コード例 #1
0
ファイル: gstcmmldec.c プロジェクト: ChinnaSuhas/ossbuild
/* parses the first CMML packet (the ident header)
 */
static void
gst_cmml_dec_parse_ident_header (GstCmmlDec * dec, GstBuffer * buffer)
{
  guint8 *data = GST_BUFFER_DATA (buffer);

  /* the ident header has a fixed length */
  if (GST_BUFFER_SIZE (buffer) != CMML_IDENT_HEADER_SIZE) {
    GST_ELEMENT_ERROR (dec, STREAM, DECODE,
        (NULL), ("wrong ident header size: %d", GST_BUFFER_SIZE (buffer)));
    dec->flow_return = GST_FLOW_ERROR;

    return;
  }

  data += 8;
  dec->major = GST_READ_UINT16_LE (data);
  data += 2;
  dec->minor = GST_READ_UINT16_LE (data);
  data += 2;
  dec->granulerate_n = GST_READ_UINT64_LE (data);
  data += 8;
  dec->granulerate_d = GST_READ_UINT64_LE (data);
  data += 8;
  dec->granuleshift = GST_READ_UINT8 (data);

  GST_INFO_OBJECT (dec, "bitstream initialized "
      "(major: %" G_GINT16_FORMAT " minor: %" G_GINT16_FORMAT
      " granulerate_n: %" G_GINT64_FORMAT " granulerate_d: %" G_GINT64_FORMAT
      " granuleshift: %d)",
      dec->major, dec->minor,
      dec->granulerate_n, dec->granulerate_d, dec->granuleshift);

  dec->flow_return = GST_FLOW_OK;
}
コード例 #2
0
ファイル: gstivfparse.c プロジェクト: 0p1pp1/gst-plugins-bad
static GstFlowReturn
gst_ivf_parse_handle_frame_start (GstIvfParse * ivf, GstBaseParseFrame * frame,
    gint * skipsize)
{
  GstBuffer *const buffer = frame->buffer;
  GstMapInfo map;
  GstFlowReturn ret = GST_FLOW_OK;

  gst_buffer_map (buffer, &map, GST_MAP_READ);
  if (map.size >= IVF_FILE_HEADER_SIZE) {
    guint32 magic = GST_READ_UINT32_LE (map.data);
    guint16 version = GST_READ_UINT16_LE (map.data + 4);
    guint16 header_size = GST_READ_UINT16_LE (map.data + 6);
    guint32 fourcc = GST_READ_UINT32_LE (map.data + 8);
    guint16 width = GST_READ_UINT16_LE (map.data + 12);
    guint16 height = GST_READ_UINT16_LE (map.data + 14);
    guint32 fps_n = GST_READ_UINT32_LE (map.data + 16);
    guint32 fps_d = GST_READ_UINT32_LE (map.data + 20);
#ifndef GST_DISABLE_GST_DEBUG
    guint32 num_frames = GST_READ_UINT32_LE (map.data + 24);
#endif

    if (magic != GST_MAKE_FOURCC ('D', 'K', 'I', 'F') ||
        version != 0 || header_size != 32 ||
        fourcc_to_media_type (fourcc) == NULL) {
      GST_ELEMENT_ERROR (ivf, STREAM, WRONG_TYPE, (NULL), (NULL));
      ret = GST_FLOW_ERROR;
      goto end;
    }

    ivf->fourcc = fourcc;

    gst_ivf_parse_set_size (ivf, width, height);
    gst_ivf_parse_set_framerate (ivf, fps_n, fps_d);

    GST_LOG_OBJECT (ivf, "Stream has %d frames", num_frames);

    /* move along */
    ivf->state = GST_IVF_PARSE_DATA;
    gst_base_parse_set_min_frame_size (GST_BASE_PARSE_CAST (ivf),
        IVF_FRAME_HEADER_SIZE);
    *skipsize = IVF_FILE_HEADER_SIZE;
  } else {
    GST_LOG_OBJECT (ivf, "Header data not yet available.");
    *skipsize = 0;
  }

end:
  gst_buffer_unmap (buffer, &map);
  return ret;
}
コード例 #3
0
ファイル: gstexiftag.c プロジェクト: genesi/gst-base-plugins
static gint
deserialize_orientation (GstExifReader * exif_reader,
    GstByteReader * reader, const GstExifTagMatch * exiftag,
    GstExifTagData * tagdata)
{
  gint ret = 1;
  const gchar *str = NULL;
  gint value;

  GST_LOG ("Starting to parse %s tag in exif 0x%x", exiftag->gst_tag,
      exiftag->exif_tag);

  /* validate tag */
  if (tagdata->tag_type != EXIF_TYPE_SHORT || tagdata->count != 1) {
    GST_WARNING ("Orientation tag has unexpected type/count");
    return ret;
  }

  if (exif_reader->byte_order == G_LITTLE_ENDIAN) {
    value = GST_READ_UINT16_LE (tagdata->offset_as_data);
  } else {
    value = GST_READ_UINT16_BE (tagdata->offset_as_data);
  }

  str = gst_tag_image_orientation_from_exif_value (value);
  if (str == NULL) {
    GST_WARNING ("Invalid value for exif orientation tag: %d", value);
    return ret;
  }
  gst_tag_list_add (exif_reader->taglist, GST_TAG_MERGE_REPLACE,
      exiftag->gst_tag, str, NULL);

  return ret;
}
コード例 #4
0
ファイル: mveaudiodec.cpp プロジェクト: AlanWasTaken/gemrb
void
ipaudio_uncompress (short *buffer, unsigned short buf_len,
    const unsigned char *data, unsigned char channels)
{
  int i, out = 0;
  int predictor[2] = {};
  int channel_number = 0;

  for (i = 0; i < channels; ++i) {
    predictor[i] = GST_READ_UINT16_LE (data);
    data += 2;
    if (predictor[i] & 0x8000)
      predictor[i] -= 0x10000;
    buffer[out++] = predictor[i];
  }

  /* we count in 16-bit ints, so adjust the buffer size */
  buf_len /= 2;
  while (out < buf_len) {
    predictor[channel_number] += delta_table[*data++];
    if (predictor[channel_number] < -32768)
      predictor[channel_number] = -32768;
    else if (predictor[channel_number] > 32767)
      predictor[channel_number] = 32767;
    buffer[out++] = predictor[channel_number];

    /* toggle channel */
    channel_number ^= channels - 1;
  }
}
コード例 #5
0
static void
asf_payload_parse_replicated_data_extensions (AsfStream * stream,
    AsfPayload * payload)
{
  AsfPayloadExtension *ext;
  guint off;

  if (!stream->ext_props.valid || stream->ext_props.payload_extensions == NULL)
    return;

  off = 8;
  for (ext = stream->ext_props.payload_extensions; ext->len > 0; ++ext) {
    if (G_UNLIKELY (off + ext->len > payload->rep_data_len)) {
      GST_WARNING ("not enough replicated data for defined extensions");
      return;
    }
    switch (ext->id) {
      case ASF_PAYLOAD_EXTENSION_DURATION:
        if (G_LIKELY (ext->len == 2)) {
          guint16 tdur = GST_READ_UINT16_LE (payload->rep_data + off);
          /* packet durations of 1ms are mostly invalid */
          if (tdur != 1)
            payload->duration = tdur * GST_MSECOND;
        } else {
          GST_WARNING ("unexpected DURATION extensions len %u", ext->len);
        }
        break;
      case ASF_PAYLOAD_EXTENSION_SYSTEM_CONTENT:
        if (G_LIKELY (ext->len == 1)) {
          guint8 data = payload->rep_data[off];

          payload->interlaced = data & 0x1;
          payload->rff = data & 0x8;
          payload->tff = (data & 0x2) || !(data & 0x4);
          GST_DEBUG ("SYSTEM_CONTENT: interlaced:%d, rff:%d, tff:%d",
              payload->interlaced, payload->rff, payload->tff);
        } else {
          GST_WARNING ("unexpected SYSTEM_CONTE extensions len %u", ext->len);
        }
        break;
      case ASF_PAYLOAD_EXTENSION_SYSTEM_PIXEL_ASPECT_RATIO:
        if (G_LIKELY (ext->len == 2)) {
          payload->par_x = payload->rep_data[off];
          payload->par_y = payload->rep_data[off + 1];
          GST_DEBUG ("PAR %d / %d", payload->par_x, payload->par_y);
        } else {
          GST_WARNING ("unexpected SYSTEM_PIXEL_ASPECT_RATIO extensions len %u",
              ext->len);
        }
        break;
      default:
        GST_WARNING ("UNKNOWN PAYLOAD EXTENSION !");
        break;
    }
    off += ext->len;
  }
}
コード例 #6
0
/**
 * gst_audio_format_fill_silence:
 * @info: a #GstAudioFormatInfo
 * @dest: (array length=length) (element-type guint8): a destination
 *   to fill
 * @length: the length to fill
 *
 * Fill @length bytes in @dest with silence samples for @info.
 */
void
gst_audio_format_fill_silence (const GstAudioFormatInfo * info,
    gpointer dest, gsize length)
{
  guint8 *dptr = dest;

  g_return_if_fail (info != NULL);
  g_return_if_fail (dest != NULL);

  if (info->flags & GST_AUDIO_FORMAT_FLAG_FLOAT ||
      info->flags & GST_AUDIO_FORMAT_FLAG_SIGNED) {
    /* float or signed always 0 */
    orc_memset (dest, 0, length);
  } else {
    gint i, j, bps = info->width >> 3;

    switch (bps) {
      case 1:
        orc_memset (dest, info->silence[0], length);
        break;
      case 2:{
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
        guint16 silence = GST_READ_UINT16_LE (info->silence);
#else
        guint16 silence = GST_READ_UINT16_BE (info->silence);
#endif
        audio_orc_splat_u16 (dest, silence, length / bps);
        break;
      }
      case 4:{
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
        guint32 silence = GST_READ_UINT32_LE (info->silence);
#else
        guint32 silence = GST_READ_UINT32_BE (info->silence);
#endif
        audio_orc_splat_u32 (dest, silence, length / bps);
        break;
      }
      case 8:{
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
        guint64 silence = GST_READ_UINT64_LE (info->silence);
#else
        guint64 silence = GST_READ_UINT64_BE (info->silence);
#endif
        audio_orc_splat_u64 (dest, silence, length / bps);
        break;
      }
      default:
        for (i = 0; i < length; i += bps) {
          for (j = 0; j < bps; j++)
            *dptr++ = info->silence[j];
        }
        break;
    }
  }
}
コード例 #7
0
static GstData *
gst_v4ljpegsrc_get (GstPad * pad)
{
  GstV4lJpegSrc *v4ljpegsrc;
  GstV4lSrc *v4lsrc;
  GstData *data;
  GstBuffer *buf;
  GstBuffer *outbuf;
  int jpeg_size;

  g_return_val_if_fail (pad != NULL, NULL);
  v4ljpegsrc = GST_V4LJPEGSRC (gst_pad_get_parent (pad));
  v4lsrc = GST_V4LSRC (v4ljpegsrc);

