void camera_calibrate_run(GstCameraCalibrate *calib, IplImage *img)
{
  cv::Mat view = cv::cvarrToMat(img);

  // For camera only take new samples after delay time
  if (calib->mode == CAPTURING) {
    // get_input
    cv::Size imageSize = view.size();

    /* find_pattern
     * FIXME find ways to reduce CPU usage
     * don't do it on all frames ? will it help ? corner display will be affected.
     * in a separate frame?
     * in a separate element that gets composited back into the main stream
     * (video is tee-d into it and can then be decimated, scaled, etc..) */

    std::vector<cv::Point2f> pointBuf;
    bool found;
    int chessBoardFlags = cv::CALIB_CB_ADAPTIVE_THRESH | cv::CALIB_CB_NORMALIZE_IMAGE;

    if (!calib->useFisheye) {
      /* fast check erroneously fails with high distortions like fisheye */
      chessBoardFlags |= cv::CALIB_CB_FAST_CHECK;
    }

    /* Find feature points on the input format */
    switch(calib->calibrationPattern) {
      case GST_CAMERA_CALIBRATION_PATTERN_CHESSBOARD:
        found = cv::findChessboardCorners(view, calib->boardSize, pointBuf, chessBoardFlags);
        break;
      case GST_CAMERA_CALIBRATION_PATTERN_CIRCLES_GRID:
        found = cv::findCirclesGrid(view, calib->boardSize, pointBuf);
        break;
      case GST_CAMERA_CALIBRATION_PATTERN_ASYMMETRIC_CIRCLES_GRID:
        found = cv::findCirclesGrid(view, calib->boardSize, pointBuf, cv::CALIB_CB_ASYMMETRIC_GRID );
        break;
      default:
        found = FALSE;
        break;
    }

    bool blinkOutput = FALSE;
    if (found) {
      /* improve the found corners' coordinate accuracy for chessboard */
      if (calib->calibrationPattern == GST_CAMERA_CALIBRATION_PATTERN_CHESSBOARD && calib->cornerSubPix) {
        /* FIXME findChessboardCorners and alike do a cv::COLOR_BGR2GRAY (and a histogram balance)
         * the color convert should be done once (if needed) and shared
         * FIXME keep viewGray around to avoid reallocating it each time... */
        cv::Mat viewGray;
        cv::cvtColor(view, viewGray, cv::COLOR_BGR2GRAY);
        cv::cornerSubPix(viewGray, pointBuf, cv::Size(11, 11), cv::Size(-1, -1),
            cv::TermCriteria(cv::TermCriteria::EPS + cv::TermCriteria::COUNT, 30, 0.1));
      }

      /* take new samples after delay time */
      if ((calib->mode == CAPTURING) && ((clock() - calib->prevTimestamp) > calib->delay * 1e-3 * CLOCKS_PER_SEC)) {
        calib->imagePoints.push_back(pointBuf);
        calib->prevTimestamp = clock();
        blinkOutput = true;
      }

      /* draw the corners */
      if (calib->showCorners) {
        cv::drawChessboardCorners(view, calib->boardSize, cv::Mat(pointBuf), found);
      }
    }

    /* if got enough frames then stop calibration and show result */
    if (calib->mode == CAPTURING && calib->imagePoints.size() >= (size_t)calib->nrFrames) {

      if (camera_calibrate_calibrate(calib, imageSize, calib->cameraMatrix, calib->distCoeffs, calib->imagePoints)) {
        calib->mode = CALIBRATED;

        GstPad *sink_pad = GST_BASE_TRANSFORM_SINK_PAD (calib);
        GstPad *src_pad = GST_BASE_TRANSFORM_SRC_PAD (calib);
        GstEvent *sink_event;
        GstEvent *src_event;

        /* set settings property */
        g_free (calib->settings);
        calib->settings = camera_serialize_undistort_settings(calib->cameraMatrix, calib->distCoeffs);

        /* create calibrated event and send upstream and downstream */
        sink_event = gst_camera_event_new_calibrated (calib->settings);
        GST_LOG_OBJECT (sink_pad, "Sending upstream event %s.", GST_EVENT_TYPE_NAME (sink_event));
        if (!gst_pad_push_event (sink_pad, sink_event)) {
          GST_WARNING_OBJECT (sink_pad, "Sending upstream event %p (%s) failed.",
              sink_event, GST_EVENT_TYPE_NAME (sink_event));
        }

        src_event = gst_camera_event_new_calibrated (calib->settings);
        GST_LOG_OBJECT (src_pad, "Sending downstream event %s.", GST_EVENT_TYPE_NAME (src_event));
        if (!gst_pad_push_event (src_pad, src_event)) {
          GST_WARNING_OBJECT (src_pad, "Sending downstream event %p (%s) failed.",
              src_event, GST_EVENT_TYPE_NAME (src_event));
        }
      } else {
        /* failed to calibrate, go back to detection mode */
        calib->mode = DETECTION;
      }
    }

    if (calib->mode == CAPTURING && blinkOutput) {
      bitwise_not(view, view);
    }

  }

  /* output text */
  /* FIXME ll additional rendering (text, corners, ...) should be done with
   * cairo or another gst framework.
   * this will relax the conditions on the input format (RBG only at the moment).
   * the calibration itself accepts more formats... */

  std::string msg = (calib->mode == CAPTURING) ? "100/100" :
      (calib->mode == CALIBRATED) ? "Calibrated" : "Waiting...";
  int baseLine = 0;
  cv::Size textSize = cv::getTextSize(msg, 1, 1, 1, &baseLine);
  cv::Point textOrigin(view.cols - 2 * textSize.width - 10, view.rows - 2 * baseLine - 10);

  if (calib->mode == CAPTURING) {
    msg = cv::format("%d/%d", (int)calib->imagePoints.size(), calib->nrFrames);
  }

  const cv::Scalar RED(0,0,255);
  const cv::Scalar GREEN(0,255,0);

  cv::putText(view, msg, textOrigin, 1, 1, calib->mode == CALIBRATED ?  GREEN : RED);
}
Beispiel #2
0
static gboolean
gst_raw_parse_sink_event (GstPad * pad, GstEvent * event)
{
  GstRawParse *rp = GST_RAW_PARSE (gst_pad_get_parent (pad));
  gboolean ret;

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_EOS:
    case GST_EVENT_FLUSH_STOP:
      /* Only happens in push mode */
      gst_raw_parse_reset (rp);
      ret = gst_pad_push_event (rp->srcpad, event);
      break;
    case GST_EVENT_NEWSEGMENT:
    {
      GstClockTimeDiff start, stop, time;
      gdouble rate, arate;
      gboolean update;
      GstFormat format;

      /* Only happens in push mode */

      gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
          &start, &stop, &time);

      if (format == GST_FORMAT_TIME) {
        gst_segment_set_newsegment_full (&rp->segment, update, rate, arate,
            GST_FORMAT_TIME, start, stop, time);
        ret = gst_pad_push_event (rp->srcpad, event);
      } else {

        gst_event_unref (event);

        ret =
            gst_raw_parse_convert (rp, format, start, GST_FORMAT_TIME, &start);
        ret &= gst_raw_parse_convert (rp, format, time, GST_FORMAT_TIME, &time);
        ret &= gst_raw_parse_convert (rp, format, stop, GST_FORMAT_TIME, &stop);
        if (!ret) {
          GST_ERROR_OBJECT (rp,
              "Failed converting to GST_FORMAT_TIME format (%d)", format);
          break;
        }

        gst_segment_set_newsegment_full (&rp->segment, update, rate, arate,
            GST_FORMAT_TIME, start, stop, time);

        /* create new segment with the fields converted to time */
        event = gst_event_new_new_segment_full (update, rate, arate,
            GST_FORMAT_TIME, start, stop, time);

        ret = gst_pad_push_event (rp->srcpad, event);
      }
      break;
    }
    default:
      ret = gst_pad_event_default (rp->sinkpad, event);
      break;
  }

  gst_object_unref (rp);

  return ret;
}
static gboolean
gst_rtp_mux_setcaps (GstPad * pad, GstRTPMux * rtp_mux, GstCaps * caps)
{
  GstStructure *structure;
  gboolean ret = FALSE;
  GstRTPMuxPadPrivate *padpriv;
  GstCaps *peercaps;

  if (!gst_caps_is_fixed (caps))
    return FALSE;

  peercaps = gst_pad_peer_query_caps (rtp_mux->srcpad, NULL);
  if (peercaps) {
    GstCaps *tcaps, *othercaps;;
    tcaps = gst_pad_get_pad_template_caps (pad);
    othercaps = gst_caps_intersect_full (peercaps, tcaps,
        GST_CAPS_INTERSECT_FIRST);

    if (gst_caps_get_size (othercaps) > 0) {
      structure = gst_caps_get_structure (othercaps, 0);
      GST_OBJECT_LOCK (rtp_mux);
      if (gst_structure_get_uint (structure, "ssrc", &rtp_mux->current_ssrc)) {
        GST_DEBUG_OBJECT (pad, "Use downstream ssrc: %x",
            rtp_mux->current_ssrc);
        rtp_mux->have_ssrc = TRUE;
      }
      GST_OBJECT_UNLOCK (rtp_mux);
    }

    gst_caps_unref (othercaps);

    gst_caps_unref (peercaps);
    gst_caps_unref (tcaps);
  }

  structure = gst_caps_get_structure (caps, 0);

  if (!structure)
    return FALSE;

  GST_OBJECT_LOCK (rtp_mux);
  padpriv = gst_pad_get_element_private (pad);
  if (padpriv &&
      gst_structure_get_uint (structure, "timestamp-offset",
          &padpriv->timestamp_offset)) {
    padpriv->have_timestamp_offset = TRUE;
  }

  caps = gst_caps_copy (caps);

  /* if we don't have a specified ssrc, first try to take one from the caps,
     and if that fails, generate one */
  if (!rtp_mux->have_ssrc) {
    if (rtp_mux->ssrc_random) {
      if (!gst_structure_get_uint (structure, "ssrc", &rtp_mux->current_ssrc))
        rtp_mux->current_ssrc = g_random_int ();
      rtp_mux->have_ssrc = TRUE;
    }
  }

  gst_caps_set_simple (caps,
      "timestamp-offset", G_TYPE_UINT, rtp_mux->ts_base,
      "seqnum-offset", G_TYPE_UINT, rtp_mux->seqnum_base,
      "ssrc", G_TYPE_UINT, rtp_mux->current_ssrc, NULL);

  GST_OBJECT_UNLOCK (rtp_mux);

  if (rtp_mux->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 (rtp_mux->srcpad, gst_event_new_stream_start (s_id));

    rtp_mux->send_stream_start = FALSE;
  }

  GST_DEBUG_OBJECT (rtp_mux,
      "setting caps %" GST_PTR_FORMAT " on src pad..", caps);
  ret = gst_pad_set_caps (rtp_mux->srcpad, caps);


  gst_caps_unref (caps);

  return ret;
}
static gboolean
gst_musepackdec_handle_seek_event (GstMusepackDec * dec, GstEvent * event)
{
  GstSeekType start_type, stop_type;
  GstSeekFlags flags;
  GstSegment segment;
  GstFormat format;
  gboolean flush;
  gdouble rate;
  gint64 start, stop;
  gint samplerate;

  gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start,
      &stop_type, &stop);

  if (format != GST_FORMAT_TIME && format != GST_FORMAT_DEFAULT) {
    GST_DEBUG_OBJECT (dec, "seek failed: only TIME or DEFAULT format allowed");
    return FALSE;
  }

  samplerate = g_atomic_int_get (&dec->rate);

  if (format == GST_FORMAT_TIME) {
    if (start_type != GST_SEEK_TYPE_NONE)
      start = gst_util_uint64_scale_int (start, samplerate, GST_SECOND);
    if (stop_type != GST_SEEK_TYPE_NONE)
      stop = gst_util_uint64_scale_int (stop, samplerate, GST_SECOND);
  }

  flush = ((flags & GST_SEEK_FLAG_FLUSH) == GST_SEEK_FLAG_FLUSH);

  if (flush)
    gst_pad_push_event (dec->srcpad, gst_event_new_flush_start ());
  else
    gst_pad_pause_task (dec->sinkpad);  /* not _stop_task()? */

