Exemplo n.º 1
0
static GstFlowReturn
gst_siren_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
{
  GstSirenDec *dec;
  GstFlowReturn ret = GST_FLOW_OK;
  GstBuffer *out_buf;
  guint8 *in_data, *out_data;
  guint i, size, num_frames;
  gint out_size, in_size;
  gint decode_ret;
  GstMapInfo inmap, outmap;

  dec = GST_SIREN_DEC (bdec);

  size = gst_buffer_get_size (buf);

  GST_LOG_OBJECT (dec, "Received buffer of size %u", size);

  g_return_val_if_fail (size % 40 == 0, GST_FLOW_ERROR);
  g_return_val_if_fail (size > 0, GST_FLOW_ERROR);

  /* process 40 input bytes into 640 output bytes */
  num_frames = size / 40;

  /* this is the input/output size */
  in_size = num_frames * 40;
  out_size = num_frames * 640;

  GST_LOG_OBJECT (dec, "we have %u frames, %u in, %u out", num_frames, in_size,
      out_size);

  out_buf = gst_audio_decoder_allocate_output_buffer (bdec, out_size);
  if (out_buf == NULL)
    goto alloc_failed;

  /* get the input data for all the frames */
  gst_buffer_map (buf, &inmap, GST_MAP_READ);
  gst_buffer_map (out_buf, &outmap, GST_MAP_WRITE);

  in_data = inmap.data;
  out_data = outmap.data;

  for (i = 0; i < num_frames; i++) {
    GST_LOG_OBJECT (dec, "Decoding frame %u/%u", i, num_frames);

    /* decode 40 input bytes to 640 output bytes */
    decode_ret = Siren7_DecodeFrame (dec->decoder, in_data, out_data);
    if (decode_ret != 0)
      goto decode_error;

    /* move to next frame */
    out_data += 640;
    in_data += 40;
  }

  gst_buffer_unmap (buf, &inmap);
  gst_buffer_unmap (out_buf, &outmap);

  GST_LOG_OBJECT (dec, "Finished decoding");

  /* might really be multiple frames,
   * but was treated as one for all purposes here */
  ret = gst_audio_decoder_finish_frame (bdec, out_buf, 1);

done:
  return ret;

  /* ERRORS */
alloc_failed:
  {
    GST_DEBUG_OBJECT (dec, "failed to pad_alloc buffer: %d (%s)", ret,
        gst_flow_get_name (ret));
    goto done;
  }
decode_error:
  {
    GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
        ("Error decoding frame: %d", decode_ret), ret);
    if (ret == GST_FLOW_OK)
      gst_audio_decoder_finish_frame (bdec, NULL, 1);
    gst_buffer_unref (out_buf);
    goto done;
  }
}
Exemplo n.º 2
0
static GstFlowReturn
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
{
  GstFlowReturn res = GST_FLOW_OK;
  gsize size;
  guint8 *data;
  GstBuffer *outbuf;
  gint16 *out_data;
  int n, err;
  int samples;
  unsigned int packet_size;
  GstBuffer *buf;
  GstMapInfo map, omap;

  if (dec->state == NULL) {
    /* If we did not get any headers, default to 2 channels */
    if (dec->n_channels == 0) {
      GST_INFO_OBJECT (dec, "No header, assuming single stream");
      dec->n_channels = 2;
      dec->sample_rate = 48000;
      /* default stereo mapping */
      dec->channel_mapping_family = 0;
      dec->channel_mapping[0] = 0;
      dec->channel_mapping[1] = 1;
      dec->n_streams = 1;
      dec->n_stereo_streams = 1;

      gst_opus_dec_negotiate (dec, NULL);
    }

    GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
        dec->n_channels, dec->sample_rate);
#ifndef GST_DISABLE_DEBUG
    gst_opus_common_log_channel_mapping_table (GST_ELEMENT (dec), opusdec_debug,
        "Mapping table", dec->n_channels, dec->channel_mapping);
#endif

    GST_DEBUG_OBJECT (dec, "%d streams, %d stereo", dec->n_streams,
        dec->n_stereo_streams);
    dec->state =
        opus_multistream_decoder_create (dec->sample_rate, dec->n_channels,
        dec->n_streams, dec->n_stereo_streams, dec->channel_mapping, &err);
    if (!dec->state || err != OPUS_OK)
      goto creation_failed;
  }

  if (buffer) {
    GST_DEBUG_OBJECT (dec, "Received buffer of size %u",
        gst_buffer_get_size (buffer));
  } else {
    GST_DEBUG_OBJECT (dec, "Received missing buffer");
  }

  /* if using in-band FEC, we introdude one extra frame's delay as we need
     to potentially wait for next buffer to decode a missing buffer */
  if (dec->use_inband_fec && !dec->primed) {
    GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out");
    gst_buffer_replace (&dec->last_buffer, buffer);
    dec->primed = TRUE;
    goto done;
  }

  /* That's the buffer we'll be sending to the opus decoder. */
  buf = (dec->use_inband_fec
      && gst_buffer_get_size (dec->last_buffer) >
      0) ? dec->last_buffer : buffer;

  if (buf && gst_buffer_get_size (buf) > 0) {
    gst_buffer_map (buf, &map, GST_MAP_READ);
    data = map.data;
    size = map.size;
    GST_DEBUG_OBJECT (dec, "Using buffer of size %u", size);
  } else {
    /* concealment data, pass NULL as the bits parameters */
    GST_DEBUG_OBJECT (dec, "Using NULL buffer");
    data = NULL;
    size = 0;
  }

  /* use maximum size (120 ms) as the number of returned samples is
     not constant over the stream. */
  samples = 120 * dec->sample_rate / 1000;
  packet_size = samples * dec->n_channels * 2;

  outbuf = gst_buffer_new_and_alloc (packet_size);
  if (!outbuf) {
    goto buffer_failed;
  }

  gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
  out_data = (gint16 *) omap.data;

  if (dec->use_inband_fec) {
    if (dec->last_buffer) {
      /* normal delayed decode */
      GST_LOG_OBJECT (dec, "FEC enabled, decoding last delayed buffer");
      n = opus_multistream_decode (dec->state, data, size, out_data, samples,
          0);
    } else {
      /* FEC reconstruction decode */
      GST_LOG_OBJECT (dec, "FEC enabled, reconstructing last buffer");
      n = opus_multistream_decode (dec->state, data, size, out_data, samples,
          1);
    }
  } else {
    /* normal decode */
    GST_LOG_OBJECT (dec, "FEC disabled, decoding buffer");
    n = opus_multistream_decode (dec->state, data, size, out_data, samples, 0);
  }
  gst_buffer_unmap (outbuf, &omap);
  if (buf)
    gst_buffer_unmap (buf, &map);

  if (n < 0) {
    GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
    gst_buffer_unref (outbuf);
    return GST_FLOW_ERROR;
  }
  GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
  gst_buffer_set_size (outbuf, n * 2 * dec->n_channels);

  /* Skip any samples that need skipping */
  if (dec->pre_skip > 0) {
    guint scaled_pre_skip = dec->pre_skip * dec->sample_rate / 48000;
    guint skip = scaled_pre_skip > n ? n : scaled_pre_skip;
    guint scaled_skip = skip * 48000 / dec->sample_rate;

    gst_buffer_resize (outbuf, skip * 2 * dec->n_channels, -1);
    dec->pre_skip -= scaled_skip;
    GST_INFO_OBJECT (dec,
        "Skipping %u samples (%u at 48000 Hz, %u left to skip)", skip,
        scaled_skip, dec->pre_skip);
  }

  if (gst_buffer_get_size (outbuf) == 0) {
    gst_buffer_unref (outbuf);
    outbuf = NULL;
  } else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) {
    gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16,
        dec->n_channels, dec->opus_pos, dec->info.position);
  }

  /* Apply gain */
  /* Would be better off leaving this to a volume element, as this is
     a naive conversion that does too many int/float conversions.
     However, we don't have control over the pipeline...
     So make it optional if the user program wants to use a volume,
     but do it by default so the correct volume goes out by default */
  if (dec->apply_gain && outbuf && dec->r128_gain) {
    gsize rsize;
    unsigned int i, nsamples;
    double volume = dec->r128_gain_volume;
    gint16 *samples;

    gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE);
    samples = (gint16 *) omap.data;
    rsize = omap.size;
    GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
    nsamples = rsize / 2;
    for (i = 0; i < nsamples; ++i) {
      int sample = (int) (samples[i] * volume + 0.5);
      samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
    }
    gst_buffer_unmap (outbuf, &omap);
  }

  if (dec->use_inband_fec) {
    gst_buffer_replace (&dec->last_buffer, buffer);
  }

  res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);

  if (res != GST_FLOW_OK)
    GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));

done:
  return res;

creation_failed:
  GST_ERROR_OBJECT (dec, "Failed to create Opus decoder: %d", err);
  return GST_FLOW_ERROR;

buffer_failed:
  GST_ERROR_OBJECT (dec, "Failed to create %u byte buffer", packet_size);
  return GST_FLOW_ERROR;
}
Exemplo n.º 3
0
static gboolean
gst_bz2enc_event (GstPad * pad, GstObject * parent, GstEvent * e)
{
  GstBz2enc *b;
  gboolean ret;

  b = GST_BZ2ENC (parent);
  switch (GST_EVENT_TYPE (e)) {
    case GST_EVENT_EOS:{
      GstFlowReturn flow;
      int r = BZ_FINISH_OK;

      do {
        GstBuffer *out;
        GstMapInfo omap;
        guint n;

        out = gst_buffer_new_and_alloc (b->buffer_size);

        gst_buffer_map (out, &omap, GST_MAP_WRITE);
        b->stream.next_out = (char *) omap.data;
        b->stream.avail_out = omap.size;
        r = BZ2_bzCompress (&b->stream, BZ_FINISH);
        gst_buffer_unmap (out, &omap);
        if ((r != BZ_FINISH_OK) && (r != BZ_STREAM_END)) {
          GST_ELEMENT_ERROR (b, STREAM, ENCODE, (NULL),
              ("Failed to finish to compress (error code %i).", r));
          gst_buffer_unref (out);
          break;
        }

        n = gst_buffer_get_size (out);
        if (b->stream.avail_out >= n) {
          gst_buffer_unref (out);
          break;
        }

        gst_buffer_resize (out, 0, n - b->stream.avail_out);
        n = gst_buffer_get_size (out);
        GST_BUFFER_OFFSET (out) = b->stream.total_out_lo32 - n;

        flow = gst_pad_push (b->src, out);

        if (flow != GST_FLOW_OK) {
          GST_DEBUG_OBJECT (b, "push on EOS failed: %s",
              gst_flow_get_name (flow));
          break;
        }
      } while (r != BZ_STREAM_END);

      ret = gst_pad_event_default (pad, parent, e);

      if (r != BZ_STREAM_END || flow != GST_FLOW_OK)
        ret = FALSE;

      gst_bz2enc_compress_init (b);
      break;
    }
    default:
      ret = gst_pad_event_default (pad, parent, e);
      break;
  }

  return ret;
}
/* Reads in buffers, parses them, reframes into one-buffer-per-ogg-page, submits
 * pages to output pad.
 */
static GstFlowReturn
gst_ogg_parse_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
  GstOggParse *ogg;
  GstFlowReturn result = GST_FLOW_OK;
  gint ret = -1;
  guint32 serialno;
  GstBuffer *pagebuffer;
  GstClockTime buffertimestamp = GST_BUFFER_TIMESTAMP (buffer);

  ogg = GST_OGG_PARSE (parent);

  GST_LOG_OBJECT (ogg,
      "Chain function received buffer of size %" G_GSIZE_FORMAT,
      gst_buffer_get_size (buffer));

  gst_ogg_parse_submit_buffer (ogg, buffer);

  while (ret != 0 && result == GST_FLOW_OK) {
    ogg_page page;

    /* We use ogg_sync_pageseek() rather than ogg_sync_pageout() so that we can
     * track how many bytes the ogg layer discarded (in the case of sync errors,
     * etc.); this allows us to accurately track the current stream offset
     */
    ret = ogg_sync_pageseek (&ogg->sync, &page);
    if (ret == 0) {
      /* need more data, that's fine... */
      break;
    } else if (ret < 0) {
      /* discontinuity; track how many bytes we skipped (-ret) */
      ogg->offset -= ret;
    } else {
      gint64 granule = ogg_page_granulepos (&page);
#ifndef GST_DISABLE_GST_DEBUG
      int bos = ogg_page_bos (&page);
#endif
      guint64 startoffset = ogg->offset;
      GstOggStream *stream;
      gboolean keyframe;

      serialno = ogg_page_serialno (&page);
      stream = gst_ogg_parse_find_stream (ogg, serialno);

      GST_LOG_OBJECT (ogg, "Timestamping outgoing buffer as %" GST_TIME_FORMAT,
          GST_TIME_ARGS (buffertimestamp));

      if (stream) {
        buffertimestamp = gst_ogg_stream_get_end_time_for_granulepos (stream,
            granule);
        if (ogg->video_stream) {
          if (stream == ogg->video_stream) {
            keyframe = gst_ogg_stream_granulepos_is_key_frame (stream, granule);
          } else {
            keyframe = FALSE;
          }
        } else {
          keyframe = TRUE;
        }
      } else {
        buffertimestamp = GST_CLOCK_TIME_NONE;
        keyframe = TRUE;
      }
      pagebuffer = gst_ogg_parse_buffer_from_page (&page, startoffset,
          buffertimestamp);

      /* We read out 'ret' bytes, so we set the next offset appropriately */
      ogg->offset += ret;

      GST_LOG_OBJECT (ogg,
          "processing ogg page (serial %08x, pageno %ld, "
          "granule pos %" G_GUINT64_FORMAT ", bos %d, offset %"
          G_GUINT64_FORMAT "-%" G_GUINT64_FORMAT ") keyframe=%d",
          serialno, ogg_page_pageno (&page),
          granule, bos, startoffset, ogg->offset, keyframe);

      if (ogg_page_bos (&page)) {
        /* If we've seen this serialno before, this is technically an error,
         * we log this case but accept it - this one replaces the previous
         * stream with this serialno. We can do this since we're streaming, and
         * not supporting seeking...
         */
        GstOggStream *stream = gst_ogg_parse_find_stream (ogg, serialno);

        if (stream != NULL) {
          GST_LOG_OBJECT (ogg, "Incorrect stream; repeats serial number %08x "
              "at offset %" G_GINT64_FORMAT, serialno, ogg->offset);
        }

        if (ogg->last_page_not_bos) {
          GST_LOG_OBJECT (ogg, "Deleting all referenced streams, found a new "
              "chain starting with serial %u", serialno);
          gst_ogg_parse_delete_all_streams (ogg);
        }

        stream = gst_ogg_parse_new_stream (ogg, &page);

        ogg->last_page_not_bos = FALSE;

        gst_buffer_ref (pagebuffer);
        stream->headers = g_list_append (stream->headers, pagebuffer);

        if (!ogg->in_headers) {
          GST_LOG_OBJECT (ogg,
              "Found start of new chain at offset %" G_GUINT64_FORMAT,
              startoffset);
          ogg->in_headers = 1;
        }

        /* For now, we just keep the header buffer in the stream->headers list;
         * it actually gets output once we've collected the entire set
         */
      } else {
        /* Non-BOS page. Either: we're outside headers, and this isn't a 
         * header (normal data), outside headers and this is (error!), inside
         * headers, this is (append header), or inside headers and this isn't 
         * (we've found the end of headers; flush the lot!)
         *
         * Before that, we flag that the last page seen (this one) was not a 
         * BOS page; that way we know that when we next see a BOS page it's a
         * new chain, and we can flush all existing streams.
         */
        page_type type;
        GstOggStream *stream = gst_ogg_parse_find_stream (ogg, serialno);

        if (!stream) {
          GST_LOG_OBJECT (ogg,
              "Non-BOS page unexpectedly found at %" G_GINT64_FORMAT,
              ogg->offset);
          goto failure;
        }

        ogg->last_page_not_bos = TRUE;

        type = gst_ogg_parse_is_header (ogg, stream, &page);

        if (type == PAGE_PENDING && ogg->in_headers) {
          gst_buffer_ref (pagebuffer);

          stream->unknown_pages = g_list_append (stream->unknown_pages,
              pagebuffer);
        } else if (type == PAGE_HEADER) {
          if (!ogg->in_headers) {
            GST_LOG_OBJECT (ogg, "Header page unexpectedly found outside "
                "headers at offset %" G_GINT64_FORMAT, ogg->offset);
            goto failure;
          } else {
            /* Append the header to the buffer list, after any unknown previous
             * pages
             */
            stream->headers = g_list_concat (stream->headers,
                stream->unknown_pages);
            g_list_free (stream->unknown_pages);
            gst_buffer_ref (pagebuffer);
            stream->headers = g_list_append (stream->headers, pagebuffer);
          }
        } else {                /* PAGE_DATA, or PAGE_PENDING but outside headers */
          if (ogg->in_headers) {
            /* First non-header page... set caps, flush headers.
             *
             * First up, we build a single GValue list of all the pagebuffers
             * we're using for the headers, in order.
             * Then we set this on the caps structure. Then we can start pushing
             * buffers for the headers, and finally we send this non-header
             * page.
             */
            GstCaps *caps;
            GstStructure *structure;
            GValue array = { 0 };
            gint count = 0;
            gboolean found_pending_headers = FALSE;
            GSList *l;

            g_value_init (&array, GST_TYPE_ARRAY);

            for (l = ogg->oggstreams; l != NULL; l = l->next) {
              GstOggStream *stream = (GstOggStream *) l->data;

              if (g_list_length (stream->headers) == 0) {
                GST_LOG_OBJECT (ogg, "No primary header found for stream %08x",
                    stream->serialno);
                goto failure;
              }

              gst_ogg_parse_append_header (&array,
                  GST_BUFFER (stream->headers->data));
              count++;
            }

            for (l = ogg->oggstreams; l != NULL; l = l->next) {
              GstOggStream *stream = (GstOggStream *) l->data;
              GList *j;

              /* already appended the first header, now do headers 2-N */
              for (j = stream->headers->next; j != NULL; j = j->next) {
                gst_ogg_parse_append_header (&array, GST_BUFFER (j->data));
                count++;
              }
            }

            caps = gst_pad_query_caps (ogg->srcpad, NULL);
            caps = gst_caps_make_writable (caps);

            structure = gst_caps_get_structure (caps, 0);
            gst_structure_take_value (structure, "streamheader", &array);

            gst_pad_set_caps (ogg->srcpad, caps);

            if (ogg->caps)
              gst_caps_unref (ogg->caps);
            ogg->caps = caps;