  /* Fetch from the v4lsrc class get fn.  */
  data = v4ljpegsrc->getfn (pad);

  /* If not a buffer, return it unchanged */
  if (!data || (!GST_IS_BUFFER (data)))
    return data;

  buf = GST_BUFFER (data);

  /* Confirm that the buffer contains jpeg data */

  /* 
   * Create a new subbuffer from the jpeg data 
   * The first 2 bytes in the buffer are the size of the jpeg data
   */
  if (GST_BUFFER_SIZE (buf) > 2) {
    jpeg_size = (int) (GST_READ_UINT16_LE (GST_BUFFER_DATA (buf))) * 8;
  } else
    jpeg_size = 0;

  /* Check that the size is sensible */
  if ((jpeg_size <= 0) || (jpeg_size > GST_BUFFER_SIZE (buf) - 2)) {
    GST_ELEMENT_ERROR (v4ljpegsrc, STREAM, FORMAT, (NULL),
        ("Invalid non-jpeg frame from camera"));
    return NULL;
  }

  GST_DEBUG_OBJECT (v4ljpegsrc, "Creating JPEG subbuffer of size %d",
      jpeg_size);
  outbuf = gst_buffer_create_sub (buf, 2, jpeg_size);

  /* Copy timestamps onto the subbuffer */
  gst_buffer_stamp (outbuf, buf);

  /* Release the main buffer */
  gst_buffer_unref (buf);

  return GST_DATA (outbuf);
}
コード例 #8
0
static void
check_rgb_buf (const guint8 * pixels, guint32 r_mask, guint32 g_mask,
    guint32 b_mask, guint32 a_mask, guint8 r_expected, guint8 g_expected,
    guint8 b_expected, guint bpp, guint depth)
{
  guint32 pixel, red, green, blue, alpha;

  switch (bpp) {
    case 32:
      pixel = GST_READ_UINT32_BE (pixels);
      break;
    case 24:
      pixel = (GST_READ_UINT8 (pixels) << 16) |
          (GST_READ_UINT8 (pixels + 1) << 8) |
          (GST_READ_UINT8 (pixels + 2) << 0);
      break;
    case 16:
      if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
        pixel = GST_READ_UINT16_LE (pixels);
      else
        pixel = GST_READ_UINT16_BE (pixels);
      break;
    default:
      g_return_if_reached ();
  }

  red = right_shift_colour (r_mask, pixel);
  green = right_shift_colour (g_mask, pixel);
  blue = right_shift_colour (b_mask, pixel);
  alpha = right_shift_colour (a_mask, pixel);

  /* can't enable this by default, valgrind will complain about accessing
   * uninitialised memory for the depth=24,bpp=32 formats ... */
  /* GST_LOG ("pixels: 0x%02x 0x%02x 0x%02x 0x%02x => pixel = 0x%08x",
     pixels[0], (guint) pixels[1], pixels[2], pixels[3], pixel); */

  /* fix up the mask (for rgb15/16) */
  if (bpp == 16) {
    r_expected = fix_expected_colour (r_mask, r_expected);
    g_expected = fix_expected_colour (g_mask, g_expected);
    b_expected = fix_expected_colour (b_mask, b_expected);
  }

  fail_unless (red == r_expected, "RED: expected 0x%02x, found 0x%02x",
      r_expected, red);
  fail_unless (green == g_expected, "GREEN: expected 0x%02x, found 0x%02x",
      g_expected, green);
  fail_unless (blue == b_expected, "BLUE: expected 0x%02x, found 0x%02x",
      b_expected, blue);

  fail_unless (a_mask == 0 || alpha != 0);      /* better than nothing */
}
コード例 #9
0
/* inspired by the original one in wavpack */
gboolean
gst_wavpack_read_metadata (GstWavpackMetadata * wpmd, guint8 * header_data,
    guint8 ** p_data)
{
  WavpackHeader hdr;
  guint8 *end;

  gst_wavpack_read_header (&hdr, header_data);
  end = header_data + hdr.ckSize + 8;

  if (end - *p_data < 2)
    return FALSE;

  wpmd->id = GST_READ_UINT8 (*p_data);
  wpmd->byte_length = 2 * (guint) GST_READ_UINT8 (*p_data + 1);

  *p_data += 2;

  if ((wpmd->id & ID_LARGE) == ID_LARGE) {
    guint extra;

    wpmd->id &= ~ID_LARGE;

    if (end - *p_data < 2)
      return FALSE;

    extra = GST_READ_UINT16_LE (*p_data);
    wpmd->byte_length += (extra << 9);
    *p_data += 2;
  }

  if ((wpmd->id & ID_ODD_SIZE) == ID_ODD_SIZE) {
    wpmd->id &= ~ID_ODD_SIZE;
    --wpmd->byte_length;
  }

  if (wpmd->byte_length > 0) {
    if (end - *p_data < wpmd->byte_length + (wpmd->byte_length & 1)) {
      wpmd->data = NULL;
      return FALSE;
    }

    wpmd->data = *p_data;
    *p_data += wpmd->byte_length + (wpmd->byte_length & 1);
  } else {
    wpmd->data = NULL;
  }

  return TRUE;
}
コード例 #10
0
ファイル: rmutils.c プロジェクト: prajnashi/gst-plugins-ugly
GstBuffer *
gst_rm_utils_descramble_dnet_buffer (GstBuffer * buf)
{
  guint8 *data, *end;

  buf = gst_buffer_make_writable (buf);

  /* dnet = byte-order swapped AC3 */
  data = GST_BUFFER_DATA (buf);
  end = GST_BUFFER_DATA (buf) + GST_BUFFER_SIZE (buf);
  while ((data + 1) < end) {
    /* byte-swap in an alignment-safe way */
    GST_WRITE_UINT16_BE (data, GST_READ_UINT16_LE (data));
    data += sizeof (guint16);
  }
  return buf;
}
コード例 #11
0
/* we are unlikely to deal with lengths > 2GB here any time soon, so just
 * return a signed int and use that for error reporting */
static inline gint
asf_packet_read_varlen_int (guint lentype_flags, guint lentype_bit_offset,
    const guint8 ** p_data, guint * p_size)
{
  static const guint lens[4] = { 0, 1, 2, 4 };
  guint len, val;

  len = lens[(lentype_flags >> lentype_bit_offset) & 0x03];

  /* will make caller bail out with a short read if there's not enough data */
  if (G_UNLIKELY (*p_size < len)) {
    GST_WARNING ("need %u bytes, but only %u bytes available", len, *p_size);
    return -1;
  }

  switch (len) {
    case 0:
      val = 0;
      break;
    case 1:
      val = GST_READ_UINT8 (*p_data);
      break;
    case 2:
      val = GST_READ_UINT16_LE (*p_data);
      break;
    case 4:
      val = GST_READ_UINT32_LE (*p_data);
      break;
    default:
      g_assert_not_reached ();
  }

  *p_data += len;
  *p_size -= len;

  return (gint) val;
}
コード例 #12
0
/* Correctly format samples with width!=depth for the wav format, i.e.
 * have the data in the highest depth bits and all others zero */
static void
gst_wavenc_format_samples (GstBuffer * buf, guint width, guint depth)
{
  guint8 *data = GST_BUFFER_DATA (buf);
  guint nsamples = (GST_BUFFER_SIZE (buf) * 8) / width;
  guint32 tmp;

  for (; nsamples; nsamples--) {
    switch (width) {

      case 8:
        tmp = *data;
        *data = *data << (width - depth);
        data += 1;
        break;
      case 16:
        tmp = GST_READ_UINT16_LE (data);
        tmp = tmp << (width - depth);
        GST_WRITE_UINT16_LE (data, tmp);
        data += 2;
        break;
      case 24:
        tmp = READ24_FROM_LE (data);
        tmp = tmp << (width - depth);
        WRITE24_TO_LE (data, tmp);
        data += 3;
        break;
      case 32:
        tmp = GST_READ_UINT32_LE (data);
        tmp = tmp << (width - depth);
        GST_WRITE_UINT32_LE (data, tmp);
        data += 4;
        break;
    }
  }
}
コード例 #13
0
gint
mve_compress_audio (guint8 * dest, const guint8 * src, guint16 len,
    guint8 channels)
{
  gint16 prev[2], s;
  gint delta, real_res;
  gint cur_chan;
  guint8 v;

  for (cur_chan = 0; cur_chan < channels; ++cur_chan) {
    prev[cur_chan] = GST_READ_UINT16_LE (src);
    GST_WRITE_UINT16_LE (dest, prev[cur_chan]);
    src += 2;
    dest += 2;
    len -= 2;
  }

  cur_chan = 0;
  while (len > 0) {
    s = GST_READ_UINT16_LE (src);
    src += 2;

    delta = s - prev[cur_chan];

    if (delta >= 0)

      v = mve_enc_delta (delta);

    else

      v = 256 - mve_enc_delta (-delta);


    real_res = dec_table[v] + prev[cur_chan];

    if (real_res < -32768 || real_res > 32767) {

      /* correct overflow */
      /* GST_DEBUG ("co:%d + %d = %d -> new v:%d, dec_table:%d will be %d",
         prev[cur_chan], dec_table[v], real_res,
         v, dec_table[v], prev[cur_chan]+dec_table[v]); */
      if (s > 0) {

        if (real_res > 32767)
          --v;