  GST_PAD_STREAM_LOCK (dec->sinkpad);

  /* operate on segment copy until we know the seek worked */
  segment = dec->segment;

  gst_segment_set_seek (&segment, rate, GST_FORMAT_DEFAULT,
      flags, start_type, start, stop_type, stop, NULL);

  gst_pad_push_event (dec->sinkpad, gst_event_new_flush_stop ());

  GST_DEBUG_OBJECT (dec, "segment: [%" G_GINT64_FORMAT "-%" G_GINT64_FORMAT
      "] = [%" GST_TIME_FORMAT "-%" GST_TIME_FORMAT "]",
      segment.start, segment.stop,
      GST_TIME_ARGS (segment.start * GST_SECOND / dec->rate),
      GST_TIME_ARGS (segment.stop * GST_SECOND / dec->rate));

  GST_DEBUG_OBJECT (dec, "performing seek to sample %" G_GINT64_FORMAT,
      segment.start);

  if (segment.start < 0 || segment.start >= segment.duration) {
    GST_WARNING_OBJECT (dec, "seek out of bounds");
    goto failed;
  }
#ifdef MPC_IS_OLD_API
  if (!mpc_decoder_seek_sample (dec->d, segment.start))
    goto failed;
#else
  if (mpc_demux_seek_sample (dec->d, segment.start) != MPC_STATUS_OK)
    goto failed;
#endif

  if ((flags & GST_SEEK_FLAG_SEGMENT) == GST_SEEK_FLAG_SEGMENT) {
    GST_DEBUG_OBJECT (dec, "posting SEGMENT_START message");

    gst_element_post_message (GST_ELEMENT (dec),
        gst_message_new_segment_start (GST_OBJECT (dec), GST_FORMAT_TIME,
            gst_util_uint64_scale_int (segment.start, GST_SECOND, dec->rate)));
  }

  if (flush) {
    gst_pad_push_event (dec->srcpad, gst_event_new_flush_stop ());
  }

  gst_segment_set_last_stop (&segment, GST_FORMAT_DEFAULT, segment.start);
  dec->segment = segment;
  gst_musepackdec_send_newsegment (dec);

  GST_DEBUG_OBJECT (dec, "seek successful");

  gst_pad_start_task (dec->sinkpad,
      (GstTaskFunction) gst_musepackdec_loop, dec->sinkpad, NULL);

  GST_PAD_STREAM_UNLOCK (dec->sinkpad);

  return TRUE;

failed:
  {
    GST_WARNING_OBJECT (dec, "seek failed");
    GST_PAD_STREAM_UNLOCK (dec->sinkpad);
    return FALSE;
  }
}
static gboolean
rsn_audiomunge_sink_event (GstPad * pad, GstEvent * event)
{
  gboolean ret = FALSE;
  RsnAudioMunge *munge = RSN_AUDIOMUNGE (gst_pad_get_parent (pad));

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_FLUSH_STOP:
      rsn_audiomunge_reset (munge);
      ret = gst_pad_push_event (munge->srcpad, event);
      break;
    case GST_EVENT_NEWSEGMENT:
    {
      GstSegment *segment;
      gboolean update;
      GstFormat format;
      gdouble rate, arate;
      gint64 start, stop, time;

      gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
          &start, &stop, &time);

      /* we need TIME format */
      if (format != GST_FORMAT_TIME)
        goto newseg_wrong_format;

      /* now configure the values */
      segment = &munge->sink_segment;

      gst_segment_set_newsegment_full (segment, update,
          rate, arate, format, start, stop, time);

      if (munge->have_audio) {
        ret = gst_pad_push_event (munge->srcpad, event);
        break;
      }

      /*
       * FIXME:
       * If the accum >= threshold or we're in a still frame and there's been
       * no audio received, then we need to generate some audio data.
       * If caused by a segment start update (time advancing in a gap) adjust
       * the new-segment and send the buffer.
       *
       * Otherwise, send the buffer before the newsegment, so that it appears
       * in the closing segment.
       */
      if (!update) {
        GST_DEBUG_OBJECT (munge, "Sending newsegment: start %" GST_TIME_FORMAT
            " stop %" GST_TIME_FORMAT " accum now %" GST_TIME_FORMAT,
            GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
            GST_TIME_ARGS (segment->accum));

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

      if (segment->accum >= AUDIO_FILL_THRESHOLD || munge->in_still) {
        g_print ("***********  Sending audio fill: accum = %" GST_TIME_FORMAT
            " still-state=%d\n", GST_TIME_ARGS (segment->accum),
            munge->in_still);

        /* Just generate a 100ms silence buffer for now. FIXME: Fill the gap */
        if (rsn_audiomunge_make_audio (munge, segment->start,
                GST_SECOND / 10) == GST_FLOW_OK)
          munge->have_audio = TRUE;
      } else {
        GST_LOG_OBJECT (munge, "Not sending audio fill buffer: "
            "segment accum below thresh: accum = %" GST_TIME_FORMAT,
            GST_TIME_ARGS (segment->accum));
      }

      if (update) {
        GST_DEBUG_OBJECT (munge, "Sending newsegment: start %" GST_TIME_FORMAT
            " stop %" GST_TIME_FORMAT " accum now %" GST_TIME_FORMAT,
            GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
            GST_TIME_ARGS (segment->accum));

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

      break;
    }
    case GST_EVENT_CUSTOM_DOWNSTREAM:
    {
      const GstStructure *s = gst_event_get_structure (event);

      if (s && gst_structure_has_name (s, "application/x-gst-dvd"))
        rsn_audiomunge_handle_dvd_event (munge, event);

      ret = gst_pad_push_event (munge->srcpad, event);
      break;
    }
    default:
      ret = gst_pad_push_event (munge->srcpad, event);
      break;
  }

  return ret;

newseg_wrong_format:

  GST_DEBUG_OBJECT (munge, "received non TIME newsegment");
  gst_event_unref (event);
  gst_object_unref (munge);
  return FALSE;
}
Beispiel #6
0
static GstFlowReturn
gst_flac_tag_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
    GstFlacTag *tag;
    GstFlowReturn ret;
    GstMapInfo map;
    gsize size;

    ret = GST_FLOW_OK;
    tag = GST_FLAC_TAG (parent);

    gst_adapter_push (tag->adapter, buffer);

    /* Initial state, we don't even know if we are dealing with a flac file */
    if (tag->state == GST_FLAC_TAG_STATE_INIT) {
        GstBuffer *id_buffer;

        if (gst_adapter_available (tag->adapter) < sizeof (FLAC_MAGIC))
            goto cleanup;

        id_buffer = gst_adapter_take_buffer (tag->adapter, FLAC_MAGIC_SIZE);
        GST_DEBUG_OBJECT (tag, "looking for " FLAC_MAGIC " identifier");
        if (gst_buffer_memcmp (id_buffer, 0, FLAC_MAGIC, FLAC_MAGIC_SIZE) == 0) {

            GST_DEBUG_OBJECT (tag, "pushing " FLAC_MAGIC " identifier buffer");
            ret = gst_pad_push (tag->srcpad, id_buffer);
            if (ret != GST_FLOW_OK)
                goto cleanup;

            tag->state = GST_FLAC_TAG_STATE_METADATA_BLOCKS;
        } else {
            /* FIXME: does that work well with FLAC files containing ID3v2 tags ? */
            gst_buffer_unref (id_buffer);
            GST_ELEMENT_ERROR (tag, STREAM, WRONG_TYPE, (NULL), (NULL));
            ret = GST_FLOW_ERROR;
        }
    }


    /* The fLaC magic string has been skipped, try to detect the beginning
     * of a metadata block
     */
    if (tag->state == GST_FLAC_TAG_STATE_METADATA_BLOCKS) {
        guint type;
        gboolean is_last;
        const guint8 *block_header;

        g_assert (tag->metadata_block_size == 0);
        g_assert (tag->metadata_last_block == FALSE);

        /* The header of a flac metadata block is 4 bytes long:
         * 1st bit: indicates whether this is the last metadata info block
         * 7 next bits: 4 if vorbis comment block
         * 24 next bits: size of the metadata to follow (big endian)
         */
        if (gst_adapter_available (tag->adapter) < 4)
            goto cleanup;

        block_header = gst_adapter_map (tag->adapter, 4);

        is_last = ((block_header[0] & 0x80) == 0x80);
        type = block_header[0] & 0x7F;
        size = (block_header[1] << 16)
               | (block_header[2] << 8)
               | block_header[3];
        gst_adapter_unmap (tag->adapter);

        /* The 4 bytes long header isn't included in the metadata size */
        tag->metadata_block_size = size + 4;
        tag->metadata_last_block = is_last;

        GST_DEBUG_OBJECT (tag,
                          "got metadata block: %" G_GSIZE_FORMAT " bytes, type %d, "
                          "is vorbiscomment: %d, is last: %d",
                          size, type, (type == 0x04), is_last);

        /* Metadata blocks of type 4 are vorbis comment blocks */
        if (type == 0x04) {
            tag->state = GST_FLAC_TAG_STATE_VC_METADATA_BLOCK;
        } else {
            tag->state = GST_FLAC_TAG_STATE_WRITING_METADATA_BLOCK;
        }
    }


    /* Reads a metadata block */
    if ((tag->state == GST_FLAC_TAG_STATE_WRITING_METADATA_BLOCK) ||
            (tag->state == GST_FLAC_TAG_STATE_VC_METADATA_BLOCK)) {
        GstBuffer *metadata_buffer;

        if (gst_adapter_available (tag->adapter) < tag->metadata_block_size)
            goto cleanup;

        metadata_buffer = gst_adapter_take_buffer (tag->adapter,
                          tag->metadata_block_size);
        /* clear the is-last flag, as the last metadata block will
         * be the vorbis comment block which we will build ourselves.
         */
        gst_buffer_map (metadata_buffer, &map, GST_MAP_READWRITE);
        map.data[0] &= (~0x80);
        gst_buffer_unmap (metadata_buffer, &map);

        if (tag->state == GST_FLAC_TAG_STATE_WRITING_METADATA_BLOCK) {
            GST_DEBUG_OBJECT (tag, "pushing metadata block buffer");
            ret = gst_pad_push (tag->srcpad, metadata_buffer);
            if (ret != GST_FLOW_OK)
                goto cleanup;
        } else {
            tag->vorbiscomment = metadata_buffer;
        }
        tag->metadata_block_size = 0;
        tag->state = GST_FLAC_TAG_STATE_METADATA_NEXT_BLOCK;
    }

    /* This state is mainly used to be able to stop as soon as we read
     * a vorbiscomment block from the flac file if we are in an only output
     * tags mode
     */
    if (tag->state == GST_FLAC_TAG_STATE_METADATA_NEXT_BLOCK) {
        /* Check if in the previous iteration we read a vorbis comment metadata
         * block, and stop now if the user only wants to read tags
         */
        if (tag->vorbiscomment != NULL) {
            guint8 id_data[4];
            /* We found some tags, try to parse them and notify the other elements
             * that we encountered some tags
             */
            GST_DEBUG_OBJECT (tag, "emitting vorbiscomment tags");
            gst_buffer_extract (tag->vorbiscomment, 0, id_data, 4);
            tag->tags = gst_tag_list_from_vorbiscomment_buffer (tag->vorbiscomment,
                        id_data, 4, NULL);
            if (tag->tags != NULL) {
                gst_pad_push_event (tag->srcpad,
                                    gst_event_new_tag (gst_tag_list_copy (tag->tags)));
            }

            gst_buffer_unref (tag->vorbiscomment);
            tag->vorbiscomment = NULL;
        }

        /* Skip to next state */
        if (tag->metadata_last_block == FALSE) {
            tag->state = GST_FLAC_TAG_STATE_METADATA_BLOCKS;
        } else {
            tag->state = GST_FLAC_TAG_STATE_ADD_VORBIS_COMMENT;
        }
    }