            GST_LOG_OBJECT (ogg, "Set \"streamheader\" caps with %d buffers "
                "(one per page)", count);

            /* Now, we do the same thing, but push buffers... */
            for (l = ogg->oggstreams; l != NULL; l = l->next) {
              GstOggStream *stream = (GstOggStream *) l->data;
              GstBuffer *buf = GST_BUFFER (stream->headers->data);

              result = gst_pad_push (ogg->srcpad, buf);
              if (result != GST_FLOW_OK)
                return result;
            }
            for (l = ogg->oggstreams; l != NULL; l = l->next) {
              GstOggStream *stream = (GstOggStream *) l->data;
              GList *j;

              /* pushed the first one for each stream already, now do 2-N */
              for (j = stream->headers->next; j != NULL; j = j->next) {
                GstBuffer *buf = GST_BUFFER (j->data);

                result = gst_pad_push (ogg->srcpad, buf);
                if (result != GST_FLOW_OK)
                  return result;
              }
            }

            ogg->in_headers = 0;

            /* And finally the pending data pages */
            for (l = ogg->oggstreams; l != NULL; l = l->next) {
              GstOggStream *stream = (GstOggStream *) l->data;
              GList *k;

              if (stream->unknown_pages == NULL)
                continue;

              if (found_pending_headers) {
                GST_WARNING_OBJECT (ogg, "Incorrectly muxed headers found at "
                    "approximate offset %" G_GINT64_FORMAT, ogg->offset);
              }
              found_pending_headers = TRUE;

              GST_LOG_OBJECT (ogg, "Pushing %d pending pages after headers",
                  g_list_length (stream->unknown_pages) + 1);

              for (k = stream->unknown_pages; k != NULL; k = k->next) {
                GstBuffer *buf = GST_BUFFER (k->data);

                result = gst_pad_push (ogg->srcpad, buf);
                if (result != GST_FLOW_OK)
                  return result;
              }
              g_list_foreach (stream->unknown_pages,
                  (GFunc) gst_mini_object_unref, NULL);
              g_list_free (stream->unknown_pages);
              stream->unknown_pages = NULL;
            }
          }

          if (granule == -1) {
            stream->stored_buffers = g_list_append (stream->stored_buffers,
                pagebuffer);
          } else {
            while (stream->stored_buffers) {
              GstBuffer *buf = stream->stored_buffers->data;

              buf = gst_buffer_make_writable (buf);

              GST_BUFFER_TIMESTAMP (buf) = buffertimestamp;
              if (!keyframe) {
                GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DELTA_UNIT);
              } else {
                keyframe = FALSE;
              }

              result = gst_pad_push (ogg->srcpad, buf);
              if (result != GST_FLOW_OK)
                return result;

              stream->stored_buffers =
                  g_list_delete_link (stream->stored_buffers,
                  stream->stored_buffers);
            }

            pagebuffer = gst_buffer_make_writable (pagebuffer);
            if (!keyframe) {
              GST_BUFFER_FLAG_SET (pagebuffer, GST_BUFFER_FLAG_DELTA_UNIT);
            } else {
              keyframe = FALSE;
            }

            result = gst_pad_push (ogg->srcpad, pagebuffer);
            if (result != GST_FLOW_OK)
              return result;
          }
        }
      }
    }
  }

  return result;

failure:
  gst_pad_push_event (GST_PAD (ogg->srcpad), gst_event_new_eos ());
  return GST_FLOW_ERROR;
}
/* for off == 4 initial code; returns TRUE if code starts a frame
 * otherwise returns TRUE if code terminates preceding frame */
static gboolean
gst_mpegv_parse_process_sc (GstMpegvParse * mpvparse,
    GstBuffer * buf, gint off, guint8 code)
{
  gboolean ret = FALSE, packet = TRUE;

  g_return_val_if_fail (buf && gst_buffer_get_size (buf) >= 4, FALSE);

  GST_LOG_OBJECT (mpvparse, "process startcode %x (%s)", code,
      picture_start_code_name (code));

  switch (code) {
    case GST_MPEG_VIDEO_PACKET_PICTURE:
      GST_LOG_OBJECT (mpvparse, "startcode is PICTURE");
      /* picture is aggregated with preceding sequence/gop, if any.
       * so, picture start code only ends if already a previous one */
      if (mpvparse->pic_offset < 0)
        mpvparse->pic_offset = off;
      else
        ret = (off != mpvparse->pic_offset);
      /* but it's a valid starting one */
      if (off == 4)
        ret = TRUE;
      break;
    case GST_MPEG_VIDEO_PACKET_SEQUENCE:
      GST_LOG_OBJECT (mpvparse, "startcode is SEQUENCE");
      if (mpvparse->seq_offset < 0)
        mpvparse->seq_offset = off;
      ret = TRUE;
      break;
    case GST_MPEG_VIDEO_PACKET_GOP:
      GST_LOG_OBJECT (mpvparse, "startcode is GOP");
      if (mpvparse->seq_offset >= 0)
        ret = mpvparse->gop_split;
      else
        ret = TRUE;
      break;
    case GST_MPEG_VIDEO_PACKET_EXTENSION:
      GST_LOG_OBJECT (mpvparse, "startcode is VIDEO PACKET EXTENSION");
      parse_picture_extension (mpvparse, buf, off);
      if (mpvparse->ext_count < G_N_ELEMENTS (mpvparse->ext_offsets))
        mpvparse->ext_offsets[mpvparse->ext_count++] = off;
      /* fall-through */
    default:
      packet = FALSE;
      break;
  }

  /* set size to avoid processing config again */
  if (mpvparse->seq_offset >= 0 && off != mpvparse->seq_offset &&
      !mpvparse->seq_size && packet) {
    /* should always be at start */
    g_assert (mpvparse->seq_offset <= 4);
    gst_mpegv_parse_process_config (mpvparse, buf, off - mpvparse->seq_offset);
    mpvparse->seq_size = off - mpvparse->seq_offset;
  }

  /* extract some picture info if there is any in the frame being terminated */
  if (ret && mpvparse->pic_offset >= 0 && mpvparse->pic_offset < off) {
    GstMapInfo map;

    gst_buffer_map (buf, &map, GST_MAP_READ);
    if (gst_mpeg_video_parse_picture_header (&mpvparse->pichdr,
            map.data, map.size, mpvparse->pic_offset))
      GST_LOG_OBJECT (mpvparse, "picture_coding_type %d (%s), ending"
          "frame of size %d", mpvparse->pichdr.pic_type,
          picture_type_name (mpvparse->pichdr.pic_type), off - 4);
    else
      GST_LOG_OBJECT (mpvparse, "Couldn't parse picture at offset %d",
          mpvparse->pic_offset);

    gst_buffer_unmap (buf, &map);
  }

  return ret;
}
static void
gst_type_find_element_loop (GstPad * pad)
{
  GstTypeFindElement *typefind;
  GstFlowReturn ret = GST_FLOW_OK;

  typefind = GST_TYPE_FIND_ELEMENT (GST_PAD_PARENT (pad));

  if (typefind->need_stream_start) {
    gchar *stream_id;
    GstEvent *event;

    stream_id = gst_pad_create_stream_id (typefind->src,
        GST_ELEMENT_CAST (typefind), NULL);

    GST_DEBUG_OBJECT (typefind, "Pushing STREAM_START");
    event = gst_event_new_stream_start (stream_id);
    gst_event_set_group_id (event, gst_util_group_id_next ());
    gst_pad_push_event (typefind->src, event);

    typefind->need_stream_start = FALSE;
    g_free (stream_id);
  }

  if (typefind->mode == MODE_TYPEFIND) {
    GstPad *peer = NULL;
    GstCaps *found_caps = NULL;
    GstTypeFindProbability probability = GST_TYPE_FIND_NONE;

    GST_DEBUG_OBJECT (typefind, "find type in pull mode");

    GST_OBJECT_LOCK (typefind);
    if (typefind->force_caps) {
      found_caps = gst_caps_ref (typefind->force_caps);
      probability = GST_TYPE_FIND_MAXIMUM;
    }
    GST_OBJECT_UNLOCK (typefind);

    if (!found_caps) {
      peer = gst_pad_get_peer (pad);
      if (peer) {
        gint64 size;
        gchar *ext;

        if (!gst_pad_query_duration (peer, GST_FORMAT_BYTES, &size)) {
          GST_WARNING_OBJECT (typefind, "Could not query upstream length!");
          gst_object_unref (peer);

          ret = GST_FLOW_ERROR;
          goto pause;
        }

        /* the size if 0, we cannot continue */
        if (size == 0) {
          /* keep message in sync with message in sink event handler */
          GST_ELEMENT_ERROR (typefind, STREAM, TYPE_NOT_FOUND,
              (_("Stream contains no data.")), ("Can't typefind empty stream"));
          gst_object_unref (peer);
          ret = GST_FLOW_ERROR;
          goto pause;
        }
        ext = gst_type_find_get_extension (typefind, pad);

        found_caps =
            gst_type_find_helper_get_range (GST_OBJECT_CAST (peer),
            GST_OBJECT_PARENT (peer),
            (GstTypeFindHelperGetRangeFunction) (GST_PAD_GETRANGEFUNC (peer)),
            (guint64) size, ext, &probability);
        g_free (ext);

        GST_DEBUG ("Found caps %" GST_PTR_FORMAT, found_caps);

        gst_object_unref (peer);
      }
    }

    if (!found_caps || probability < typefind->min_probability) {
      GST_DEBUG ("Trying to guess using extension");
      gst_caps_replace (&found_caps, NULL);
      found_caps =
          gst_type_find_guess_by_extension (typefind, pad, &probability);
    }

    if (!found_caps || probability < typefind->min_probability) {
      GST_ELEMENT_ERROR (typefind, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
      gst_caps_replace (&found_caps, NULL);
      ret = GST_FLOW_ERROR;
      goto pause;
    }

    GST_DEBUG ("Emiting found caps %" GST_PTR_FORMAT, found_caps);
    gst_type_find_element_emit_have_type (typefind, probability, found_caps);
    typefind->mode = MODE_NORMAL;
    gst_caps_unref (found_caps);
  } else if (typefind->mode == MODE_NORMAL) {
    GstBuffer *outbuf = NULL;

    if (typefind->need_segment) {
      typefind->need_segment = FALSE;
      gst_pad_push_event (typefind->src,
          gst_event_new_segment (&typefind->segment));
    }

    /* Pull 4k blocks and send downstream */
    ret = gst_pad_pull_range (typefind->sink, typefind->offset, 4096, &outbuf);
    if (ret != GST_FLOW_OK)
      goto pause;

    typefind->offset += gst_buffer_get_size (outbuf);

    ret = gst_pad_push (typefind->src, outbuf);
    if (ret != GST_FLOW_OK)
      goto pause;
  } else {
    /* Error out */
    ret = GST_FLOW_ERROR;
    goto pause;
  }

  return;

pause:
  {
    const gchar *reason = gst_flow_get_name (ret);
    gboolean push_eos = FALSE;

    GST_LOG_OBJECT (typefind, "pausing task, reason %s", reason);
    gst_pad_pause_task (typefind->sink);

    if (ret == GST_FLOW_EOS) {
      /* perform EOS logic */

      if (typefind->segment.flags & GST_SEGMENT_FLAG_SEGMENT) {
        gint64 stop;

        /* for segment playback we need to post when (in stream time)
         * we stopped, this is either stop (when set) or the duration. */
        if ((stop = typefind->segment.stop) == -1)
          stop = typefind->offset;

        GST_LOG_OBJECT (typefind, "Sending segment done, at end of segment");
        gst_element_post_message (GST_ELEMENT (typefind),
            gst_message_new_segment_done (GST_OBJECT (typefind),
                GST_FORMAT_BYTES, stop));
        gst_pad_push_event (typefind->src,
            gst_event_new_segment_done (GST_FORMAT_BYTES, stop));
      } else {
        push_eos = TRUE;
      }
    } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
      /* for fatal errors we post an error message */
      GST_ELEMENT_ERROR (typefind, STREAM, FAILED, (NULL),
          ("stream stopped, reason %s", reason));
      push_eos = TRUE;
    }
    if (push_eos) {
      /* send EOS, and prevent hanging if no streams yet */
      GST_LOG_OBJECT (typefind, "Sending EOS, at end of stream");
      gst_pad_push_event (typefind->src, gst_event_new_eos ());
    }
    return;
  }
}
Exemplo n.º 7
0
static GstFlowReturn
gst_dvd_spu_subpic_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
  GstDVDSpu *dvdspu = (GstDVDSpu *) parent;
  GstFlowReturn ret = GST_FLOW_OK;
  gsize size;

  g_return_val_if_fail (dvdspu != NULL, GST_FLOW_ERROR);

  GST_INFO_OBJECT (dvdspu, "Have subpicture buffer with timestamp %"
      GST_TIME_FORMAT " and size %" G_GSIZE_FORMAT,
      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), gst_buffer_get_size (buf));

  DVD_SPU_LOCK (dvdspu);

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

  if (GST_BUFFER_IS_DISCONT (buf) && dvdspu->partial_spu) {
    gst_buffer_unref (dvdspu->partial_spu);
    dvdspu->partial_spu = NULL;
  }

  if (dvdspu->partial_spu != NULL) {
    if (GST_BUFFER_TIMESTAMP_IS_VALID (buf))
      GST_WARNING_OBJECT (dvdspu,
          "Joining subpicture buffer with timestamp to previous");
    dvdspu->partial_spu = gst_buffer_append (dvdspu->partial_spu, buf);
  } else {
    /* If we don't yet have a buffer, wait for one with a timestamp,
     * since that will avoid collecting the 2nd half of a partial buf */
    if (GST_BUFFER_TIMESTAMP_IS_VALID (buf))
      dvdspu->partial_spu = buf;
    else
      gst_buffer_unref (buf);
  }

  if (dvdspu->partial_spu == NULL)
    goto done;

  size = gst_buffer_get_size (dvdspu->partial_spu);

  switch (dvdspu->spu_input_type) {
    case SPU_INPUT_TYPE_VOBSUB:
      if (size > 4) {
        guint8 header[2];
        guint16 packet_size;

        gst_buffer_extract (dvdspu->partial_spu, 0, header, 2);
        packet_size = GST_READ_UINT16_BE (header);
        if (packet_size == size) {
          submit_new_spu_packet (dvdspu, dvdspu->partial_spu);
          dvdspu->partial_spu = NULL;
        } else if (packet_size < size) {
          /* Somehow we collected too much - something is wrong. Drop the
           * packet entirely and wait for a new one */
          GST_DEBUG_OBJECT (dvdspu,
              "Discarding invalid SPU buffer of size %" G_GSIZE_FORMAT, size);

          gst_buffer_unref (dvdspu->partial_spu);
          dvdspu->partial_spu = NULL;
        } else {
          GST_LOG_OBJECT (dvdspu,
              "SPU buffer claims to be of size %u. Collected %" G_GSIZE_FORMAT
              " so far.", packet_size, size);
        }
      }
      break;
    case SPU_INPUT_TYPE_PGS:{
      /* Collect until we have a command buffer that ends exactly at the size
       * we've collected */
      guint8 packet_type;
      guint16 packet_size;
      GstMapInfo map;
      guint8 *ptr, *end;
      gboolean invalid = FALSE;

      gst_buffer_map (dvdspu->partial_spu, &map, GST_MAP_READ);

      ptr = map.data;
      end = ptr + map.size;

      /* FIXME: There's no need to walk the command set each time. We can set a
       * marker and resume where we left off next time */
      /* FIXME: Move the packet parsing and sanity checking into the format-specific modules */
      while (ptr != end) {
        if (ptr + 3 > end)
          break;
        packet_type = *ptr++;
        packet_size = GST_READ_UINT16_BE (ptr);
        ptr += 2;
        if (ptr + packet_size > end)
          break;
        ptr += packet_size;
        /* 0x80 is the END command for PGS packets */
        if (packet_type == 0x80 && ptr != end) {
          /* Extra cruft on the end of the packet -> assume invalid */
          invalid = TRUE;
          break;
        }
      }
      gst_buffer_unmap (dvdspu->partial_spu, &map);

      if (invalid) {
        gst_buffer_unref (dvdspu->partial_spu);
        dvdspu->partial_spu = NULL;
      } else if (ptr == end) {
        GST_DEBUG_OBJECT (dvdspu,
            "Have complete PGS packet of size %" G_GSIZE_FORMAT ". Enqueueing.",
            map.size);
        submit_new_spu_packet (dvdspu, dvdspu->partial_spu);
        dvdspu->partial_spu = NULL;
      }
      break;
    }
    default:
      GST_ERROR_OBJECT (dvdspu, "Input type not configured before SPU passing");
      goto caps_not_set;
  }

done:
  DVD_SPU_UNLOCK (dvdspu);

  return ret;

  /* ERRORS */
caps_not_set:
  {
    GST_ELEMENT_ERROR (dvdspu, RESOURCE, NO_SPACE_LEFT,
        (_("Subpicture format was not configured before data flow")), (NULL));
    ret = GST_FLOW_ERROR;
    goto done;
  }
}
Exemplo n.º 8
0
static GstFlowReturn
gst_interleave_collected (GstCollectPads * pads, GstInterleave * self)
{
  guint size;
  GstBuffer *outbuf = NULL;
  GstFlowReturn ret = GST_FLOW_OK;
  GSList *collected;
  guint nsamples;
  guint ncollected = 0;
  gboolean empty = TRUE;
  gint width = self->width / 8;
  GstMapInfo write_info;
  GstClockTime timestamp = -1;

  /* FIXME: send caps and tags after stream-start */
#if 0
  if (self->send_stream_start) {
    gchar s_id[32];

    /* stream-start (FIXME: create id based on input ids) */
    g_snprintf (s_id, sizeof (s_id), "interleave-%08x", g_random_int ());
    gst_pad_push_event (self->src, gst_event_new_stream_start (s_id));
    self->send_stream_start = FALSE;
  }
#endif

  size = gst_collect_pads_available (pads);
  if (size == 0)
    goto eos;

  g_return_val_if_fail (self->func != NULL, GST_FLOW_NOT_NEGOTIATED);
  g_return_val_if_fail (self->width > 0, GST_FLOW_NOT_NEGOTIATED);
  g_return_val_if_fail (self->channels > 0, GST_FLOW_NOT_NEGOTIATED);
  g_return_val_if_fail (self->rate > 0, GST_FLOW_NOT_NEGOTIATED);

  g_return_val_if_fail (size % width == 0, GST_FLOW_ERROR);

