay->packetsize = GST_READ_UINT32_BE (payload + pos + 24);
            /* 16 bit empty block (0x02 0x00) */
            pos += 30;
            GST_DEBUG
                ("channs:%d, samplerate:%d, bitrate:%d, blocksize:%d, framesize:%d, packetsize:%d",
                rtpqdm2depay->channs, rtpqdm2depay->samplerate,
                rtpqdm2depay->bitrate, rtpqdm2depay->blocksize,
                rtpqdm2depay->framesize, rtpqdm2depay->packetsize);

            /* Caps */
            codecdata = gst_buffer_new_and_alloc (48);
            gst_buffer_map (codecdata, &cmap, GST_MAP_WRITE);
            memcpy (cmap.data, headheader, 20);
            memcpy (cmap.data + 20, ourdata, 28);
            gst_buffer_unmap (codecdata, &cmap);

            caps = gst_caps_new_simple ("audio/x-qdm2",
                "samplesize", G_TYPE_INT, 16,
                "rate", G_TYPE_INT, rtpqdm2depay->samplerate,
                "channels", G_TYPE_INT, rtpqdm2depay->channs,
                "codec_data", GST_TYPE_BUFFER, codecdata, NULL);
            gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), caps);
            gst_caps_unref (caps);
            rtpqdm2depay->configured = TRUE;
          } else {
            GST_DEBUG ("Already configured, skipping headers");
            pos += 40;
          }
          break;
        default:{
          /* Shuffled packet contents */
//.........这里部分代码省略.........

static GstBuffer *
gst_rtp_dtmf_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{

  GstRtpDTMFDepay *rtpdtmfdepay = NULL;
  GstBuffer *outbuf = NULL;
  gint payload_len;
  guint8 *payload = NULL;
  guint32 timestamp;
  GstRTPDTMFPayload dtmf_payload;
  gboolean marker;
  GstStructure *structure = NULL;
  GstMessage *dtmf_message = NULL;

  rtpdtmfdepay = GST_RTP_DTMF_DEPAY (depayload);

  if (!gst_rtp_buffer_validate (buf))
    goto bad_packet;

  payload_len = gst_rtp_buffer_get_payload_len (buf);
  payload = gst_rtp_buffer_get_payload (buf);

  if (payload_len != sizeof (GstRTPDTMFPayload))
    goto bad_packet;

  memcpy (&dtmf_payload, payload, sizeof (GstRTPDTMFPayload));

  if (dtmf_payload.event > MAX_EVENT)
    goto bad_packet;


  marker = gst_rtp_buffer_get_marker (buf);

  timestamp = gst_rtp_buffer_get_timestamp (buf);

  dtmf_payload.duration = g_ntohs (dtmf_payload.duration);

  /* clip to whole units of unit_time */
  if (rtpdtmfdepay->unit_time) {
    guint unit_time_clock =
        (rtpdtmfdepay->unit_time * depayload->clock_rate) / 1000;
    if (dtmf_payload.duration % unit_time_clock) {
      /* Make sure we don't overflow the duration */
      if (dtmf_payload.duration < G_MAXUINT16 - unit_time_clock)
        dtmf_payload.duration += unit_time_clock -
            (dtmf_payload.duration % unit_time_clock);
      else
        dtmf_payload.duration -= dtmf_payload.duration % unit_time_clock;
    }
  }

  /* clip to max duration */
  if (rtpdtmfdepay->max_duration) {
    guint max_duration_clock =
        (rtpdtmfdepay->max_duration * depayload->clock_rate) / 1000;

    if (max_duration_clock < G_MAXUINT16 &&
        dtmf_payload.duration > max_duration_clock)
      dtmf_payload.duration = max_duration_clock;
  }

  GST_DEBUG_OBJECT (depayload, "Received new RTP DTMF packet : "
      "marker=%d - timestamp=%u - event=%d - duration=%d",
      marker, timestamp, dtmf_payload.event, dtmf_payload.duration);