      } else {

        if (real_res < -32768)
          ++v;

      }

      real_res = dec_table[v] + prev[cur_chan];

    }

    if (G_UNLIKELY (abs (real_res - s) > 32767)) {
      GST_ERROR ("sign loss left unfixed in audio stream, deviation:%d",
          real_res - s);
      return -1;
    }


    *dest++ = v;

    --len;
    /* use previous output instead of input. That way output will not go too far from input. */
    prev[cur_chan] += dec_table[v];
    cur_chan = channels - 1 - cur_chan;

  }

  return 0;
}
コード例 #14
0
static GstFlowReturn
gst_avi_subtitle_parse_gab2_chunk (GstAviSubtitle * sub, GstBuffer * buf)
{
  const guint8 *data;
  gchar *name_utf8;
  guint name_length;
  guint file_length;
  guint size;

  data = GST_BUFFER_DATA (buf);
  size = GST_BUFFER_SIZE (buf);

  /* check the magic word "GAB2\0", and the next word must be 2 */
  if (size < 12 || memcmp (data, "GAB2\0\2\0", 5 + 2) != 0)
    goto wrong_magic_word;

  /* read 'name' of subtitle */
  name_length = GST_READ_UINT32_LE (data + 5 + 2);
  GST_LOG_OBJECT (sub, "length of name: %u", name_length);
  if (size <= 17 + name_length)
    goto wrong_name_length;

  name_utf8 = g_convert ((gchar *) data + 11, name_length, "UTF-8", "UTF-16LE",
      NULL, NULL, NULL);

  if (name_utf8) {
    GST_LOG_OBJECT (sub, "subtitle name: %s", name_utf8);
    gst_avi_subtitle_title_tag (sub, name_utf8);
    g_free (name_utf8);
  }

  /* next word must be 4 */
  if (GST_READ_UINT16_LE (data + 11 + name_length) != 0x4)
    goto wrong_fixed_word_2;

  file_length = GST_READ_UINT32_LE (data + 13 + name_length);
  GST_LOG_OBJECT (sub, "length srt/ssa file: %u", file_length);

  if (size < (17 + name_length + file_length))
    goto wrong_total_length;

  /* store this, so we can send it again after a seek; note that we shouldn't
   * assume all the remaining data in the chunk is subtitle data, there may
   * be padding at the end for some reason, so only parse file_length bytes */
  sub->subfile =
      gst_avi_subtitle_extract_file (sub, buf, 17 + name_length, file_length);

  if (sub->subfile == NULL)
    goto extract_failed;

  return GST_FLOW_OK;

  /* ERRORS */
wrong_magic_word:
  {
    GST_ELEMENT_ERROR (sub, STREAM, DECODE, (NULL), ("Wrong magic word"));
    return GST_FLOW_ERROR;
  }
wrong_name_length:
  {
    GST_ELEMENT_ERROR (sub, STREAM, DECODE, (NULL),
        ("name doesn't fit in buffer (%d < %d)", size, 17 + name_length));
    return GST_FLOW_ERROR;
  }
wrong_fixed_word_2:
  {
    GST_ELEMENT_ERROR (sub, STREAM, DECODE, (NULL),
        ("wrong fixed word: expected %u, got %u", 4,
            GST_READ_UINT16_LE (data + 11 + name_length)));
    return GST_FLOW_ERROR;
  }
wrong_total_length:
  {
    GST_ELEMENT_ERROR (sub, STREAM, DECODE, (NULL),
        ("buffer size is wrong: need %d bytes, have %d bytes",
            17 + name_length + file_length, size));
    return GST_FLOW_ERROR;
  }
extract_failed:
  {
    GST_ELEMENT_ERROR (sub, STREAM, DECODE, (NULL),
        ("could not extract subtitles"));
    return GST_FLOW_ERROR;
  }
}
コード例 #15
0
ファイル: gstivfparse.c プロジェクト: ylatuya/gst-plugins-bad
/* chain function
 * this function does the actual processing
 */
static GstFlowReturn
gst_ivf_parse_chain (GstPad * pad, GstBuffer * buf)
{
  GstIvfParse *ivf = GST_IVF_PARSE (GST_OBJECT_PARENT (pad));
  gboolean res;

  /* lazy creation of the adapter */
  if (G_UNLIKELY (ivf->adapter == NULL)) {
    ivf->adapter = gst_adapter_new ();
  }

  GST_LOG_OBJECT (ivf, "Pushing buffer of size %u to adapter",
      GST_BUFFER_SIZE (buf));

  gst_adapter_push (ivf->adapter, buf); /* adapter takes ownership of buf */

  res = GST_FLOW_OK;

  switch (ivf->state) {
    case GST_IVF_PARSE_START:
      if (gst_adapter_available (ivf->adapter) >= 32) {
        GstCaps *caps;

        const guint8 *data = gst_adapter_peek (ivf->adapter, 32);
        guint32 magic = GST_READ_UINT32_LE (data);
        guint16 version = GST_READ_UINT16_LE (data + 4);
        guint16 header_size = GST_READ_UINT16_LE (data + 6);
        guint32 fourcc = GST_READ_UINT32_LE (data + 8);
        guint16 width = GST_READ_UINT16_LE (data + 12);
        guint16 height = GST_READ_UINT16_LE (data + 14);
        guint32 rate_num = GST_READ_UINT32_LE (data + 16);
        guint32 rate_den = GST_READ_UINT32_LE (data + 20);
#ifndef GST_DISABLE_GST_DEBUG
        guint32 num_frames = GST_READ_UINT32_LE (data + 24);
#endif

        /* last 4 bytes unused */
        gst_adapter_flush (ivf->adapter, 32);

        if (magic != GST_MAKE_FOURCC ('D', 'K', 'I', 'F') ||
            version != 0 || header_size != 32 ||
            fourcc != GST_MAKE_FOURCC ('V', 'P', '8', '0')) {
          GST_ELEMENT_ERROR (ivf, STREAM, WRONG_TYPE, (NULL), (NULL));
          return GST_FLOW_ERROR;
        }

        /* create src pad caps */
        caps = gst_caps_new_simple ("video/x-vp8",
            "width", G_TYPE_INT, width, "height", G_TYPE_INT, height,
            "framerate", GST_TYPE_FRACTION, rate_num, rate_den, NULL);

        GST_INFO_OBJECT (ivf, "Found stream: %" GST_PTR_FORMAT, caps);

        GST_LOG_OBJECT (ivf, "Stream has %d frames", num_frames);

        gst_pad_set_caps (ivf->srcpad, caps);
        gst_caps_unref (caps);

        /* keep framerate in instance for convenience */
        ivf->rate_num = rate_num;
        ivf->rate_den = rate_den;

        gst_pad_push_event (ivf->srcpad, gst_event_new_new_segment (FALSE, 1.0,
                GST_FORMAT_TIME, 0, -1, 0));

        /* move along */
        ivf->state = GST_IVF_PARSE_DATA;
      } else {
        GST_LOG_OBJECT (ivf, "Header data not yet available.");
        break;
      }

      /* fall through */

    case GST_IVF_PARSE_DATA:
      while (gst_adapter_available (ivf->adapter) > 12) {
        const guint8 *data = gst_adapter_peek (ivf->adapter, 12);
        guint32 frame_size = GST_READ_UINT32_LE (data);
        guint64 frame_pts = GST_READ_UINT64_LE (data + 4);

        GST_LOG_OBJECT (ivf,
            "Read frame header: size %u, pts %" G_GUINT64_FORMAT, frame_size,
            frame_pts);

        if (gst_adapter_available (ivf->adapter) >= 12 + frame_size) {
          GstBuffer *frame;

          gst_adapter_flush (ivf->adapter, 12);

          frame = gst_adapter_take_buffer (ivf->adapter, frame_size);
          gst_buffer_set_caps (frame, GST_PAD_CAPS (ivf->srcpad));
          GST_BUFFER_TIMESTAMP (frame) =
              gst_util_uint64_scale_int (GST_SECOND * frame_pts, ivf->rate_den,
              ivf->rate_num);
          GST_BUFFER_DURATION (frame) =
              gst_util_uint64_scale_int (GST_SECOND, ivf->rate_den,
              ivf->rate_num);