    /* Creates a vorbis comment block from the metadata which was set
     * on the gstreamer element, and add it to the flac stream
     */
    if (tag->state == GST_FLAC_TAG_STATE_ADD_VORBIS_COMMENT) {
        GstBuffer *buffer;
        const GstTagList *user_tags;
        GstTagList *merged_tags;

        /* merge the tag lists */
        user_tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (tag));
        if (user_tags != NULL) {
            merged_tags = gst_tag_list_merge (user_tags, tag->tags,
                                              gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (tag)));
        } else {
            merged_tags = gst_tag_list_copy (tag->tags);
        }

        if (merged_tags == NULL) {
            /* If we get a NULL list of tags, we must generate a padding block
             * which is marked as the last metadata block, otherwise we'll
             * end up with a corrupted flac file.
             */
            GST_WARNING_OBJECT (tag, "No tags found");
            buffer = gst_buffer_new_and_alloc (12);
            if (buffer == NULL)
                goto no_buffer;

            gst_buffer_map (buffer, &map, GST_MAP_WRITE);
            memset (map.data, 0, map.size);
            map.data[0] = 0x81;       /* 0x80 = Last metadata block,
                                 * 0x01 = padding block */
            gst_buffer_unmap (buffer, &map);
        } else {
            guchar header[4];
            guint8 fbit[1];

            memset (header, 0, sizeof (header));
            header[0] = 0x84;         /* 0x80 = Last metadata block,
                                 * 0x04 = vorbiscomment block */
            buffer = gst_tag_list_to_vorbiscomment_buffer (merged_tags, header,
                     sizeof (header), NULL);
            GST_DEBUG_OBJECT (tag, "Writing tags %" GST_PTR_FORMAT, merged_tags);
            gst_tag_list_free (merged_tags);
            if (buffer == NULL)
                goto no_comment;

            size = gst_buffer_get_size (buffer);
            if ((size < 4) || ((size - 4) > 0xFFFFFF))
                goto comment_too_long;

            fbit[0] = 1;
            /* Get rid of the framing bit at the end of the vorbiscomment buffer
             * if it exists since libFLAC seems to lose sync because of this
             * bit in gstflacdec
             */
            if (gst_buffer_memcmp (buffer, size - 1, fbit, 1) == 0) {
                buffer = gst_buffer_make_writable (buffer);
                gst_buffer_resize (buffer, 0, size - 1);
            }
        }

        /* The 4 byte metadata block header isn't accounted for in the total
         * size of the metadata block
         */
        gst_buffer_map (buffer, &map, GST_MAP_WRITE);
        map.data[1] = (((map.size - 4) & 0xFF0000) >> 16);
        map.data[2] = (((map.size - 4) & 0x00FF00) >> 8);
        map.data[3] = ((map.size - 4) & 0x0000FF);
        gst_buffer_unmap (buffer, &map);

        GST_DEBUG_OBJECT (tag, "pushing %" G_GSIZE_FORMAT " byte vorbiscomment "
                          "buffer", map.size);

        ret = gst_pad_push (tag->srcpad, buffer);
        if (ret != GST_FLOW_OK) {
            goto cleanup;
        }
        tag->state = GST_FLAC_TAG_STATE_AUDIO_DATA;
    }
static gboolean
gst_type_find_element_seek (GstTypeFindElement * typefind, GstEvent * event)
{
  GstSeekFlags flags;
  GstSeekType start_type, stop_type;
  GstFormat format;
  gboolean flush;
  gdouble rate;
  gint64 start, stop;
  GstSegment seeksegment = { 0, };

  gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start,
      &stop_type, &stop);

  /* we can only seek on bytes */
  if (format != GST_FORMAT_BYTES) {
    GST_DEBUG_OBJECT (typefind, "Can only seek on BYTES");
    return FALSE;
  }

  /* copy segment, we need this because we still need the old
   * segment when we close the current segment. */
  memcpy (&seeksegment, &typefind->segment, sizeof (GstSegment));

  GST_DEBUG_OBJECT (typefind, "configuring seek");
  gst_segment_do_seek (&seeksegment, rate, format, flags,
      start_type, start, stop_type, stop, NULL);

  flush = ! !(flags & GST_SEEK_FLAG_FLUSH);

  GST_DEBUG_OBJECT (typefind, "New segment %" GST_SEGMENT_FORMAT, &seeksegment);

  if (flush) {
    GST_DEBUG_OBJECT (typefind, "Starting flush");
    gst_pad_push_event (typefind->sink, gst_event_new_flush_start ());
    gst_pad_push_event (typefind->src, gst_event_new_flush_start ());
  } else {
    GST_DEBUG_OBJECT (typefind, "Non-flushing seek, pausing task");
    gst_pad_pause_task (typefind->sink);
  }

  /* now grab the stream lock so that streaming cannot continue, for
   * non flushing seeks when the element is in PAUSED this could block
   * forever. */
  GST_DEBUG_OBJECT (typefind, "Waiting for streaming to stop");
  GST_PAD_STREAM_LOCK (typefind->sink);

  if (flush) {
    GST_DEBUG_OBJECT (typefind, "Stopping flush");
    gst_pad_push_event (typefind->sink, gst_event_new_flush_stop (TRUE));
    gst_pad_push_event (typefind->src, gst_event_new_flush_stop (TRUE));
  }

  /* now update the real segment info */
  GST_DEBUG_OBJECT (typefind, "Committing new seek segment");
  memcpy (&typefind->segment, &seeksegment, sizeof (GstSegment));
  typefind->offset = typefind->segment.start;

  /* notify start of new segment */
  if (typefind->segment.flags & GST_SEGMENT_FLAG_SEGMENT) {
    GstMessage *msg;

    msg = gst_message_new_segment_start (GST_OBJECT (typefind),
        GST_FORMAT_BYTES, typefind->segment.start);
    gst_element_post_message (GST_ELEMENT (typefind), msg);
  }

  typefind->need_segment = TRUE;

  /* restart our task since it might have been stopped when we did the
   * flush. */
  gst_pad_start_task (typefind->sink,
      (GstTaskFunction) gst_type_find_element_loop, typefind->sink, NULL);

  /* streaming can continue now */
  GST_PAD_STREAM_UNLOCK (typefind->sink);

  return TRUE;
}
/* sinkpad functions */
static gboolean
gst_stream_synchronizer_sink_event (GstPad * pad, GstObject * parent,
    GstEvent * event)
{
  GstStreamSynchronizer *self = GST_STREAM_SYNCHRONIZER (parent);
  gboolean ret = FALSE;

  GST_LOG_OBJECT (pad, "Handling event %s: %" GST_PTR_FORMAT,
      GST_EVENT_TYPE_NAME (event), event);

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_STREAM_START:
    {
      GstSyncStream *stream, *ostream;
      guint32 seqnum = gst_event_get_seqnum (event);
      guint group_id;
      gboolean have_group_id;
      GList *l;
      gboolean all_wait = TRUE;
      gboolean new_stream = TRUE;

      have_group_id = gst_event_parse_group_id (event, &group_id);

      GST_STREAM_SYNCHRONIZER_LOCK (self);
      self->have_group_id &= have_group_id;
      have_group_id = self->have_group_id;

      stream = gst_pad_get_element_private (pad);

      if (!stream) {
        GST_DEBUG_OBJECT (self, "No stream or STREAM_START from same source");
        GST_STREAM_SYNCHRONIZER_UNLOCK (self);
        break;
      }

      gst_event_parse_stream_flags (event, &stream->flags);

      if ((have_group_id && stream->group_id != group_id) || (!have_group_id
              && stream->stream_start_seqnum != seqnum)) {
        stream->is_eos = FALSE;
        stream->eos_sent = FALSE;
        stream->flushing = FALSE;
        stream->stream_start_seqnum = seqnum;
        stream->group_id = group_id;

        if (!have_group_id) {
          /* Check if this belongs to a stream that is already there,
           * e.g. we got the visualizations for an audio stream */
          for (l = self->streams; l; l = l->next) {
            ostream = l->data;

            if (ostream != stream && ostream->stream_start_seqnum == seqnum
                && !ostream->wait) {
              new_stream = FALSE;
              break;
            }
          }

          if (!new_stream) {
            GST_DEBUG_OBJECT (pad,
                "Stream %d belongs to running stream %d, no waiting",
                stream->stream_number, ostream->stream_number);
            stream->wait = FALSE;

            GST_STREAM_SYNCHRONIZER_UNLOCK (self);
            break;
          }
        } else if (group_id == self->group_id) {
          GST_DEBUG_OBJECT (pad, "Stream %d belongs to running group %d, "
              "no waiting", stream->stream_number, group_id);
          GST_STREAM_SYNCHRONIZER_UNLOCK (self);
          break;
        }

        GST_DEBUG_OBJECT (pad, "Stream %d changed", stream->stream_number);

        stream->wait = TRUE;

        for (l = self->streams; l; l = l->next) {
          GstSyncStream *ostream = l->data;

          all_wait = all_wait && ((ostream->flags & GST_STREAM_FLAG_SPARSE)
              || (ostream->wait && (!have_group_id
                      || ostream->group_id == group_id)));
          if (!all_wait)
            break;
        }

        if (all_wait) {
          gint64 position = 0;

          if (have_group_id)
            GST_DEBUG_OBJECT (self,
                "All streams have changed to group id %u -- unblocking",
                group_id);
          else
            GST_DEBUG_OBJECT (self, "All streams have changed -- unblocking");

          self->group_id = group_id;

          for (l = self->streams; l; l = l->next) {
            GstSyncStream *ostream = l->data;
            gint64 stop_running_time;
            gint64 position_running_time;

            ostream->wait = FALSE;

            if (ostream->segment.format == GST_FORMAT_TIME) {
              if (ostream->segment.rate > 0)
                stop_running_time =
                    gst_segment_to_running_time (&ostream->segment,
                    GST_FORMAT_TIME, ostream->segment.stop);
              else
                stop_running_time =
                    gst_segment_to_running_time (&ostream->segment,
                    GST_FORMAT_TIME, ostream->segment.start);

              position_running_time =
                  gst_segment_to_running_time (&ostream->segment,
                  GST_FORMAT_TIME, ostream->segment.position);

              position_running_time =
                  MAX (position_running_time, stop_running_time);

              if (ostream->segment.rate > 0)
                position_running_time -=
                    gst_segment_to_running_time (&ostream->segment,
                    GST_FORMAT_TIME, ostream->segment.start);
              else
                position_running_time -=
                    gst_segment_to_running_time (&ostream->segment,
                    GST_FORMAT_TIME, ostream->segment.stop);

              position_running_time = MAX (0, position_running_time);

              position = MAX (position, position_running_time);
            }
          }

          self->group_start_time += position;

          GST_DEBUG_OBJECT (self, "New group start time: %" GST_TIME_FORMAT,
              GST_TIME_ARGS (self->group_start_time));

          for (l = self->streams; l; l = l->next) {
            GstSyncStream *ostream = l->data;
            ostream->wait = FALSE;
            g_cond_broadcast (&ostream->stream_finish_cond);
          }
        }
      }

      GST_STREAM_SYNCHRONIZER_UNLOCK (self);
      break;
    }
    case GST_EVENT_SEGMENT:{
      GstSyncStream *stream;
      GstSegment segment;

      gst_event_copy_segment (event, &segment);

      GST_STREAM_SYNCHRONIZER_LOCK (self);

      gst_stream_synchronizer_wait (self, pad);

      if (self->shutdown) {
        GST_STREAM_SYNCHRONIZER_UNLOCK (self);
        gst_event_unref (event);
        goto done;
      }

      stream = gst_pad_get_element_private (pad);
      if (stream && segment.format == GST_FORMAT_TIME) {
        GST_DEBUG_OBJECT (pad,
            "New stream, updating base from %" GST_TIME_FORMAT " to %"
            GST_TIME_FORMAT, GST_TIME_ARGS (segment.base),
            GST_TIME_ARGS (segment.base + self->group_start_time));
        segment.base += self->group_start_time;