  GST_DEBUG_OBJECT (depayload,
      "Previous information : timestamp=%u - duration=%d",
      rtpdtmfdepay->previous_ts, rtpdtmfdepay->previous_duration);

  /* First packet */
  if (marker || rtpdtmfdepay->previous_ts != timestamp) {
    rtpdtmfdepay->sample = 0;
    rtpdtmfdepay->previous_ts = timestamp;
    rtpdtmfdepay->previous_duration = dtmf_payload.duration;
    rtpdtmfdepay->first_gst_ts = GST_BUFFER_TIMESTAMP (buf);

    structure = gst_structure_new ("dtmf-event",
        "number", G_TYPE_INT, dtmf_payload.event,
        "volume", G_TYPE_INT, dtmf_payload.volume,
        "type", G_TYPE_INT, 1, "method", G_TYPE_INT, 1, NULL);
    if (structure) {
      dtmf_message =
          gst_message_new_element (GST_OBJECT (depayload), structure);
      if (dtmf_message) {
        if (!gst_element_post_message (GST_ELEMENT (depayload), dtmf_message)) {
          GST_ERROR_OBJECT (depayload,
              "Unable to send dtmf-event message to bus");
        }
      } else {
        GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event message");
      }
    } else {
      GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event structure");
    }
  } else {
    guint16 duration = dtmf_payload.duration;
    dtmf_payload.duration -= rtpdtmfdepay->previous_duration;
    /* If late buffer, ignore */
    if (duration > rtpdtmfdepay->previous_duration)
      rtpdtmfdepay->previous_duration = duration;
//.........这里部分代码省略.........

static MpegPsPadData *
mpegpsmux_choose_best_stream (MpegPsMux * mux)
{
  /* Choose from which stream to mux with */

  MpegPsPadData *best = NULL;
  GstCollectData *c_best = NULL;
  GSList *walk;

  for (walk = mux->collect->data; walk != NULL; walk = g_slist_next (walk)) {
    GstCollectData *c_data = (GstCollectData *) walk->data;
    MpegPsPadData *ps_data = (MpegPsPadData *) walk->data;

    if (ps_data->eos == FALSE) {
      if (ps_data->queued_buf == NULL) {
        GstBuffer *buf;

        ps_data->queued_buf = buf =
            gst_collect_pads_peek (mux->collect, c_data);

        if (buf != NULL) {
          if (ps_data->prepare_func) {
            buf = ps_data->prepare_func (buf, ps_data, mux);
            if (buf) {          /* Take the prepared buffer instead */
              gst_buffer_unref (ps_data->queued_buf);
              ps_data->queued_buf = buf;
            } else {            /* If data preparation returned NULL, use unprepared one */
              buf = ps_data->queued_buf;
            }
          }
          if (GST_BUFFER_TIMESTAMP (buf) != GST_CLOCK_TIME_NONE) {
            /* Ignore timestamps that go backward for now. FIXME: Handle all
             * incoming PTS */
            if (ps_data->last_ts == GST_CLOCK_TIME_NONE ||
                ps_data->last_ts < GST_BUFFER_TIMESTAMP (buf)) {
              ps_data->cur_ts = ps_data->last_ts =
                  gst_segment_to_running_time (&c_data->segment,
                  GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (buf));
            } else {
              GST_DEBUG_OBJECT (mux, "Ignoring PTS that has gone backward");
            }
          } else
            ps_data->cur_ts = GST_CLOCK_TIME_NONE;

          GST_DEBUG_OBJECT (mux, "Pulled buffer with ts %" GST_TIME_FORMAT
              " (uncorrected ts %" GST_TIME_FORMAT " %" G_GUINT64_FORMAT
              ") for PID 0x%04x",
              GST_TIME_ARGS (ps_data->cur_ts),
              GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
              GST_BUFFER_TIMESTAMP (buf), ps_data->stream_id);

