static void
gst_a52dec_get_property (GObject * object, guint prop_id, GValue * value,
    GParamSpec * pspec)
{
  GstA52Dec *src = GST_A52DEC (object);

  switch (prop_id) {
    case ARG_DRC:
      GST_OBJECT_LOCK (src);
      g_value_set_boolean (value, src->dynamic_range_compression);
      GST_OBJECT_UNLOCK (src);
      break;
    case ARG_MODE:
      GST_OBJECT_LOCK (src);
      g_value_set_enum (value, src->request_channels & A52_CHANNEL_MASK);
      GST_OBJECT_UNLOCK (src);
      break;
    case ARG_LFE:
      GST_OBJECT_LOCK (src);
      g_value_set_boolean (value, src->request_channels & A52_LFE);
      GST_OBJECT_UNLOCK (src);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}
示例#2
0
文件: gsta52dec.c 项目: zsx/ossbuild
static GstStateChangeReturn
gst_a52dec_change_state (GstElement * element, GstStateChange transition)
{
  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
  GstA52Dec *a52dec = GST_A52DEC (element);

  switch (transition) {
    case GST_STATE_CHANGE_NULL_TO_READY:{
      GstA52DecClass *klass;

      klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
      a52dec->state = a52_init (klass->a52_cpuflags);
      break;
    }
    case GST_STATE_CHANGE_READY_TO_PAUSED:
      a52dec->samples = a52_samples (a52dec->state);
      a52dec->bit_rate = -1;
      a52dec->sample_rate = -1;
      a52dec->stream_channels = A52_CHANNEL;
      a52dec->using_channels = A52_CHANNEL;
      a52dec->level = 1;
      a52dec->bias = 0;
      a52dec->time = 0;
      a52dec->sent_segment = FALSE;
      a52dec->flag_update = TRUE;
      gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
      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:
      a52dec->samples = NULL;
      if (a52dec->cache) {
        gst_buffer_unref (a52dec->cache);
        a52dec->cache = NULL;
      }
      clear_queued (a52dec);
      break;
    case GST_STATE_CHANGE_READY_TO_NULL:
      a52_free (a52dec->state);
      a52dec->state = NULL;
      break;
    default:
      break;
  }

  return ret;
}
static gboolean
gst_a52dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{
  GstA52Dec *a52dec = GST_A52DEC (bdec);
  GstStructure *structure;

  structure = gst_caps_get_structure (caps, 0);

  if (structure && gst_structure_has_name (structure, "audio/x-private1-ac3"))
    a52dec->dvdmode = TRUE;
  else
    a52dec->dvdmode = FALSE;

  return TRUE;
}
static gboolean
gst_a52dec_stop (GstAudioDecoder * dec)
{
  GstA52Dec *a52dec = GST_A52DEC (dec);

  GST_DEBUG_OBJECT (dec, "stop");

  a52dec->samples = NULL;
  if (a52dec->state) {
    a52_free (a52dec->state);
    a52dec->state = NULL;
  }

  return TRUE;
}
示例#5
0
文件: gsta52dec.c 项目: zsx/ossbuild
static gboolean
gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps)
{
  GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
  GstStructure *structure;

  structure = gst_caps_get_structure (caps, 0);

  if (structure && gst_structure_has_name (structure, "audio/x-private1-ac3"))
    a52dec->dvdmode = TRUE;
  else
    a52dec->dvdmode = FALSE;

  gst_object_unref (a52dec);

  return TRUE;
}
static GstFlowReturn
gst_a52dec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
    gint * _offset, gint * len)
{
  GstA52Dec *a52dec;
  const guint8 *data;
  gint av, size;
  gint length = 0, flags, sample_rate, bit_rate;
  GstFlowReturn result = GST_FLOW_EOS;

  a52dec = GST_A52DEC (bdec);

  size = av = gst_adapter_available (adapter);
  data = (const guint8 *) gst_adapter_map (adapter, av);

  /* find and read header */
  bit_rate = a52dec->bit_rate;
  sample_rate = a52dec->sample_rate;
  flags = 0;
  while (size >= 7) {
    length = a52_syncinfo ((guint8 *) data, &flags, &sample_rate, &bit_rate);

    if (length == 0) {
      /* shift window to re-find sync */
      data++;
      size--;
    } else if (length <= size) {
      GST_LOG_OBJECT (a52dec, "Sync: frame size %d", length);
      result = GST_FLOW_OK;
      break;
    } else {
      GST_LOG_OBJECT (a52dec, "Not enough data available (needed %d had %d)",
          length, size);
      break;
    }
  }
  gst_adapter_unmap (adapter);

  *_offset = av - size;
  *len = length;

  return result;
}
static gboolean
gst_a52dec_start (GstAudioDecoder * dec)
{
  GstA52Dec *a52dec = GST_A52DEC (dec);
  GstA52DecClass *klass;