        GST_DEBUG_OBJECT (pad, "Segment was: %" GST_SEGMENT_FORMAT,
            &stream->segment);
        gst_segment_copy_into (&segment, &stream->segment);
        GST_DEBUG_OBJECT (pad, "Segment now is: %" GST_SEGMENT_FORMAT,
            &stream->segment);
        stream->segment_seqnum = gst_event_get_seqnum (event);

        GST_DEBUG_OBJECT (pad, "Stream start running time: %" GST_TIME_FORMAT,
            GST_TIME_ARGS (stream->segment.base));
        {
          GstEvent *tmpev;

          tmpev = gst_event_new_segment (&stream->segment);
          gst_event_set_seqnum (tmpev, stream->segment_seqnum);
          gst_event_unref (event);
          event = tmpev;
        }
      } else if (stream) {
        GST_WARNING_OBJECT (pad, "Non-TIME segment: %s",
            gst_format_get_name (segment.format));
        gst_segment_init (&stream->segment, GST_FORMAT_UNDEFINED);
      }
      GST_STREAM_SYNCHRONIZER_UNLOCK (self);
      break;
    }
    case GST_EVENT_FLUSH_START:{
      GstSyncStream *stream;

      GST_STREAM_SYNCHRONIZER_LOCK (self);
      stream = gst_pad_get_element_private (pad);
      self->eos = FALSE;
      if (stream) {
        GST_DEBUG_OBJECT (pad, "Flushing streams");
        stream->flushing = TRUE;
        g_cond_broadcast (&stream->stream_finish_cond);
      }
      GST_STREAM_SYNCHRONIZER_UNLOCK (self);
      break;
    }
    case GST_EVENT_FLUSH_STOP:{
      GstSyncStream *stream;
      GList *l;
      GstClockTime new_group_start_time = 0;

      GST_STREAM_SYNCHRONIZER_LOCK (self);
      stream = gst_pad_get_element_private (pad);
      if (stream) {
        GST_DEBUG_OBJECT (pad, "Resetting segment for stream %d",
            stream->stream_number);
        gst_segment_init (&stream->segment, GST_FORMAT_UNDEFINED);

        stream->is_eos = FALSE;
        stream->eos_sent = FALSE;
        stream->flushing = FALSE;
        stream->wait = FALSE;
        g_cond_broadcast (&stream->stream_finish_cond);
      }

      for (l = self->streams; l; l = l->next) {
        GstSyncStream *ostream = l->data;
        GstClockTime start_running_time;

        if (ostream == stream || ostream->flushing)
          continue;

        if (ostream->segment.format == GST_FORMAT_TIME) {
          if (ostream->segment.rate > 0)
            start_running_time =
                gst_segment_to_running_time (&ostream->segment,
                GST_FORMAT_TIME, ostream->segment.start);
          else
            start_running_time =
                gst_segment_to_running_time (&ostream->segment,
                GST_FORMAT_TIME, ostream->segment.stop);

          new_group_start_time = MAX (new_group_start_time, start_running_time);
        }
      }

      GST_DEBUG_OBJECT (pad,
          "Updating group start time from %" GST_TIME_FORMAT " to %"
          GST_TIME_FORMAT, GST_TIME_ARGS (self->group_start_time),
          GST_TIME_ARGS (new_group_start_time));
      self->group_start_time = new_group_start_time;
      GST_STREAM_SYNCHRONIZER_UNLOCK (self);
      break;
    }
      /* unblocking EOS wait when track switch. */
    case GST_EVENT_CUSTOM_DOWNSTREAM_OOB:{
      if (gst_event_has_name (event, "playsink-custom-video-flush")
          || gst_event_has_name (event, "playsink-custom-audio-flush")
          || gst_event_has_name (event, "playsink-custom-subtitle-flush")) {
        GstSyncStream *stream;

        GST_STREAM_SYNCHRONIZER_LOCK (self);
        stream = gst_pad_get_element_private (pad);
        if (stream) {
          stream->is_eos = FALSE;
          stream->eos_sent = FALSE;
          stream->wait = FALSE;
          g_cond_broadcast (&stream->stream_finish_cond);
        }
        GST_STREAM_SYNCHRONIZER_UNLOCK (self);
      }
      break;
    }
    case GST_EVENT_EOS:{
      GstSyncStream *stream;
      GList *l;
      gboolean all_eos = TRUE;
      gboolean seen_data;
      GSList *pads = NULL;
      GstPad *srcpad;
      GstClockTime timestamp;

      GST_STREAM_SYNCHRONIZER_LOCK (self);
      stream = gst_pad_get_element_private (pad);
      if (!stream) {
        GST_STREAM_SYNCHRONIZER_UNLOCK (self);
        GST_WARNING_OBJECT (pad, "EOS for unknown stream");
        break;
      }

      GST_DEBUG_OBJECT (pad, "Have EOS for stream %d", stream->stream_number);
      stream->is_eos = TRUE;

      seen_data = stream->seen_data;
      srcpad = gst_object_ref (stream->srcpad);

      if (seen_data && stream->segment.position != -1)
        timestamp = stream->segment.position;
      else if (stream->segment.rate < 0.0 || stream->segment.stop == -1)
        timestamp = stream->segment.start;
      else
        timestamp = stream->segment.stop;

      stream->segment.position = timestamp;

      for (l = self->streams; l; l = l->next) {
        GstSyncStream *ostream = l->data;

        all_eos = all_eos && ostream->is_eos;
        if (!all_eos)
          break;
      }

      if (all_eos) {
        GST_DEBUG_OBJECT (self, "All streams are EOS -- forwarding");
        self->eos = TRUE;
        for (l = self->streams; l; l = l->next) {
          GstSyncStream *ostream = l->data;
          /* local snapshot of current pads */
          gst_object_ref (ostream->srcpad);
          pads = g_slist_prepend (pads, ostream->srcpad);
        }
      }
      if (pads) {
        GstPad *pad;
        GSList *epad;
        GstSyncStream *ostream;

        ret = TRUE;
        epad = pads;
        while (epad) {
          pad = epad->data;
          ostream = gst_pad_get_element_private (pad);
          if (ostream) {
            g_cond_broadcast (&ostream->stream_finish_cond);
          }

          gst_object_unref (pad);
          epad = g_slist_next (epad);
        }
        g_slist_free (pads);
      } else {
        if (seen_data) {
          self->send_gap_event = TRUE;
          stream->gap_duration = GST_CLOCK_TIME_NONE;
          stream->wait = TRUE;
          ret = gst_stream_synchronizer_wait (self, srcpad);
        }
      }

      /* send eos if haven't seen data. seen_data will be true if data buffer
       * of the track have received in anytime. sink is ready if seen_data is
       * true, so can send GAP event. Will send EOS if sink isn't ready. The
       * scenario for the case is one track haven't any media data and then
       * send EOS. Or no any valid media data in one track, so decoder can't
       * get valid CAPS for the track. sink can't ready without received CAPS.*/
      if (!seen_data || self->eos) {
        GST_DEBUG_OBJECT (pad, "send EOS event");
        /* drop lock when sending eos, which may block in e.g. preroll */
        GST_STREAM_SYNCHRONIZER_UNLOCK (self);
        ret = gst_pad_push_event (srcpad, gst_event_new_eos ());
        GST_STREAM_SYNCHRONIZER_LOCK (self);
        stream = gst_pad_get_element_private (pad);
        if (stream) {
          stream->eos_sent = TRUE;
        }
      }

      gst_object_unref (srcpad);
      gst_event_unref (event);
      GST_STREAM_SYNCHRONIZER_UNLOCK (self);
      goto done;
    }
    default:
      break;
  }

  ret = gst_pad_event_default (pad, parent, event);

done:

  return ret;
}
Beispiel #9
0
static GstFlowReturn
gst_gdp_depay_chain (GstPad * pad, GstBuffer * buffer)
{
  GstGDPDepay *this;
  GstFlowReturn ret = GST_FLOW_OK;
  GstCaps *caps;
  GstBuffer *buf;
  GstEvent *event;
  guint available;

  this = GST_GDP_DEPAY (gst_pad_get_parent (pad));

  /* On DISCONT, get rid of accumulated data. We assume a buffer after the
   * DISCONT contains (part of) a new valid header, if not we error because we
   * lost sync */
  if (GST_BUFFER_IS_DISCONT (buffer)) {
    gst_adapter_clear (this->adapter);
    this->state = GST_GDP_DEPAY_STATE_HEADER;
  }
  gst_adapter_push (this->adapter, buffer);

  while (TRUE) {
    switch (this->state) {
      case GST_GDP_DEPAY_STATE_HEADER:
      {
        guint8 *header;

        /* collect a complete header, validate and store the header. Figure out
         * the payload length and switch to the PAYLOAD state */
        available = gst_adapter_available (this->adapter);
        if (available < GST_DP_HEADER_LENGTH)
          goto done;

        GST_LOG_OBJECT (this, "reading GDP header from adapter");
        header = gst_adapter_take (this->adapter, GST_DP_HEADER_LENGTH);
        if (!gst_dp_validate_header (GST_DP_HEADER_LENGTH, header)) {
          g_free (header);
          goto header_validate_error;
        }

        /* store types and payload length. Also store the header, which we need
         * to make the payload. */
        this->payload_length = gst_dp_header_payload_length (header);
        this->payload_type = gst_dp_header_payload_type (header);
        /* free previous header and store new one. */
        g_free (this->header);
        this->header = header;

        GST_LOG_OBJECT (this,
            "read GDP header, payload size %d, payload type %d, switching to state PAYLOAD",
            this->payload_length, this->payload_type);
        this->state = GST_GDP_DEPAY_STATE_PAYLOAD;
        break;
      }
      case GST_GDP_DEPAY_STATE_PAYLOAD:
      {
        /* in this state we wait for all the payload data to be available in the
         * adapter. Then we switch to the state where we actually process the
         * payload. */
        available = gst_adapter_available (this->adapter);
        if (available < this->payload_length)
          goto done;

        /* change state based on type */
        if (this->payload_type == GST_DP_PAYLOAD_BUFFER) {
          GST_LOG_OBJECT (this, "switching to state BUFFER");
          this->state = GST_GDP_DEPAY_STATE_BUFFER;
        } else if (this->payload_type == GST_DP_PAYLOAD_CAPS) {
          GST_LOG_OBJECT (this, "switching to state CAPS");
          this->state = GST_GDP_DEPAY_STATE_CAPS;
        } else if (this->payload_type >= GST_DP_PAYLOAD_EVENT_NONE) {
          GST_LOG_OBJECT (this, "switching to state EVENT");
          this->state = GST_GDP_DEPAY_STATE_EVENT;
        } else {
          goto wrong_type;
        }

        if (this->payload_length
            && (!gst_dp_validate_payload (GST_DP_HEADER_LENGTH, this->header,
                    gst_adapter_peek (this->adapter, this->payload_length)))) {
          goto payload_validate_error;
        }

        break;
      }
      case GST_GDP_DEPAY_STATE_BUFFER:
      {

        /* if we receive a buffer without caps first, we error out */
        if (!this->caps)
          goto no_caps;

        GST_LOG_OBJECT (this, "reading GDP buffer from adapter");
        buf = gst_dp_buffer_from_header (GST_DP_HEADER_LENGTH, this->header);
        if (!buf)
          goto buffer_failed;

        /* now take the payload if there is any */
        if (this->payload_length > 0) {
          guint8 *payload;

          payload = gst_adapter_take (this->adapter, this->payload_length);
          memcpy (GST_BUFFER_DATA (buf), payload, this->payload_length);
          g_free (payload);
        }

        /* set caps and push */
        gst_buffer_set_caps (buf, this->caps);
        GST_LOG_OBJECT (this, "deserialized buffer %p, pushing, timestamp %"
            GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT
            ", offset %" G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT
            ", size %d, flags 0x%x",
            buf,
            GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
            GST_TIME_ARGS (GST_BUFFER_DURATION (buf)),
            GST_BUFFER_OFFSET (buf), GST_BUFFER_OFFSET_END (buf),
            GST_BUFFER_SIZE (buf), GST_BUFFER_FLAGS (buf));
        ret = gst_pad_push (this->srcpad, buf);
        if (ret != GST_FLOW_OK)
          goto push_error;