          /* Choose a stream we've never seen a timestamp for to ensure
           * we push enough buffers from it to reach a timestamp */
          if (ps_data->last_ts == GST_CLOCK_TIME_NONE) {
            best = ps_data;
            c_best = c_data;
          }
        } else {
          ps_data->eos = TRUE;
          continue;
        }
      }

      /* If we don't yet have a best pad, take this one, otherwise take
       * whichever has the oldest timestamp */
      if (best != NULL) {
        if (ps_data->last_ts != GST_CLOCK_TIME_NONE &&
            best->last_ts != GST_CLOCK_TIME_NONE &&
            ps_data->last_ts < best->last_ts) {
          best = ps_data;
          c_best = c_data;
        }
      } else {
        best = ps_data;
        c_best = c_data;
      }
    }
  }
  if (c_best) {
    gst_buffer_unref (gst_collect_pads_pop (mux->collect, c_best));
  }

  return best;
}

static GstFlowReturn
gst_vader_chain(GstPad * pad, GstBuffer * buf)
{
    GstVader *filter;
    gint16 *in_data;
    guint num_samples;
    gint i, vote;
    guint power, rms;

    g_return_val_if_fail(pad != NULL, GST_FLOW_ERROR);
    g_return_val_if_fail(GST_IS_PAD(pad), GST_FLOW_ERROR);
    g_return_val_if_fail(buf != NULL, GST_FLOW_ERROR);

    filter = GST_VADER(GST_OBJECT_PARENT(pad));
    g_return_val_if_fail(filter != NULL, GST_FLOW_ERROR);
    g_return_val_if_fail(GST_IS_VADER(filter), GST_FLOW_ERROR);

    in_data = (gint16 *) GST_BUFFER_DATA(buf);
    num_samples = GST_BUFFER_SIZE(buf) / 2;

    /* Enter a critical section. */
    g_static_rec_mutex_lock(&filter->mtx);
    filter->silent_prev = filter->silent;
    /* If we are in auto-threshold mode, check to see if we have
     * enough data to estimate a threshold.  (FIXME: we should be
     * estimating at the sample level rather than the frame level,
     * probably) */
    if (filter->threshold_level == -1) {
        if (filter->silence_frames > 5) {
            filter->silence_mean /= filter->silence_frames;
            filter->silence_stddev /= filter->silence_frames;
            filter->silence_stddev -= filter->silence_mean * filter->silence_mean;
            filter->silence_stddev = fixpoint_bogus_sqrt(filter->silence_stddev);
            /* Set threshold three standard deviations from the mean. */
            filter->threshold_level = filter->silence_mean + 3 * filter->silence_stddev;
            GST_DEBUG_OBJECT(filter, "silence_mean %d stddev %d auto_threshold %d\n",
                             filter->silence_mean, filter->silence_stddev,
                             filter->threshold_level);
        }
    }

    /* Divide buffer into reasonably sized parts. */
    for (i = 0; i < num_samples; i += VADER_FRAME) {
        gint frame_len, j;

        frame_len = MIN(num_samples - i, VADER_FRAME);
        power = compute_normed_power(in_data + i, frame_len, &filter->prior_sample);
        rms = fixpoint_sqrt_q15(power);

        /* If we are in auto-threshold mode, don't do any voting etc. */
        if (filter->threshold_level == -1) {
            filter->silence_mean += rms;
            filter->silence_stddev += rms * rms;
            filter->silence_frames += 1;
            GST_DEBUG_OBJECT(filter, "silence_mean_acc %d silence_stddev_acc %d frames %d\n",
                             filter->silence_mean, filter->silence_stddev, filter->silence_frames);
            continue;
        }
        /* Shift back window values. */
        memmove(filter->window, filter->window + 1,
                (VADER_WINDOW - 1) * sizeof(*filter->window));

