예제 #1
0
static GstFlowReturn
gst_dtsdec_chain_raw (GstPad * pad, GstBuffer * buf)
{
  GstDtsDec *dts;
  guint8 *data;
  gint size;
  gint length, flags, sample_rate, bit_rate, frame_length;
  GstFlowReturn result = GST_FLOW_OK;

  dts = GST_DTSDEC (GST_PAD_PARENT (pad));

  if (dts->cache) {
    buf = gst_buffer_join (dts->cache, buf);
    dts->cache = NULL;
  }

  data = GST_BUFFER_DATA (buf);
  size = GST_BUFFER_SIZE (buf);
  length = 0;
  while (size >= 7) {
    length = dts_syncinfo (dts->state, data, &flags,
        &sample_rate, &bit_rate, &frame_length);
    if (length == 0) {
      /* shift window to re-find sync */
      data++;
      size--;
    } else if (length <= size) {
      GST_DEBUG ("Sync: frame size %d", length);
      result = gst_dtsdec_handle_frame (dts, data, length,
          flags, sample_rate, bit_rate);
      if (result != GST_FLOW_OK) {
        size = 0;
        break;
      }
      size -= length;
      data += length;
    } else {
      GST_LOG ("Not enough data available (needed %d had %d)", length, size);
      break;
    }
  }

  /* keep cache */
  if (length == 0) {
    GST_LOG ("No sync found");
  }
  if (size > 0) {
    dts->cache = gst_buffer_create_sub (buf,
        GST_BUFFER_SIZE (buf) - size, size);
  }

  gst_buffer_unref (buf);

  return result;
}
예제 #2
0
static GstStateChangeReturn
gst_dtsdec_change_state (GstElement * element, GstStateChange transition)
{
  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
  GstDtsDec *dts = GST_DTSDEC (element);

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

      klass = GST_DTSDEC_CLASS (G_OBJECT_GET_CLASS (dts));
      dts->state = dts_init (klass->dts_cpuflags);
      break;
    }
    case GST_STATE_CHANGE_READY_TO_PAUSED:
      dts->samples = dts_samples (dts->state);
      dts->bit_rate = -1;
      dts->sample_rate = -1;
      dts->stream_channels = 0;
      /* FIXME force stereo for now */
      dts->request_channels = DTS_CHANNEL;
      dts->using_channels = 0;
      dts->level = 1;
      dts->bias = 0;
      dts->current_ts = 0;
      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:
      dts->samples = NULL;
      if (dts->cache) {
        gst_buffer_unref (dts->cache);
        dts->cache = NULL;
      }
      break;
    case GST_STATE_CHANGE_READY_TO_NULL:
      dts_free (dts->state);
      dts->state = NULL;
      break;
    default:
      break;
  }

  return ret;
}
예제 #3
0
static void
gst_dtsdec_get_property (GObject * object, guint prop_id, GValue * value,
    GParamSpec * pspec)
{
  GstDtsDec *dts = GST_DTSDEC (object);

  switch (prop_id) {
    case PROP_DRC:
      g_value_set_boolean (value, dts->dynamic_range_compression);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}
예제 #4
0
static gboolean
gst_dtsdec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{
  GstDtsDec *dts = GST_DTSDEC (bdec);
  GstStructure *structure;

  structure = gst_caps_get_structure (caps, 0);

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

  return TRUE;
}
예제 #5
0
static gboolean
gst_dtsdec_stop (GstAudioDecoder * dec)
{
  GstDtsDec *dts = GST_DTSDEC (dec);

  GST_DEBUG_OBJECT (dec, "stop");

  dts->samples = NULL;
  if (dts->state) {
    dca_free (dts->state);
    dts->state = NULL;
  }

  return TRUE;
}
예제 #6
0
static gboolean
gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps)
{
  GstDtsDec *dts = GST_DTSDEC (gst_pad_get_parent (pad));
  GstStructure *structure;

  structure = gst_caps_get_structure (caps, 0);