  GST_DEBUG_OBJECT (self, "Starting to collect %u bytes from %d channels", size,
      self->channels);

  nsamples = size / width;

  outbuf = gst_buffer_new_allocate (NULL, size * self->channels, NULL);

  if (outbuf == NULL || gst_buffer_get_size (outbuf) < size * self->channels) {
    gst_buffer_unref (outbuf);
    return GST_FLOW_NOT_NEGOTIATED;
  }

  gst_buffer_map (outbuf, &write_info, GST_MAP_WRITE);
  memset (write_info.data, 0, size * self->channels);

  for (collected = pads->data; collected != NULL; collected = collected->next) {
    GstCollectData *cdata;
    GstBuffer *inbuf;
    guint8 *outdata;
    GstMapInfo input_info;

    cdata = (GstCollectData *) collected->data;

    inbuf = gst_collect_pads_take_buffer (pads, cdata, size);
    if (inbuf == NULL) {
      GST_DEBUG_OBJECT (cdata->pad, "No buffer available");
      goto next;
    }
    ncollected++;
    gst_buffer_map (inbuf, &input_info, GST_MAP_READ);

    if (timestamp == -1)
      timestamp = GST_BUFFER_TIMESTAMP (inbuf);

    if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP))
      goto next;

    empty = FALSE;
    outdata =
        write_info.data + width * GST_INTERLEAVE_PAD_CAST (cdata->pad)->channel;

    self->func (outdata, input_info.data, self->channels, nsamples);
    gst_buffer_unmap (inbuf, &input_info);

  next:
    if (inbuf)
      gst_buffer_unref (inbuf);
  }

  if (ncollected == 0) {
    gst_buffer_unmap (outbuf, &write_info);
    goto eos;
  }

  GST_OBJECT_LOCK (self);
  if (self->pending_segment) {
    GstEvent *event;
    GstSegment segment;

    event = self->pending_segment;
    self->pending_segment = NULL;
    GST_OBJECT_UNLOCK (self);

    /* convert the input segment to time now */
    gst_event_copy_segment (event, &segment);

    if (segment.format != GST_FORMAT_TIME) {
      gst_event_unref (event);

      /* not time, convert */
      switch (segment.format) {
        case GST_FORMAT_BYTES:
          segment.start *= width;
          if (segment.stop != -1)
            segment.stop *= width;
          if (segment.position != -1)
            segment.position *= width;
          /* fallthrough for the samples case */
        case GST_FORMAT_DEFAULT:
          segment.start =
              gst_util_uint64_scale_int (segment.start, GST_SECOND, self->rate);
          if (segment.stop != -1)
            segment.stop =
                gst_util_uint64_scale_int (segment.stop, GST_SECOND,
                self->rate);
          if (segment.position != -1)
            segment.position =
                gst_util_uint64_scale_int (segment.position, GST_SECOND,
                self->rate);
          break;
        default:
          GST_WARNING ("can't convert segment values");
          segment.start = 0;
          segment.stop = -1;
          segment.position = 0;
          break;
      }
      event = gst_event_new_segment (&segment);
    }
    gst_pad_push_event (self->src, event);

    GST_OBJECT_LOCK (self);
  }
  GST_OBJECT_UNLOCK (self);

  if (timestamp != -1) {
    self->offset = gst_util_uint64_scale_int (timestamp, self->rate,
        GST_SECOND);
    self->timestamp = timestamp;
  }

  GST_BUFFER_TIMESTAMP (outbuf) = self->timestamp;
  GST_BUFFER_OFFSET (outbuf) = self->offset;

  self->offset += nsamples;
  self->timestamp = gst_util_uint64_scale_int (self->offset,
      GST_SECOND, self->rate);

  GST_BUFFER_DURATION (outbuf) =
      self->timestamp - GST_BUFFER_TIMESTAMP (outbuf);

  if (empty)
    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);

  gst_buffer_unmap (outbuf, &write_info);

  GST_LOG_OBJECT (self, "pushing outbuf, timestamp %" GST_TIME_FORMAT,
      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
  ret = gst_pad_push (self->src, outbuf);

  return ret;

eos:
  {
    GST_DEBUG_OBJECT (self, "no data available, must be EOS");
    if (outbuf)
      gst_buffer_unref (outbuf);
    gst_pad_push_event (self->src, gst_event_new_eos ());
    return GST_FLOW_EOS;
  }
}
Exemplo n.º 9
0
static GstFlowReturn
mpegpsmux_create_stream (MpegPsMux * mux, MpegPsPadData * ps_data, GstPad * pad)
{
  /* Create a steam. Fill in codec specific information */

  GstFlowReturn ret = GST_FLOW_ERROR;
  GstCaps *caps;
  GstStructure *s;
  gboolean is_video = FALSE;

  caps = gst_pad_get_current_caps (pad);
  if (caps == NULL) {
    GST_DEBUG_OBJECT (pad, "Sink pad caps were not set before pushing");
    return GST_FLOW_NOT_NEGOTIATED;
  }

  s = gst_caps_get_structure (caps, 0);
  g_return_val_if_fail (s != NULL, FALSE);

  if (gst_structure_has_name (s, "video/x-dirac")) {
    GST_DEBUG_OBJECT (pad, "Creating Dirac stream");
    ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_VIDEO_DIRAC);
    is_video = TRUE;
  } else if (gst_structure_has_name (s, "audio/x-ac3")) {
    GST_DEBUG_OBJECT (pad, "Creating AC3 stream");
    ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_PS_AUDIO_AC3);
  } else if (gst_structure_has_name (s, "audio/x-dts")) {
    GST_DEBUG_OBJECT (pad, "Creating DTS stream");
    ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_PS_AUDIO_DTS);
  } else if (gst_structure_has_name (s, "audio/x-lpcm")) {
    GST_DEBUG_OBJECT (pad, "Creating LPCM stream");
    ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_PS_AUDIO_LPCM);
  } else if (gst_structure_has_name (s, "video/x-h264")) {
    const GValue *value;
    GST_DEBUG_OBJECT (pad, "Creating H264 stream");
    /* Codec data contains SPS/PPS which need to go in stream for valid ES */
    value = gst_structure_get_value (s, "codec_data");
    if (value) {
      ps_data->codec_data = gst_buffer_ref (gst_value_get_buffer (value));
      GST_DEBUG_OBJECT (pad, "%" G_GSIZE_FORMAT " bytes of codec data",
          gst_buffer_get_size (ps_data->codec_data));
      ps_data->prepare_func = mpegpsmux_prepare_h264;
    } else {
      ps_data->codec_data = NULL;
    }
    ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_VIDEO_H264);
    is_video = TRUE;
  } else if (gst_structure_has_name (s, "audio/mpeg")) {
    gint mpegversion;
    if (!gst_structure_get_int (s, "mpegversion", &mpegversion)) {
      GST_ELEMENT_ERROR (pad, STREAM, FORMAT,
          ("Invalid data format presented"),
          ("Caps with type audio/mpeg did not have mpegversion"));
      goto beach;
    }

    switch (mpegversion) {
      case 1:
        GST_DEBUG_OBJECT (pad, "Creating MPEG Audio, version 1 stream");
        ps_data->stream =
            psmux_create_stream (mux->psmux, PSMUX_ST_AUDIO_MPEG1);
        break;
      case 2:
        GST_DEBUG_OBJECT (pad, "Creating MPEG Audio, version 2 stream");
        ps_data->stream =
            psmux_create_stream (mux->psmux, PSMUX_ST_AUDIO_MPEG2);
        break;
      case 4:
      {
        const GValue *value;
        /* Codec data contains SPS/PPS which need to go in stream for valid ES */
        GST_DEBUG_OBJECT (pad, "Creating MPEG Audio, version 4 stream");
        value = gst_structure_get_value (s, "codec_data");
        if (value) {
          ps_data->codec_data = gst_buffer_ref (gst_value_get_buffer (value));
          GST_DEBUG_OBJECT (pad, "%" G_GSIZE_FORMAT " bytes of codec data",
              gst_buffer_get_size (ps_data->codec_data));
          ps_data->prepare_func = mpegpsmux_prepare_aac;
        } else {
          ps_data->codec_data = NULL;
        }
        ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_AUDIO_AAC);
        break;
      }
      default:
        GST_WARNING_OBJECT (pad, "unsupported mpegversion %d", mpegversion);
        goto beach;
    }
  } else if (gst_structure_has_name (s, "video/mpeg")) {
    gint mpegversion;
    if (!gst_structure_get_int (s, "mpegversion", &mpegversion)) {
      GST_ELEMENT_ERROR (mux, STREAM, FORMAT,
          ("Invalid data format presented"),
          ("Caps with type video/mpeg did not have mpegversion"));
      goto beach;
    }

    if (mpegversion == 1) {
      GST_DEBUG_OBJECT (pad, "Creating MPEG Video, version 1 stream");
      ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_VIDEO_MPEG1);
    } else if (mpegversion == 2) {
      GST_DEBUG_OBJECT (pad, "Creating MPEG Video, version 2 stream");
      ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_VIDEO_MPEG2);
    } else {
      GST_DEBUG_OBJECT (pad, "Creating MPEG Video, version 4 stream");
      ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_VIDEO_MPEG4);
    }
    is_video = TRUE;
  }

  if (ps_data->stream != NULL) {
    ps_data->stream_id = ps_data->stream->stream_id;
    ps_data->stream_id_ext = ps_data->stream->stream_id_ext;
    GST_DEBUG_OBJECT (pad, "Stream created, stream_id=%04x, stream_id_ext=%04x",
        ps_data->stream_id, ps_data->stream_id_ext);

    gst_structure_get_int (s, "rate", &ps_data->stream->audio_sampling);
    gst_structure_get_int (s, "channels", &ps_data->stream->audio_channels);
    gst_structure_get_int (s, "bitrate", &ps_data->stream->audio_bitrate);

    ret = GST_FLOW_OK;

    if (is_video && mux->video_stream_id == 0) {
      mux->video_stream_id = ps_data->stream_id;
      GST_INFO_OBJECT (mux, "video pad stream_id 0x%02x", mux->video_stream_id);
    }
  }

beach:
  return ret;
}
Exemplo n.º 10
0
static GstFlowReturn
gst_raw_base_parse_handle_frame (GstBaseParse * parse,
    GstBaseParseFrame * frame, gint * skipsize)
{
  gsize in_size, out_size;
  guint frame_size;
  guint num_out_frames;
  gsize units_n, units_d;
  guint64 buffer_duration;
  GstFlowReturn flow_ret = GST_FLOW_OK;
  GstEvent *new_caps_event = NULL;
  GstRawBaseParse *raw_base_parse = GST_RAW_BASE_PARSE (parse);
  GstRawBaseParseClass *klass = GST_RAW_BASE_PARSE_GET_CLASS (parse);

  g_assert (klass->is_config_ready);
  g_assert (klass->get_caps_from_config);
  g_assert (klass->get_config_frame_size);
  g_assert (klass->get_units_per_second);


  /* We never skip any bytes this way. Instead, subclass takes care
   * of skipping any overhead (necessary, since the way it needs to
   * be skipped is completely subclass specific). */
  *skipsize = 0;


  /* The operations below access the current config. Protect
   * against race conditions by using the object lock. */
  GST_RAW_BASE_PARSE_CONFIG_MUTEX_LOCK (raw_base_parse);


  /* If the source pad caps haven't been set yet, or need to be
   * set again, do so now, BEFORE any buffers are pushed out */
  if (G_UNLIKELY (!raw_base_parse->src_caps_set)) {
    GstCaps *new_src_caps;

    if (G_UNLIKELY (!klass->is_config_ready (raw_base_parse,
                GST_RAW_BASE_PARSE_CONFIG_CURRENT))) {
      /* The current configuration is not ready. No caps can be
       * generated out of it.
       * The most likely reason for this is that the sink caps config
       * is the current one and no valid sink caps have been pushed
       * by upstream. Report the problem and exit. */

      if (gst_raw_base_parse_is_using_sink_caps (raw_base_parse)) {
        goto config_not_ready;
      } else {
        /* This should not be reached if the property config is active */
        g_assert_not_reached ();
      }
    }

    GST_DEBUG_OBJECT (parse,
        "setting src caps since this has not been done yet");

    /* Convert the current config to a caps structure to
     * inform downstream about the new format */
    if (!klass->get_caps_from_config (raw_base_parse,
            GST_RAW_BASE_PARSE_CONFIG_CURRENT, &new_src_caps)) {
      GST_ERROR_OBJECT (raw_base_parse,
          "could not get src caps from current config");
      flow_ret = GST_FLOW_NOT_NEGOTIATED;
      goto error_locked;
    }

    new_caps_event = gst_event_new_caps (new_src_caps);
    gst_caps_unref (new_src_caps);

    raw_base_parse->src_caps_set = TRUE;
  }

  frame_size =
      klass->get_config_frame_size (raw_base_parse,
      GST_RAW_BASE_PARSE_CONFIG_CURRENT);


  in_size = gst_buffer_get_size (frame->buffer);

  /* gst_base_parse_set_min_frame_size() is called when the current
   * configuration changes and the change affects the frame size. This
   * means that a buffer must contain at least as many bytes as indicated
   * by the frame size. If there are fewer inside an error occurred;
   * either something in the parser went wrong, or the min frame size
   * wasn't updated properly. */
  g_assert (in_size >= frame_size);

  /* Determine how many complete frames would fit in the input buffer.
   * Then check if this amount exceeds the maximum number of frames
   * as indicated by the subclass. */
  num_out_frames = (in_size / frame_size);
  if (klass->get_max_frames_per_buffer) {
    guint max_num_out_frames = klass->get_max_frames_per_buffer (raw_base_parse,
        GST_RAW_BASE_PARSE_CONFIG_CURRENT);
    num_out_frames = MIN (num_out_frames, max_num_out_frames);
  }

  /* Ensure that the size of the buffers that get pushed downstream
   * is always an integer multiple of the frame size to prevent cases
   * where downstream gets buffers with incomplete frames. */
  out_size = num_out_frames * frame_size;

  /* Set the overhead size to ensure that timestamping excludes these
   * extra overhead bytes. */
  frame->overhead =
      klass->get_overhead_size ? klass->get_overhead_size (raw_base_parse,
      GST_RAW_BASE_PARSE_CONFIG_CURRENT) : 0;

  g_assert (out_size >= (guint) (frame->overhead));
  out_size -= frame->overhead;

  GST_LOG_OBJECT (raw_base_parse,
      "%" G_GSIZE_FORMAT " bytes input  %" G_GSIZE_FORMAT
      " bytes output (%u frame(s))  %d bytes overhead", in_size, out_size,
      num_out_frames, frame->overhead);

  /* Calculate buffer duration */
  klass->get_units_per_second (raw_base_parse, GST_FORMAT_BYTES,
      GST_RAW_BASE_PARSE_CONFIG_CURRENT, &units_n, &units_d);
  if (units_n == 0 || units_d == 0)
    buffer_duration = GST_CLOCK_TIME_NONE;
  else
    buffer_duration =
        gst_util_uint64_scale (out_size, GST_SECOND * units_d, units_n);

  if (klass->process) {
    GstBuffer *processed_data = NULL;

    if (!klass->process (raw_base_parse, GST_RAW_BASE_PARSE_CONFIG_CURRENT,
            frame->buffer, in_size, out_size, &processed_data))
      goto process_error;

    frame->out_buffer = processed_data;
  } else {
    frame->out_buffer = NULL;
  }

  /* Set the duration of the output buffer, or if none exists, of
   * the input buffer. Do this after the process() call, since in
   * case out_buffer is set, the subclass has created a new buffer.
   * Instead of requiring subclasses to set the duration (which
   * anyway must always be buffer_duration), let's do it here. */
  if (frame->out_buffer != NULL)
    GST_BUFFER_DURATION (frame->out_buffer) = buffer_duration;
  else
    GST_BUFFER_DURATION (frame->buffer) = buffer_duration;

  /* Access to the current config is not needed in subsequent
   * operations, so the lock can be released */
  GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (raw_base_parse);


  /* If any new caps have to be pushed downstrean, do so
   * *before* the frame is finished */
  if (G_UNLIKELY (new_caps_event != NULL)) {
    gst_pad_push_event (GST_BASE_PARSE_SRC_PAD (raw_base_parse),
        new_caps_event);
    new_caps_event = NULL;
  }

  gst_base_parse_finish_frame (parse, frame, out_size + frame->overhead);


  return flow_ret;


config_not_ready:
  GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (raw_base_parse);
  GST_ELEMENT_ERROR (parse, STREAM, FORMAT,
      ("sink caps config is the current config, and it is not ready -"
          "upstream may not have pushed a caps event yet"), (NULL));
  flow_ret = GST_FLOW_ERROR;
  goto error_end;

process_error:
  GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (raw_base_parse);
  GST_ELEMENT_ERROR (parse, STREAM, DECODE, ("could not process data"), (NULL));
  flow_ret = GST_FLOW_ERROR;
  goto error_end;

error_locked:
  GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (raw_base_parse);
  goto error_end;

error_end:
  frame->flags |= GST_BASE_PARSE_FRAME_FLAG_DROP;
  if (new_caps_event != NULL)
    gst_event_unref (new_caps_event);
  return flow_ret;
}
Exemplo n.º 11
0
static GstFlowReturn
gst_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
    guint length, GstBuffer ** buffer)
{
  GstBuffer *buf = NULL;
  GstDTMFSrcEvent *event;
  GstDTMFSrc *dtmfsrc;
  GstClock *clock;
  GstClockID *clockid;
  GstClockReturn clockret;

  dtmfsrc = GST_DTMF_SRC (basesrc);

  do {

    if (dtmfsrc->last_event == NULL) {
      GST_DEBUG_OBJECT (dtmfsrc, "popping");
      event = g_async_queue_pop (dtmfsrc->event_queue);

      GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);

      switch (event->event_type) {
        case DTMF_EVENT_TYPE_STOP:
          GST_WARNING_OBJECT (dtmfsrc,
              "Received a DTMF stop event when already stopped");
          gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
          break;
        case DTMF_EVENT_TYPE_START:
          gst_dtmf_prepare_timestamps (dtmfsrc);

          event->packet_count = 0;
          dtmfsrc->last_event = event;
          event = NULL;
          gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed",
              dtmfsrc->last_event);
          break;
        case DTMF_EVENT_TYPE_PAUSE_TASK:
          /*
           * We're pushing it back because it has to stay in there until
           * the task is really paused (and the queue will then be flushed)
           */
          GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
          GST_OBJECT_LOCK (dtmfsrc);
          if (dtmfsrc->paused) {
            g_async_queue_push (dtmfsrc->event_queue, event);
            goto paused_locked;
          }
          GST_OBJECT_UNLOCK (dtmfsrc);
          break;
      }
      if (event)
        g_slice_free (GstDTMFSrcEvent, event);
    } else if (dtmfsrc->last_event->packet_count * dtmfsrc->interval >=
        MIN_DUTY_CYCLE) {
      event = g_async_queue_try_pop (dtmfsrc->event_queue);

      if (event != NULL) {

        switch (event->event_type) {
          case DTMF_EVENT_TYPE_START:
            GST_WARNING_OBJECT (dtmfsrc,
                "Received two consecutive DTMF start events");
            gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
            break;
          case DTMF_EVENT_TYPE_STOP:
            g_slice_free (GstDTMFSrcEvent, dtmfsrc->last_event);
            dtmfsrc->last_event = NULL;
            gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed", event);
            break;
          case DTMF_EVENT_TYPE_PAUSE_TASK:
            /*
             * We're pushing it back because it has to stay in there until
             * the task is really paused (and the queue will then be flushed)
             */
            GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");

            GST_OBJECT_LOCK (dtmfsrc);
            if (dtmfsrc->paused) {
              g_async_queue_push (dtmfsrc->event_queue, event);
              goto paused_locked;
            }
            GST_OBJECT_UNLOCK (dtmfsrc);

            break;
        }
        g_slice_free (GstDTMFSrcEvent, event);
      }
    }
  } while (dtmfsrc->last_event == NULL);

  GST_LOG_OBJECT (dtmfsrc, "end event check, now wait for the proper time");

  clock = gst_element_get_clock (GST_ELEMENT (basesrc));

  clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
      gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
  gst_object_unref (clock);

  GST_OBJECT_LOCK (dtmfsrc);
  if (!dtmfsrc->paused) {
    dtmfsrc->clockid = clockid;
    GST_OBJECT_UNLOCK (dtmfsrc);

    clockret = gst_clock_id_wait (clockid, NULL);

    GST_OBJECT_LOCK (dtmfsrc);
    if (dtmfsrc->paused)
      clockret = GST_CLOCK_UNSCHEDULED;
  } else {
    clockret = GST_CLOCK_UNSCHEDULED;
  }
  gst_clock_id_unref (clockid);
  dtmfsrc->clockid = NULL;
  GST_OBJECT_UNLOCK (dtmfsrc);

  if (clockret == GST_CLOCK_UNSCHEDULED) {
    goto paused;
  }

  buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc, dtmfsrc->last_event);

  GST_LOG_OBJECT (dtmfsrc, "Created buffer of size %" G_GSIZE_FORMAT,
      gst_buffer_get_size (buf));
  *buffer = buf;

  return GST_FLOW_OK;

paused_locked:
  GST_OBJECT_UNLOCK (dtmfsrc);

paused:

  if (dtmfsrc->last_event) {
    GST_DEBUG_OBJECT (dtmfsrc, "Stopping current event");
    /* Don't forget to release the stream lock */
    g_slice_free (GstDTMFSrcEvent, dtmfsrc->last_event);
    dtmfsrc->last_event = NULL;
  }

  return GST_FLOW_FLUSHING;

}
Exemplo n.º 12
0
static GstFlowReturn
gst_tcp_mix_src_pad_read (GstTCPMixSrcPad * pad, GstBuffer ** outbuf)
{
  GstTCPMixSrc *src = GST_TCP_MIX_SRC (GST_PAD_PARENT (pad));
  gssize avail, receivedBytes;
  GstMapInfo map;
  GError *err = NULL;

  /* if we have a client, wait for read */
  GST_LOG_OBJECT (pad, "asked for a buffer");

  if (!pad->client) {
    if (src->mode == MODE_LOOP)
      goto loop_read;
    else
      goto no_client;
  }

  /* read the buffer header */

read_available_bytes:
  avail = g_socket_get_available_bytes (pad->client);
  if (avail < 0) {
    goto socket_get_available_bytes_error;
  } else if (avail == 0) {
    GIOCondition condition;

    if (!g_socket_condition_wait (pad->client,
            G_IO_IN | G_IO_PRI | G_IO_ERR | G_IO_HUP, pad->cancellable, &err))
      goto socket_condition_wait_error;

    condition = g_socket_condition_check (pad->client,
        G_IO_IN | G_IO_PRI | G_IO_ERR | G_IO_HUP);

    if ((condition & G_IO_ERR))
      goto socket_condition_error;
    else if ((condition & G_IO_HUP))
      goto socket_condition_hup;

    avail = g_socket_get_available_bytes (pad->client);
    if (avail < 0)
      goto socket_get_available_bytes_error;
  }

  if (0 < avail) {
    gsize readBytes = MIN (avail, MAX_READ_SIZE);
    *outbuf = gst_buffer_new_and_alloc (readBytes);
    gst_buffer_map (*outbuf, &map, GST_MAP_READWRITE);
    receivedBytes = g_socket_receive (pad->client, (gchar *) map.data,
        readBytes, pad->cancellable, &err);
  } else {
    /* Connection closed */
    receivedBytes = 0;
    *outbuf = NULL;
  }

  if (receivedBytes == 0)
    goto socket_connection_closed;
  else if (receivedBytes < 0)
    goto socket_receive_error;

  gst_buffer_unmap (*outbuf, &map);
  gst_buffer_resize (*outbuf, 0, receivedBytes);

#if 0
  GST_LOG_OBJECT (pad,
      "Returning buffer from _get of size %" G_GSIZE_FORMAT
      ", ts %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT
      ", offset %" G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
      gst_buffer_get_size (*outbuf),
      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (*outbuf)),
      GST_TIME_ARGS (GST_BUFFER_DURATION (*outbuf)),
      GST_BUFFER_OFFSET (*outbuf), GST_BUFFER_OFFSET_END (*outbuf));
#endif

  g_clear_error (&err);

  return GST_FLOW_OK;

  /* Handling Errors */
no_client:
  {
    GST_ELEMENT_ERROR (pad, RESOURCE, READ, (NULL),
        ("No client socket (%s)", GST_PAD_NAME (pad)));

    if (src->mode == MODE_LOOP)
      goto loop_read;
    return GST_FLOW_ERROR;
  }

socket_get_available_bytes_error:
  {
    GST_ELEMENT_ERROR (pad, RESOURCE, READ, (NULL),
        ("Failed to get available bytes from socket"));

    gst_tcp_mix_src_pad_reset (pad);

    if (src->mode == MODE_LOOP)
      goto loop_read;
    return GST_FLOW_ERROR;
  }

socket_condition_wait_error:
  {
    GST_ELEMENT_ERROR (pad, RESOURCE, READ, (NULL),
        ("Select failed: %s", err->message));
    g_clear_error (&err);

    gst_tcp_mix_src_pad_reset (pad);

    if (src->mode == MODE_LOOP)
      goto loop_read;
    return GST_FLOW_ERROR;
  }

socket_condition_error:
  {
    GST_ELEMENT_ERROR (pad, RESOURCE, READ, (NULL), ("Socket in error state"));
    *outbuf = NULL;

    gst_tcp_mix_src_pad_reset (pad);

    if (src->mode == MODE_LOOP)
      goto loop_read;
    return GST_FLOW_ERROR;
  }

socket_condition_hup:
  {
    GST_DEBUG_OBJECT (pad, "Connection closed");
    *outbuf = NULL;

    gst_tcp_mix_src_pad_reset (pad);

    if (src->mode == MODE_LOOP)
      goto loop_read;
    return GST_FLOW_EOS;
  }

socket_connection_closed:
  {
    GST_DEBUG_OBJECT (pad, "Connection closed");
    if (*outbuf) {
      gst_buffer_unmap (*outbuf, &map);
      gst_buffer_unref (*outbuf);
    }
    *outbuf = NULL;

    gst_tcp_mix_src_pad_reset (pad);

    if (src->mode == MODE_LOOP)
      goto loop_read;
    return GST_FLOW_EOS;
  }

socket_receive_error:
  {
    gst_buffer_unmap (*outbuf, &map);
    gst_buffer_unref (*outbuf);
    *outbuf = NULL;

    if (g_error_matches (err, G_IO_ERROR, G_IO_ERROR_CANCELLED)) {
      GST_DEBUG_OBJECT (pad, "Cancelled reading from socket");

      if (src->mode == MODE_LOOP)
        goto loop_read;
      return GST_FLOW_FLUSHING;
    } else {
      GST_ELEMENT_ERROR (pad, RESOURCE, READ, (NULL),
          ("Failed to read from socket: %s", err->message));

      if (src->mode == MODE_LOOP)
        goto loop_read;
      return GST_FLOW_ERROR;
    }
  }

loop_read:
  {
#if 0
    GstEvent *event = gst_event_new_flush_start ();

    if (!gst_pad_push_event (pad, event)) {
      GST_ERROR_OBJECT (src, "Failed to flush data on %s.%s",
          GST_ELEMENT_NAME (src), GST_PAD_NAME (pad));
    }
#endif

#if 0
    GST_DEBUG_OBJECT (pad, "Looping");
#endif

    if (src->fill == FILL_NONE) {
      gst_tcp_mix_src_pad_wait_for_client (pad);
      goto read_available_bytes;
    }

    enum
    { buffer_size = 1024 };
    *outbuf = gst_buffer_new_and_alloc (buffer_size);

    switch (src->fill) {
      case FILL_ZERO:
        break;
      case FILL_RAND:
      {
        guchar *p;
        gst_buffer_map (*outbuf, &map, GST_MAP_READWRITE);
        for (p = map.data; p < map.data + buffer_size; p += 4) {
          *((int *) p) = rand ();
        }
      } break;
    }
    return GST_FLOW_OK;
  }
}
Exemplo n.º 13
0
static GstFlowReturn
gst_decklink_sink_videosink_chain (GstPad * pad, GstObject * parent,
    GstBuffer * buffer)
{
  GstDecklinkSink *decklinksink;
  IDeckLinkMutableVideoFrame *frame;
  void *data;
  GstFlowReturn ret;
  const GstDecklinkMode *mode;

  decklinksink = GST_DECKLINK_SINK (parent);

#if 0
  if (!decklinksink->video_enabled) {
    HRESULT ret;
    ret = decklinksink->output->EnableVideoOutput (decklinksink->display_mode,
        bmdVideoOutputFlagDefault);
    if (ret != S_OK) {
      GST_WARNING ("failed to enable video output");
      //return FALSE;
    }
    decklinksink->video_enabled = TRUE;
  }
#endif

  mode = gst_decklink_get_mode (decklinksink->mode);

  decklinksink->output->CreateVideoFrame (mode->width,
      mode->height, mode->width * 2, decklinksink->pixel_format,
      bmdFrameFlagDefault, &frame);

  frame->GetBytes (&data);
  gst_buffer_extract (buffer, 0, data, gst_buffer_get_size (buffer));
  gst_buffer_unref (buffer);

  g_mutex_lock (&decklinksink->mutex);
  while (decklinksink->queued_frames > 2 && !decklinksink->stop) {
    g_cond_wait (&decklinksink->cond, &decklinksink->mutex);
  }
  if (!decklinksink->stop) {
    decklinksink->queued_frames++;
  }
  g_mutex_unlock (&decklinksink->mutex);

  if (!decklinksink->stop) {
    decklinksink->output->ScheduleVideoFrame (frame,
        decklinksink->num_frames * mode->fps_d, mode->fps_d, mode->fps_n);
    decklinksink->num_frames++;

    if (!decklinksink->sched_started) {
      decklinksink->output->StartScheduledPlayback (0, mode->fps_d, 1.0);
      decklinksink->sched_started = TRUE;
    }

    ret = GST_FLOW_OK;
  } else {
    ret = GST_FLOW_FLUSHING;
  }

  frame->Release ();

  return ret;
}
Exemplo n.º 14
0
static GstFlowReturn
gst_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload *
    basepayload, GstBuffer * buffer)
{
  GstRTPBaseAudioPayload *payload;
  GstRTPBaseAudioPayloadPrivate *priv;
  guint payload_len;
  GstFlowReturn ret;
  guint available;
  guint min_payload_len;
  guint max_payload_len;
  guint align;
  guint size;
  gboolean discont;
  GstClockTime timestamp;

  ret = GST_FLOW_OK;

  payload = GST_RTP_BASE_AUDIO_PAYLOAD_CAST (basepayload);
  priv = payload->priv;

  timestamp = GST_BUFFER_PTS (buffer);
  discont = GST_BUFFER_IS_DISCONT (buffer);
  if (discont) {

    GST_DEBUG_OBJECT (payload, "Got DISCONT");
    /* flush everything out of the adapter, mark DISCONT */
    ret = gst_rtp_base_audio_payload_flush (payload, -1, -1);
    priv->discont = TRUE;

    /* get the distance between the timestamp gap and produce the same gap in
     * the RTP timestamps */
    if (priv->last_timestamp != -1 && timestamp != -1) {
      /* we had a last timestamp, compare it to the new timestamp and update the
       * offset counter for RTP timestamps. The effect is that we will produce
       * output buffers containing the same RTP timestamp gap as the gap
       * between the GST timestamps. */
      if (timestamp > priv->last_timestamp) {
        GstClockTime diff;
        guint64 bytes;
        /* we're only going to apply a positive gap, otherwise we let the marker
         * bit do its thing. simply convert to bytes and add the current
         * offset */
        diff = timestamp - priv->last_timestamp;
        bytes = priv->time_to_bytes (payload, diff);
        priv->offset += bytes;

        GST_DEBUG_OBJECT (payload,
            "elapsed time %" GST_TIME_FORMAT ", bytes %" G_GUINT64_FORMAT
            ", new offset %" G_GUINT64_FORMAT, GST_TIME_ARGS (diff), bytes,
            priv->offset);
      }
    }
  }

  if (!gst_rtp_base_audio_payload_get_lengths (basepayload,
          gst_rtp_base_payload_get_source_count (basepayload, buffer),
          &min_payload_len, &max_payload_len, &align))
    goto config_error;

  GST_DEBUG_OBJECT (payload,
      "Calculated min_payload_len %u and max_payload_len %u",
      min_payload_len, max_payload_len);

  size = gst_buffer_get_size (buffer);

  /* shortcut, we don't need to use the adapter when the packet can be pushed
   * through directly. */
  available = gst_adapter_available (priv->adapter);

  GST_DEBUG_OBJECT (payload, "got buffer size %u, available %u",
      size, available);

  if (available == 0 && (size >= min_payload_len && size <= max_payload_len) &&
      (size % align == 0)) {
    /* If buffer fits on an RTP packet, let's just push it through
     * this will check against max_ptime and max_mtu */
    GST_DEBUG_OBJECT (payload, "Fast packet push");
    ret = gst_rtp_base_audio_payload_push_buffer (payload, buffer, timestamp);
  } else {
    /* push the buffer in the adapter */
    gst_adapter_push (priv->adapter, buffer);
    available += size;

    GST_DEBUG_OBJECT (payload, "available now %u", available);

    /* as long as we have full frames */
    /* TODO: Use buffer lists here */
    while (available >= min_payload_len) {
      /* get multiple of alignment */
      payload_len = MIN (max_payload_len, available);
      payload_len = ALIGN_DOWN (payload_len, align);

      /* and flush out the bytes from the adapter, automatically set the
       * timestamp. */
      ret = gst_rtp_base_audio_payload_flush (payload, payload_len, -1);

      available -= payload_len;
      GST_DEBUG_OBJECT (payload, "available after push %u", available);
    }
  }
  return ret;

  /* ERRORS */
config_error:
  {
    GST_ELEMENT_ERROR (payload, STREAM, NOT_IMPLEMENTED, (NULL),
        ("subclass did not configure us properly"));
    gst_buffer_unref (buffer);
    return GST_FLOW_ERROR;
  }
}
Exemplo n.º 15
0
static GstFlowReturn
gst_monoscope_chain (GstPad * pad, GstObject * parent, GstBuffer * inbuf)
{
  GstFlowReturn flow_ret = GST_FLOW_OK;
  GstMonoscope *monoscope;

  monoscope = GST_MONOSCOPE (parent);

  if (monoscope->rate == 0) {
    gst_buffer_unref (inbuf);
    flow_ret = GST_FLOW_NOT_NEGOTIATED;
    goto out;
  }

  /* Make sure have an output format */
  flow_ret = ensure_negotiated (monoscope);
  if (flow_ret != GST_FLOW_OK) {
    gst_buffer_unref (inbuf);
    goto out;
  }

  /* don't try to combine samples from discont buffer */
  if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DISCONT)) {
    gst_adapter_clear (monoscope->adapter);
    monoscope->next_ts = GST_CLOCK_TIME_NONE;
  }

  /* Match timestamps from the incoming audio */
  if (GST_BUFFER_TIMESTAMP (inbuf) != GST_CLOCK_TIME_NONE)
    monoscope->next_ts = GST_BUFFER_TIMESTAMP (inbuf);

  GST_LOG_OBJECT (monoscope, "in buffer has %d samples, ts=%" GST_TIME_FORMAT,
      gst_buffer_get_size (inbuf) / monoscope->bps,
      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)));

  gst_adapter_push (monoscope->adapter, inbuf);
  inbuf = NULL;

  /* Collect samples until we have enough for an output frame */
  while (flow_ret == GST_FLOW_OK) {
    gint16 *samples;
    GstBuffer *outbuf = NULL;
    guint32 *pixels, avail, bytesperframe;

    avail = gst_adapter_available (monoscope->adapter);
    GST_LOG_OBJECT (monoscope, "bytes avail now %u", avail);

    bytesperframe = monoscope->spf * monoscope->bps;
    if (avail < bytesperframe)
      break;