        /* Decide if this buffer is silence or not. */
        if (rms > filter->threshold_level)
            filter->window[VADER_WINDOW-1] = TRUE;
        else
            filter->window[VADER_WINDOW-1] = FALSE;

        /* Vote on whether we have entered a region of non-silence. */
        vote = 0;
        for (j = 0; j < VADER_WINDOW; ++j)
            vote += filter->window[j];

        GST_DEBUG_OBJECT(filter, "frame_len %d rms power %d threshold %d vote %d\n",
                         frame_len, rms, filter->threshold_level, vote);

        if (vote > VADER_WINDOW / 2) {
            filter->silent_run_length = 0;
            filter->silent = FALSE;
        }
        else {
            filter->silent_run_length
                += gst_audio_duration_from_pad_buffer(filter->sinkpad, buf);
        }

        if (filter->silent_run_length > filter->threshold_length)
            /* it has been silent long enough, flag it */
            filter->silent = TRUE;
    }

    /* Handle transitions between silence and non-silence. */
    if (filter->silent != filter->silent_prev) {
        gst_vader_transition(filter, GST_BUFFER_TIMESTAMP(buf));
    }
    /* Handling of silence detection is done. */
    g_static_rec_mutex_unlock(&filter->mtx);

    /* now check if we have to send the new buffer to the internal buffer cache
     * or to the srcpad */
    if (filter->silent) {
//.........这里部分代码省略.........

static void
gst_vader_transition(GstVader *filter, GstClockTime ts)
{
    /* NOTE: This function MUST be called with filter->mtx held! */
    /* has the silent status changed ? if so, send right signal
     * and, if from silent -> not silent, flush pre_record buffer
     */
    if (filter->silent) {
        /* Sound to silence transition. */
        GstMessage *m =
            gst_vader_message_new(filter, FALSE, ts);
        GstEvent *e =
            gst_vader_event_new(filter, GST_EVENT_VADER_STOP, ts);
        GST_DEBUG_OBJECT(filter, "signaling CUT_STOP");
        gst_element_post_message(GST_ELEMENT(filter), m);
        /* Insert a custom event in the stream to mark the end of a cut. */
        /* This will block if the pipeline is paused so we have to unlock. */
        g_static_rec_mutex_unlock(&filter->mtx);
        gst_pad_push_event(filter->srcpad, e);
        g_static_rec_mutex_lock(&filter->mtx);
        /* FIXME: That event's timestamp is wrong... as is this one. */
        g_signal_emit(filter, gst_vader_signals[SIGNAL_VADER_STOP], 0, ts);
        /* Stop dumping audio */
        if (filter->dumpfile) {
            fclose(filter->dumpfile);
            filter->dumpfile = NULL;
            ++filter->dumpidx;
        }
    } else {
        /* Silence to sound transition. */
        gint count = 0;
        GstMessage *m;
        GstEvent *e;

        GST_DEBUG_OBJECT(filter, "signaling CUT_START");
        /* Use the first pre_buffer's timestamp for the signal if possible. */
        if (filter->pre_buffer) {
            GstBuffer *prebuf;

            prebuf = (g_list_first(filter->pre_buffer))->data;
            ts = GST_BUFFER_TIMESTAMP(prebuf);
        }

        g_signal_emit(filter, gst_vader_signals[SIGNAL_VADER_START],
                      0, ts);
        m = gst_vader_message_new(filter, TRUE, ts);
        e = gst_vader_event_new(filter, GST_EVENT_VADER_START, ts);
        gst_element_post_message(GST_ELEMENT(filter), m);

        /* Insert a custom event in the stream to mark the beginning of a cut. */
        /* This will block if the pipeline is paused so we have to unlock. */
        g_static_rec_mutex_unlock(&filter->mtx);
        gst_pad_push_event(filter->srcpad, e);
        g_static_rec_mutex_lock(&filter->mtx);