  GST_DEBUG_OBJECT (dec, "start");

  klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
#if defined(A52_ACCEL_DETECT)
  a52dec->state = a52_init ();
  /* This line is just to avoid being accused of not using klass */
  a52_accel (klass->a52_cpuflags & A52_ACCEL_DETECT);
#else
  a52dec->state = a52_init (klass->a52_cpuflags);
#endif

  if (!a52dec->state) {
    GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), LIBRARY, INIT, (NULL),
        ("failed to initialize a52 state"));
    return FALSE;
  }

  a52dec->samples = a52_samples (a52dec->state);
  a52dec->bit_rate = -1;
  a52dec->sample_rate = -1;
  a52dec->stream_channels = A52_CHANNEL;
  a52dec->using_channels = A52_CHANNEL;
  a52dec->level = 1;
  a52dec->bias = 0;
  a52dec->flag_update = TRUE;

  /* call upon legacy upstream byte support (e.g. seeking) */
  gst_audio_decoder_set_estimate_rate (dec, TRUE);

  return TRUE;
}
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_a52dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
{
  GstA52Dec *a52dec;
  gint channels, i;
  gboolean need_reneg = FALSE;
  gint chans;
  gint length = 0, flags, sample_rate, bit_rate;
  GstMapInfo map;
  GstFlowReturn result = GST_FLOW_OK;
  GstBuffer *outbuf;
  const gint num_blocks = 6;

  a52dec = GST_A52DEC (bdec);

  /* no fancy draining */
  if (G_UNLIKELY (!buffer))
    return GST_FLOW_OK;

  /* parsed stuff already, so this should work out fine */
  gst_buffer_map (buffer, &map, GST_MAP_READ);
  g_assert (map.size >= 7);

  /* re-obtain some sync header info,
   * should be same as during _parse and could also be cached there,
   * but anyway ... */
  bit_rate = a52dec->bit_rate;
  sample_rate = a52dec->sample_rate;
  flags = 0;
  length = a52_syncinfo (map.data, &flags, &sample_rate, &bit_rate);
  g_assert (length == map.size);

  /* update stream information, renegotiate or re-streaminfo if needed */
  need_reneg = FALSE;
  if (a52dec->sample_rate != sample_rate) {
    GST_DEBUG_OBJECT (a52dec, "sample rate changed");
    need_reneg = TRUE;
    a52dec->sample_rate = sample_rate;
  }

  if (flags) {
    if (a52dec->stream_channels != (flags & (A52_CHANNEL_MASK | A52_LFE))) {
      GST_DEBUG_OBJECT (a52dec, "stream channel flags changed, marking update");
      a52dec->flag_update = TRUE;
    }
    a52dec->stream_channels = flags & (A52_CHANNEL_MASK | A52_LFE);
  }

  if (bit_rate != a52dec->bit_rate) {
    a52dec->bit_rate = bit_rate;
    gst_a52dec_update_streaminfo (a52dec);
  }

  /* If we haven't had an explicit number of channels chosen through properties
   * at this point, choose what to downmix to now, based on what the peer will
   * accept - this allows a52dec to do downmixing in preference to a
   * downstream element such as audioconvert.
   */
  if (a52dec->request_channels != A52_CHANNEL) {
    flags = a52dec->request_channels;
  } else if (a52dec->flag_update) {
    GstCaps *caps;

    a52dec->flag_update = FALSE;

    caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (a52dec));
    if (caps && gst_caps_get_size (caps) > 0) {
      GstCaps *copy = gst_caps_copy_nth (caps, 0);
      GstStructure *structure = gst_caps_get_structure (copy, 0);
      gint orig_channels = flags ? gst_a52dec_channels (flags, NULL) : 6;
      gint fixed_channels = 0;
      const int a52_channels[6] = {
        A52_MONO,
        A52_STEREO,
        A52_STEREO | A52_LFE,
        A52_2F2R,
        A52_2F2R | A52_LFE,
        A52_3F2R | A52_LFE,
      };

      /* Prefer the original number of channels, but fixate to something
       * preferred (first in the caps) downstream if possible.
       */
      gst_structure_fixate_field_nearest_int (structure, "channels",
          orig_channels);

      if (gst_structure_get_int (structure, "channels", &fixed_channels)
          && fixed_channels <= 6) {
        if (fixed_channels < orig_channels)
          flags = a52_channels[fixed_channels - 1];
      } else {
        flags = a52_channels[5];
      }

      gst_caps_unref (copy);
    } else if (flags)
      flags = a52dec->stream_channels;
    else
      flags = A52_3F2R | A52_LFE;

    if (caps)
      gst_caps_unref (caps);
  } else {
    flags = a52dec->using_channels;
  }