          GST_DEBUG_OBJECT (ivf, "Pushing frame of size %u, ts %"
              GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT ", off %"
              G_GUINT64_FORMAT ", off_end %" G_GUINT64_FORMAT,
              GST_BUFFER_SIZE (frame),
              GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (frame)),
              GST_TIME_ARGS (GST_BUFFER_DURATION (frame)),
              GST_BUFFER_OFFSET (frame), GST_BUFFER_OFFSET_END (frame));

          res = gst_pad_push (ivf->srcpad, frame);
          if (res != GST_FLOW_OK)
            break;
        } else {
          GST_LOG_OBJECT (ivf, "Frame data not yet available.");
          break;
        }
      }
      break;

    default:
      g_return_val_if_reached (GST_FLOW_ERROR);
  }

  return res;
}
コード例 #16
0
/**
 * gst_riff_parse_strf_auds:
 * @element: caller element (used for debugging/error).
 * @buf: input data to be used for parsing, stripped from header.
 * @strf: a pointer (returned by this function) to a filled-in
 *        strf/auds structure. Caller should free it.
 * @data: a pointer (returned by this function) to a buffer
 *        containing extradata for this particular stream (e.g.
 *        codec initialization data).
 *
 * Parses an audio stream´s strf structure plus optionally some
 * extradata from input data. This function takes ownership of @buf.
 * use.
 *
 * Returns: TRUE if parsing succeeded, otherwise FALSE. The stream
 *          should be skipped on error, but it is not fatal.
 */
gboolean
gst_riff_parse_strf_auds (GstElement * element,
    GstBuffer * buf, gst_riff_strf_auds ** _strf, GstBuffer ** data)
{
  gst_riff_strf_auds *strf;
  guint bufsize;

  g_return_val_if_fail (buf != NULL, FALSE);
  g_return_val_if_fail (_strf != NULL, FALSE);
  g_return_val_if_fail (data != NULL, FALSE);

  bufsize = GST_BUFFER_SIZE (buf);

  if (bufsize < sizeof (gst_riff_strf_auds))
    goto too_small;

  strf = g_memdup (GST_BUFFER_DATA (buf), bufsize);

#if (G_BYTE_ORDER == G_BIG_ENDIAN)
  strf->format = GUINT16_FROM_LE (strf->format);
  strf->channels = GUINT16_FROM_LE (strf->channels);
  strf->rate = GUINT32_FROM_LE (strf->rate);
  strf->av_bps = GUINT32_FROM_LE (strf->av_bps);
  strf->blockalign = GUINT16_FROM_LE (strf->blockalign);
  strf->size = GUINT16_FROM_LE (strf->size);
#endif

  /* size checking */
  *data = NULL;
  if (bufsize > sizeof (gst_riff_strf_auds) + 2) {
    gint len;

    len = GST_READ_UINT16_LE (&GST_BUFFER_DATA (buf)[16]);
    if (len + 2 + sizeof (gst_riff_strf_auds) > bufsize) {
      GST_WARNING_OBJECT (element,
          "Extradata indicated %d bytes, but only %" G_GSSIZE_FORMAT
          " available", len, bufsize - 2 - sizeof (gst_riff_strf_auds));
      len = bufsize - 2 - sizeof (gst_riff_strf_auds);
    }
    if (len)
      *data = gst_buffer_create_sub (buf, sizeof (gst_riff_strf_auds) + 2, len);
  }

  /* debug */
  GST_INFO_OBJECT (element, "strf tag found in context auds:");
  GST_INFO_OBJECT (element, " format      %d", strf->format);
  GST_INFO_OBJECT (element, " channels    %d", strf->channels);
  GST_INFO_OBJECT (element, " rate        %d", strf->rate);
  GST_INFO_OBJECT (element, " av_bps      %d", strf->av_bps);
  GST_INFO_OBJECT (element, " blockalign  %d", strf->blockalign);
  GST_INFO_OBJECT (element, " size        %d", strf->size);
  if (*data)
    GST_INFO_OBJECT (element, " %d bytes extradata", GST_BUFFER_SIZE (*data));

  gst_buffer_unref (buf);

  *_strf = strf;

  return TRUE;

  /* ERROR */
too_small:
  {
    GST_ERROR_OBJECT (element,
        "Too small strf_auds (%d available, %" G_GSSIZE_FORMAT " needed)",
        bufsize, sizeof (gst_riff_strf_auds));
    gst_buffer_unref (buf);
    return FALSE;
  }
}
コード例 #17
0
static GstFlowReturn
gst_mve_video_create_buffer (GstMveDemux * mve, guint8 version,
    const guint8 * data, guint16 len)
{
  GstBuffer *buf;
  guint16 w, h, n, true_color, bpp;
  guint required, size;

  GST_DEBUG_OBJECT (mve, "create video buffer");

  if (mve->video_stream == NULL) {
    GST_ELEMENT_ERROR (mve, STREAM, DECODE, (NULL),
        ("trying to create video buffer for uninitialized stream"));
    return GST_FLOW_ERROR;
  }

  /* need 4 to 8 more bytes */
  required = (version > 1) ? 8 : (version * 2);
  if (len < required)
    return gst_mve_stream_error (mve, required, len);

  w = GST_READ_UINT16_LE (data) << 3;
  h = GST_READ_UINT16_LE (data + 2) << 3;

  if (version > 0)
    n = GST_READ_UINT16_LE (data + 4);
  else
    n = 1;

  if (version > 1)
    true_color = GST_READ_UINT16_LE (data + 6);
  else
    true_color = 0;

  bpp = (true_color ? 2 : 1);
  size = w * h * bpp;

  if (mve->video_stream->buffer != NULL) {
    GST_DEBUG_OBJECT (mve, "video buffer already created");

    if (GST_BUFFER_SIZE (mve->video_stream->buffer) == size * 2)
      return GST_FLOW_OK;

    GST_DEBUG_OBJECT (mve, "video buffer size has changed");
    gst_buffer_unref (mve->video_stream->buffer);
  }

  GST_DEBUG_OBJECT (mve,
      "allocating video buffer, w:%u, h:%u, n:%u, true_color:%u", w, h, n,
      true_color);

  /* we need a buffer to keep the last 2 frames, since those may be
     needed for decoding the next one */
  buf = gst_buffer_new_and_alloc (size * 2);

  mve->video_stream->bpp = bpp;
  mve->video_stream->width = w;
  mve->video_stream->height = h;
  mve->video_stream->buffer = buf;
  mve->video_stream->back_buf1 = GST_BUFFER_DATA (buf);
  mve->video_stream->back_buf2 = mve->video_stream->back_buf1 + size;
  mve->video_stream->max_block_offset = (h - 7) * w - 8;
  memset (mve->video_stream->back_buf1, 0, size * 2);

  return GST_FLOW_OK;
}
コード例 #18
0
static GstFlowReturn
gst_mve_video_data (GstMveDemux * mve, const guint8 * data, guint16 len,
    GstBuffer ** output)
{
  GstFlowReturn ret = GST_FLOW_OK;
  gint16 cur_frame, last_frame;
  gint16 x_offset, y_offset;
  gint16 x_size, y_size;
  guint16 flags;
  gint dec;
  GstBuffer *buf = NULL;
  GstMveDemuxStream *s = mve->video_stream;

  GST_LOG_OBJECT (mve, "video data");

  if (s == NULL) {
    GST_ELEMENT_ERROR (mve, STREAM, DECODE, (NULL),
        ("trying to decode video data before stream was initialized"));
    return GST_FLOW_ERROR;
  }

  if (GST_CLOCK_TIME_IS_VALID (mve->frame_duration)) {
    if (GST_CLOCK_TIME_IS_VALID (s->last_ts))
      s->last_ts += mve->frame_duration;
    else
      s->last_ts = 0;
  }

  if (!s->code_map_avail) {
    GST_ELEMENT_ERROR (mve, STREAM, DECODE, (NULL),
        ("no code map available for decoding"));
    return GST_FLOW_ERROR;
  }

  /* need at least 14 more bytes */
  if (len < 14)
    return gst_mve_stream_error (mve, 14, len);

  len -= 14;

  cur_frame = GST_READ_UINT16_LE (data);
  last_frame = GST_READ_UINT16_LE (data + 2);
  x_offset = GST_READ_UINT16_LE (data + 4);
  y_offset = GST_READ_UINT16_LE (data + 6);
  x_size = GST_READ_UINT16_LE (data + 8);
  y_size = GST_READ_UINT16_LE (data + 10);
  flags = GST_READ_UINT16_LE (data + 12);
  data += 14;

  GST_DEBUG_OBJECT (mve,
      "video data hot:%d, cold:%d, xoff:%d, yoff:%d, w:%d, h:%d, flags:%x",
      cur_frame, last_frame, x_offset, y_offset, x_size, y_size, flags);

  if (flags & MVE_VIDEO_DELTA_FRAME) {
    guint8 *temp = s->back_buf1;

    s->back_buf1 = s->back_buf2;
    s->back_buf2 = temp;
  }

  ret = gst_mve_buffer_alloc_for_pad (s, s->width * s->height * s->bpp, &buf);
  if (ret != GST_FLOW_OK)
    return ret;

  if (s->bpp == 2) {
    dec = ipvideo_decode_frame16 (s, data, len);
  } else {
    if (s->palette == NULL) {
      GST_ELEMENT_ERROR (mve, STREAM, DECODE, (NULL), ("no palette available"));
      goto error;
    }

    dec = ipvideo_decode_frame8 (s, data, len);
  }
  if (dec != 0)
    goto error;

  memcpy (GST_BUFFER_DATA (buf), s->back_buf1, GST_BUFFER_SIZE (buf));
  GST_BUFFER_DURATION (buf) = mve->frame_duration;
  GST_BUFFER_OFFSET_END (buf) = ++s->offset;

  if (s->bpp == 1) {
    GstCaps *caps;

    /* set the palette on the outgoing buffer */
    caps = gst_caps_copy (s->caps);
    gst_caps_set_simple (caps,
        "palette_data", GST_TYPE_BUFFER, s->palette, NULL);
    gst_buffer_set_caps (buf, caps);
    gst_caps_unref (caps);
  }

  *output = buf;
  return GST_FLOW_OK;

error:
  gst_buffer_unref (buf);
  return GST_FLOW_ERROR;
}
コード例 #19
0
static GstBuffer *
gst_rtp_vp8_depay_process (GstRTPBaseDepayload * depay, GstRTPBuffer * rtp)
{
  GstRtpVP8Depay *self = GST_RTP_VP8_DEPAY (depay);
  GstBuffer *payload;
  guint8 *data;
  guint hdrsize;
  guint size;

  if (G_UNLIKELY (GST_BUFFER_IS_DISCONT (rtp->buffer))) {
    GST_LOG_OBJECT (self, "Discontinuity, flushing adapter");
    gst_adapter_clear (self->adapter);
    self->started = FALSE;
  }

  size = gst_rtp_buffer_get_payload_len (rtp);