    /* FIXME: something is wrong with QoS, we are skipping way too much
     * stuff even with very low CPU loads */
#if 0
    if (monoscope->next_ts != -1) {
      gboolean need_skip;
      gint64 qostime;

      qostime = gst_segment_to_running_time (&monoscope->segment,
          GST_FORMAT_TIME, monoscope->next_ts);

      GST_OBJECT_LOCK (monoscope);
      /* check for QoS, don't compute buffers that are known to be late */
      need_skip =
          GST_CLOCK_TIME_IS_VALID (monoscope->earliest_time) &&
          qostime <= monoscope->earliest_time;
      GST_OBJECT_UNLOCK (monoscope);

      if (need_skip) {
        GST_WARNING_OBJECT (monoscope,
            "QoS: skip ts: %" GST_TIME_FORMAT ", earliest: %" GST_TIME_FORMAT,
            GST_TIME_ARGS (qostime), GST_TIME_ARGS (monoscope->earliest_time));
        goto skip;
      }
    }
#endif

    samples = (gint16 *) gst_adapter_map (monoscope->adapter, bytesperframe);

    if (monoscope->spf < 512) {
      gint16 in_data[512], i;

      for (i = 0; i < 512; ++i) {
        gdouble off;

        off = ((gdouble) i * (gdouble) monoscope->spf) / 512.0;
        in_data[i] = samples[MIN ((guint) off, monoscope->spf)];
      }
      pixels = monoscope_update (monoscope->visstate, in_data);
    } else {
      /* not really correct, but looks much prettier */
      pixels = monoscope_update (monoscope->visstate, samples);
    }

    GST_LOG_OBJECT (monoscope, "allocating output buffer");
    flow_ret = gst_buffer_pool_acquire_buffer (monoscope->pool, &outbuf, NULL);
    if (flow_ret != GST_FLOW_OK) {
      gst_adapter_unmap (monoscope->adapter);
      goto out;
    }

    gst_buffer_fill (outbuf, 0, pixels, monoscope->outsize);

    GST_BUFFER_TIMESTAMP (outbuf) = monoscope->next_ts;
    GST_BUFFER_DURATION (outbuf) = monoscope->frame_duration;

    flow_ret = gst_pad_push (monoscope->srcpad, outbuf);

#if 0
  skip:
#endif

    if (GST_CLOCK_TIME_IS_VALID (monoscope->next_ts))
      monoscope->next_ts += monoscope->frame_duration;

    gst_adapter_flush (monoscope->adapter, bytesperframe);
  }

out:

  return flow_ret;
}
Exemplo n.º 16
0
static void
gst_wildmidi_loop (GstPad * sinkpad)
{
  GstWildmidi *wildmidi = GST_WILDMIDI (GST_PAD_PARENT (sinkpad));
  GstFlowReturn ret;

  switch (wildmidi->state) {
    case GST_WILDMIDI_STATE_LOAD:
    {
      GstBuffer *buffer = NULL;

      GST_DEBUG_OBJECT (wildmidi, "loading song");

      ret =
          gst_pad_pull_range (wildmidi->sinkpad, wildmidi->offset, -1, &buffer);

      if (ret == GST_FLOW_EOS) {
        GST_DEBUG_OBJECT (wildmidi, "Song loaded");
        wildmidi->state = GST_WILDMIDI_STATE_PARSE;
      } else if (ret != GST_FLOW_OK) {
        GST_ELEMENT_ERROR (wildmidi, STREAM, DECODE, (NULL),
            ("Unable to read song"));
        goto pause;
      } else {
        GST_DEBUG_OBJECT (wildmidi, "pushing buffer");
        gst_adapter_push (wildmidi->adapter, buffer);
        wildmidi->offset += gst_buffer_get_size (buffer);
      }
      break;
    }
    case GST_WILDMIDI_STATE_PARSE:
      ret = gst_wildmidi_parse_song (wildmidi);
      if (ret != GST_FLOW_OK)
        goto pause;
      wildmidi->state = GST_WILDMIDI_STATE_PLAY;
      break;
    case GST_WILDMIDI_STATE_PLAY:
      ret = gst_wildmidi_do_play (wildmidi);
      if (ret != GST_FLOW_OK)
        goto pause;
      break;
    default:
      break;
  }
  return;

pause:
  {
    const gchar *reason = gst_flow_get_name (ret);
    GstEvent *event;

    GST_DEBUG_OBJECT (wildmidi, "pausing task, reason %s", reason);
    gst_pad_pause_task (sinkpad);
    if (ret == GST_FLOW_EOS) {
      /* perform EOS logic */
      event = gst_event_new_eos ();
      gst_pad_push_event (wildmidi->srcpad, event);
    } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
      event = gst_event_new_eos ();
      /* for fatal errors we post an error message, post the error
       * first so the app knows about the error first. */
      GST_ELEMENT_ERROR (wildmidi, STREAM, FAILED,
          ("Internal data flow error."),
          ("streaming task paused, reason %s (%d)", reason, ret));
      gst_pad_push_event (wildmidi->srcpad, event);
    }
  }
}
Exemplo n.º 17
0
/* FIXME: this entire function looks completely wrong, if there is or
 * was a bug in GStreamer it should be fixed there (tpm) */
static GstPadProbeReturn
brasero_transcode_buffer_handler (GstPad *pad,
                                  GstPadProbeInfo *info,
                                  gpointer user_data)
{
	BraseroTranscodePrivate *priv;
	BraseroTranscode *self = user_data;
	GstBuffer *buffer = GST_PAD_PROBE_INFO_BUFFER (info);
	GstPad *peer;
	gint64 size;

	priv = BRASERO_TRANSCODE_PRIVATE (self);

	size = gst_buffer_get_size (buffer);

	if (priv->segment_start <= 0 && priv->segment_end <= 0)
		return GST_PAD_PROBE_OK;

	/* what we do here is more or less what gstreamer does when seeking:
	 * it reads and process from 0 to the seek position (I tried).
	 * It even forwards the data before the seek position to the sink (which
	 * is a problem in our case as it would be written) */
	if (priv->size > priv->segment_end) {
		priv->size += size;
		return GST_PAD_PROBE_DROP;
	}

	if (priv->size + size > priv->segment_end) {
		GstBuffer *new_buffer;
		int data_size;

		/* the entire the buffer is not interesting for us */
		/* create a new buffer and push it on the pad:
		 * NOTE: we're going to receive it ... */
		data_size = priv->segment_end - priv->size;
		new_buffer = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_METADATA, 0, data_size);

		/* FIXME: we can now modify the probe buffer in 0.11 */
		/* Recursive: the following calls ourselves BEFORE we finish */
		peer = gst_pad_get_peer (pad);
		gst_pad_push (peer, new_buffer);

		priv->size += size - data_size;

		/* post an EOS event to stop pipeline */
		gst_pad_push_event (peer, gst_event_new_eos ());
		gst_object_unref (peer);
		return GST_PAD_PROBE_DROP;
	}

	/* see if the buffer is in the segment */
	if (priv->size < priv->segment_start) {
		GstBuffer *new_buffer;
		gint data_size;

		/* see if all the buffer is interesting for us */
		if (priv->size + size < priv->segment_start) {
			priv->size += size;
			return GST_PAD_PROBE_DROP;
		}

		/* create a new buffer and push it on the pad:
		 * NOTE: we're going to receive it ... */
		data_size = priv->size + size - priv->segment_start;
		new_buffer = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_METADATA, size - data_size, data_size);
		/* FIXME: this looks dodgy (tpm) */
		GST_BUFFER_TIMESTAMP (new_buffer) = GST_BUFFER_TIMESTAMP (buffer) + data_size;

		/* move forward by the size of bytes we dropped */
		priv->size += size - data_size;

		/* FIXME: we can now modify the probe buffer in 0.11 */
		/* this is recursive the following calls ourselves 
		 * BEFORE we finish */
		peer = gst_pad_get_peer (pad);
		gst_pad_push (peer, new_buffer);
		gst_object_unref (peer);

		return GST_PAD_PROBE_DROP;
	}

	priv->size += size;
	priv->pos += size;

	return GST_PAD_PROBE_OK;
}
Exemplo n.º 18
0
static GstFlowReturn
gst_identity_transform_ip (GstBaseTransform * trans, GstBuffer * buf)
{
  GstFlowReturn ret = GST_FLOW_OK;
  GstIdentity *identity = GST_IDENTITY (trans);
  GstClockTime rundts = GST_CLOCK_TIME_NONE;
  GstClockTime runpts = GST_CLOCK_TIME_NONE;
  GstClockTime ts, duration, runtimestamp;
  gsize size;

  size = gst_buffer_get_size (buf);

  if (identity->check_imperfect_timestamp)
    gst_identity_check_imperfect_timestamp (identity, buf);
  if (identity->check_imperfect_offset)
    gst_identity_check_imperfect_offset (identity, buf);

  /* update prev values */
  identity->prev_timestamp = GST_BUFFER_TIMESTAMP (buf);
  identity->prev_duration = GST_BUFFER_DURATION (buf);
  identity->prev_offset_end = GST_BUFFER_OFFSET_END (buf);
  identity->prev_offset = GST_BUFFER_OFFSET (buf);

  if (identity->error_after >= 0) {
    identity->error_after--;
    if (identity->error_after == 0)
      goto error_after;
  }

  if (identity->drop_probability > 0.0) {
    if ((gfloat) (1.0 * rand () / (RAND_MAX)) < identity->drop_probability)
      goto dropped;
  }

  if (identity->dump) {
    GstMapInfo info;

    gst_buffer_map (buf, &info, GST_MAP_READ);
    gst_util_dump_mem (info.data, info.size);
    gst_buffer_unmap (buf, &info);
  }

  if (!identity->silent) {
    gst_identity_update_last_message_for_buffer (identity, "chain", buf, size);
  }

  if (identity->datarate > 0) {
    GstClockTime time = gst_util_uint64_scale_int (identity->offset,
        GST_SECOND, identity->datarate);

    GST_BUFFER_PTS (buf) = GST_BUFFER_DTS (buf) = time;
    GST_BUFFER_DURATION (buf) = size * GST_SECOND / identity->datarate;
  }

  if (identity->signal_handoffs)
    g_signal_emit (identity, gst_identity_signals[SIGNAL_HANDOFF], 0, buf);

  if (trans->segment.format == GST_FORMAT_TIME) {
    rundts = gst_segment_to_running_time (&trans->segment,
        GST_FORMAT_TIME, GST_BUFFER_DTS (buf));
    runpts = gst_segment_to_running_time (&trans->segment,
        GST_FORMAT_TIME, GST_BUFFER_PTS (buf));
  }

  if (GST_CLOCK_TIME_IS_VALID (rundts))
    runtimestamp = rundts;
  else if (GST_CLOCK_TIME_IS_VALID (runpts))
    runtimestamp = runpts;
  else
    runtimestamp = 0;
  ret = gst_identity_do_sync (identity, runtimestamp);

  identity->offset += size;

  if (identity->sleep_time && ret == GST_FLOW_OK)
    g_usleep (identity->sleep_time);

  if (identity->single_segment && (trans->segment.format == GST_FORMAT_TIME)
      && (ret == GST_FLOW_OK)) {
    GST_BUFFER_DTS (buf) = rundts;
    GST_BUFFER_PTS (buf) = runpts;
    GST_BUFFER_OFFSET (buf) = GST_CLOCK_TIME_NONE;
    GST_BUFFER_OFFSET_END (buf) = GST_CLOCK_TIME_NONE;
  }

  return ret;

  /* ERRORS */
error_after:
  {
    GST_ELEMENT_ERROR (identity, CORE, FAILED,
        (_("Failed after iterations as requested.")), (NULL));
    return GST_FLOW_ERROR;
  }
dropped:
  {
    if (!identity->silent) {
      gst_identity_update_last_message_for_buffer (identity, "dropping", buf,
          size);
    }

    ts = GST_BUFFER_TIMESTAMP (buf);
    if (GST_CLOCK_TIME_IS_VALID (ts)) {
      duration = GST_BUFFER_DURATION (buf);
      gst_pad_push_event (GST_BASE_TRANSFORM_SRC_PAD (identity),
          gst_event_new_gap (ts, duration));
    }

    /* return DROPPED to basetransform. */
    return GST_BASE_TRANSFORM_FLOW_DROPPED;
  }
}
static GstFlowReturn
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
{
  GstFlowReturn res = GST_FLOW_OK;
  gsize size;
  guint8 *data;
  GstBuffer *outbuf, *bufd;
  gint16 *out_data;
  int n, err;
  int samples;
  unsigned int packet_size;
  GstBuffer *buf;
  GstMapInfo map, omap;
  GstAudioClippingMeta *cmeta = NULL;

  if (dec->state == NULL) {
    /* If we did not get any headers, default to 2 channels */
    if (dec->n_channels == 0) {
      GST_INFO_OBJECT (dec, "No header, assuming single stream");
      dec->n_channels = 2;
      dec->sample_rate = 48000;
      /* default stereo mapping */
      dec->channel_mapping_family = 0;
      dec->channel_mapping[0] = 0;
      dec->channel_mapping[1] = 1;
      dec->n_streams = 1;
      dec->n_stereo_streams = 1;

      if (!gst_opus_dec_negotiate (dec, NULL))
        return GST_FLOW_NOT_NEGOTIATED;
    }

    if (dec->n_channels == 2 && dec->n_streams == 1
        && dec->n_stereo_streams == 0) {
      /* if we are automatically decoding 2 channels, but only have
         a single encoded one, direct both channels to it */
      dec->channel_mapping[1] = 0;
    }

    GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
        dec->n_channels, dec->sample_rate);
#ifndef GST_DISABLE_GST_DEBUG
    gst_opus_common_log_channel_mapping_table (GST_ELEMENT (dec), opusdec_debug,
        "Mapping table", dec->n_channels, dec->channel_mapping);
#endif

    GST_DEBUG_OBJECT (dec, "%d streams, %d stereo", dec->n_streams,
        dec->n_stereo_streams);
    dec->state =
        opus_multistream_decoder_create (dec->sample_rate, dec->n_channels,
        dec->n_streams, dec->n_stereo_streams, dec->channel_mapping, &err);
    if (!dec->state || err != OPUS_OK)
      goto creation_failed;
  }

  if (buffer) {
    GST_DEBUG_OBJECT (dec, "Received buffer of size %" G_GSIZE_FORMAT,
        gst_buffer_get_size (buffer));
  } else {
    GST_DEBUG_OBJECT (dec, "Received missing buffer");
  }

  /* if using in-band FEC, we introdude one extra frame's delay as we need
     to potentially wait for next buffer to decode a missing buffer */
  if (dec->use_inband_fec && !dec->primed) {
    GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out");
    gst_buffer_replace (&dec->last_buffer, buffer);
    dec->primed = TRUE;
    goto done;
  }

  /* That's the buffer we'll be sending to the opus decoder. */
  buf = (dec->use_inband_fec
      && gst_buffer_get_size (dec->last_buffer) >
      0) ? dec->last_buffer : buffer;

  /* That's the buffer we get duration from */
  bufd = dec->use_inband_fec ? dec->last_buffer : buffer;

  if (buf && gst_buffer_get_size (buf) > 0) {
    gst_buffer_map (buf, &map, GST_MAP_READ);
    data = map.data;
    size = map.size;
    GST_DEBUG_OBJECT (dec, "Using buffer of size %" G_GSIZE_FORMAT, size);
  } else {
    /* concealment data, pass NULL as the bits parameters */
    GST_DEBUG_OBJECT (dec, "Using NULL buffer");
    data = NULL;
    size = 0;
  }

  if (gst_buffer_get_size (bufd) == 0) {
    GstClockTime const opus_plc_alignment = 2500 * GST_USECOND;
    GstClockTime aligned_missing_duration;
    GstClockTime missing_duration = GST_BUFFER_DURATION (bufd);

    if (!GST_CLOCK_TIME_IS_VALID (missing_duration) || missing_duration == 0) {
      if (GST_CLOCK_TIME_IS_VALID (dec->last_known_buffer_duration)) {
        missing_duration = dec->last_known_buffer_duration;
        GST_WARNING_OBJECT (dec,
            "Missing duration, using last duration %" GST_TIME_FORMAT,
            GST_TIME_ARGS (missing_duration));
      } else {
        GST_WARNING_OBJECT (dec,
            "Missing buffer, but unknown duration, and no previously known duration, assuming 20 ms");
        missing_duration = 20 * GST_MSECOND;
      }
    }

    GST_DEBUG_OBJECT (dec,
        "missing buffer, doing PLC duration %" GST_TIME_FORMAT
        " plus leftover %" GST_TIME_FORMAT, GST_TIME_ARGS (missing_duration),
        GST_TIME_ARGS (dec->leftover_plc_duration));

    /* add the leftover PLC duration to that of the buffer */
    missing_duration += dec->leftover_plc_duration;

    /* align the combined buffer and leftover PLC duration to multiples
     * of 2.5ms, rounding to nearest, and store excess duration for later */
    aligned_missing_duration =
        ((missing_duration +
            opus_plc_alignment / 2) / opus_plc_alignment) * opus_plc_alignment;
    dec->leftover_plc_duration = missing_duration - aligned_missing_duration;

    /* Opus' PLC cannot operate with less than 2.5ms; skip PLC
     * and accumulate the missing duration in the leftover_plc_duration
     * for the next PLC attempt */
    if (aligned_missing_duration < opus_plc_alignment) {
      GST_DEBUG_OBJECT (dec,
          "current duration %" GST_TIME_FORMAT
          " of missing data not enough for PLC (minimum needed: %"
          GST_TIME_FORMAT ") - skipping", GST_TIME_ARGS (missing_duration),
          GST_TIME_ARGS (opus_plc_alignment));
      goto done;
    }

    /* convert the duration (in nanoseconds) to sample count */
    samples =
        gst_util_uint64_scale_int (aligned_missing_duration, dec->sample_rate,
        GST_SECOND);