  /* process */
  flags |= A52_ADJUST_LEVEL;
  a52dec->level = 1;
  if (a52_frame (a52dec->state, map.data, &flags, &a52dec->level, a52dec->bias)) {
    gst_buffer_unmap (buffer, &map);
    GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
        ("a52_frame error"), result);
    goto exit;
  }
  gst_buffer_unmap (buffer, &map);

  channels = flags & (A52_CHANNEL_MASK | A52_LFE);
  if (a52dec->using_channels != channels) {
    need_reneg = TRUE;
    a52dec->using_channels = channels;
  }

  /* negotiate if required */
  if (need_reneg) {
    GST_DEBUG_OBJECT (a52dec,
        "a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d",
        a52dec->sample_rate, a52dec->stream_channels, a52dec->using_channels);
    if (!gst_a52dec_reneg (a52dec))
      goto failed_negotiation;
  }

  if (a52dec->dynamic_range_compression == FALSE) {
    a52_dynrng (a52dec->state, NULL, NULL);
  }

  flags &= (A52_CHANNEL_MASK | A52_LFE);
  chans = gst_a52dec_channels (flags, NULL);
  if (!chans)
    goto invalid_flags;

  /* handle decoded data;
   * each frame has 6 blocks, one block is 256 samples, ea */
  outbuf =
      gst_buffer_new_and_alloc (256 * chans * (SAMPLE_WIDTH / 8) * num_blocks);

  gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
  {
    guint8 *ptr = map.data;
    for (i = 0; i < num_blocks; i++) {
      if (a52_block (a52dec->state)) {
        /* also marks discont */
        GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
            ("error decoding block %d", i), result);
        if (result != GST_FLOW_OK) {
          gst_buffer_unmap (outbuf, &map);
          goto exit;
        }
      } else {
        gint n, c;
        gint *reorder_map = a52dec->channel_reorder_map;

        for (n = 0; n < 256; n++) {
          for (c = 0; c < chans; c++) {
            ((sample_t *) ptr)[n * chans + reorder_map[c]] =
                a52dec->samples[c * 256 + n];
          }
        }
      }
      ptr += 256 * chans * (SAMPLE_WIDTH / 8);
    }
  }
  gst_buffer_unmap (outbuf, &map);

  result = gst_audio_decoder_finish_frame (bdec, outbuf, 1);

exit:
  return result;

  /* ERRORS */
failed_negotiation:
  {
    GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
    return GST_FLOW_ERROR;
  }
invalid_flags:
  {
    GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
        ("Invalid channel flags: %d", flags));
    return GST_FLOW_ERROR;
  }
}
示例#10
0
文件: gsta52dec.c 项目: zsx/ossbuild
static GstFlowReturn
gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf)
{
  GstA52Dec *a52dec;
  guint8 *data;
  guint size;
  gint length = 0, flags, sample_rate, bit_rate;
  GstFlowReturn result = GST_FLOW_OK;

  a52dec = GST_A52DEC (GST_PAD_PARENT (pad));

  if (!a52dec->sent_segment) {
    GstSegment segment;

    /* Create a basic segment. Usually, we'll get a new-segment sent by 
     * another element that will know more information (a demuxer). If we're
     * just looking at a raw AC3 stream, we won't - so we need to send one
     * here, but we don't know much info, so just send a minimal TIME 
     * new-segment event
     */
    gst_segment_init (&segment, GST_FORMAT_TIME);
    gst_pad_push_event (a52dec->srcpad, gst_event_new_new_segment (FALSE,
            segment.rate, segment.format, segment.start,
            segment.duration, segment.start));
    a52dec->sent_segment = TRUE;
  }

  /* merge with cache, if any. Also make sure timestamps match */
  if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
    a52dec->time = GST_BUFFER_TIMESTAMP (buf);
    GST_DEBUG_OBJECT (a52dec,
        "Received buffer with ts %" GST_TIME_FORMAT " duration %"
        GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
        GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
  }

  if (a52dec->cache) {
    buf = gst_buffer_join (a52dec->cache, buf);
    a52dec->cache = NULL;
  }
  data = GST_BUFFER_DATA (buf);
  size = GST_BUFFER_SIZE (buf);

  /* find and read header */
  bit_rate = a52dec->bit_rate;
  sample_rate = a52dec->sample_rate;
  flags = 0;
  while (size >= 7) {
    length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);

    if (length == 0) {
      /* no sync */
      data++;
      size--;
    } else if (length <= size) {
      GST_DEBUG ("Sync: %d", length);

      if (flags != a52dec->prev_flags)
        a52dec->flag_update = TRUE;
      a52dec->prev_flags = flags;

      result = gst_a52dec_handle_frame (a52dec, data,
          length, flags, sample_rate, bit_rate);
      if (result != GST_FLOW_OK) {
        size = 0;
        break;
      }
      size -= length;
      data += length;
    } else {
      /* not enough data */
      GST_LOG ("Not enough data available");
      break;
    }
  }