        GST_LOG_OBJECT (this, "switching to state HEADER");
        this->state = GST_GDP_DEPAY_STATE_HEADER;
        break;
      }
      case GST_GDP_DEPAY_STATE_CAPS:
      {
        guint8 *payload;

        /* take the payload of the caps */
        GST_LOG_OBJECT (this, "reading GDP caps from adapter");
        payload = gst_adapter_take (this->adapter, this->payload_length);
        caps = gst_dp_caps_from_packet (GST_DP_HEADER_LENGTH, this->header,
            payload);
        g_free (payload);
        if (!caps)
          goto caps_failed;

        GST_DEBUG_OBJECT (this, "deserialized caps %" GST_PTR_FORMAT, caps);
        gst_caps_replace (&(this->caps), caps);
        gst_pad_set_caps (this->srcpad, caps);
        /* drop the creation ref we still have */
        gst_caps_unref (caps);

        GST_LOG_OBJECT (this, "switching to state HEADER");
        this->state = GST_GDP_DEPAY_STATE_HEADER;
        break;
      }
      case GST_GDP_DEPAY_STATE_EVENT:
      {
        guint8 *payload;

        GST_LOG_OBJECT (this, "reading GDP event from adapter");

        /* adapter doesn't like 0 length payload */
        if (this->payload_length > 0)
          payload = gst_adapter_take (this->adapter, this->payload_length);
        else
          payload = NULL;
        event = gst_dp_event_from_packet (GST_DP_HEADER_LENGTH, this->header,
            payload);
        g_free (payload);
        if (!event)
          goto event_failed;

        GST_DEBUG_OBJECT (this, "deserialized event %p of type %s, pushing",
            event, gst_event_type_get_name (event->type));
        gst_pad_push_event (this->srcpad, event);

        GST_LOG_OBJECT (this, "switching to state HEADER");
        this->state = GST_GDP_DEPAY_STATE_HEADER;
        break;
      }
    }
  }

done:
  gst_object_unref (this);
  return ret;

  /* ERRORS */
header_validate_error:
  {
    GST_ELEMENT_ERROR (this, STREAM, DECODE, (NULL),
        ("GDP packet header does not validate"));
    ret = GST_FLOW_ERROR;
    goto done;
  }
payload_validate_error:
  {
    GST_ELEMENT_ERROR (this, STREAM, DECODE, (NULL),
        ("GDP packet payload does not validate"));
    ret = GST_FLOW_ERROR;
    goto done;
  }
wrong_type:
  {
    GST_ELEMENT_ERROR (this, STREAM, DECODE, (NULL),
        ("GDP packet header is of wrong type"));
    ret = GST_FLOW_ERROR;
    goto done;
  }
no_caps:
  {
    GST_ELEMENT_ERROR (this, STREAM, DECODE, (NULL),
        ("Received a buffer without first receiving caps"));
    ret = GST_FLOW_NOT_NEGOTIATED;
    goto done;
  }
buffer_failed:
  {
    GST_ELEMENT_ERROR (this, STREAM, DECODE, (NULL),
        ("could not create buffer from GDP packet"));
    ret = GST_FLOW_ERROR;
    goto done;
  }
push_error:
  {
    GST_WARNING_OBJECT (this, "pushing depayloaded buffer returned %d", ret);
    goto done;
  }
caps_failed:
  {
    GST_ELEMENT_ERROR (this, STREAM, DECODE, (NULL),
        ("could not create caps from GDP packet"));
    ret = GST_FLOW_ERROR;
    goto done;
  }
event_failed:
  {
    GST_ELEMENT_ERROR (this, STREAM, DECODE, (NULL),
        ("could not create event from GDP packet"));
    ret = GST_FLOW_ERROR;
    goto done;
  }
}
static gboolean
gst_dvbsub_overlay_event_text (GstPad * pad, GstObject * parent,
    GstEvent * event)
{
  gboolean ret = FALSE;
  GstDVBSubOverlay *render = GST_DVBSUB_OVERLAY (parent);

  GST_DEBUG_OBJECT (pad, "received text event %s", GST_EVENT_TYPE_NAME (event));

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_SEGMENT:
    {
      GstSegment seg;

      GST_DEBUG_OBJECT (render, "received new segment");

      gst_event_copy_segment (event, &seg);

      if (seg.format == GST_FORMAT_TIME) {
        GST_DEBUG_OBJECT (render, "SUBTITLE SEGMENT now: %" GST_SEGMENT_FORMAT,
            &render->subtitle_segment);

        render->subtitle_segment = seg;

        GST_DEBUG_OBJECT (render,
            "SUBTITLE SEGMENT after: %" GST_SEGMENT_FORMAT,
            &render->subtitle_segment);
        ret = TRUE;
        gst_event_unref (event);
      } else {
        GST_ELEMENT_WARNING (render, STREAM, MUX, (NULL),
            ("received non-TIME newsegment event on subtitle sinkpad"));
        ret = FALSE;
        gst_event_unref (event);
      }
      break;
    }
    case GST_EVENT_FLUSH_STOP:
      GST_DEBUG_OBJECT (render, "stop flushing");
      gst_dvbsub_overlay_flush_subtitles (render);
      gst_segment_init (&render->subtitle_segment, GST_FORMAT_TIME);
      gst_event_unref (event);
      ret = TRUE;
      break;
    case GST_EVENT_FLUSH_START:
      GST_DEBUG_OBJECT (render, "begin flushing");
      gst_event_unref (event);
      ret = TRUE;
      break;
    case GST_EVENT_EOS:
      GST_INFO_OBJECT (render, "text EOS");
      gst_event_unref (event);
      ret = TRUE;
      break;
    case GST_EVENT_GAP:
      gst_event_unref (event);
      ret = TRUE;
      break;
    case GST_EVENT_CAPS:
      /* don't want to forward the subtitle caps */
      gst_event_unref (event);
      ret = TRUE;
      break;
    default:
      ret = gst_pad_push_event (render->srcpad, event);
      break;
  }

  return ret;
}
Beispiel #11
0
static gboolean
pad_event (GstPad *pad,
           GstEvent *event)
{
    GstOmxBaseFilter *self;
    GOmxCore *gomx;
    GOmxPort *in_port;
    GOmxPort *out_port;
    gboolean ret;

    self = GST_OMX_BASE_FILTER (GST_OBJECT_PARENT (pad));
    gomx = self->gomx;
    in_port = self->in_port;
    out_port = self->out_port;

    GST_LOG_OBJECT (self, "begin");

    GST_INFO_OBJECT (self, "event: %s", GST_EVENT_TYPE_NAME (event));

    switch (GST_EVENT_TYPE (event))
    {
        case GST_EVENT_EOS:
            if (self->initialized)
            {
                /* send buffer with eos flag */
                /** @todo move to util */
                {
                    OMX_BUFFERHEADERTYPE *omx_buffer;

                    GST_LOG_OBJECT (self, "request buffer");
                    omx_buffer = g_omx_port_request_buffer (in_port);

                    if (G_LIKELY (omx_buffer))
                    {
                        omx_buffer->nFlags |= OMX_BUFFERFLAG_EOS;

                        GST_LOG_OBJECT (self, "release_buffer");
                        /* foo_buffer_untaint (omx_buffer); */
                        g_omx_port_release_buffer (in_port, omx_buffer);
                    }
                    else
                    {
                        g_omx_core_set_done (gomx);
                    }
                }

                /* Wait for the output port to get the EOS. */
                g_omx_core_wait_for_done (gomx);
            }

            ret = gst_pad_push_event (self->srcpad, event);
            break;

        case GST_EVENT_FLUSH_START:
            gst_pad_push_event (self->srcpad, event);
            self->last_pad_push_return = GST_FLOW_WRONG_STATE;

            g_omx_core_flush_start (gomx);

            gst_pad_pause_task (self->srcpad);

            ret = TRUE;
            break;

        case GST_EVENT_FLUSH_STOP:
            gst_pad_push_event (self->srcpad, event);
            self->last_pad_push_return = GST_FLOW_OK;

            g_omx_core_flush_stop (gomx);

            gst_pad_start_task (self->srcpad, output_loop, self->srcpad);

            ret = TRUE;
            break;

        case GST_EVENT_NEWSEGMENT:
            ret = gst_pad_push_event (self->srcpad, event);
            break;

        default:
            ret = gst_pad_push_event (self->srcpad, event);
            break;
    }

    GST_LOG_OBJECT (self, "end");

    return ret;
}
Beispiel #12
0
static void
test_basic (const gchar * elem_name, const gchar * sink2, int count,
    check_cb cb)
{
  GstElement *rtpmux = NULL;
  GstPad *reqpad1 = NULL;
  GstPad *reqpad2 = NULL;
  GstPad *src1 = NULL;
  GstPad *src2 = NULL;
  GstPad *sink = NULL;
  GstBuffer *inbuf = NULL;
  GstCaps *src1caps = NULL;
  GstCaps *src2caps = NULL;
  GstCaps *sinkcaps = NULL;
  GstCaps *caps;
  GstSegment segment;
  int i;

  rtpmux = gst_check_setup_element (elem_name);

  reqpad1 = gst_element_get_request_pad (rtpmux, "sink_1");
  fail_unless (reqpad1 != NULL);
  reqpad2 = gst_element_get_request_pad (rtpmux, sink2);
  fail_unless (reqpad2 != NULL);
  sink = gst_check_setup_sink_pad_by_name (rtpmux, &sinktemplate, "src");

  src1 = gst_pad_new_from_static_template (&srctemplate, "src");
  src2 = gst_pad_new_from_static_template (&srctemplate, "src");
  fail_unless (gst_pad_link (src1, reqpad1) == GST_PAD_LINK_OK);
  fail_unless (gst_pad_link (src2, reqpad2) == GST_PAD_LINK_OK);
  gst_pad_set_query_function (src1, query_func);
  gst_pad_set_query_function (src2, query_func);
  gst_pad_set_query_function (sink, query_func);
  gst_pad_set_event_function (sink, event_func);
  g_object_set_data (G_OBJECT (src1), "caps", &src1caps);
  g_object_set_data (G_OBJECT (src2), "caps", &src2caps);
  g_object_set_data (G_OBJECT (sink), "caps", &sinkcaps);

  src1caps = gst_caps_new_simple ("application/x-rtp",
      "clock-rate", G_TYPE_INT, 1, "ssrc", G_TYPE_UINT, 11, NULL);
  src2caps = gst_caps_new_simple ("application/x-rtp",
      "clock-rate", G_TYPE_INT, 2, "ssrc", G_TYPE_UINT, 12, NULL);
  sinkcaps = gst_caps_new_simple ("application/x-rtp",
      "clock-rate", G_TYPE_INT, 3, "ssrc", G_TYPE_UINT, 13, NULL);

  caps = gst_pad_peer_query_caps (src1, NULL);
  fail_unless (gst_caps_is_empty (caps));
  gst_caps_unref (caps);

  gst_caps_set_simple (src2caps, "clock-rate", G_TYPE_INT, 3, NULL);
  caps = gst_pad_peer_query_caps (src1, NULL);
  gst_caps_unref (caps);

  g_object_set (rtpmux, "seqnum-offset", 100, "timestamp-offset", 1000,
      "ssrc", 55, NULL);

  fail_unless (gst_element_set_state (rtpmux,
          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
  gst_pad_set_active (sink, TRUE);
  gst_pad_set_active (src1, TRUE);
  gst_pad_set_active (src2, TRUE);

  fail_unless (gst_pad_push_event (src1,
          gst_event_new_stream_start ("stream1")));
  fail_unless (gst_pad_push_event (src2,
          gst_event_new_stream_start ("stream2")));

  gst_caps_set_simple (sinkcaps,
      "payload", G_TYPE_INT, 98, "seqnum-offset", G_TYPE_UINT, 100,
      "timestamp-offset", G_TYPE_UINT, 1000, "ssrc", G_TYPE_UINT, 66, NULL);
  caps = gst_caps_new_simple ("application/x-rtp",
      "payload", G_TYPE_INT, 98, "clock-rate", G_TYPE_INT, 3,
      "seqnum-offset", G_TYPE_UINT, 56, "timestamp-offset", G_TYPE_UINT, 57,
      "ssrc", G_TYPE_UINT, 66, NULL);
  fail_unless (gst_pad_set_caps (src1, caps));
  gst_caps_unref (caps);