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

  gst_object_unref (dts);

  return TRUE;
}
예제 #7
0
static GstFlowReturn
gst_dtsdec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
    gint * _offset, gint * len)
{
  GstDtsDec *dts;
  guint8 *data;
  gint av, size;
  gint length = 0, flags, sample_rate, bit_rate, frame_length;
  GstFlowReturn result = GST_FLOW_EOS;

  dts = GST_DTSDEC (bdec);

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

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

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

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

  return result;
}
예제 #8
0
static gboolean
gst_dtsdec_sink_event (GstPad * pad, GstEvent * event)
{
  GstDtsDec *dtsdec = GST_DTSDEC (gst_pad_get_parent (pad));
  gboolean ret = FALSE;

  GST_LOG_OBJECT (dtsdec, "%s event", GST_EVENT_TYPE_NAME (event));

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_NEWSEGMENT:{
      GstFormat format;
      gint64 val;

      gst_event_parse_new_segment (event, NULL, NULL, &format, &val, NULL,
          NULL);
      if (format != GST_FORMAT_TIME || !GST_CLOCK_TIME_IS_VALID (val)) {
        GST_WARNING ("No time in newsegment event %p", event);
      } else {
        dtsdec->current_ts = val;
      }

      if (dtsdec->cache) {
        gst_buffer_unref (dtsdec->cache);
        dtsdec->cache = NULL;
      }
      ret = gst_pad_event_default (pad, event);
      break;
    }
    case GST_EVENT_FLUSH_STOP:
      if (dtsdec->cache) {
        gst_buffer_unref (dtsdec->cache);
        dtsdec->cache = NULL;
      }
      ret = gst_pad_event_default (pad, event);
      break;
    default:
      ret = gst_pad_event_default (pad, event);
      break;
  }

  gst_object_unref (dtsdec);
  return ret;
}
예제 #9
0
static gboolean
gst_dtsdec_start (GstAudioDecoder * dec)
{
  GstDtsDec *dts = GST_DTSDEC (dec);
  GstDtsDecClass *klass;

  GST_DEBUG_OBJECT (dec, "start");

  klass = GST_DTSDEC_CLASS (G_OBJECT_GET_CLASS (dts));
  dts->state = dca_init (klass->dts_cpuflags);
  dts->samples = dca_samples (dts->state);
  dts->bit_rate = -1;
  dts->sample_rate = -1;
  dts->stream_channels = DCA_CHANNEL;
  dts->using_channels = DCA_CHANNEL;
  dts->level = 1;
  dts->bias = 0;
  dts->flag_update = TRUE;

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

  return TRUE;
}
예제 #10
0
static GstFlowReturn
gst_dtsdec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
  GstFlowReturn ret = GST_FLOW_OK;
  GstDtsDec *dts = GST_DTSDEC (parent);
  gint first_access;

  if (dts->dvdmode) {
    guint8 data[2];
    gsize size;
    gint offset, 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 = dts->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 = dts->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);
      ret = dts->base_chain (pad, parent, subbuf);
      gst_buffer_unref (buf);
    }
  } else {
    ret = dts->base_chain (pad, parent, buf);
  }

done:
  return ret;

/* ERRORS */
not_enough_data:
  {
    GST_ELEMENT_ERROR (GST_ELEMENT (dts), 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 (dts), STREAM, DECODE, (NULL),
        ("Bad first_access parameter (%d) in buffer", first_access));
    gst_buffer_unref (buf);
    return GST_FLOW_ERROR;
  }
}
예제 #11
0
static GstFlowReturn
gst_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
{
  GstDtsDec *dts;
  gint channels, i, num_blocks;
  gboolean need_renegotiation = FALSE;
  guint8 *data;
  gsize size;
  GstMapInfo map;
  gint chans;
  gint length = 0, flags, sample_rate, bit_rate, frame_length;
  GstFlowReturn result = GST_FLOW_OK;
  GstBuffer *outbuf;

  dts = GST_DTSDEC (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);
  data = map.data;
  size = map.size;
  g_assert (size >= 7);

  bit_rate = dts->bit_rate;
  sample_rate = dts->sample_rate;
  flags = 0;
  length = dca_syncinfo (dts->state, data, &flags, &sample_rate, &bit_rate,
      &frame_length);
  g_assert (length == size);

  if (flags != dts->prev_flags) {
    dts->prev_flags = flags;
    dts->flag_update = TRUE;
  }