  /* At least one header and one vp8 byte */
  if (G_UNLIKELY (size < 2))
    goto too_small;

  data = gst_rtp_buffer_get_payload (rtp);

  if (G_UNLIKELY (!self->started)) {
    /* Check if this is the start of a VP8 frame, otherwise bail */
    /* S=1 and PartID= 0 */
    if ((data[0] & 0x17) != 0x10)
      goto done;

    self->started = TRUE;
  }

  hdrsize = 1;
  /* Check X optional header */
  if ((data[0] & 0x80) != 0) {
    hdrsize++;
    /* Check I optional header */
    if ((data[1] & 0x80) != 0) {
      if (G_UNLIKELY (size < 3))
        goto too_small;
      hdrsize++;
      /* Check for 16 bits PictureID */
      if ((data[2] & 0x80) != 0)
        hdrsize++;
    }
    /* Check L optional header */
    if ((data[1] & 0x40) != 0)
      hdrsize++;
    /* Check T or K optional headers */
    if ((data[1] & 0x20) != 0 || (data[1] & 0x10) != 0)
      hdrsize++;
  }
  GST_DEBUG_OBJECT (depay, "hdrsize %u, size %u", hdrsize, size);

  if (G_UNLIKELY (hdrsize >= size))
    goto too_small;

  payload = gst_rtp_buffer_get_payload_subbuffer (rtp, hdrsize, -1);
  gst_adapter_push (self->adapter, payload);

  /* Marker indicates that it was the last rtp packet for this frame */
  if (gst_rtp_buffer_get_marker (rtp)) {
    GstBuffer *out;
    guint8 header[10];

    gst_adapter_copy (self->adapter, &header, 0, 10);

    out = gst_adapter_take_buffer (self->adapter,
        gst_adapter_available (self->adapter));

    self->started = FALSE;

    /* mark keyframes */
    out = gst_buffer_make_writable (out);
    if ((header[0] & 0x01)) {
      GST_BUFFER_FLAG_SET (out, GST_BUFFER_FLAG_DELTA_UNIT);

      if (!self->caps_sent) {
        gst_buffer_unref (out);
        out = NULL;
        GST_INFO_OBJECT (self, "Dropping inter-frame before intra-frame");
        gst_pad_push_event (GST_RTP_BASE_DEPAYLOAD_SINKPAD (depay),
            gst_video_event_new_upstream_force_key_unit (GST_CLOCK_TIME_NONE,
                TRUE, 0));
      }
    } else {
      guint profile, width, height;

      GST_BUFFER_FLAG_UNSET (out, GST_BUFFER_FLAG_DELTA_UNIT);

      profile = (header[0] & 0x0e) >> 1;
      width = GST_READ_UINT16_LE (header + 6) & 0x3fff;
      height = GST_READ_UINT16_LE (header + 8) & 0x3fff;

      if (G_UNLIKELY (self->last_width != width ||
              self->last_height != height || self->last_profile != profile)) {
        gchar profile_str[3];
        GstCaps *srccaps;

        snprintf (profile_str, 3, "%u", profile);
        srccaps = gst_caps_new_simple ("video/x-vp8",
            "framerate", GST_TYPE_FRACTION, 0, 1,
            "height", G_TYPE_INT, height,
            "width", G_TYPE_INT, width,
            "profile", G_TYPE_STRING, profile_str, NULL);

        gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depay), srccaps);
        gst_caps_unref (srccaps);

        self->caps_sent = TRUE;
        self->last_width = width;
        self->last_height = height;
        self->last_profile = profile;
      }
    }

    return out;
  }

done:
  return NULL;

too_small:
  GST_LOG_OBJECT (self, "Invalid rtp packet (too small), ignoring");
  gst_adapter_clear (self->adapter);
  self->started = FALSE;

  goto done;
}
コード例 #20
0
ファイル: gstivfparse.c プロジェクト: 0p1pp1/gst-plugins-bad
static GstFlowReturn
gst_ivf_parse_handle_frame_data (GstIvfParse * ivf, GstBaseParseFrame * frame,
    gint * skipsize)
{
  GstBuffer *const buffer = frame->buffer;
  GstMapInfo map;
  GstFlowReturn ret = GST_FLOW_OK;
  GstBuffer *out_buffer;

  gst_buffer_map (buffer, &map, GST_MAP_READ);
  if (map.size >= IVF_FILE_HEADER_SIZE) {
    guint32 frame_size = GST_READ_UINT32_LE (map.data);
    guint64 frame_pts = GST_READ_UINT64_LE (map.data + 4);

    GST_LOG_OBJECT (ivf,
        "Read frame header: size %u, pts %" G_GUINT64_FORMAT, frame_size,
        frame_pts);

    if (map.size < IVF_FRAME_HEADER_SIZE + frame_size) {
      gst_base_parse_set_min_frame_size (GST_BASE_PARSE_CAST (ivf),
          IVF_FRAME_HEADER_SIZE + frame_size);
      gst_buffer_unmap (buffer, &map);
      *skipsize = 0;
      goto end;
    }

    gst_buffer_unmap (buffer, &map);

    /* Eventually, we would need the buffer memory in a merged state anyway */
    out_buffer = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_FLAGS |
        GST_BUFFER_COPY_TIMESTAMPS | GST_BUFFER_COPY_META |
        GST_BUFFER_COPY_MEMORY | GST_BUFFER_COPY_MERGE,
        IVF_FRAME_HEADER_SIZE, frame_size);
    if (!out_buffer) {
      GST_ERROR_OBJECT (ivf, "Failed to copy frame buffer");
      ret = GST_FLOW_ERROR;
      *skipsize = IVF_FRAME_HEADER_SIZE + frame_size;
      goto end;
    }
    gst_buffer_replace (&frame->out_buffer, out_buffer);
    gst_buffer_unref (out_buffer);

    /* Detect resolution changes on key frames */
    if (gst_buffer_map (frame->out_buffer, &map, GST_MAP_READ)) {
      guint32 width, height;

      if (ivf->fourcc == GST_MAKE_FOURCC ('V', 'P', '8', '0')) {
        guint32 frame_tag;
        frame_tag = GST_READ_UINT24_LE (map.data);
        if (!(frame_tag & 0x01) && map.size >= 10) {    /* key frame */
          GST_DEBUG_OBJECT (ivf, "key frame detected");

          width = GST_READ_UINT16_LE (map.data + 6) & 0x3fff;
          height = GST_READ_UINT16_LE (map.data + 8) & 0x3fff;
          gst_ivf_parse_set_size (ivf, width, height);
        }
      } else if (ivf->fourcc == GST_MAKE_FOURCC ('V', 'P', '9', '0')) {
        /* Fixme: Add vp9 frame header parsing? */
      } else if (ivf->fourcc == GST_MAKE_FOURCC ('A', 'V', '0', '1')) {
        /* Fixme: Add av1 frame header parsing? */
        /* This would allow to parse dynamic resolution changes */
        /* implement when gstav1parser is ready */
      }
      gst_buffer_unmap (frame->out_buffer, &map);
    }

    if (ivf->fps_n > 0) {
      GST_BUFFER_TIMESTAMP (out_buffer) =
          gst_util_uint64_scale_int (GST_SECOND * frame_pts, ivf->fps_d,
          ivf->fps_n);
    }

    gst_ivf_parse_update_src_caps (ivf);

    ret = gst_base_parse_finish_frame (GST_BASE_PARSE_CAST (ivf), frame,
        IVF_FRAME_HEADER_SIZE + frame_size);
    *skipsize = 0;
  } else {
    GST_LOG_OBJECT (ivf, "Frame data not yet available.");
    gst_buffer_unmap (buffer, &map);
    *skipsize = 0;
  }

end:
  return ret;
}
コード例 #21
0
static GstFlowReturn
gst_mve_timer_create (GstMveDemux * mve, const guint8 * data, guint16 len,
    GstBuffer ** buf)
{
  guint32 t_rate;
  guint16 t_subdiv;
  GstMveDemuxStream *s;
  GstTagList *list;
  gint rate_nom, rate_den;

  g_return_val_if_fail (mve->video_stream != NULL, GST_FLOW_ERROR);

  /* need 6 more bytes */
  if (len < 6)
    return gst_mve_stream_error (mve, 6, len);

  t_rate = GST_READ_UINT32_LE (data);
  t_subdiv = GST_READ_UINT16_LE (data + 4);

  GST_DEBUG_OBJECT (mve, "found timer:%ux%u", t_rate, t_subdiv);
  mve->frame_duration = t_rate * t_subdiv * GST_USECOND;