    GST_DEBUG_OBJECT (dec,
        "calculated PLC frame length: %" GST_TIME_FORMAT
        " num frame samples: %d new leftover: %" GST_TIME_FORMAT,
        GST_TIME_ARGS (aligned_missing_duration), samples,
        GST_TIME_ARGS (dec->leftover_plc_duration));
  } else {
    /* use maximum size (120 ms) as the number of returned samples is
       not constant over the stream. */
    samples = 120 * dec->sample_rate / 1000;
  }
  packet_size = samples * dec->n_channels * 2;

  outbuf =
      gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec),
      packet_size);
  if (!outbuf) {
    goto buffer_failed;
  }

  if (size > 0)
    dec->last_known_buffer_duration = packet_duration_opus (data, size);

  gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
  out_data = (gint16 *) omap.data;

  do {
    if (dec->use_inband_fec) {
      if (gst_buffer_get_size (dec->last_buffer) > 0) {
        /* normal delayed decode */
        GST_LOG_OBJECT (dec, "FEC enabled, decoding last delayed buffer");
        n = opus_multistream_decode (dec->state, data, size, out_data, samples,
            0);
      } else {
        /* FEC reconstruction decode */
        GST_LOG_OBJECT (dec, "FEC enabled, reconstructing last buffer");
        n = opus_multistream_decode (dec->state, data, size, out_data, samples,
            1);
      }
    } else {
      /* normal decode */
      GST_LOG_OBJECT (dec, "FEC disabled, decoding buffer");
      n = opus_multistream_decode (dec->state, data, size, out_data, samples,
          0);
    }
    if (n == OPUS_BUFFER_TOO_SMALL) {
      /* if too small, add 2.5 milliseconds and try again, up to the
       * Opus max size of 120 milliseconds */
      if (samples >= 120 * dec->sample_rate / 1000)
        break;
      samples += 25 * dec->sample_rate / 10000;
      packet_size = samples * dec->n_channels * 2;
      gst_buffer_unmap (outbuf, &omap);
      gst_buffer_unref (outbuf);
      outbuf =
          gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec),
          packet_size);
      if (!outbuf) {
        goto buffer_failed;
      }
      gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
      out_data = (gint16 *) omap.data;
    }
  } while (n == OPUS_BUFFER_TOO_SMALL);
  gst_buffer_unmap (outbuf, &omap);
  if (data != NULL)
    gst_buffer_unmap (buf, &map);

  if (n < 0) {
    GstFlowReturn ret = GST_FLOW_ERROR;

    gst_buffer_unref (outbuf);
    GST_AUDIO_DECODER_ERROR (dec, 1, STREAM, DECODE, (NULL),
        ("Decoding error (%d): %s", n, opus_strerror (n)), ret);
    return ret;
  }
  GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
  gst_buffer_set_size (outbuf, n * 2 * dec->n_channels);
  GST_BUFFER_DURATION (outbuf) = samples * GST_SECOND / dec->sample_rate;
  samples = n;

  cmeta = gst_buffer_get_audio_clipping_meta (buf);

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

  /* Skip any samples that need skipping */
  if (cmeta && cmeta->start) {
    guint pre_skip = cmeta->start;
    guint scaled_pre_skip = pre_skip * dec->sample_rate / 48000;
    guint skip = scaled_pre_skip > n ? n : scaled_pre_skip;
    guint scaled_skip = skip * 48000 / dec->sample_rate;

    gst_buffer_resize (outbuf, skip * 2 * dec->n_channels, -1);

    GST_INFO_OBJECT (dec,
        "Skipping %u samples at the beginning (%u at 48000 Hz)",
        skip, scaled_skip);
  }

  if (cmeta && cmeta->end) {
    guint post_skip = cmeta->end;
    guint scaled_post_skip = post_skip * dec->sample_rate / 48000;
    guint skip = scaled_post_skip > n ? n : scaled_post_skip;
    guint scaled_skip = skip * 48000 / dec->sample_rate;
    guint outsize = gst_buffer_get_size (outbuf);
    guint skip_bytes = skip * 2 * dec->n_channels;

    if (outsize > skip_bytes)
      outsize -= skip_bytes;
    else
      outsize = 0;

    gst_buffer_resize (outbuf, 0, outsize);

    GST_INFO_OBJECT (dec,
        "Skipping %u samples at the end (%u at 48000 Hz)", skip, scaled_skip);
  }

  if (gst_buffer_get_size (outbuf) == 0) {
    gst_buffer_unref (outbuf);
    outbuf = NULL;
  } else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) {
    gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16,
        dec->n_channels, dec->opus_pos, dec->info.position);
  }

  /* Apply gain */
  /* Would be better off leaving this to a volume element, as this is
     a naive conversion that does too many int/float conversions.
     However, we don't have control over the pipeline...
     So make it optional if the user program wants to use a volume,
     but do it by default so the correct volume goes out by default */
  if (dec->apply_gain && outbuf && dec->r128_gain) {
    gsize rsize;
    unsigned int i, nsamples;
    double volume = dec->r128_gain_volume;
    gint16 *samples;

    gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE);
    samples = (gint16 *) omap.data;
    rsize = omap.size;
    GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
    nsamples = rsize / 2;
    for (i = 0; i < nsamples; ++i) {
      int sample = (int) (samples[i] * volume + 0.5);
      samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
    }
    gst_buffer_unmap (outbuf, &omap);
  }

  if (dec->use_inband_fec) {
    gst_buffer_replace (&dec->last_buffer, buffer);
  }

  res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);

  if (res != GST_FLOW_OK)
    GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));

done:
  return res;

creation_failed:
  GST_ELEMENT_ERROR (dec, LIBRARY, INIT, ("Failed to create Opus decoder"),
      ("Failed to create Opus decoder (%d): %s", err, opus_strerror (err)));
  return GST_FLOW_ERROR;

buffer_failed:
  GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
      ("Failed to create %u byte buffer", packet_size));
  return GST_FLOW_ERROR;
}
Exemplo n.º 20
0
static GstFlowReturn
gst_cutter_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
  GstFlowReturn ret = GST_FLOW_OK;
  GstCutter *filter;
  GstMapInfo map;
  gint16 *in_data;
  gint bpf, rate;
  gsize in_size;
  guint num_samples;
  gdouble NCS = 0.0;            /* Normalized Cumulative Square of buffer */
  gdouble RMS = 0.0;            /* RMS of signal in buffer */
  gdouble NMS = 0.0;            /* Normalized Mean Square of buffer */
  GstBuffer *prebuf;            /* pointer to a prebuffer element */
  GstClockTime duration;

  filter = GST_CUTTER (parent);

  if (GST_AUDIO_INFO_FORMAT (&filter->info) == GST_AUDIO_FORMAT_UNKNOWN)
    goto not_negotiated;

  bpf = GST_AUDIO_INFO_BPF (&filter->info);
  rate = GST_AUDIO_INFO_RATE (&filter->info);

  gst_buffer_map (buf, &map, GST_MAP_READ);
  in_data = (gint16 *) map.data;
  in_size = map.size;

  GST_LOG_OBJECT (filter, "length of prerec buffer: %" GST_TIME_FORMAT,
      GST_TIME_ARGS (filter->pre_run_length));

  /* calculate mean square value on buffer */
  switch (GST_AUDIO_INFO_FORMAT (&filter->info)) {
    case GST_AUDIO_FORMAT_S16:
      num_samples = in_size / 2;
      gst_cutter_calculate_gint16 (in_data, num_samples, &NCS);
      NMS = NCS / num_samples;
      break;
    case GST_AUDIO_FORMAT_S8:
      num_samples = in_size;
      gst_cutter_calculate_gint8 ((gint8 *) in_data, num_samples, &NCS);
      NMS = NCS / num_samples;
      break;
    default:
      /* this shouldn't happen */
      g_warning ("no mean square function for format");
      break;
  }

  gst_buffer_unmap (buf, &map);

  filter->silent_prev = filter->silent;

  duration = gst_util_uint64_scale (in_size / bpf, GST_SECOND, rate);

  RMS = sqrt (NMS);
  /* if RMS below threshold, add buffer length to silent run length count
   * if not, reset
   */
  GST_LOG_OBJECT (filter, "buffer stats: NMS %f, RMS %f, audio length %f", NMS,
      RMS, gst_guint64_to_gdouble (duration));

  if (RMS < filter->threshold_level)
    filter->silent_run_length += gst_guint64_to_gdouble (duration);
  else {
    filter->silent_run_length = 0 * GST_SECOND;
    filter->silent = FALSE;
  }

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

  /* has the silent status changed ? if so, send right signal
   * and, if from silent -> not silent, flush pre_record buffer
   */
  if (filter->silent != filter->silent_prev) {
    if (filter->silent) {
      GstMessage *m =
          gst_cutter_message_new (filter, FALSE, GST_BUFFER_TIMESTAMP (buf));
      GST_DEBUG_OBJECT (filter, "signaling CUT_STOP");
      gst_element_post_message (GST_ELEMENT (filter), m);
    } else {
      gint count = 0;
      GstMessage *m =
          gst_cutter_message_new (filter, TRUE, GST_BUFFER_TIMESTAMP (buf));

      GST_DEBUG_OBJECT (filter, "signaling CUT_START");
      gst_element_post_message (GST_ELEMENT (filter), m);
      /* first of all, flush current buffer */
      GST_DEBUG_OBJECT (filter, "flushing buffer of length %" GST_TIME_FORMAT,
          GST_TIME_ARGS (filter->pre_run_length));

      while (filter->pre_buffer) {
        prebuf = (g_list_first (filter->pre_buffer))->data;
        filter->pre_buffer = g_list_remove (filter->pre_buffer, prebuf);
        gst_pad_push (filter->srcpad, prebuf);
        ++count;
      }
      GST_DEBUG_OBJECT (filter, "flushed %d buffers", count);
      filter->pre_run_length = 0 * GST_SECOND;
    }
  }
  /* now check if we have to send the new buffer to the internal buffer cache
   * or to the srcpad */
  if (filter->silent) {
    filter->pre_buffer = g_list_append (filter->pre_buffer, buf);
    filter->pre_run_length += gst_guint64_to_gdouble (duration);

    while (filter->pre_run_length > filter->pre_length) {
      GstClockTime pduration;
      gsize psize;

      prebuf = (g_list_first (filter->pre_buffer))->data;
      g_assert (GST_IS_BUFFER (prebuf));

      psize = gst_buffer_get_size (prebuf);
      pduration = gst_util_uint64_scale (psize / bpf, GST_SECOND, rate);

      filter->pre_buffer = g_list_remove (filter->pre_buffer, prebuf);
      filter->pre_run_length -= gst_guint64_to_gdouble (pduration);

      /* only pass buffers if we don't leak */
      if (!filter->leaky)
        ret = gst_pad_push (filter->srcpad, prebuf);
      else
        gst_buffer_unref (prebuf);
    }
  } else
    ret = gst_pad_push (filter->srcpad, buf);

  return ret;

  /* ERRORS */
not_negotiated:
  {
    return GST_FLOW_NOT_NEGOTIATED;
  }
}
Exemplo n.º 21
0
/*
 * Adds buffer sizes together.
 */
static void
buffer_count_size (void *buffer, void *user_data)
{
  guint *sum = (guint *) user_data;
  *sum += gst_buffer_get_size (buffer);
}
Exemplo n.º 22
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;
  }
}
static gboolean
gst_mpegv_parse_process_config (GstMpegvParse * mpvparse, GstBuffer * buf,
    guint size)
{
  guint8 *data;
  guint8 *data_with_prefix;
  GstMapInfo map;
  gint i, offset;

  if (mpvparse->seq_offset < 4) {
    /* This shouldn't happen, but just in case... */
    GST_WARNING_OBJECT (mpvparse, "Sequence header start code missing.");
    return FALSE;
  }

  gst_buffer_map (buf, &map, GST_MAP_READ);
  data = map.data + mpvparse->seq_offset;
  g_assert (size <= map.size);
  /* pointer to sequence header data including the start code prefix -
     used for codec private data */
  data_with_prefix = data - 4;

  /* only do stuff if something new; only compare first 11 bytes, changes in
     quantiser matrix doesn't matter here. Also changing the matrices in
     codec_data seems to cause problem with decoders */
  if (mpvparse->config && size == gst_buffer_get_size (mpvparse->config) &&
      gst_buffer_memcmp (mpvparse->config, 0, data_with_prefix, MIN (size,
              11)) == 0) {
    gst_buffer_unmap (buf, &map);
    return TRUE;
  }

  if (!gst_mpeg_video_parse_sequence_header (&mpvparse->sequencehdr, data,
          size - mpvparse->seq_offset, 0)) {
    GST_DEBUG_OBJECT (mpvparse,
        "failed to parse config data (size %d) at offset %d",
        size, mpvparse->seq_offset);
    gst_buffer_unmap (buf, &map);
    return FALSE;
  }

  GST_LOG_OBJECT (mpvparse, "accepting parsed config size %d", size);

  /* Set mpeg version, and parse sequence extension */
  mpvparse->config_flags = FLAG_NONE;
  for (i = 0; i < mpvparse->ext_count; ++i) {
    offset = mpvparse->ext_offsets[i];
    mpvparse->config_flags |= FLAG_MPEG2;
    if (offset < size) {
      if (gst_mpeg_video_parse_sequence_extension (&mpvparse->sequenceext,
              map.data, size, offset)) {
        GST_LOG_OBJECT (mpvparse, "Read Sequence Extension");
        mpvparse->config_flags |= FLAG_SEQUENCE_EXT;
      } else
          if (gst_mpeg_video_parse_sequence_display_extension
          (&mpvparse->sequencedispext, map.data, size, offset)) {
        GST_LOG_OBJECT (mpvparse, "Read Sequence Display Extension");
        mpvparse->config_flags |= FLAG_SEQUENCE_DISPLAY_EXT;
      }
    }
  }
  if (mpvparse->config_flags & FLAG_MPEG2) {
    /* Update the sequence header based on extensions */
    GstMpegVideoSequenceExt *seqext = NULL;
    GstMpegVideoSequenceDisplayExt *seqdispext = NULL;

    if (mpvparse->config_flags & FLAG_SEQUENCE_EXT)
      seqext = &mpvparse->sequenceext;
    if (mpvparse->config_flags & FLAG_SEQUENCE_DISPLAY_EXT)
      seqdispext = &mpvparse->sequencedispext;

    gst_mpeg_video_finalise_mpeg2_sequence_header (&mpvparse->sequencehdr,
        seqext, seqdispext);
  }

  if (mpvparse->fps_num == 0 || mpvparse->fps_den == 0) {
    mpvparse->fps_num = mpvparse->sequencehdr.fps_n;
    mpvparse->fps_den = mpvparse->sequencehdr.fps_d;
  }

  /* parsing ok, so accept it as new config */
  if (mpvparse->config != NULL)
    gst_buffer_unref (mpvparse->config);

  mpvparse->config = gst_buffer_new_and_alloc (size);
  gst_buffer_fill (mpvparse->config, 0, data_with_prefix, size);

  /* trigger src caps update */
  mpvparse->update_caps = TRUE;

  gst_buffer_unmap (buf, &map);

  return TRUE;
}
/* Fill */
static GstFlowReturn gst_android_video_source_fill(GstPushSrc * p_pushsrc, GstBuffer * p_buf)
{
    GstAndroidVideoSource *p_src;
    int vcd_ret;
    static struct timeval time_of_day;
    static struct timeval window_time;
    static struct timeval start_time;
    int timeDiffUsec;
    static gint frame_count = 0;
    static gint frame_count_window = 0;
    GstBuffer *p_outbuf;
    GstMapInfo mem_info;
    gboolean ok;

    GA_LOGTRACE("ENTER %s --xx--> thread(%ld)", __FUNCTION__, pthread_self());

    p_src = GST_ANDROIDVIDEOSOURCE(p_pushsrc);

    if (gst_buffer_get_size(p_buf) != p_src->m_bufSize) {
        GA_LOGWARN("%s: WARNING: gst_buffer_get_size(p_buf)==%d != p_src->m_bufSize==%d", __FUNCTION__, gst_buffer_get_size(p_buf), p_src->m_bufSize);
        goto fill_error_negotiation;
    }

    VCD_checkChangeCamera(p_src->m_devHandle);

    if (!p_src->vcdStarted) {
        AV_CHECK_ERR(VCD_start(p_src->m_devHandle), fill_error_vcd_start);
        p_src->vcdStarted = TRUE;
    }

    if (!frame_count) { // Only first time
        gettimeofday(&start_time, NULL);
        gettimeofday(&window_time, NULL);
    }

    frame_count++;
    frame_count_window++;

    gettimeofday(&time_of_day, NULL);
    timeDiffUsec = time_diff_usec(&time_of_day, &window_time);
    if (timeDiffUsec > p_src->log_interval || !p_src->log_interval) {
        int framerate;
        int framerateWindow;
        int timeDiffSec;
        timeDiffSec = time_diff_sec(&time_of_day, &start_time);
        framerate = frame_count / (timeDiffSec > 0 ? timeDiffSec : 1);
        framerateWindow = frame_count_window * 1000000 / timeDiffUsec;
        GA_LOGVERB("%s ------> has now been called %d times --Create--> framerate since start: %d fps, framerate last %d usec: %d fps", __FUNCTION__, frame_count, framerate, timeDiffUsec, framerateWindow);
        gettimeofday(&window_time, NULL);
        frame_count_window = 0;
    }

    g_warn_if_fail(gst_buffer_is_writable(p_buf)); /* g_warn_if_fail() used for internal error (exception to our rules for this special "buffer copying case") */
    g_assert(gst_buffer_is_writable(p_buf)); /* this buf should be allocated in the base class and should always be writable */
    p_outbuf = gst_buffer_make_writable(p_buf); /* do this cause we never wanna crash in release even if somebody makes a mistake somewhere... */

    ok = gst_buffer_map(p_outbuf, &mem_info, GST_MAP_WRITE);
    if (!ok) {
        goto fill_error_gst_buffer_map;
    }
    vcd_ret = VCD_read(p_src->m_devHandle, &(mem_info.data), mem_info.size);
    if (vcd_ret == VCD_ERR_NO_DATA) {
        // This should never happen. There should always be more data from the device.
        // In any case, if it happens we don't want to end or lock or anything, we just
        // want to go on as if there actually were data. Specifically, we do not want
        // to block the streaming thread...
        memset(mem_info.data, 0, mem_info.size);
        gst_buffer_unmap(p_outbuf, &mem_info);
        GST_BUFFER_OFFSET(p_outbuf) = GST_BUFFER_OFFSET_NONE;
        GST_BUFFER_OFFSET_END(p_outbuf) = GST_BUFFER_OFFSET_NONE;
        GST_BUFFER_PTS(p_outbuf) = GST_CLOCK_TIME_NONE;
        GST_BUFFER_DTS(p_outbuf) = GST_CLOCK_TIME_NONE;
        GST_BUFFER_DURATION(p_outbuf) = GST_CLOCK_TIME_NONE;
        GA_LOGWARN("%s: WARNING: Returning GST_FLOW_OK with a buffer with zeros...", __FUNCTION__);
        return GST_FLOW_OK;
    }
    if (vcd_ret != VCD_NO_ERROR) {
        gst_buffer_unmap(p_outbuf, &mem_info);
        goto fill_error_read;
    }
    gst_buffer_unmap(p_outbuf, &mem_info);

    set_gstbuf_time_and_offset(p_src, p_outbuf);