  /* go over stream properties, renegotiate or update streaminfo if needed */
  if (dts->sample_rate != sample_rate) {
    need_renegotiation = TRUE;
    dts->sample_rate = sample_rate;
  }

  if (flags) {
    dts->stream_channels = flags & (DCA_CHANNEL_MASK | DCA_LFE);
  }

  if (bit_rate != dts->bit_rate) {
    dts->bit_rate = bit_rate;
    gst_dtsdec_update_streaminfo (dts);
  }

  /* 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.
   * FIXME: Add the property back in for forcing output channels.
   */
  if (dts->request_channels != DCA_CHANNEL) {
    flags = dts->request_channels;
  } else if (dts->flag_update) {
    GstCaps *caps;

    dts->flag_update = FALSE;

    caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dts));
    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 channels;
      const int dts_channels[6] = {
        DCA_MONO,
        DCA_STEREO,
        DCA_STEREO | DCA_LFE,
        DCA_2F2R,
        DCA_2F2R | DCA_LFE,
        DCA_3F2R | DCA_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",
          flags ? gst_dtsdec_channels (flags, NULL) : 6);
      gst_structure_get_int (structure, "channels", &channels);
      if (channels <= 6)
        flags = dts_channels[channels - 1];
      else
        flags = dts_channels[5];

      gst_caps_unref (copy);
    } else if (flags) {
      flags = dts->stream_channels;
    } else {
      flags = DCA_3F2R | DCA_LFE;
    }

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

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

  channels = flags & (DCA_CHANNEL_MASK | DCA_LFE);
  if (dts->using_channels != channels) {
    need_renegotiation = TRUE;
    dts->using_channels = channels;
  }

  /* negotiate if required */
  if (need_renegotiation) {
    GST_DEBUG_OBJECT (dts,
        "dtsdec: sample_rate:%d stream_chans:0x%x using_chans:0x%x",
        dts->sample_rate, dts->stream_channels, dts->using_channels);
    if (!gst_dtsdec_renegotiate (dts))
      goto failed_negotiation;
  }

  if (dts->dynamic_range_compression == FALSE) {
    dca_dynrng (dts->state, NULL, NULL);
  }

  flags &= (DCA_CHANNEL_MASK | DCA_LFE);
  chans = gst_dtsdec_channels (flags, NULL);
  if (!chans)
    goto invalid_flags;

  /* handle decoded data, one block is 256 samples */
  num_blocks = dca_blocks_num (dts->state);
  outbuf =
      gst_buffer_new_and_alloc (256 * chans * (SAMPLE_WIDTH / 8) * num_blocks);

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

        for (n = 0; n < 256; n++) {
          for (c = 0; c < chans; c++) {
            ((sample_t *) ptr)[n * chans + reorder_map[c]] =
                dts->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 (dts, CORE, NEGOTIATION, (NULL), (NULL));
    return GST_FLOW_ERROR;
  }
invalid_flags:
  {
    GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL),
        ("Invalid channel flags: %d", flags));
    return GST_FLOW_ERROR;
  }
}
예제 #12
0
static GstFlowReturn
gst_dtsdec_chain (GstPad * pad, GstBuffer * buf)
{
  GstFlowReturn res = GST_FLOW_OK;
  GstDtsDec *dts = GST_DTSDEC (GST_PAD_PARENT (pad));
  gint first_access;

  if (dts->dvdmode) {
    gint size = GST_BUFFER_SIZE (buf);
    guint8 *data = GST_BUFFER_DATA (buf);
    gint offset, 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;
      res = gst_dtsdec_chain_raw (pad, subbuf);
      if (res != 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);

        res = gst_dtsdec_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);
      res = gst_dtsdec_chain_raw (pad, subbuf);
    }
  } else {
    res = gst_dtsdec_chain_raw (pad, buf);
  }

done:
  return res;

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

}