        /* Start dumping audio */
        if (filter->dumpdir) {
            gchar *filename = g_strdup_printf("%s/%08d.raw", filter->dumpdir,
                                              filter->dumpidx);
            filter->dumpfile = fopen(filename, "wb");
            g_free(filename);
        }

        /* first of all, flush current buffer */
        GST_DEBUG_OBJECT(filter, "flushing buffer of length %" GST_TIME_FORMAT,
                         GST_TIME_ARGS(filter->pre_run_length));
        while (filter->pre_buffer) {
            GstBuffer *prebuf;

            prebuf = (g_list_first(filter->pre_buffer))->data;
            filter->pre_buffer = g_list_remove(filter->pre_buffer, prebuf);
            if (filter->dumpfile)
                fwrite(GST_BUFFER_DATA(prebuf), 1, GST_BUFFER_SIZE(prebuf),
                       filter->dumpfile);
            /* This will block if the pipeline is paused so we have to unlock. */
            g_static_rec_mutex_unlock(&filter->mtx);
            gst_pad_push(filter->srcpad, prebuf);
            g_static_rec_mutex_lock(&filter->mtx);
            ++count;
        }
        GST_DEBUG_OBJECT(filter, "flushed %d buffers", count);
        filter->pre_run_length = 0;
    }
}

static GstFlowReturn
audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
    GstBuffer * outbuf)
{
  GstAudioresample *audioresample;
  ResampleState *r;
  guchar *data, *datacopy;
  gulong size;
  GstClockTime timestamp;

  audioresample = GST_AUDIORESAMPLE (base);
  r = audioresample->resample;

  data = GST_BUFFER_DATA (inbuf);
  size = GST_BUFFER_SIZE (inbuf);
  timestamp = GST_BUFFER_TIMESTAMP (inbuf);

  GST_LOG_OBJECT (audioresample, "transforming buffer of %ld bytes, ts %"
      GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
      G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
      size, GST_TIME_ARGS (timestamp),
      GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)),
      GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));

  /* check for timestamp discontinuities and flush/reset if needed */
  if (G_UNLIKELY (audioresample_check_discont (audioresample, timestamp))) {
    /* Flush internal samples */
    audioresample_pushthrough (audioresample);
    /* Inform downstream element about discontinuity */
    audioresample->need_discont = TRUE;
    /* We want to recalculate the offset */
    audioresample->ts_offset = -1;
  }

  if (audioresample->ts_offset == -1) {
    /* if we don't know the initial offset yet, calculate it based on the 
     * input timestamp. */
    if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
      GstClockTime stime;

      /* offset used to calculate the timestamps. We use the sample offset for
       * this to make it more accurate. We want the first buffer to have the
       * same timestamp as the incoming timestamp. */
      audioresample->next_ts = timestamp;
      audioresample->ts_offset =
          gst_util_uint64_scale_int (timestamp, r->o_rate, GST_SECOND);
      /* offset used to set as the buffer offset, this offset is always
       * relative to the stream time, note that timestamp is not... */
      stime = (timestamp - base->segment.start) + base->segment.time;
      audioresample->offset =
          gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND);
    }
  }
  audioresample->prev_ts = timestamp;
  audioresample->prev_duration = GST_BUFFER_DURATION (inbuf);

  /* need to memdup, resample takes ownership. */
  datacopy = g_memdup (data, size);
  resample_add_input_data (r, datacopy, size, g_free, datacopy);

  return audioresample_do_output (audioresample, outbuf);
}

static GstFlowReturn
gst_a52dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
  GstA52Dec *a52dec = GST_A52DEC (parent);
  GstFlowReturn ret = GST_FLOW_OK;
  gint first_access;

  if (a52dec->dvdmode) {
    gsize size;
    guint8 data[2];
    gint offset;
    gint len;
    GstBuffer *subbuf;

    size = gst_buffer_get_size (buf);
    if (size < 2)
      goto not_enough_data;

    gst_buffer_extract (buf, 0, data, 2);
    first_access = (data[0] << 8) | data[1];