  /* keep cache */
  if (length == 0) {
    GST_LOG ("No sync found");
  }

  if (size > 0) {
    a52dec->cache = gst_buffer_create_sub (buf,
        GST_BUFFER_SIZE (buf) - size, size);
  }

  gst_buffer_unref (buf);

  return result;
}
示例#11
0
文件: gsta52dec.c 项目: zsx/ossbuild
static GstFlowReturn
gst_a52dec_chain (GstPad * pad, GstBuffer * buf)
{
  GstA52Dec *a52dec = GST_A52DEC (GST_PAD_PARENT (pad));
  GstFlowReturn ret;
  gint first_access;

  if (GST_BUFFER_IS_DISCONT (buf)) {
    GST_LOG_OBJECT (a52dec, "received DISCONT");
    gst_a52dec_drain (a52dec);
    /* clear cache on discont and mark a discont in the element */
    if (a52dec->cache) {
      gst_buffer_unref (a52dec->cache);
      a52dec->cache = NULL;
    }
    a52dec->discont = TRUE;
  }

  if (a52dec->dvdmode) {
    gint size = GST_BUFFER_SIZE (buf);
    guchar *data = GST_BUFFER_DATA (buf);
    gint offset;
    gint len;
    GstBuffer *subbuf;

    if (size < 2)
      goto not_enough_data;

    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_create_sub (buf, offset, len);
      GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
      ret = gst_a52dec_chain_raw (pad, subbuf);
      if (ret != GST_FLOW_OK)
        goto done;

      offset += len;
      len = size - offset;

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

        ret = gst_a52dec_chain_raw (pad, subbuf);
      }
    } else {
      /* first_access = 0 or 1, so if there's a timestamp it applies to the first byte */
      subbuf = gst_buffer_create_sub (buf, offset, size - offset);
      GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
      ret = gst_a52dec_chain_raw (pad, subbuf);
    }
  } else {
    gst_buffer_ref (buf);
    ret = gst_a52dec_chain_raw (pad, buf);
  }

done:
  gst_buffer_unref (buf);
  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;
  }
}
示例#12
0
文件: gsta52dec.c 项目: zsx/ossbuild
static gboolean
gst_a52dec_sink_event (GstPad * pad, GstEvent * event)
{
  GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
  gboolean ret = FALSE;

  GST_LOG ("Handling %s event", GST_EVENT_TYPE_NAME (event));

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_NEWSEGMENT:
    {
      GstFormat fmt;
      gboolean update;
      gint64 start, end, pos;
      gdouble rate, arate;

      gst_event_parse_new_segment_full (event, &update, &rate, &arate, &fmt,
          &start, &end, &pos);

      /* drain queued buffers before activating the segment so that we can clip
       * against the old segment first */
      gst_a52dec_drain (a52dec);

      if (fmt != GST_FORMAT_TIME || !GST_CLOCK_TIME_IS_VALID (start)) {
        GST_WARNING ("No time in newsegment event %p (format is %s)",
            event, gst_format_get_name (fmt));
        gst_event_unref (event);
        a52dec->sent_segment = FALSE;
        /* set some dummy values, FIXME: do proper conversion */
        a52dec->time = start = pos = 0;
        fmt = GST_FORMAT_TIME;
        end = -1;
      } else {
        a52dec->time = start;
        a52dec->sent_segment = TRUE;
        GST_DEBUG_OBJECT (a52dec,
            "Pushing newseg rate %g, applied rate %g, format %d, start %"
            G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT ", pos %"
            G_GINT64_FORMAT, rate, arate, fmt, start, end, time);

        ret = gst_pad_push_event (a52dec->srcpad, event);
      }

      gst_segment_set_newsegment (&a52dec->segment, update, rate, fmt, start,
          end, pos);
      break;
    }
    case GST_EVENT_TAG:
      ret = gst_pad_push_event (a52dec->srcpad, event);
      break;
    case GST_EVENT_EOS:
      gst_a52dec_drain (a52dec);
      ret = gst_pad_push_event (a52dec->srcpad, event);
      break;
    case GST_EVENT_FLUSH_START:
      ret = gst_pad_push_event (a52dec->srcpad, event);
      break;
    case GST_EVENT_FLUSH_STOP:
      if (a52dec->cache) {
        gst_buffer_unref (a52dec->cache);
        a52dec->cache = NULL;
      }
      clear_queued (a52dec);
      gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
      ret = gst_pad_push_event (a52dec->srcpad, event);
      break;
    default:
      ret = gst_pad_push_event (a52dec->srcpad, event);
      break;
  }

  gst_object_unref (a52dec);
  return ret;
}