    GA_LOGTRACE("EXIT %s", __FUNCTION__);
    return GST_FLOW_OK;

    /*
     * propagate unhandled errors
     */
fill_error_read:
    {
        GA_LOGERROR("%s: Error when reading data from device!", __FUNCTION__);
        return GST_FLOW_ERROR;
    }
fill_error_negotiation:
    {
        GA_LOGERROR("%s: ERROR: Strange buffer size. Negotiation not done? Disallowed renegotiation done?", __FUNCTION__);
        return GST_FLOW_ERROR;
    }
fill_error_gst_buffer_map:
    {
        GA_LOGERROR("%s: gst_buffer_map() failed!", __FUNCTION__);
        return GST_FLOW_ERROR;
    }
fill_error_vcd_start:
    {
        GA_LOGERROR("%s: FATAL ERROR: Could not start the video device!", __FUNCTION__);
        return GST_FLOW_ERROR;
    }
}
Exemplo n.º 25
0
static GstFlowReturn
gst_fake_sink_render (GstBaseSink * bsink, GstBuffer * buf)
{
  GstFakeSink *sink = GST_FAKE_SINK_CAST (bsink);

  if (sink->num_buffers_left == 0)
    goto eos;

  if (sink->num_buffers_left != -1)
    sink->num_buffers_left--;

  if (!sink->silent) {
    gchar dts_str[64], pts_str[64], dur_str[64];
    gchar *flag_str, *meta_str;

    GST_OBJECT_LOCK (sink);
    g_free (sink->last_message);

    if (GST_BUFFER_DTS (buf) != GST_CLOCK_TIME_NONE) {
      g_snprintf (dts_str, sizeof (dts_str), "%" GST_TIME_FORMAT,
          GST_TIME_ARGS (GST_BUFFER_DTS (buf)));
    } else {
      g_strlcpy (dts_str, "none", sizeof (dts_str));
    }

    if (GST_BUFFER_PTS (buf) != GST_CLOCK_TIME_NONE) {
      g_snprintf (pts_str, sizeof (pts_str), "%" GST_TIME_FORMAT,
          GST_TIME_ARGS (GST_BUFFER_PTS (buf)));
    } else {
      g_strlcpy (pts_str, "none", sizeof (pts_str));
    }

    if (GST_BUFFER_DURATION (buf) != GST_CLOCK_TIME_NONE) {
      g_snprintf (dur_str, sizeof (dur_str), "%" GST_TIME_FORMAT,
          GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
    } else {
      g_strlcpy (dur_str, "none", sizeof (dur_str));
    }

    flag_str = gst_buffer_get_flags_string (buf);
    meta_str = gst_buffer_get_meta_string (buf);

    sink->last_message =
        g_strdup_printf ("chain   ******* (%s:%s) (%u bytes, dts: %s, pts: %s"
        ", duration: %s, offset: %" G_GINT64_FORMAT ", offset_end: %"
        G_GINT64_FORMAT ", flags: %08x %s, meta: %s) %p",
        GST_DEBUG_PAD_NAME (GST_BASE_SINK_CAST (sink)->sinkpad),
        (guint) gst_buffer_get_size (buf), dts_str, pts_str,
        dur_str, GST_BUFFER_OFFSET (buf), GST_BUFFER_OFFSET_END (buf),
        GST_MINI_OBJECT_CAST (buf)->flags, flag_str,
        meta_str ? meta_str : "none", buf);
    g_free (flag_str);
    g_free (meta_str);
    GST_OBJECT_UNLOCK (sink);

    gst_fake_sink_notify_last_message (sink);
  }
  if (sink->signal_handoffs)
    g_signal_emit (sink, gst_fake_sink_signals[SIGNAL_HANDOFF], 0, buf,
        bsink->sinkpad);

  if (sink->dump) {
    GstMapInfo info;

    if (gst_buffer_map (buf, &info, GST_MAP_READ)) {
      gst_util_dump_mem (info.data, info.size);
      gst_buffer_unmap (buf, &info);
    }
  }
  if (sink->num_buffers_left == 0)
    goto eos;

  return GST_FLOW_OK;

  /* ERRORS */
eos:
  {
    GST_DEBUG_OBJECT (sink, "we are EOS");
    return GST_FLOW_EOS;
  }
}
Exemplo n.º 26
0
static GstFlowReturn
gst_wavenc_write_toc (GstWavEnc * wavenc)
{
  GList *list;
  GstToc *toc;
  GstTocEntry *entry, *subentry;
  GstBuffer *buf;
  GstMapInfo map;
  guint8 *data;
  guint32 ncues, size, cues_size, labls_size, notes_size;

  if (!wavenc->toc) {
    GST_DEBUG_OBJECT (wavenc, "have no toc, checking toc_setter");
    wavenc->toc = gst_toc_setter_get_toc (GST_TOC_SETTER (wavenc));
  }
  if (!wavenc->toc) {
    GST_WARNING_OBJECT (wavenc, "have no toc");
    return GST_FLOW_OK;
  }

  toc = gst_toc_ref (wavenc->toc);
  size = 0;
  cues_size = 0;
  labls_size = 0;
  notes_size = 0;

  /* check if the TOC entries is valid */
  list = gst_toc_get_entries (toc);
  entry = list->data;
  if (gst_toc_entry_is_alternative (entry)) {
    list = gst_toc_entry_get_sub_entries (entry);
    while (list) {
      subentry = list->data;
      if (!gst_toc_entry_is_sequence (subentry))
        return FALSE;
      list = g_list_next (list);
    }
    list = gst_toc_entry_get_sub_entries (entry);
  }
  if (gst_toc_entry_is_sequence (entry)) {
    while (list) {
      entry = list->data;
      if (!gst_toc_entry_is_sequence (entry))
        return FALSE;
      list = g_list_next (list);
    }
    list = gst_toc_get_entries (toc);
  }

  ncues = g_list_length (list);
  GST_DEBUG_OBJECT (wavenc, "number of cue entries: %d", ncues);

  while (list) {
    guint32 id = 0;
    gint64 id64;
    const gchar *uid;

    entry = list->data;
    uid = gst_toc_entry_get_uid (entry);
    id64 = g_ascii_strtoll (uid, NULL, 0);
    /* check if id unique compatible with guint32 else generate random */
    if (id64 >= 0 && gst_wavenc_is_cue_id_unique (id64, wavenc->cues)) {
      id = (guint32) id64;
    } else {
      do {
        id = g_random_int ();
      } while (!gst_wavenc_is_cue_id_unique (id, wavenc->cues));
    }
    gst_wavenc_parse_cue (wavenc, id, entry);
    gst_wavenc_parse_labl (wavenc, id, entry);
    gst_wavenc_parse_note (wavenc, id, entry);
    list = g_list_next (list);
  }

  /* count cues size */
  if (wavenc->cues) {
    cues_size = 24 * g_list_length (wavenc->cues);
    size += 12 + cues_size;
  } else {
    GST_WARNING_OBJECT (wavenc, "cue's not found");
    return FALSE;
  }
  /* count labls size */
  if (wavenc->labls) {
    list = wavenc->labls;
    while (list) {
      GstWavEncLabl *labl;
      labl = list->data;
      labls_size += 8 + GST_ROUND_UP_2 (labl->chunk_data_size);
      list = g_list_next (list);
    }
    size += labls_size;
  }
  /* count notes size */
  if (wavenc->notes) {
    list = wavenc->notes;
    while (list) {
      GstWavEncNote *note;
      note = list->data;
      notes_size += 8 + GST_ROUND_UP_2 (note->chunk_data_size);
      list = g_list_next (list);
    }
    size += notes_size;
  }
  if (wavenc->labls || wavenc->notes) {
    size += 12;
  }

  buf = gst_buffer_new_and_alloc (size);
  gst_buffer_map (buf, &map, GST_MAP_WRITE);
  data = map.data;
  memset (data, 0, size);

  /* write Cue Chunk */
  if (wavenc->cues) {
    memcpy (data, (gchar *) "cue ", 4);
    GST_WRITE_UINT32_LE (data + 4, 4 + cues_size);
    GST_WRITE_UINT32_LE (data + 8, ncues);
    data += 12;
    gst_wavenc_write_cues (&data, wavenc->cues);

    /* write Associated Data List Chunk */
    if (wavenc->labls || wavenc->notes) {
      memcpy (data, (gchar *) "LIST", 4);
      GST_WRITE_UINT32_LE (data + 4, 4 + labls_size + notes_size);
      memcpy (data + 8, (gchar *) "adtl", 4);
      data += 12;
      if (wavenc->labls)
        gst_wavenc_write_labls (&data, wavenc->labls);
      if (wavenc->notes)
        gst_wavenc_write_notes (&data, wavenc->notes);
    }
  }

  /* free resources */
  if (toc)
    gst_toc_unref (toc);
  if (wavenc->cues)
    g_list_free_full (wavenc->cues, g_free);
  if (wavenc->labls)
    g_list_free_full (wavenc->labls, g_free);
  if (wavenc->notes)
    g_list_free_full (wavenc->notes, g_free);

  gst_buffer_unmap (buf, &map);
  wavenc->meta_length += gst_buffer_get_size (buf);

  return gst_pad_push (wavenc->srcpad, buf);
}
static GstFlowReturn
gst_rtp_vraw_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer)
{
  GstRtpVRawPay *rtpvrawpay;
  GstFlowReturn ret = GST_FLOW_OK;
  gfloat packets_per_packline;
  guint pgroups_per_packet;
  guint packlines_per_list, buffers_per_list;
  guint lines_delay;            /* after how many packed lines we push out a buffer list */
  guint last_line;              /* last pack line number we pushed out a buffer list     */
  guint line, offset;
  guint8 *p0, *yp, *up, *vp;
  guint ystride, uvstride;
  guint xinc, yinc;
  guint pgroup;
  guint mtu;
  guint width, height;
  gint field, fields;
  GstVideoFormat format;
  GstVideoFrame frame;
  gint interlaced;
  gboolean use_buffer_lists;
  GstBufferList *list = NULL;
  GstRTPBuffer rtp = { NULL, };

  rtpvrawpay = GST_RTP_VRAW_PAY (payload);

  if (!gst_video_frame_map (&frame, &rtpvrawpay->vinfo, buffer, GST_MAP_READ)) {
    gst_buffer_unref (buffer);
    return GST_FLOW_ERROR;
  }

  GST_LOG_OBJECT (rtpvrawpay, "new frame of %" G_GSIZE_FORMAT " bytes",
      gst_buffer_get_size (buffer));

  /* get pointer and strides of the planes */
  p0 = GST_VIDEO_FRAME_PLANE_DATA (&frame, 0);
  yp = GST_VIDEO_FRAME_COMP_DATA (&frame, 0);
  up = GST_VIDEO_FRAME_COMP_DATA (&frame, 1);
  vp = GST_VIDEO_FRAME_COMP_DATA (&frame, 2);

  ystride = GST_VIDEO_FRAME_COMP_STRIDE (&frame, 0);
  uvstride = GST_VIDEO_FRAME_COMP_STRIDE (&frame, 1);

  mtu = GST_RTP_BASE_PAYLOAD_MTU (payload);

  /* amount of bytes for one pixel */
  pgroup = rtpvrawpay->pgroup;
  width = GST_VIDEO_INFO_WIDTH (&rtpvrawpay->vinfo);
  height = GST_VIDEO_INFO_HEIGHT (&rtpvrawpay->vinfo);

  interlaced = GST_VIDEO_INFO_IS_INTERLACED (&rtpvrawpay->vinfo);

  format = GST_VIDEO_INFO_FORMAT (&rtpvrawpay->vinfo);

  yinc = rtpvrawpay->yinc;
  xinc = rtpvrawpay->xinc;

  /* after how many packed lines we push out a buffer list */
  lines_delay = GST_ROUND_UP_4 (height / rtpvrawpay->chunks_per_frame);

  /* calculate how many buffers we expect to store in a single buffer list */
  pgroups_per_packet = (mtu - (12 + 14)) / pgroup;
  packets_per_packline = width / (xinc * pgroups_per_packet * 1.0);
  packlines_per_list = height / (yinc * rtpvrawpay->chunks_per_frame);
  buffers_per_list = packlines_per_list * packets_per_packline;
  buffers_per_list = GST_ROUND_UP_8 (buffers_per_list);

  use_buffer_lists = (rtpvrawpay->chunks_per_frame < (height / yinc));

  fields = 1 + interlaced;

  /* start with line 0, offset 0 */
  for (field = 0; field < fields; field++) {
    line = field;
    offset = 0;
    last_line = 0;

    if (use_buffer_lists)
      list = gst_buffer_list_new_sized (buffers_per_list);

    /* write all lines */
    while (line < height) {
      guint left, pack_line;
      GstBuffer *out;
      guint8 *outdata, *headers;
      gboolean next_line, complete = FALSE;
      guint length, cont, pixels;

      /* get the max allowed payload length size, we try to fill the complete MTU */
      left = gst_rtp_buffer_calc_payload_len (mtu, 0, 0);
      out = gst_rtp_buffer_new_allocate (left, 0, 0);

      if (field == 0) {
        GST_BUFFER_PTS (out) = GST_BUFFER_PTS (buffer);
      } else {
        GST_BUFFER_PTS (out) = GST_BUFFER_PTS (buffer) +
            GST_BUFFER_DURATION (buffer) / 2;
      }

      gst_rtp_buffer_map (out, GST_MAP_WRITE, &rtp);
      outdata = gst_rtp_buffer_get_payload (&rtp);

      GST_LOG_OBJECT (rtpvrawpay, "created buffer of size %u for MTU %u", left,
          mtu);

      /*
       *   0                   1                   2                   3
       *   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
       *  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       *  |   Extended Sequence Number    |            Length             |
       *  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       *  |F|          Line No            |C|           Offset            |
       *  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       *  |            Length             |F|          Line No            |
       *  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       *  |C|           Offset            |                               .
       *  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               .
       *  .                                                               .
       *  .                 Two (partial) lines of video data             .
       *  .                                                               .
       *  +---------------------------------------------------------------+
       */

      /* need 2 bytes for the extended sequence number */
      *outdata++ = 0;
      *outdata++ = 0;
      left -= 2;

      /* the headers start here */
      headers = outdata;

      /* make sure we can fit at least *one* header and pixel */
      if (!(left > (6 + pgroup))) {
        gst_rtp_buffer_unmap (&rtp);
        gst_buffer_unref (out);
        goto too_small;
      }

      /* while we can fit at least one header and one pixel */
      while (left > (6 + pgroup)) {
        /* we need a 6 bytes header */
        left -= 6;

        /* get how may bytes we need for the remaining pixels */
        pixels = width - offset;
        length = (pixels * pgroup) / xinc;

        if (left >= length) {
          /* pixels and header fit completely, we will write them and skip to the
           * next line. */
          next_line = TRUE;
        } else {
          /* line does not fit completely, see how many pixels fit */
          pixels = (left / pgroup) * xinc;
          length = (pixels * pgroup) / xinc;
          next_line = FALSE;
        }
        GST_LOG_OBJECT (rtpvrawpay, "filling %u bytes in %u pixels", length,
            pixels);
        left -= length;

        /* write length */
        *outdata++ = (length >> 8) & 0xff;
        *outdata++ = length & 0xff;

        /* write line no */
        *outdata++ = ((line >> 8) & 0x7f) | ((field << 7) & 0x80);
        *outdata++ = line & 0xff;

        if (next_line) {
          /* go to next line we do this here to make the check below easier */
          line += yinc;
        }

        /* calculate continuation marker */
        cont = (left > (6 + pgroup) && line < height) ? 0x80 : 0x00;

        /* write offset and continuation marker */
        *outdata++ = ((offset >> 8) & 0x7f) | cont;
        *outdata++ = offset & 0xff;

        if (next_line) {
          /* reset offset */
          offset = 0;
          GST_LOG_OBJECT (rtpvrawpay, "go to next line %u", line);
        } else {
          offset += pixels;
          GST_LOG_OBJECT (rtpvrawpay, "next offset %u", offset);
        }

        if (!cont)
          break;
      }
      GST_LOG_OBJECT (rtpvrawpay, "consumed %u bytes",
          (guint) (outdata - headers));

      /* second pass, read headers and write the data */
      while (TRUE) {
        guint offs, lin;

        /* read length and cont */
        length = (headers[0] << 8) | headers[1];
        lin = ((headers[2] & 0x7f) << 8) | headers[3];
        offs = ((headers[4] & 0x7f) << 8) | headers[5];
        cont = headers[4] & 0x80;
        pixels = length / pgroup;
        headers += 6;