  caps = gst_pad_peer_query_caps (sink, NULL);
  fail_if (gst_caps_is_empty (caps));

  gst_segment_init (&segment, GST_FORMAT_TIME);
  segment.start = 100000;
  fail_unless (gst_pad_push_event (src1, gst_event_new_segment (&segment)));
  segment.start = 0;
  fail_unless (gst_pad_push_event (src2, gst_event_new_segment (&segment)));


  for (i = 0; i < count; i++) {
    GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;

    inbuf = gst_rtp_buffer_new_allocate (10, 0, 0);
    GST_BUFFER_PTS (inbuf) = i * 1000 + 100000;
    GST_BUFFER_DURATION (inbuf) = 1000;

    gst_rtp_buffer_map (inbuf, GST_MAP_WRITE, &rtpbuffer);

    gst_rtp_buffer_set_version (&rtpbuffer, 2);
    gst_rtp_buffer_set_payload_type (&rtpbuffer, 98);
    gst_rtp_buffer_set_ssrc (&rtpbuffer, 44);
    gst_rtp_buffer_set_timestamp (&rtpbuffer, 200 + i);
    gst_rtp_buffer_set_seq (&rtpbuffer, 2000 + i);
    gst_rtp_buffer_unmap (&rtpbuffer);
    fail_unless (gst_pad_push (src1, inbuf) == GST_FLOW_OK);

    if (buffers)
      fail_unless (GST_BUFFER_PTS (buffers->data) == i * 1000, "%lld",
          GST_BUFFER_PTS (buffers->data));

    cb (src2, i);

    g_list_foreach (buffers, (GFunc) gst_buffer_unref, NULL);
    g_list_free (buffers);
    buffers = NULL;
  }


  gst_pad_set_active (sink, FALSE);
  gst_pad_set_active (src1, FALSE);
  gst_pad_set_active (src2, FALSE);
  fail_unless (gst_element_set_state (rtpmux,
          GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS);
  gst_check_teardown_pad_by_name (rtpmux, "src");
  gst_object_unref (reqpad1);
  gst_object_unref (reqpad2);
  gst_check_teardown_pad_by_name (rtpmux, "sink_1");
  gst_check_teardown_pad_by_name (rtpmux, sink2);
  gst_element_release_request_pad (rtpmux, reqpad1);
  gst_element_release_request_pad (rtpmux, reqpad2);

  gst_caps_unref (caps);
  gst_caps_replace (&src1caps, NULL);
  gst_caps_replace (&src2caps, NULL);
  gst_caps_replace (&sinkcaps, NULL);

  gst_check_teardown_element (rtpmux);
}
Beispiel #13
0
static void
gst_timidity_loop (GstPad * sinkpad)
{
  GstTimidity *timidity = GST_TIMIDITY (GST_PAD_PARENT (sinkpad));
  GstBuffer *out;
  GstFlowReturn ret;

  if (timidity->mididata_size == 0) {
    if (!gst_timidity_get_upstream_size (timidity, &timidity->mididata_size)) {
      GST_ELEMENT_ERROR (timidity, STREAM, DECODE, (NULL),
          ("Unable to get song length"));
      goto paused;
    }

    if (timidity->mididata)
      g_free (timidity->mididata);

    timidity->mididata = g_malloc (timidity->mididata_size);
    timidity->mididata_offset = 0;
    return;
  }

  if (timidity->mididata_offset < timidity->mididata_size) {
    GstBuffer *buffer;
    gint64 size;

    GST_DEBUG_OBJECT (timidity, "loading song");

    ret =
        gst_pad_pull_range (timidity->sinkpad, timidity->mididata_offset,
        -1, &buffer);
    if (ret != GST_FLOW_OK) {
      GST_ELEMENT_ERROR (timidity, STREAM, DECODE, (NULL),
          ("Unable to load song"));
      goto paused;
    }

    size = timidity->mididata_size - timidity->mididata_offset;
    if (GST_BUFFER_SIZE (buffer) < size)
      size = GST_BUFFER_SIZE (buffer);

    memmove (timidity->mididata + timidity->mididata_offset,
        GST_BUFFER_DATA (buffer), size);
    gst_buffer_unref (buffer);

    timidity->mididata_offset += size;
    GST_DEBUG_OBJECT (timidity, "Song loaded");
    return;
  }

  if (!timidity->song) {
    MidIStream *stream;
    GstTagList *tags = NULL;
    gchar *text;

    GST_DEBUG_OBJECT (timidity, "Parsing song");

    stream =
        mid_istream_open_mem (timidity->mididata, timidity->mididata_size, 0);

    timidity->song = mid_song_load (stream, timidity->song_options);
    mid_istream_close (stream);

    if (!timidity->song) {
      GST_ELEMENT_ERROR (timidity, STREAM, DECODE, (NULL),
          ("Unable to parse midi"));
      goto paused;
    }

    mid_song_start (timidity->song);
    timidity->o_len = (GST_MSECOND *
        (GstClockTime) mid_song_get_total_time (timidity->song)) /
        timidity->time_per_frame;
    gst_segment_set_newsegment (timidity->o_segment, FALSE, 1.0,
        GST_FORMAT_DEFAULT, 0, GST_CLOCK_TIME_NONE, 0);


    gst_pad_push_event (timidity->srcpad,
        gst_timidity_get_new_segment_event (timidity, GST_FORMAT_TIME, FALSE));

    /* extract tags */
    text = mid_song_get_meta (timidity->song, MID_SONG_TEXT);
    if (text) {
      tags = gst_tag_list_new ();
      gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, text, NULL);

      //g_free (text);
    }

    text = mid_song_get_meta (timidity->song, MID_SONG_COPYRIGHT);
    if (text) {
      if (tags == NULL)
        tags = gst_tag_list_new ();
      gst_tag_list_add (tags, GST_TAG_MERGE_APPEND,
          GST_TAG_COPYRIGHT, text, NULL);

      //g_free (text);
    }

    if (tags) {
      gst_element_found_tags (GST_ELEMENT (timidity), tags);
    }

    GST_DEBUG_OBJECT (timidity, "Parsing song done");
    return;
  }

  if (timidity->o_segment_changed) {
    GstSegment *segment = gst_timidity_get_segment (timidity, GST_FORMAT_TIME,
        !timidity->o_new_segment);

    GST_LOG_OBJECT (timidity,
        "sending newsegment from %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT
        ", pos=%" GST_TIME_FORMAT, GST_TIME_ARGS ((guint64) segment->start),
        GST_TIME_ARGS ((guint64) segment->stop),
        GST_TIME_ARGS ((guint64) segment->time));

    if (timidity->o_segment->flags & GST_SEEK_FLAG_SEGMENT) {
      gst_element_post_message (GST_ELEMENT (timidity),
          gst_message_new_segment_start (GST_OBJECT (timidity),
              segment->format, segment->start));
    }

    gst_segment_free (segment);
    timidity->o_segment_changed = FALSE;
    return;
  }

  if (timidity->o_seek) {
    /* perform a seek internally */
    timidity->o_segment->last_stop = timidity->o_segment->time;
    mid_song_seek (timidity->song,
        (timidity->o_segment->last_stop * timidity->time_per_frame) /
        GST_MSECOND);
  }

  out = gst_timidity_get_buffer (timidity);
  if (!out) {
    GST_LOG_OBJECT (timidity, "Song ended, generating eos");
    gst_pad_push_event (timidity->srcpad, gst_event_new_eos ());
    timidity->o_seek = FALSE;
    goto paused;
  }

  if (timidity->o_seek) {
    GST_BUFFER_FLAG_SET (out, GST_BUFFER_FLAG_DISCONT);
    timidity->o_seek = FALSE;
  }

  gst_buffer_set_caps (out, timidity->out_caps);
  ret = gst_pad_push (timidity->srcpad, out);

  if (ret == GST_FLOW_UNEXPECTED)
    goto eos;
  else if (ret < GST_FLOW_UNEXPECTED || ret == GST_FLOW_NOT_LINKED)
    goto error;

  return;

paused:
  {
    GST_DEBUG_OBJECT (timidity, "pausing task");
    gst_pad_pause_task (timidity->sinkpad);
    return;
  }
eos:
  {
    gst_pad_push_event (timidity->srcpad, gst_event_new_eos ());
    goto paused;
  }
error:
  {
    GST_ELEMENT_ERROR (timidity, STREAM, FAILED,
        ("Internal data stream error"),
        ("Streaming stopped, reason %s", gst_flow_get_name (ret)));
    gst_pad_push_event (timidity->srcpad, gst_event_new_eos ());
    goto paused;
  }
}
Beispiel #14
0
static gboolean
gst_timidity_src_event (GstPad * pad, GstEvent * event)
{
  gboolean res = FALSE;
  GstTimidity *timidity = GST_TIMIDITY (gst_pad_get_parent (pad));

  GST_DEBUG_OBJECT (pad, "%s event received", GST_EVENT_TYPE_NAME (event));

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_SEEK:
    {
      gdouble rate;
      GstFormat src_format, dst_format;
      GstSeekFlags flags;
      GstSeekType start_type, stop_type;
      gint64 orig_start, start = 0, stop = 0;
      gboolean flush, update;

      if (!timidity->song)
        break;

      gst_event_parse_seek (event, &rate, &src_format, &flags,
          &start_type, &orig_start, &stop_type, &stop);

      dst_format = GST_FORMAT_DEFAULT;

      gst_timidity_src_convert (timidity, src_format, orig_start,
          &dst_format, &start);
      gst_timidity_src_convert (timidity, src_format, stop, &dst_format, &stop);

      flush = ((flags & GST_SEEK_FLAG_FLUSH) == GST_SEEK_FLAG_FLUSH);

      if (flush) {
        GST_DEBUG ("performing flush");
        gst_pad_push_event (timidity->srcpad, gst_event_new_flush_start ());
      } else {
        gst_pad_stop_task (timidity->sinkpad);
      }

      GST_PAD_STREAM_LOCK (timidity->sinkpad);

      if (flush) {
        gst_pad_push_event (timidity->srcpad, gst_event_new_flush_stop ());
      }

      gst_segment_set_seek (timidity->o_segment, rate, dst_format, flags,
          start_type, start, stop_type, stop, &update);

      if (flags & GST_SEEK_FLAG_SEGMENT) {
        GST_DEBUG_OBJECT (timidity, "received segment seek %d, %d",
            (gint) start_type, (gint) stop_type);
      } else {
        GST_DEBUG_OBJECT (timidity, "received normal seek %d",
            (gint) start_type);
        update = FALSE;
      }

      gst_pad_push_event (timidity->srcpad,
          gst_timidity_get_new_segment_event (timidity, GST_FORMAT_TIME,
              update));

      timidity->o_seek = TRUE;

      gst_pad_start_task (timidity->sinkpad,
          (GstTaskFunction) gst_timidity_loop, timidity->sinkpad);

      GST_PAD_STREAM_UNLOCK (timidity->sinkpad);
      GST_DEBUG ("seek done");
    }
      res = TRUE;
      break;
    default:
      break;
  }

  g_object_unref (timidity);
  return res;
}
static GstFlowReturn
gst_multipart_demux_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
  GstMultipartDemux *multipart;
  GstAdapter *adapter;
  gint size = 1;
  GstFlowReturn res;

  multipart = GST_MULTIPART_DEMUX (parent);
  adapter = multipart->adapter;

  res = GST_FLOW_OK;

  if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
    GSList *l;

    for (l = multipart->srcpads; l != NULL; l = l->next) {
      GstMultipartPad *srcpad = l->data;

      srcpad->discont = TRUE;
    }
    gst_adapter_clear (adapter);
  }
  gst_adapter_push (adapter, buf);

  while (gst_adapter_available (adapter) > 0) {
    GstMultipartPad *srcpad;
    GstBuffer *outbuf;
    gboolean created;
    gint datalen;

    if (G_UNLIKELY (!multipart->header_completed)) {
      if ((size = multipart_parse_header (multipart)) < 0) {
        goto nodata;
      } else {
        gst_adapter_flush (adapter, size);
        multipart->header_completed = TRUE;
      }
    }
    if ((size = multipart_find_boundary (multipart, &datalen)) < 0) {
      goto nodata;
    }