  /* now really start rolling... */
  s = mve->video_stream;

  if ((s->buffer == NULL) || (s->width == 0) || (s->height == 0)) {
    GST_ELEMENT_ERROR (mve, STREAM, DECODE, (NULL),
        ("missing or invalid create-video-buffer segment (%dx%d)",
            s->width, s->height));
    return GST_FLOW_ERROR;
  }

  if (s->pad != NULL) {
    if (s->caps != NULL) {
      gst_caps_unref (s->caps);
      s->caps = NULL;
    }
    if (s->code_map != NULL) {
      g_free (s->code_map);
      s->code_map = NULL;
    }
    list = NULL;
  } else {
    list = gst_tag_list_new ();
    gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
        GST_TAG_VIDEO_CODEC, "Raw RGB video", NULL);
  }

  s->caps = gst_caps_from_string ("video/x-raw-rgb");
  if (s->caps == NULL)
    return GST_FLOW_ERROR;

  rate_nom = GST_SECOND / GST_USECOND;
  rate_den = mve->frame_duration / GST_USECOND;

  gst_caps_set_simple (s->caps,
      "bpp", G_TYPE_INT, s->bpp * 8,
      "depth", G_TYPE_INT, (s->bpp == 1) ? 8 : 15,
      "width", G_TYPE_INT, s->width,
      "height", G_TYPE_INT, s->height,
      "framerate", GST_TYPE_FRACTION, rate_nom, rate_den,
      "endianness", G_TYPE_INT, G_BYTE_ORDER, NULL);
  if (s->bpp > 1) {
    gst_caps_set_simple (s->caps, "red_mask", G_TYPE_INT, 0x7C00,       /* 31744 */
        "green_mask", G_TYPE_INT, 0x03E0,       /*   992 */
        "blue_mask", G_TYPE_INT, 0x001F,        /*    31 */
        NULL);
  }

  s->code_map = g_malloc ((s->width * s->height) / (8 * 8 * 2));

  if (gst_mve_add_stream (mve, s, list))
    return gst_pad_push_event (s->pad,
        gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME,
            0, GST_CLOCK_TIME_NONE, 0)) ? GST_FLOW_OK : GST_FLOW_ERROR;
  else
    return GST_FLOW_OK;
}
コード例 #22
0
static GstFlowReturn
gst_mve_audio_data (GstMveDemux * mve, guint8 type, const guint8 * data,
    guint16 len, GstBuffer ** output)
{
  GstFlowReturn ret;
  GstMveDemuxStream *s = mve->audio_stream;
  GstBuffer *buf = NULL;
  guint16 stream_mask;
  guint16 size;

  GST_LOG_OBJECT (mve, "audio data");

  if (s == NULL) {
    GST_ELEMENT_ERROR (mve, STREAM, DECODE, (NULL),
        ("trying to queue samples with no audio stream"));
    return GST_FLOW_ERROR;
  }

  /* need at least 6 more bytes */
  if (len < 6)
    return gst_mve_stream_error (mve, 6, len);

  len -= 6;

  stream_mask = GST_READ_UINT16_LE (data + 2);
  size = GST_READ_UINT16_LE (data + 4);
  data += 6;

  if (stream_mask & MVE_DEFAULT_AUDIO_STREAM) {
    guint16 n_samples = size / s->n_channels / (s->sample_size / 8);
    GstClockTime duration = (GST_SECOND / s->sample_rate) * n_samples;

    if (type == MVE_OC_AUDIO_DATA) {
      guint16 required = (s->compression ? size / 2 + s->n_channels : size);

      if (len < required)
        return gst_mve_stream_error (mve, required, len);

      ret = gst_mve_buffer_alloc_for_pad (s, size, &buf);

      if (ret != GST_FLOW_OK)
        return ret;

      if (s->compression)
        ipaudio_uncompress ((gint16 *) GST_BUFFER_DATA (buf), size,
            data, s->n_channels);
      else
        memcpy (GST_BUFFER_DATA (buf), data, size);

      GST_DEBUG_OBJECT (mve, "created audio buffer, size:%u, stream_mask:%x",
          size, stream_mask);
    } else {
      /* silence - create a minimal buffer with no sound */
      size = s->n_channels * (s->sample_size / 8);
      ret = gst_mve_buffer_alloc_for_pad (s, size, &buf);
      memset (GST_BUFFER_DATA (buf), 0, size);
    }

    GST_BUFFER_DURATION (buf) = duration;
    GST_BUFFER_OFFSET_END (buf) = s->offset + n_samples;
    *output = buf;

    s->offset += n_samples;
    s->last_ts += duration;
  } else {
    /* alternate audio streams not supported.
       are there any movies which use them? */
    if (type == MVE_OC_AUDIO_DATA)
      GST_WARNING_OBJECT (mve, "found non-empty alternate audio stream");
  }

  return GST_FLOW_OK;
}
コード例 #23
0
static GstFlowReturn
gst_mve_audio_init (GstMveDemux * mve, guint8 version, const guint8 * data,
    guint16 len)
{
  GstMveDemuxStream *stream;
  guint16 flags;
  guint32 requested_buffer;
  GstTagList *list;
  gchar *name;

  GST_DEBUG_OBJECT (mve, "init audio");

  /* need 8 more bytes */
  if (len < 8)
    return gst_mve_stream_error (mve, 8, len);

  if (mve->audio_stream == NULL) {
    stream = g_new0 (GstMveDemuxStream, 1);
    stream->offset = 0;
    stream->last_ts = 0;
    stream->last_flow = GST_FLOW_OK;
    mve->audio_stream = stream;
  } else {
    stream = mve->audio_stream;
    gst_caps_unref (stream->caps);
  }

  flags = GST_READ_UINT16_LE (data + 2);
  stream->sample_rate = GST_READ_UINT16_LE (data + 4);
  requested_buffer = GST_READ_UINT32_LE (data + 6);

  /* bit 0: 0 = mono, 1 = stereo */
  stream->n_channels = (flags & MVE_AUDIO_STEREO) + 1;
  /* bit 1: 0 = 8 bit, 1 = 16 bit */
  stream->sample_size = (((flags & MVE_AUDIO_16BIT) >> 1) + 1) * 8;
  /* bit 2: 0 = uncompressed, 1 = compressed */
  stream->compression = ((version > 0) && (flags & MVE_AUDIO_COMPRESSED)) ?
      TRUE : FALSE;

  GST_DEBUG_OBJECT (mve, "audio init, sample_rate:%d, channels:%d, "
      "bits_per_sample:%d, compression:%d, buffer:%u",
      stream->sample_rate, stream->n_channels,
      stream->sample_size, stream->compression, requested_buffer);

  stream->caps = gst_caps_from_string ("audio/x-raw-int");
  if (stream->caps == NULL)
    return GST_FLOW_ERROR;

  gst_caps_set_simple (stream->caps,
      "signed", G_TYPE_BOOLEAN, (stream->sample_size == 8) ? FALSE : TRUE,
      "depth", G_TYPE_INT, stream->sample_size,
      "width", G_TYPE_INT, stream->sample_size,
      "channels", G_TYPE_INT, stream->n_channels,
      "rate", G_TYPE_INT, stream->sample_rate, NULL);
  if (stream->sample_size > 8) {
    /* for uncompressed audio we can simply copy the incoming buffer
       which is always in little endian format */
    gst_caps_set_simple (stream->caps, "endianness", G_TYPE_INT,
        (stream->compression ? G_BYTE_ORDER : G_LITTLE_ENDIAN), NULL);
  } else if (stream->compression) {
    GST_WARNING_OBJECT (mve,
        "compression is only supported for 16-bit samples");
    stream->compression = FALSE;
  }

  list = gst_tag_list_new ();
  name = g_strdup_printf ("Raw %d-bit PCM audio", stream->sample_size);
  gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
      GST_TAG_AUDIO_CODEC, name, NULL);
  g_free (name);

  if (gst_mve_add_stream (mve, stream, list))
    return gst_pad_push_event (mve->audio_stream->pad,
        gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME,
            0, GST_CLOCK_TIME_NONE, 0)) ? GST_FLOW_OK : GST_FLOW_ERROR;
  else
    return GST_FLOW_OK;
}
コード例 #24
0
static void
check_1x1_buffer (GstBuffer * buf, GstCaps * caps)
{
  GstVideoInfo info;
  GstVideoFrame frame;
  /* the exact values we check for come from videotestsrc */
  static const guint yuv_values[] = { 81, 90, 240, 255 };
  static const guint rgb_values[] = { 0xff, 0, 0, 255 };
  static const guint gray8_values[] = { 0x51 };
  static const guint gray16_values[] = { 0x5151 };
  const guint *values;
  guint i;
  const GstVideoFormatInfo *finfo;

  fail_unless (buf != NULL);
  fail_unless (caps != NULL);

  fail_unless (gst_video_info_from_caps (&info, caps));
  fail_unless (gst_video_frame_map (&frame, &info, buf, GST_MAP_READ));

  finfo = info.finfo;

  if (GST_VIDEO_INFO_IS_YUV (&info))
    values = yuv_values;
  else if (GST_VIDEO_INFO_IS_GRAY (&info))
    if (GST_VIDEO_FORMAT_INFO_BITS (finfo) == 8)
      values = gray8_values;
    else
      values = gray16_values;
  else
    values = rgb_values;

  GST_MEMDUMP ("buffer", GST_VIDEO_FRAME_PLANE_DATA (&frame, 0), 8);

  for (i = 0; i < GST_VIDEO_FRAME_N_COMPONENTS (&frame); i++) {
    guint8 *data = GST_VIDEO_FRAME_COMP_DATA (&frame, i);

    GST_DEBUG ("W: %d", GST_VIDEO_FORMAT_INFO_W_SUB (finfo, i));
    GST_DEBUG ("H: %d", GST_VIDEO_FORMAT_INFO_H_SUB (finfo, i));

    if (GST_VIDEO_FORMAT_INFO_W_SUB (finfo,
            i) >= GST_VIDEO_FRAME_WIDTH (&frame))
      continue;
    if (GST_VIDEO_FORMAT_INFO_H_SUB (finfo,
            i) >= GST_VIDEO_FRAME_HEIGHT (&frame))
      continue;

    if (GST_VIDEO_FORMAT_INFO_BITS (finfo) == 8) {
      fail_unless_equals_int (data[0], values[i]);
    } else if (GST_VIDEO_FORMAT_INFO_BITS (finfo) == 16) {
      guint16 pixels, val;
      gint depth;

      if (GST_VIDEO_FORMAT_INFO_IS_LE (finfo))
        pixels = GST_READ_UINT16_LE (data);
      else
        pixels = GST_READ_UINT16_BE (data);

      depth = GST_VIDEO_FORMAT_INFO_DEPTH (finfo, i);
      val = pixels >> GST_VIDEO_FORMAT_INFO_SHIFT (finfo, i);
      val = val & ((1 << depth) - 1);