        GST_LOG_OBJECT (payload,
            "writing length %u, line %u, offset %u, cont %d", length, lin, offs,
            cont);

        switch (format) {
          case GST_VIDEO_FORMAT_RGB:
          case GST_VIDEO_FORMAT_RGBA:
          case GST_VIDEO_FORMAT_BGR:
          case GST_VIDEO_FORMAT_BGRA:
          case GST_VIDEO_FORMAT_UYVY:
          case GST_VIDEO_FORMAT_UYVP:
            offs /= xinc;
            memcpy (outdata, p0 + (lin * ystride) + (offs * pgroup), length);
            outdata += length;
            break;
          case GST_VIDEO_FORMAT_AYUV:
          {
            gint i;
            guint8 *datap;

            datap = p0 + (lin * ystride) + (offs * 4);

            for (i = 0; i < pixels; i++) {
              *outdata++ = datap[2];
              *outdata++ = datap[1];
              *outdata++ = datap[3];
              datap += 4;
            }
            break;
          }
          case GST_VIDEO_FORMAT_I420:
          {
            gint i;
            guint uvoff;
            guint8 *yd1p, *yd2p, *udp, *vdp;

            yd1p = yp + (lin * ystride) + (offs);
            yd2p = yd1p + ystride;
            uvoff = (lin / yinc * uvstride) + (offs / xinc);
            udp = up + uvoff;
            vdp = vp + uvoff;

            for (i = 0; i < pixels; i++) {
              *outdata++ = *yd1p++;
              *outdata++ = *yd1p++;
              *outdata++ = *yd2p++;
              *outdata++ = *yd2p++;
              *outdata++ = *udp++;
              *outdata++ = *vdp++;
            }
            break;
          }
          case GST_VIDEO_FORMAT_Y41B:
          {
            gint i;
            guint uvoff;
            guint8 *ydp, *udp, *vdp;

            ydp = yp + (lin * ystride) + offs;
            uvoff = (lin / yinc * uvstride) + (offs / xinc);
            udp = up + uvoff;
            vdp = vp + uvoff;

            for (i = 0; i < pixels; i++) {
              *outdata++ = *udp++;
              *outdata++ = *ydp++;
              *outdata++ = *ydp++;
              *outdata++ = *vdp++;
              *outdata++ = *ydp++;
              *outdata++ = *ydp++;
            }
            break;
          }
          default:
            gst_rtp_buffer_unmap (&rtp);
            gst_buffer_unref (out);
            goto unknown_sampling;
        }

        if (!cont)
          break;
      }

      if (line >= height) {
        GST_LOG_OBJECT (rtpvrawpay, "field/frame complete, set marker");
        gst_rtp_buffer_set_marker (&rtp, TRUE);
        complete = TRUE;
      }
      gst_rtp_buffer_unmap (&rtp);
      if (left > 0) {
        GST_LOG_OBJECT (rtpvrawpay, "we have %u bytes left", left);
        gst_buffer_resize (out, 0, gst_buffer_get_size (out) - left);
      }

      gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpvrawpay), out, buffer,
          g_quark_from_static_string (GST_META_TAG_VIDEO_STR));


      /* Now either push out the buffer directly */
      if (!use_buffer_lists) {
        ret = gst_rtp_base_payload_push (payload, out);
        continue;
      }

      /* or add the buffer to buffer list ... */
      gst_buffer_list_add (list, out);

      /* .. and check if we need to push out the list */
      pack_line = (line - field) / fields;
      if (complete || (pack_line > last_line && pack_line % lines_delay == 0)) {
        GST_LOG_OBJECT (rtpvrawpay, "pushing list of %u buffers up to pack "
            "line %u", gst_buffer_list_length (list), pack_line);
        ret = gst_rtp_base_payload_push_list (payload, list);
        list = NULL;
        if (!complete)
          list = gst_buffer_list_new_sized (buffers_per_list);
        last_line = pack_line;
      }
    }

  }

  gst_video_frame_unmap (&frame);
  gst_buffer_unref (buffer);

  return ret;

  /* ERRORS */
unknown_sampling:
  {
    GST_ELEMENT_ERROR (payload, STREAM, FORMAT,
        (NULL), ("unimplemented sampling"));
    gst_video_frame_unmap (&frame);
    gst_buffer_unref (buffer);
    return GST_FLOW_NOT_SUPPORTED;
  }
too_small:
  {
    GST_ELEMENT_ERROR (payload, RESOURCE, NO_SPACE_LEFT,
        (NULL), ("not enough space to send at least one pixel"));
    gst_video_frame_unmap (&frame);
    gst_buffer_unref (buffer);
    return GST_FLOW_NOT_SUPPORTED;
  }
}
Exemplo n.º 28
0
GstFlowReturn
gst_kate_util_decoder_base_chain_kate_packet (GstKateDecoderBase * decoder,
    GstElement * element, GstPad * pad, GstBuffer * buf, GstPad * srcpad,
    GstPad * tagpad, GstCaps ** src_caps, const kate_event ** ev)
{
  kate_packet kp;
  int ret;
  GstFlowReturn rflow = GST_FLOW_OK;
  gboolean is_header;
  guint8 *data;
  gsize size;
  guint8 header[1];

  size = gst_buffer_extract (buf, 0, header, 1);

  GST_DEBUG_OBJECT (element, "got kate packet, %u bytes, type %02x",
      gst_buffer_get_size (buf), size == 0 ? -1 : header[0]);

  is_header = size > 0 && (header[0] & 0x80);

  if (!is_header && decoder->tags) {
    /* after we've processed headers, send any tags before processing the data packet */
    GST_DEBUG_OBJECT (element, "Not a header, sending tags for pad %s:%s",
        GST_DEBUG_PAD_NAME (tagpad));
    gst_element_found_tags_for_pad (element, tagpad, decoder->tags);
    decoder->tags = NULL;
  }

  data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
  kate_packet_wrap (&kp, size, data);
  ret = kate_high_decode_packetin (&decoder->k, &kp, ev);
  gst_buffer_unmap (buf, data, size);

  if (G_UNLIKELY (ret < 0)) {
    GST_ELEMENT_ERROR (element, STREAM, DECODE, (NULL),
        ("Failed to decode Kate packet: %s",
            gst_kate_util_get_error_message (ret)));
    return GST_FLOW_ERROR;
  }

  if (G_UNLIKELY (ret > 0)) {
    GST_DEBUG_OBJECT (element,
        "kate_high_decode_packetin has received EOS packet");
  }

  /* headers may be interesting to retrieve information from */
  if (G_UNLIKELY (is_header)) {
    switch (header[0]) {
      case 0x80:               /* ID header */
        GST_INFO_OBJECT (element, "Parsed ID header: language %s, category %s",
            decoder->k.ki->language, decoder->k.ki->category);
        if (src_caps) {
          if (*src_caps) {
            gst_caps_unref (*src_caps);
            *src_caps = NULL;
          }
          if (strcmp (decoder->k.ki->category, "K-SPU") == 0 ||
              strcmp (decoder->k.ki->category, "spu-subtitles") == 0) {
            *src_caps = gst_caps_new_empty_simple ("video/x-dvd-subpicture");
          } else if (decoder->k.ki->text_markup_type == kate_markup_none) {
            *src_caps = gst_caps_new_empty_simple ("text/plain");
          } else {
            *src_caps = gst_caps_new_empty_simple ("text/x-pango-markup");
          }
          GST_INFO_OBJECT (srcpad, "Setting caps: %" GST_PTR_FORMAT, *src_caps);
          if (!gst_pad_set_caps (srcpad, *src_caps)) {
            GST_ERROR_OBJECT (srcpad, "Failed to set caps %" GST_PTR_FORMAT,
                *src_caps);
          }
        }
        if (decoder->k.ki->language && *decoder->k.ki->language) {
          GstTagList *old = decoder->tags, *tags = gst_tag_list_new_empty ();
          if (tags) {
            gchar *lang_code;

            /* en_GB -> en */
            lang_code = g_ascii_strdown (decoder->k.ki->language, -1);
            g_strdelimit (lang_code, NULL, '\0');
            gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_LANGUAGE_CODE,
                lang_code, NULL);
            g_free (lang_code);
            /* TODO: category - where should it go ? */
            decoder->tags =
                gst_tag_list_merge (decoder->tags, tags, GST_TAG_MERGE_REPLACE);
            gst_tag_list_free (tags);
            if (old)
              gst_tag_list_free (old);
          }
        }

        /* update properties */
        if (decoder->language)
          g_free (decoder->language);
        decoder->language = g_strdup (decoder->k.ki->language);
        if (decoder->category)
          g_free (decoder->category);
        decoder->category = g_strdup (decoder->k.ki->category);
        decoder->original_canvas_width = decoder->k.ki->original_canvas_width;
        decoder->original_canvas_height = decoder->k.ki->original_canvas_height;

        /* we can now send away any event we've delayed, as the src pad now has caps */
        gst_kate_util_decoder_base_drain_event_queue (decoder);

        break;

      case 0x81:               /* Vorbis comments header */
        GST_INFO_OBJECT (element, "Parsed comments header");
        {
          gchar *encoder = NULL;
          GstTagList *old = decoder->tags, *list =
              gst_tag_list_from_vorbiscomment_buffer (buf,
              (const guint8 *) "\201kate\0\0\0\0", 9, &encoder);
          if (list) {
            decoder->tags =
                gst_tag_list_merge (decoder->tags, list, GST_TAG_MERGE_REPLACE);
            gst_tag_list_free (list);
          }

          if (!decoder->tags) {
            GST_ERROR_OBJECT (element, "failed to decode comment header");
            decoder->tags = gst_tag_list_new_empty ();
          }
          if (encoder) {
            gst_tag_list_add (decoder->tags, GST_TAG_MERGE_REPLACE,
                GST_TAG_ENCODER, encoder, NULL);
            g_free (encoder);
          }
          gst_tag_list_add (decoder->tags, GST_TAG_MERGE_REPLACE,
              GST_TAG_SUBTITLE_CODEC, "Kate", NULL);
          gst_tag_list_add (decoder->tags, GST_TAG_MERGE_REPLACE,
              GST_TAG_ENCODER_VERSION, decoder->k.ki->bitstream_version_major,
              NULL);

          if (old)
            gst_tag_list_free (old);

          if (decoder->initialized) {
            gst_element_found_tags_for_pad (element, tagpad, decoder->tags);
            decoder->tags = NULL;
          } else {
            /* Only push them as messages for the time being. *
             * They will be pushed on the pad once the decoder is initialized */
            gst_element_post_message (element,
                gst_message_new_tag (GST_OBJECT (element),
                    gst_tag_list_copy (decoder->tags)));
          }
        }
        break;

      default:
        break;
    }
  }
#if ((KATE_VERSION_MAJOR<<16)|(KATE_VERSION_MINOR<<8)|KATE_VERSION_PATCH) >= 0x000400
  else if (*ev && (*ev)->meta) {
    int count = kate_meta_query_count ((*ev)->meta);
    if (count > 0) {
      GstTagList *evtags = gst_tag_list_new_empty ();
      int idx;
      GST_DEBUG_OBJECT (decoder, "Kate event has %d attached metadata", count);
      for (idx = 0; idx < count; ++idx) {
        const char *tag, *value;
        size_t len;
        if (kate_meta_query ((*ev)->meta, idx, &tag, &value, &len) < 0) {
          GST_WARNING_OBJECT (decoder, "Failed to retrieve metadata %d", idx);
        } else {
          if (gst_kate_util_is_utf8_string (value, len)) {
            gchar *compound = g_strdup_printf ("%s=%s", tag, value);
            GST_DEBUG_OBJECT (decoder, "Metadata %d: %s=%s (%zu bytes)", idx,
                tag, value, len);
            gst_tag_list_add (evtags, GST_TAG_MERGE_APPEND,
                GST_TAG_EXTENDED_COMMENT, compound, NULL);
            g_free (compound);
          } else {
            GST_INFO_OBJECT (decoder,
                "Metadata %d, (%s, %zu bytes) is binary, ignored", idx, tag,
                len);
          }
        }
      }
      if (gst_tag_list_is_empty (evtags))
        gst_tag_list_free (evtags);
      else
        gst_element_found_tags_for_pad (element, tagpad, evtags);
    }
  }
#endif

  return rflow;
}
Exemplo n.º 29
0
static GstFlowReturn
gst_bz2enc_chain (GstPad * pad, GstObject * parent, GstBuffer * in)
{
  GstFlowReturn flow = GST_FLOW_OK;
  GstBuffer *out;
  GstBz2enc *b;
  guint n;
  int bz2_ret;
  GstMapInfo map, omap;

  b = GST_BZ2ENC (parent);

  if (!b->ready)
    goto not_ready;

  gst_buffer_map (in, &map, GST_MAP_READ);
  b->stream.next_in = (char *) map.data;
  b->stream.avail_in = map.size;
  while (b->stream.avail_in) {
    out = gst_buffer_new_and_alloc (b->buffer_size);

    gst_buffer_map (out, &omap, GST_MAP_WRITE);
    b->stream.next_out = (char *) omap.data;
    b->stream.avail_out = omap.size;
    bz2_ret = BZ2_bzCompress (&b->stream, BZ_RUN);
    gst_buffer_unmap (out, &omap);
    if (bz2_ret != BZ_RUN_OK)
      goto compress_error;

    n = gst_buffer_get_size (out);
    if (b->stream.avail_out >= n) {
      gst_buffer_unref (out);
      break;
    }

    gst_buffer_resize (out, 0, n - b->stream.avail_out);
    n = gst_buffer_get_size (out);
    GST_BUFFER_OFFSET (out) = b->stream.total_out_lo32 - n;

    flow = gst_pad_push (b->src, out);

    if (flow != GST_FLOW_OK)
      break;

    b->offset += n;
  }

done:

  gst_buffer_unmap (in, &map);
  gst_buffer_unref (in);
  return flow;

/* ERRORS */
not_ready:
  {
    GST_ELEMENT_ERROR (b, LIBRARY, FAILED, (NULL), ("Compressor not ready."));
    flow = GST_FLOW_FLUSHING;
    goto done;
  }
compress_error:
  {
    GST_ELEMENT_ERROR (b, STREAM, ENCODE, (NULL),
        ("Failed to compress data (error code %i)", bz2_ret));
    gst_bz2enc_compress_init (b);
    gst_buffer_unref (out);
    flow = GST_FLOW_ERROR;
    goto done;
  }
}
Exemplo n.º 30
0
/**
 * gst_audio_buffer_clip:
 * @buffer: (transfer full): The buffer to clip.
 * @segment: Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which
 *           the buffer should be clipped.
 * @rate: sample rate.
 * @bpf: size of one audio frame in bytes. This is the size of one sample
 * * channels.
 *
 * Clip the buffer to the given %GstSegment.
 *
 * After calling this function the caller does not own a reference to
 * @buffer anymore.
 *
 * Returns: (transfer full): %NULL if the buffer is completely outside the configured segment,
 * otherwise the clipped buffer is returned.
 *
 * If the buffer has no timestamp, it is assumed to be inside the segment and
 * is not clipped
 */
GstBuffer *
gst_audio_buffer_clip (GstBuffer * buffer, GstSegment * segment, gint rate,
    gint bpf)
{
  GstBuffer *ret;
  GstClockTime timestamp = GST_CLOCK_TIME_NONE, duration = GST_CLOCK_TIME_NONE;
  guint64 offset = GST_BUFFER_OFFSET_NONE, offset_end = GST_BUFFER_OFFSET_NONE;
  gsize trim, size, osize;
  gboolean change_duration = TRUE, change_offset = TRUE, change_offset_end =
      TRUE;

  g_return_val_if_fail (segment->format == GST_FORMAT_TIME ||
      segment->format == GST_FORMAT_DEFAULT, buffer);
  g_return_val_if_fail (GST_IS_BUFFER (buffer), NULL);

  if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
    /* No timestamp - assume the buffer is completely in the segment */
    return buffer;

  /* Get copies of the buffer metadata to change later.
   * Calculate the missing values for the calculations,
   * they won't be changed later though. */

  trim = 0;
  osize = size = gst_buffer_get_size (buffer);

  /* no data, nothing to clip */
  if (!size)
    return buffer;

  timestamp = GST_BUFFER_TIMESTAMP (buffer);
  GST_DEBUG ("timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp));
  if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
    duration = GST_BUFFER_DURATION (buffer);
  } else {
    change_duration = FALSE;
    duration = gst_util_uint64_scale (size / bpf, GST_SECOND, rate);
  }

  if (GST_BUFFER_OFFSET_IS_VALID (buffer)) {
    offset = GST_BUFFER_OFFSET (buffer);
  } else {
    change_offset = FALSE;
    offset = 0;
  }

  if (GST_BUFFER_OFFSET_END_IS_VALID (buffer)) {
    offset_end = GST_BUFFER_OFFSET_END (buffer);
  } else {
    change_offset_end = FALSE;
    offset_end = offset + size / bpf;
  }

  if (segment->format == GST_FORMAT_TIME) {
    /* Handle clipping for GST_FORMAT_TIME */

    guint64 start, stop, cstart, cstop, diff;

    start = timestamp;
    stop = timestamp + duration;

    if (gst_segment_clip (segment, GST_FORMAT_TIME,
            start, stop, &cstart, &cstop)) {

      diff = cstart - start;
      if (diff > 0) {
        timestamp = cstart;

        if (change_duration)
          duration -= diff;

        diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
        if (change_offset)
          offset += diff;
        trim += diff * bpf;
        size -= diff * bpf;
      }

      diff = stop - cstop;
      if (diff > 0) {
        /* duration is always valid if stop is valid */
        duration -= diff;

        diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
        if (change_offset_end)
          offset_end -= diff;
        size -= diff * bpf;
      }
    } else {
      gst_buffer_unref (buffer);
      return NULL;
    }
  } else {
    /* Handle clipping for GST_FORMAT_DEFAULT */
    guint64 start, stop, cstart, cstop, diff;

    g_return_val_if_fail (GST_BUFFER_OFFSET_IS_VALID (buffer), buffer);

    start = offset;
    stop = offset_end;

    if (gst_segment_clip (segment, GST_FORMAT_DEFAULT,
            start, stop, &cstart, &cstop)) {

      diff = cstart - start;
      if (diff > 0) {
        offset = cstart;

        timestamp = gst_util_uint64_scale (cstart, GST_SECOND, rate);

        if (change_duration)
          duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);

        trim += diff * bpf;
        size -= diff * bpf;
      }

      diff = stop - cstop;
      if (diff > 0) {
        offset_end = cstop;

        if (change_duration)
          duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);

        size -= diff * bpf;
      }
    } else {
      gst_buffer_unref (buffer);
      return NULL;
    }
  }

  if (trim == 0 && size == osize) {
    ret = buffer;

    if (GST_BUFFER_TIMESTAMP (ret) != timestamp) {
      ret = gst_buffer_make_writable (ret);
      GST_BUFFER_TIMESTAMP (ret) = timestamp;
    }
    if (GST_BUFFER_DURATION (ret) != duration) {
      ret = gst_buffer_make_writable (ret);
      GST_BUFFER_DURATION (ret) = duration;
    }
  } else {
    /* Get a writable buffer and apply all changes */
    GST_DEBUG ("trim %" G_GSIZE_FORMAT " size %" G_GSIZE_FORMAT, trim, size);
    ret = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, trim, size);
    gst_buffer_unref (buffer);

    GST_DEBUG ("timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp));
    GST_BUFFER_TIMESTAMP (ret) = timestamp;

    if (change_duration)
      GST_BUFFER_DURATION (ret) = duration;
    if (change_offset)
      GST_BUFFER_OFFSET (ret) = offset;
    if (change_offset_end)
      GST_BUFFER_OFFSET_END (ret) = offset_end;
  }
  return ret;
}