    /* Invalidate header info */
    multipart->header_completed = FALSE;
    multipart->content_length = -1;

    if (G_UNLIKELY (datalen <= 0)) {
      GST_DEBUG_OBJECT (multipart, "skipping empty content.");
      gst_adapter_flush (adapter, size - datalen);
    } else if (G_UNLIKELY (!multipart->mime_type)) {
      GST_DEBUG_OBJECT (multipart, "content has no MIME type.");
      gst_adapter_flush (adapter, size - datalen);
    } else {
      GstClockTime ts;

      srcpad =
          gst_multipart_find_pad_by_mime (multipart,
          multipart->mime_type, &created);

      ts = gst_adapter_prev_pts (adapter, NULL);
      outbuf = gst_adapter_take_buffer (adapter, datalen);
      gst_adapter_flush (adapter, size - datalen);

      if (created) {
        GstTagList *tags;
        GstSegment segment;

        gst_segment_init (&segment, GST_FORMAT_TIME);

        /* Push new segment, first buffer has 0 timestamp */
        gst_pad_push_event (srcpad->pad, gst_event_new_segment (&segment));

        tags = gst_tag_list_new (GST_TAG_CONTAINER_FORMAT, "Multipart", NULL);
        gst_tag_list_set_scope (tags, GST_TAG_SCOPE_GLOBAL);
        gst_pad_push_event (srcpad->pad, gst_event_new_tag (tags));
      }

      outbuf = gst_buffer_make_writable (outbuf);
      if (srcpad->last_ts == GST_CLOCK_TIME_NONE || srcpad->last_ts != ts) {
        GST_BUFFER_TIMESTAMP (outbuf) = ts;
        srcpad->last_ts = ts;
      } else {
        GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE;
      }

      if (srcpad->discont) {
        GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
        srcpad->discont = FALSE;
      } else {
        GST_BUFFER_FLAG_UNSET (outbuf, GST_BUFFER_FLAG_DISCONT);
      }

      GST_DEBUG_OBJECT (multipart,
          "pushing buffer with timestamp %" GST_TIME_FORMAT,
          GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
      res = gst_pad_push (srcpad->pad, outbuf);
      res = gst_multipart_combine_flows (multipart, srcpad, res);
      if (res != GST_FLOW_OK)
        break;
    }
  }

nodata:
  if (G_UNLIKELY (size == MULTIPART_DATA_ERROR))
    return GST_FLOW_ERROR;
  if (G_UNLIKELY (size == MULTIPART_DATA_EOS))
    return GST_FLOW_EOS;

  return res;
}
Beispiel #16
0
static GstFlowReturn
gst_rsvg_dec_chain (GstPad * pad, GstBuffer * buffer)
{
  GstRsvgDec *rsvg = GST_RSVG_DEC (GST_PAD_PARENT (pad));
  gboolean completed = FALSE;
  const guint8 *data;
  guint size;
  gboolean ret = GST_FLOW_OK;

  /* first_timestamp is used slightly differently where a framerate
     is given or not.
     If there is a frame rate, it will be used as a base.
     If there is not, it will be used to keep track of the timestamp
     of the first buffer, to be used as the timestamp of the output
     buffer. When a buffer is output, first timestamp will resync to
     the next buffer's timestamp. */
  if (rsvg->first_timestamp == GST_CLOCK_TIME_NONE) {
    if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
      rsvg->first_timestamp = GST_BUFFER_TIMESTAMP (buffer);
    else if (rsvg->fps_n != 0)
      rsvg->first_timestamp = 0;
  }

  gst_adapter_push (rsvg->adapter, buffer);

  size = gst_adapter_available (rsvg->adapter);

  /* "<svg></svg>" */
  while (size >= 5 + 6 && ret == GST_FLOW_OK) {
    guint i;

    data = gst_adapter_peek (rsvg->adapter, size);
    for (i = size - 6; i >= 5; i--) {
      if (memcmp (data + i, "</svg>", 6) == 0) {
        completed = TRUE;
        size = i + 6;
        break;
      }
    }

    if (completed) {
      GstBuffer *outbuf = NULL;

      GST_LOG_OBJECT (rsvg, "have complete svg of %u bytes", size);

      data = gst_adapter_peek (rsvg->adapter, size);

      ret = gst_rsvg_decode_image (rsvg, data, size, &outbuf);
      if (ret != GST_FLOW_OK)
        break;


      if (rsvg->first_timestamp != GST_CLOCK_TIME_NONE) {
        GST_BUFFER_TIMESTAMP (outbuf) = rsvg->first_timestamp;
        GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
        if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
          GstClockTime end =
              GST_BUFFER_TIMESTAMP_IS_VALID (buffer) ?
              GST_BUFFER_TIMESTAMP (buffer) : rsvg->first_timestamp;
          end += GST_BUFFER_DURATION (buffer);
          GST_BUFFER_DURATION (outbuf) = end - GST_BUFFER_TIMESTAMP (outbuf);
        }
        if (rsvg->fps_n == 0) {
          rsvg->first_timestamp = GST_CLOCK_TIME_NONE;
        } else {
          GST_BUFFER_DURATION (outbuf) =
              gst_util_uint64_scale (rsvg->frame_count, rsvg->fps_d,
              rsvg->fps_n * GST_SECOND);
        }
      } else if (rsvg->fps_n != 0) {
        GST_BUFFER_TIMESTAMP (outbuf) =
            rsvg->first_timestamp + gst_util_uint64_scale (rsvg->frame_count,
            rsvg->fps_d, rsvg->fps_n * GST_SECOND);
        GST_BUFFER_DURATION (outbuf) =
            gst_util_uint64_scale (rsvg->frame_count, rsvg->fps_d,
            rsvg->fps_n * GST_SECOND);
      } else {
        GST_BUFFER_TIMESTAMP (outbuf) = rsvg->first_timestamp;
        GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
      }
      rsvg->frame_count++;

      if (rsvg->need_newsegment) {
        gst_pad_push_event (rsvg->srcpad,
            gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0));
        rsvg->need_newsegment = FALSE;
      }

      if (rsvg->pending_events) {
        GList *l;

        for (l = rsvg->pending_events; l; l = l->next)
          gst_pad_push_event (rsvg->srcpad, l->data);
        g_list_free (rsvg->pending_events);
        rsvg->pending_events = NULL;
      }

      if (rsvg->pending_tags) {
        gst_element_found_tags (GST_ELEMENT_CAST (rsvg), rsvg->pending_tags);
        rsvg->pending_tags = NULL;
      }

      GST_LOG_OBJECT (rsvg, "image rendered okay");

      ret = gst_pad_push (rsvg->srcpad, outbuf);
      if (ret != GST_FLOW_OK)
        break;

      gst_adapter_flush (rsvg->adapter, size);
      size = gst_adapter_available (rsvg->adapter);
      continue;
    } else {
      break;
    }
  }

  return GST_FLOW_OK;
}
Beispiel #17
0
static GstFlowReturn
mpegpsmux_collected (GstCollectPads * pads, MpegPsMux * mux)
{
  /* main muxing function */

  GstFlowReturn ret = GST_FLOW_OK;
  MpegPsPadData *best = NULL;
  gboolean keyunit;

  GST_DEBUG_OBJECT (mux, "Pads collected");

  if (mux->first) {             /* process block for the first mux */
    /* iterate through the collect pads and add streams to @mux */
    ret = mpegpsmux_create_streams (mux);
    /* Assumption : all pads are already added at this time */

    if (G_UNLIKELY (ret != GST_FLOW_OK))
      return ret;

    best = mpegpsmux_choose_best_stream (mux);

    /* prepare the src pad (output), return if failed */
    if (!mpegpsdemux_prepare_srcpad (mux)) {
      GST_DEBUG_OBJECT (mux, "Failed to send new segment");
      goto new_seg_fail;
    }

    mux->first = FALSE;
  } else {
    best = mpegpsmux_choose_best_stream (mux);
  }

  if (best != NULL) {
    GstBuffer *buf = best->queued.buf;
    gint64 pts, dts;

    g_assert (buf != NULL);

    GST_LOG_OBJECT (mux,
        "Chose stream from pad %" GST_PTR_FORMAT " for output (PID: 0x%04x): "
        "adjusted pts: %" GST_TIME_FORMAT ", dts: %" GST_TIME_FORMAT,
        best->collect.pad, best->stream_id,
        GST_TIME_ARGS (best->queued.pts), GST_TIME_ARGS (best->queued.dts));

    /* and convert to mpeg time stamps */
    pts = GSTTIME_TO_MPEGTIME (best->queued.pts);
    dts = GSTTIME_TO_MPEGTIME (best->queued.dts);

    /* start of new GOP? */
    keyunit = !GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DELTA_UNIT);

    if (keyunit && best->stream_id == mux->video_stream_id
        && mux->gop_list != NULL) {
      ret = mpegpsmux_push_gop_list (mux);
      if (ret != GST_FLOW_OK)
        goto done;
    }

    /* give the buffer to libpsmux for processing */
    psmux_stream_add_data (best->stream, buf, pts, dts, keyunit);

    best->queued.buf = NULL;

    /* write the data from libpsmux to stream */
    while (psmux_stream_bytes_in_buffer (best->stream) > 0) {
      GST_LOG_OBJECT (mux, "Before @psmux_write_stream_packet");
      if (!psmux_write_stream_packet (mux->psmux, best->stream)) {
        GST_DEBUG_OBJECT (mux, "Failed to write data packet");
        goto write_fail;
      }
    }
    mux->last_ts = best->last_ts;
  } else {
    /* FIXME: Drain all remaining streams */
    /* At EOS */
    if (mux->gop_list != NULL)
      mpegpsmux_push_gop_list (mux);

    if (!psmux_write_end_code (mux->psmux)) {
      GST_WARNING_OBJECT (mux, "Writing MPEG PS Program end code failed.");
    }
    gst_pad_push_event (mux->srcpad, gst_event_new_eos ());

    ret = GST_FLOW_EOS;
  }

done:

  return ret;
new_seg_fail:
  return GST_FLOW_ERROR;
write_fail:
  /* FIXME: Failed writing data for some reason. Should set appropriate error */
  return mux->last_flow_ret;
}
static GstFlowReturn
xing_mp3_encoder_chain(GstPad *pad, GstBuffer *buf)
{
    XingMp3Encoder *encoder;
    GstFlowReturn ret = GST_FLOW_OK;
  
    g_return_val_if_fail(pad != NULL, GST_FLOW_ERROR);
    g_return_val_if_fail(GST_IS_PAD(pad), GST_FLOW_ERROR);
    g_return_val_if_fail(buf != NULL, GST_FLOW_ERROR);

    encoder = XING_MP3_ENCODER(gst_pad_get_parent(pad));

    if(GST_IS_EVENT(buf)) {
        GstEvent *event = GST_EVENT(buf);

        switch(GST_EVENT_TYPE(event)) {
            case GST_EVENT_EOS:
                encoder->at_end_of_stream = TRUE;
                gst_event_unref(event);
                break;

            default:
                gst_pad_event_default(pad, event);
                return ret;
        }
    } else {
        guchar *data;
        gulong size;
        gulong i, j;
        float **buffer;
        gint buf_size;
        gint buf_pos;

        if(!encoder->is_initialized) {
            gst_buffer_unref(buf);
            GST_ELEMENT_ERROR(encoder, CORE, NEGOTIATION, (NULL), ("Encoder not initialized"));
            return GST_FLOW_UNEXPECTED;
        }

        if(!encoder->header_sent) {
            E_CONTROL control;
            MPEG_HEAD head;
            guchar output_buffer[OUTPUT_BUFFER_SIZE];
            gint buf_len;

            if(!encoder->use_cbr) {
                GstFlowReturn push_ret;
                
                hx_mp3enc_l3_info_ec(encoder->xing_encoder, &control);
                hx_mp3enc_l3_info_head(encoder->xing_encoder, &head);

                buf_len = XingHeader(control.samprate, head.mode, control.cr_bit, control.original,
                    VBR_SCALE_FLAG /* FRAMES_FLAG | BYTES_FLAG | TOC_FLAG */, 0, 0,
                    control.vbr_flag ? control.vbr_mnr : -1, NULL,
                    output_buffer, 0, 0, 0);
            
                if((push_ret = xing_mp3_encoder_push_buffer(encoder, output_buffer,
                    buf_len)) != GST_FLOW_OK) {
                    gst_buffer_unref(buf);
                    return push_ret;
                }
            }
            
            encoder->header_sent = TRUE;
        }

        data = (guchar *)GST_BUFFER_DATA(buf);
        buf_size = GST_BUFFER_SIZE(buf);
        buf_pos = 0;