      GST_DEBUG ("val %08x %d : %d", pixels, i, val);
      if (depth <= 8) {
        fail_unless_equals_int (val, values[i] >> (8 - depth));
      } else {
static GstFlowReturn
gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
{
  const guint8 *data = GST_BUFFER_DATA (buf);
  GstCaps *caps;
  const GstAudioChannelPosition *pos = NULL;

  if (!gst_opus_header_is_id_header (buf)) {
    GST_ERROR_OBJECT (dec, "Header is not an Opus ID header");
    return GST_FLOW_ERROR;
  }
  if (!(dec->n_channels == 0 || dec->n_channels == data[9])) {
    GST_ERROR_OBJECT (dec, "Opus ID header has invalid channels");
    return GST_FLOW_ERROR;
  }

  dec->n_channels = data[9];
  dec->pre_skip = GST_READ_UINT16_LE (data + 10);
  dec->r128_gain = GST_READ_UINT16_LE (data + 16);
  dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain);
  GST_INFO_OBJECT (dec,
      "Found pre-skip of %u samples, R128 gain %d (volume %f)",
      dec->pre_skip, dec->r128_gain, dec->r128_gain_volume);

  dec->channel_mapping_family = data[18];
  if (dec->channel_mapping_family == 0) {
    /* implicit mapping */
    GST_INFO_OBJECT (dec, "Channel mapping family 0, implicit mapping");
    dec->n_streams = dec->n_stereo_streams = 1;
    dec->channel_mapping[0] = 0;
    dec->channel_mapping[1] = 1;
  } else {
    dec->n_streams = data[19];
    dec->n_stereo_streams = data[20];
    memcpy (dec->channel_mapping, data + 21, dec->n_channels);

    if (dec->channel_mapping_family == 1) {
      GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
      switch (dec->n_channels) {
        case 1:
        case 2:
          /* nothing */
          break;
        case 3:
        case 4:
        case 5:
        case 6:
        case 7:
        case 8:
          pos = gst_opus_channel_positions[dec->n_channels - 1];
          break;
        default:{
          gint i;
          GstAudioChannelPosition *posn =
              g_new (GstAudioChannelPosition, dec->n_channels);

          GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
              (NULL), ("Using NONE channel layout for more than 8 channels"));

          for (i = 0; i < dec->n_channels; i++)
            posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE;

          pos = posn;
        }
      }
    } else {
      GST_INFO_OBJECT (dec, "Channel mapping family %d",
          dec->channel_mapping_family);
    }
  }

  caps = gst_opus_dec_negotiate (dec);

  if (pos) {
    GST_DEBUG_OBJECT (dec, "Setting channel positions on caps");
    gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
  }

  if (dec->n_channels > 8) {
    g_free ((GstAudioChannelPosition *) pos);
  }

  GST_INFO_OBJECT (dec, "Setting src caps to %" GST_PTR_FORMAT, caps);
  gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
  gst_caps_unref (caps);

  return GST_FLOW_OK;
}
コード例 #26
0
ファイル: gstopusparse.c プロジェクト: pexip/gst-plugins-bad
static GstFlowReturn
gst_opus_parse_parse_frame (GstBaseParse * base, GstBaseParseFrame * frame)
{
  guint64 duration;
  GstOpusParse *parse;
  gboolean is_idheader, is_commentheader;
  GstMapInfo map;
  GstAudioClippingMeta *cmeta =
      gst_buffer_get_audio_clipping_meta (frame->buffer);

  parse = GST_OPUS_PARSE (base);

  g_assert (!cmeta || cmeta->format == GST_FORMAT_DEFAULT);

  is_idheader = gst_opus_header_is_id_header (frame->buffer);
  is_commentheader = gst_opus_header_is_comment_header (frame->buffer);

  if (!parse->got_headers || !parse->header_sent) {
    GstCaps *caps;

    /* Opus streams can decode to 1 or 2 channels, so use the header
       value if we have one, or 2 otherwise */
    if (is_idheader) {
      gst_buffer_replace (&parse->id_header, frame->buffer);
      GST_DEBUG_OBJECT (parse, "Found ID header, keeping");
      return GST_BASE_PARSE_FLOW_DROPPED;
    } else if (is_commentheader) {
      gst_buffer_replace (&parse->comment_header, frame->buffer);
      GST_DEBUG_OBJECT (parse, "Found comment header, keeping");
      return GST_BASE_PARSE_FLOW_DROPPED;
    }

    parse->got_headers = TRUE;

    if (cmeta && cmeta->start) {
      parse->pre_skip += cmeta->start;

      gst_buffer_map (frame->buffer, &map, GST_MAP_READ);
      duration = packet_duration_opus (map.data, map.size);
      gst_buffer_unmap (frame->buffer, &map);

      /* Queue frame for later once we know all initial padding */
      if (duration == cmeta->start) {
        frame->flags |= GST_BASE_PARSE_FRAME_FLAG_QUEUE;
      }
    }

    if (!(frame->flags & GST_BASE_PARSE_FRAME_FLAG_QUEUE)) {
      if (FALSE && parse->id_header && parse->comment_header) {
        guint16 pre_skip;

        gst_buffer_map (parse->id_header, &map, GST_MAP_READWRITE);
        pre_skip = GST_READ_UINT16_LE (map.data + 10);
        if (pre_skip != parse->pre_skip) {
          GST_DEBUG_OBJECT (parse,
              "Fixing up pre-skip %u -> %" G_GUINT64_FORMAT, pre_skip,
              parse->pre_skip);
          GST_WRITE_UINT16_LE (map.data + 10, parse->pre_skip);
        }
        gst_buffer_unmap (parse->id_header, &map);

        caps =
            gst_codec_utils_opus_create_caps_from_header (parse->id_header,
            parse->comment_header);
      } else {
        GstCaps *sink_caps;
        guint32 sample_rate = 48000;
        guint8 n_channels, n_streams, n_stereo_streams, channel_mapping_family;
        guint8 channel_mapping[256];
        GstBuffer *id_header;

        sink_caps = gst_pad_get_current_caps (GST_BASE_PARSE_SINK_PAD (parse));
        if (!sink_caps
            || !gst_codec_utils_opus_parse_caps (sink_caps, &sample_rate,
                &n_channels, &channel_mapping_family, &n_streams,
                &n_stereo_streams, channel_mapping)) {
          GST_INFO_OBJECT (parse,
              "No headers and no caps, blindly setting up canonical stereo");
          n_channels = 2;
          n_streams = 1;
          n_stereo_streams = 1;
          channel_mapping_family = 0;
          channel_mapping[0] = 0;
          channel_mapping[1] = 1;
        }
        if (sink_caps)
          gst_caps_unref (sink_caps);

        id_header =
            gst_codec_utils_opus_create_header (sample_rate, n_channels,
            channel_mapping_family, n_streams, n_stereo_streams,
            channel_mapping, parse->pre_skip, 0);
        caps = gst_codec_utils_opus_create_caps_from_header (id_header, NULL);
        gst_buffer_unref (id_header);
      }

      gst_buffer_replace (&parse->id_header, NULL);
      gst_buffer_replace (&parse->comment_header, NULL);

      gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
      gst_caps_unref (caps);
      parse->header_sent = TRUE;
    }
  }

  GST_BUFFER_TIMESTAMP (frame->buffer) = parse->next_ts;

  gst_buffer_map (frame->buffer, &map, GST_MAP_READ);
  duration = packet_duration_opus (map.data, map.size);
  gst_buffer_unmap (frame->buffer, &map);
  parse->next_ts += duration;

  GST_BUFFER_DURATION (frame->buffer) = duration;
  GST_BUFFER_OFFSET_END (frame->buffer) =
      gst_util_uint64_scale (parse->next_ts, 48000, GST_SECOND);
  GST_BUFFER_OFFSET (frame->buffer) = parse->next_ts;

  return GST_FLOW_OK;
}
コード例 #27
0
static GstFlowReturn
gst_mve_demux_chain (GstPad * sinkpad, GstBuffer * inbuf)
{
  GstMveDemux *mve = GST_MVE_DEMUX (GST_PAD_PARENT (sinkpad));
  GstFlowReturn ret = GST_FLOW_OK;

  gst_adapter_push (mve->adapter, inbuf);

  GST_DEBUG_OBJECT (mve, "queuing buffer, needed:%d, available:%u",
      mve->needed_bytes, gst_adapter_available (mve->adapter));

  while ((gst_adapter_available (mve->adapter) >= mve->needed_bytes) &&
      (ret == GST_FLOW_OK)) {
    GstMveDemuxStream *stream = NULL;
    GstBuffer *outbuf = NULL;

    switch (mve->state) {
      case MVEDEMUX_STATE_INITIAL:
        gst_adapter_flush (mve->adapter, mve->needed_bytes);

        mve->chunk_offset += mve->needed_bytes;
        mve->needed_bytes = 4;
        mve->state = MVEDEMUX_STATE_NEXT_CHUNK;
        break;

      case MVEDEMUX_STATE_NEXT_CHUNK:{
        const guint8 *data;
        guint16 size;

        data = gst_adapter_peek (mve->adapter, mve->needed_bytes);
        size = GST_MVE_SEGMENT_SIZE (data);

        if (mve->chunk_offset >= mve->chunk_size) {
          /* new chunk, flush buffer and proceed with next segment */
          guint16 chunk_type = GST_READ_UINT16_LE (data + 2);

          gst_adapter_flush (mve->adapter, mve->needed_bytes);
          mve->chunk_size = size;
          mve->chunk_offset = 0;

          if (chunk_type > MVE_CHUNK_END) {
            GST_WARNING_OBJECT (mve,
                "skipping unknown chunk type 0x%02x of size:%u", chunk_type,
                size);
            mve->needed_bytes += size;
            mve->state = MVEDEMUX_STATE_SKIP;
          } else {
            GST_DEBUG_OBJECT (mve, "found new chunk type 0x%02x of size:%u",
                chunk_type, size);
          }
        } else if (mve->chunk_offset <= mve->chunk_size) {
          /* new segment */
          GST_DEBUG_OBJECT (mve, "found segment type 0x%02x of size:%u",
              GST_MVE_SEGMENT_TYPE (data), size);

          mve->needed_bytes += size;
          mve->state = MVEDEMUX_STATE_MOVIE;
        }
      }
        break;

      case MVEDEMUX_STATE_MOVIE:
        ret = gst_mve_parse_segment (mve, &stream, &outbuf);

        if ((ret == GST_FLOW_OK) && (outbuf != NULL)) {
          /* send buffer */
          GST_DEBUG_OBJECT (mve,
              "pushing buffer with time %" GST_TIME_FORMAT
              " (%u bytes) on pad %s",
              GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
              GST_BUFFER_SIZE (outbuf), GST_PAD_NAME (stream->pad));

          ret = gst_pad_push (stream->pad, outbuf);
          stream->last_flow = ret;
        }

        if (ret == GST_FLOW_NOT_LINKED) {
          if (mve->audio_stream
              && mve->audio_stream->last_flow != GST_FLOW_NOT_LINKED)
            ret = GST_FLOW_OK;
          if (mve->video_stream
              && mve->video_stream->last_flow != GST_FLOW_NOT_LINKED)
            ret = GST_FLOW_OK;
        }