    /* Skip the first_access header */
    offset = 2;

    if (first_access > 1) {
      /* Length of data before first_access */
      len = first_access - 1;

      if (len <= 0 || offset + len > size)
        goto bad_first_access_parameter;

      subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, len);
      GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
      ret = a52dec->base_chain (pad, parent, subbuf);
      if (ret != GST_FLOW_OK) {
        gst_buffer_unref (buf);
        goto done;
      }

      offset += len;
      len = size - offset;

      if (len > 0) {
        subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, len);
        GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);

        ret = a52dec->base_chain (pad, parent, subbuf);
      }
      gst_buffer_unref (buf);
    } else {
      /* first_access = 0 or 1, so if there's a timestamp it applies to the first byte */
      subbuf =
          gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset,
          size - offset);
      GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
      gst_buffer_unref (buf);
      ret = a52dec->base_chain (pad, parent, subbuf);
    }
  } else {
    ret = a52dec->base_chain (pad, parent, buf);
  }

done:
  return ret;

/* ERRORS */
not_enough_data:
  {
    GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
        ("Insufficient data in buffer. Can't determine first_acess"));
    gst_buffer_unref (buf);
    return GST_FLOW_ERROR;
  }
bad_first_access_parameter:
  {
    GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
        ("Bad first_access parameter (%d) in buffer", first_access));
    gst_buffer_unref (buf);
    return GST_FLOW_ERROR;
  }
}

static GstFlowReturn
gst_rdt_manager_chain_rdt (GstPad * pad, GstBuffer * buffer)
{
  GstFlowReturn res;
  GstRDTManager *rdtmanager;
  GstRDTManagerSession *session;
  GstClockTime timestamp;
  GstRDTPacket packet;
  guint32 ssrc;
  guint8 pt;
  gboolean more;

  rdtmanager = GST_RDT_MANAGER (GST_PAD_PARENT (pad));

  GST_DEBUG_OBJECT (rdtmanager, "got RDT packet");

  ssrc = 0;
  pt = 0;

  GST_DEBUG_OBJECT (rdtmanager, "SSRC %08x, PT %d", ssrc, pt);

  /* find session */
  session = gst_pad_get_element_private (pad);

  /* see if we have the pad */
  if (!session->active) {
    activate_session (rdtmanager, session, ssrc, pt);
    session->active = TRUE;
  }

  if (GST_BUFFER_IS_DISCONT (buffer)) {
    GST_DEBUG_OBJECT (rdtmanager, "received discont");
    session->discont = TRUE;
  }

  res = GST_FLOW_OK;

  /* take the timestamp of the buffer. This is the time when the packet was
   * received and is used to calculate jitter and clock skew. We will adjust
   * this timestamp with the smoothed value after processing it in the
   * jitterbuffer. */
  timestamp = GST_BUFFER_TIMESTAMP (buffer);
  /* bring to running time */
  timestamp = gst_segment_to_running_time (&session->segment, GST_FORMAT_TIME,
      timestamp);

  more = gst_rdt_buffer_get_first_packet (buffer, &packet);
  while (more) {
    GstRDTType type;

    type = gst_rdt_packet_get_type (&packet);
    GST_DEBUG_OBJECT (rdtmanager, "Have packet of type %04x", type);

    if (GST_RDT_IS_DATA_TYPE (type)) {
      GST_DEBUG_OBJECT (rdtmanager, "We have a data packet");
      res = gst_rdt_manager_handle_data_packet (session, timestamp, &packet);
    } else {
      switch (type) {
        default:
          GST_DEBUG_OBJECT (rdtmanager, "Ignoring packet");
          break;
      }
    }
    if (res != GST_FLOW_OK)
      break;

    more = gst_rdt_packet_move_to_next (&packet);
  }

  gst_buffer_unref (buffer);

  return res;
}