        /* update current offset */
        mve->chunk_offset += mve->needed_bytes;

        mve->state = MVEDEMUX_STATE_NEXT_CHUNK;
        mve->needed_bytes = 4;
        break;

      case MVEDEMUX_STATE_SKIP:
        mve->chunk_offset += mve->needed_bytes;
        gst_adapter_flush (mve->adapter, mve->needed_bytes);
        mve->state = MVEDEMUX_STATE_NEXT_CHUNK;
        mve->needed_bytes = 4;
        break;

      default:
        GST_ERROR_OBJECT (mve, "invalid state: %d", mve->state);
        break;
    }
  }

  return ret;
}
コード例 #28
0
static GstFlowReturn
gst_mve_video_palette (GstMveDemux * mve, const guint8 * data, guint16 len)
{
  GstBuffer *buf;
  guint16 start, count;
  const guint8 *pal;
  guint32 *pal_ptr;
  gint i;

  GST_DEBUG_OBJECT (mve, "video palette");

  if (mve->video_stream == NULL) {
    GST_ELEMENT_ERROR (mve, STREAM, DECODE, (NULL),
        ("found palette before video stream was initialized"));
    return GST_FLOW_ERROR;
  }

  /* need 4 more bytes now, more later */
  if (len < 4)
    return gst_mve_stream_error (mve, 4, len);

  len -= 4;

  start = GST_READ_UINT16_LE (data);
  count = GST_READ_UINT16_LE (data + 2);
  GST_DEBUG_OBJECT (mve, "found palette start:%u, count:%u", start, count);

  /* need more bytes */
  if (len < count * 3)
    return gst_mve_stream_error (mve, count * 3, len);

  /* make sure we don't exceed the buffer */
  if (start + count > MVE_PALETTE_COUNT) {
    GST_ELEMENT_ERROR (mve, STREAM, DECODE, (NULL),
        ("palette too large for buffer"));
    return GST_FLOW_ERROR;
  }

  if (mve->video_stream->palette != NULL) {
    /* older buffers floating around might still use the old
       palette, so make sure we can update it */
    buf = gst_buffer_make_writable (mve->video_stream->palette);
  } else {
    buf = gst_buffer_new_and_alloc (MVE_PALETTE_COUNT * 4);
    memset (GST_BUFFER_DATA (buf), 0, GST_BUFFER_SIZE (buf));
  }

  mve->video_stream->palette = buf;

  pal = data + 4;
  pal_ptr = ((guint32 *) GST_BUFFER_DATA (buf)) + start;
  for (i = 0; i < count; ++i) {
    /* convert from 6-bit VGA to 8-bit palette */
    guint8 r, g, b;

    r = (*pal) << 2;
    ++pal;
    g = (*pal) << 2;
    ++pal;
    b = (*pal) << 2;
    ++pal;
    *pal_ptr = (r << 16) | (g << 8) | (b);
    ++pal_ptr;
  }
  return GST_FLOW_OK;
}
コード例 #29
0
static GstFlowReturn
gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
{
  const guint8 *data;
  GstAudioChannelPosition pos[64];
  const GstAudioChannelPosition *posn = NULL;
  GstMapInfo map;

  if (!gst_opus_header_is_id_header (buf)) {
    GST_ERROR_OBJECT (dec, "Header is not an Opus ID header");
    return GST_FLOW_ERROR;
  }

  gst_buffer_map (buf, &map, GST_MAP_READ);
  data = map.data;

  if (!(dec->n_channels == 0 || dec->n_channels == data[9])) {
    gst_buffer_unmap (buf, &map);
    GST_ERROR_OBJECT (dec, "Opus ID header has invalid channels");
    return GST_FLOW_ERROR;
  }

  dec->n_channels = data[9];
  dec->sample_rate = GST_READ_UINT32_LE (data + 12);
  dec->pre_skip = GST_READ_UINT16_LE (data + 10);
  dec->r128_gain = GST_READ_UINT16_LE (data + 16);
  dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain);
  GST_INFO_OBJECT (dec,
      "Found pre-skip of %u samples, R128 gain %d (volume %f)",
      dec->pre_skip, dec->r128_gain, dec->r128_gain_volume);

  dec->channel_mapping_family = data[18];
  if (dec->channel_mapping_family == 0) {
    /* implicit mapping */
    GST_INFO_OBJECT (dec, "Channel mapping family 0, implicit mapping");
    dec->n_streams = dec->n_stereo_streams = 1;
    dec->channel_mapping[0] = 0;
    dec->channel_mapping[1] = 1;
  } else {
    dec->n_streams = data[19];
    dec->n_stereo_streams = data[20];
    memcpy (dec->channel_mapping, data + 21, dec->n_channels);

    if (dec->channel_mapping_family == 1) {
      GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
      switch (dec->n_channels) {
        case 1:
        case 2:
          /* nothing */
          break;
        case 3:
        case 4:
        case 5:
        case 6:
        case 7:
        case 8:
          posn = gst_opus_channel_positions[dec->n_channels - 1];
          break;
        default:{
          gint i;

          GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
              (NULL), ("Using NONE channel layout for more than 8 channels"));

          for (i = 0; i < dec->n_channels; i++)
            pos[i] = GST_AUDIO_CHANNEL_POSITION_NONE;

          posn = pos;
        }
      }
    } else {
      GST_INFO_OBJECT (dec, "Channel mapping family %d",
          dec->channel_mapping_family);
    }
  }

  gst_opus_dec_negotiate (dec, posn);

  gst_buffer_unmap (buf, &map);

  return GST_FLOW_OK;
}
コード例 #30
0
static GstFlowReturn
gst_tta_parse_parse_header (GstTtaParse * ttaparse)
{
  guchar *data;
  GstBuffer *buf = NULL;
  guint32 crc;
  double frame_length;
  int num_frames;
  GstCaps *caps;
  int i;
  guint32 offset;
  GstEvent *discont;

  if (gst_pad_pull_range (ttaparse->sinkpad, 0, 22, &buf) != GST_FLOW_OK)
    goto pull_fail;
  data = GST_BUFFER_DATA (buf);
  ttaparse->channels = GST_READ_UINT16_LE (data + 6);
  ttaparse->bits = GST_READ_UINT16_LE (data + 8);
  ttaparse->samplerate = GST_READ_UINT32_LE (data + 10);
  ttaparse->data_length = GST_READ_UINT32_LE (data + 14);
  crc = crc32 (data, 18);
  if (crc != GST_READ_UINT32_LE (data + 18)) {
    GST_DEBUG ("Header CRC wrong!");
  }
  frame_length = FRAME_TIME * ttaparse->samplerate;
  num_frames = (ttaparse->data_length / frame_length) + 1;
  ttaparse->num_frames = num_frames;
  gst_buffer_unref (buf);

  ttaparse->index =
      (GstTtaIndex *) g_malloc (num_frames * sizeof (GstTtaIndex));
  if (gst_pad_pull_range (ttaparse->sinkpad,
          22, num_frames * 4 + 4, &buf) != GST_FLOW_OK)
    goto pull_fail;
  data = GST_BUFFER_DATA (buf);

  offset = 22 + num_frames * 4 + 4;     // header size + seektable size
  for (i = 0; i < num_frames; i++) {
    ttaparse->index[i].size = GST_READ_UINT32_LE (data + i * 4);
    ttaparse->index[i].pos = offset;
    offset += ttaparse->index[i].size;
    ttaparse->index[i].time = i * FRAME_TIME * GST_SECOND;
  }
  crc = crc32 (data, num_frames * 4);
  if (crc != GST_READ_UINT32_LE (data + num_frames * 4)) {
    GST_DEBUG ("Seektable CRC wrong!");
  }

  GST_DEBUG
      ("channels: %u, bits: %u, samplerate: %u, data_length: %u, num_frames: %u",
      ttaparse->channels, ttaparse->bits, ttaparse->samplerate,
      ttaparse->data_length, num_frames);

  ttaparse->header_parsed = TRUE;
  caps = gst_caps_new_simple ("audio/x-tta",
      "width", G_TYPE_INT, ttaparse->bits,
      "channels", G_TYPE_INT, ttaparse->channels,
      "rate", G_TYPE_INT, ttaparse->samplerate, NULL);
  gst_pad_set_caps (ttaparse->srcpad, caps);

  discont =
      gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0,
      num_frames * FRAME_TIME * GST_SECOND, 0);

  gst_pad_push_event (ttaparse->srcpad, discont);

  return GST_FLOW_OK;

pull_fail:
  {
    GST_ELEMENT_ERROR (ttaparse, STREAM, DEMUX, (NULL),
        ("Couldn't read header"));
    return GST_FLOW_ERROR;
  }
}