        /* Transfer incoming data to internal buffer.
         * TODO: Use a ring buffer, avoid memmove () */

        while(buf_pos < buf_size) {
            gint gulp = MIN(buf_size - buf_pos, INPUT_BUFFER_SIZE - encoder->input_buffer_pos);

            memcpy(encoder->input_buffer + encoder->input_buffer_pos, data + buf_pos, gulp);

            encoder->samples_in += gulp / (2 * encoder->channels);
            encoder->input_buffer_pos += gulp;
            buf_pos += gulp;

            /* Pass data on to encoder */
            while(encoder->input_buffer_pos >= encoder->bytes_per_frame) {
                guchar output_buffer[OUTPUT_BUFFER_SIZE];
                IN_OUT x;

                x = hx_mp3enc_mp3_encode_frame(encoder->xing_encoder, 
                    encoder->input_buffer, output_buffer);

                if(x.in_bytes == 0 && x.out_bytes == 0) {
                    break;
                }

                memmove(encoder->input_buffer, encoder->input_buffer + x.in_bytes,
                    encoder->input_buffer_pos - x.in_bytes);
                encoder->input_buffer_pos -= x.in_bytes;

                /* Accept output from encoder and pass it on.
                 * TODO: Do this less often and save CPU */
                if(x.out_bytes > 0) {
                    GstFlowReturn push_ret;
                    
                    if((push_ret = xing_mp3_encoder_push_buffer(encoder, output_buffer,
                        x.out_bytes)) != GST_FLOW_OK) {
                        gst_buffer_unref(buf);
                        return push_ret;
                    }
                }
            }
        }

        gst_buffer_unref(buf);
    }

    if(encoder->at_end_of_stream) {
        gst_pad_push_event(encoder->srcpad, gst_event_new_eos());
    }
  
    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_FLOW_ERROR (typefind, ret);
      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;
  }
}
static gboolean
gst_wildmidi_do_seek (GstWildmidi * wildmidi, GstEvent * event)
{
  gdouble rate;
  GstFormat src_format, dst_format;
  GstSeekFlags flags;
  GstSeekType start_type, stop_type;
  gint64 start, stop;
  gboolean flush, update;
#ifdef HAVE_WILDMIDI_0_2_2
  gboolean accurate;
#endif
  gboolean res;
  unsigned long int sample;
  GstSegment *segment;

  if (!wildmidi->song)
    return FALSE;

  gst_event_parse_seek (event, &rate, &src_format, &flags,
      &start_type, &start, &stop_type, &stop);

  /* convert the input format to samples */
  dst_format = GST_FORMAT_DEFAULT;
  res = TRUE;
  if (start_type != GST_SEEK_TYPE_NONE) {
    res =
        gst_wildmidi_src_convert (wildmidi, src_format, start, &dst_format,
        &start);
  }
  if (res && stop_type != GST_SEEK_TYPE_NONE) {
    res =
        gst_wildmidi_src_convert (wildmidi, src_format, stop, &dst_format,
        &stop);
  }
  /* unsupported format */
  if (!res)
    return res;

  flush = ((flags & GST_SEEK_FLAG_FLUSH) == GST_SEEK_FLAG_FLUSH);
#ifdef HAVE_WILDMIDI_0_2_2
  accurate = ((flags & GST_SEEK_FLAG_ACCURATE) == GST_SEEK_FLAG_ACCURATE);
#endif

  if (flush) {
    GST_DEBUG ("performing flush");
    gst_pad_push_event (wildmidi->srcpad, gst_event_new_flush_start ());
  } else {
    gst_pad_stop_task (wildmidi->sinkpad);
  }

  segment = wildmidi->o_segment;

  GST_PAD_STREAM_LOCK (wildmidi->sinkpad);

  if (flush) {
    gst_pad_push_event (wildmidi->srcpad, gst_event_new_flush_stop (TRUE));
  }

  /* update the segment now */
  gst_segment_do_seek (segment, rate, dst_format, flags,
      start_type, start, stop_type, stop, &update);

  /* we need to seek to position in the segment now, sample will be updated */
  sample = segment->position;

  GST_OBJECT_LOCK (wildmidi);
#ifdef HAVE_WILDMIDI_0_2_2
  if (accurate) {
    WildMidi_SampledSeek (wildmidi->song, &sample);
  } else {
    WildMidi_FastSeek (wildmidi->song, &sample);
  }
#else
  WildMidi_FastSeek (wildmidi->song, &sample);
#endif

  GST_OBJECT_UNLOCK (wildmidi);

  segment->start = segment->time = segment->position = sample;

  gst_pad_push_event (wildmidi->srcpad,
      gst_wildmidi_get_new_segment_event (wildmidi, GST_FORMAT_TIME));

  gst_pad_start_task (wildmidi->sinkpad,
      (GstTaskFunction) gst_wildmidi_loop, wildmidi->sinkpad, NULL);

  wildmidi->discont = TRUE;
  GST_PAD_STREAM_UNLOCK (wildmidi->sinkpad);
  GST_DEBUG ("seek done");

  return TRUE;
}
static gboolean
gst_type_find_element_sink_event (GstPad * pad, GstObject * parent,
    GstEvent * event)
{
  gboolean res = FALSE;
  GstTypeFindElement *typefind = GST_TYPE_FIND_ELEMENT (parent);

  GST_DEBUG_OBJECT (typefind, "got %s event in mode %d",
      GST_EVENT_TYPE_NAME (event), typefind->mode);

  switch (typefind->mode) {
    case MODE_TYPEFIND:
      switch (GST_EVENT_TYPE (event)) {
        case GST_EVENT_CAPS:
        {
          GstCaps *caps;

          /* Parse and push out our caps and data */
          gst_event_parse_caps (event, &caps);
          res = gst_type_find_element_setcaps (typefind, caps);

          gst_event_unref (event);
          break;
        }
        case GST_EVENT_GAP:
        {
          GST_FIXME_OBJECT (typefind,
              "GAP events during typefinding not handled properly");

          /* FIXME: These would need to be inserted in the stream at
           * the right position between buffers, but we combine all
           * buffers with a GstAdapter. Drop the GAP event for now,
           * which will only cause an implicit GAP between buffers.
           */
          gst_event_unref (event);
          res = TRUE;
          break;
        }
        case GST_EVENT_EOS:
        {
          GST_INFO_OBJECT (typefind, "Got EOS and no type found yet");
          gst_type_find_element_chain_do_typefinding (typefind, FALSE, TRUE);

          res = gst_pad_push_event (typefind->src, event);
          break;
        }
        case GST_EVENT_FLUSH_STOP:{
          GList *l;

          GST_OBJECT_LOCK (typefind);

          for (l = typefind->cached_events; l; l = l->next) {
            if (GST_EVENT_IS_STICKY (l->data) &&
                GST_EVENT_TYPE (l->data) != GST_EVENT_SEGMENT &&
                GST_EVENT_TYPE (l->data) != GST_EVENT_EOS) {
              gst_pad_store_sticky_event (typefind->src, l->data);
            }
            gst_event_unref (l->data);
          }

          g_list_free (typefind->cached_events);
          typefind->cached_events = NULL;
          gst_adapter_clear (typefind->adapter);
          GST_OBJECT_UNLOCK (typefind);
          /* fall through */
        }
        case GST_EVENT_FLUSH_START:
          res = gst_pad_push_event (typefind->src, event);
          break;
        default:
          /* Forward events that would happen before the caps event
           * directly instead of storing them. There's no reason not
           * to send them directly and we should only store events
           * for later sending that would need to come after the caps
           * event */
          if (GST_EVENT_TYPE (event) < GST_EVENT_CAPS) {
            res = gst_pad_push_event (typefind->src, event);
          } else {
            GST_DEBUG_OBJECT (typefind, "Saving %s event to send later",
                GST_EVENT_TYPE_NAME (event));
            GST_OBJECT_LOCK (typefind);
            typefind->cached_events =
                g_list_append (typefind->cached_events, event);
            GST_OBJECT_UNLOCK (typefind);
            res = TRUE;
          }
          break;
      }
      break;
    case MODE_NORMAL:
      res = gst_pad_push_event (typefind->src, event);
      break;
    case MODE_ERROR:
      break;
    default:
      g_assert_not_reached ();
  }
  return res;
}
static GstFlowReturn
gst_wildmidi_parse_song (GstWildmidi * wildmidi)
{
  struct _WM_Info *info;
  GstCaps *outcaps;
  guint8 *data;
  guint size;

  GST_DEBUG_OBJECT (wildmidi, "Parsing song");

  size = gst_adapter_available (wildmidi->adapter);
  data = gst_adapter_take (wildmidi->adapter, size);

  /* this method takes our memory block */
  GST_OBJECT_LOCK (wildmidi);
  wildmidi->song = WildMidi_OpenBuffer (data, size);

  if (!wildmidi->song)
    goto open_failed;

#ifdef HAVE_WILDMIDI_0_2_2
  WildMidi_LoadSamples (wildmidi->song);
#endif

#ifdef HAVE_WILDMIDI_0_2_2
  WildMidi_SetOption (wildmidi->song, WM_MO_LINEAR_VOLUME,
      wildmidi->linear_volume);
  WildMidi_SetOption (wildmidi->song, WM_MO_EXPENSIVE_INTERPOLATION,
      wildmidi->high_quality);
#else
  WildMidi_SetOption (wildmidi->song, WM_MO_LOG_VOLUME,
      !wildmidi->linear_volume);
  WildMidi_SetOption (wildmidi->song, WM_MO_ENHANCED_RESAMPLING,
      wildmidi->high_quality);
#endif

  info = WildMidi_GetInfo (wildmidi->song);
  GST_OBJECT_UNLOCK (wildmidi);

  wildmidi->o_len = info->approx_total_samples;

  outcaps = gst_caps_copy (gst_pad_get_pad_template_caps (wildmidi->srcpad));
  gst_pad_set_caps (wildmidi->srcpad, outcaps);
  gst_caps_unref (outcaps);

  /* we keep an internal segment in samples */
  gst_segment_init (wildmidi->o_segment, GST_FORMAT_DEFAULT);

  gst_pad_push_event (wildmidi->srcpad,
      gst_wildmidi_get_new_segment_event (wildmidi, GST_FORMAT_TIME));

  GST_DEBUG_OBJECT (wildmidi, "Parsing song done");

  return GST_FLOW_OK;

  /* ERRORS */
open_failed:
  {
    GST_OBJECT_UNLOCK (wildmidi);
    GST_ELEMENT_ERROR (wildmidi, STREAM, DECODE, (NULL),
        ("Unable to parse midi data"));
    return GST_FLOW_ERROR;
  }
}
static void
gst_musepackdec_loop (GstPad * sinkpad)
{
  GstMusepackDec *musepackdec;
  GstFlowReturn flow;
  GstBuffer *out;

#ifdef MPC_IS_OLD_API
  guint32 update_acc, update_bits;
#else
  mpc_frame_info frame;
  mpc_status err;
#endif
  gint num_samples, samplerate, bitspersample;

  musepackdec = GST_MUSEPACK_DEC (GST_PAD_PARENT (sinkpad));

  samplerate = g_atomic_int_get (&musepackdec->rate);

  if (samplerate == 0) {
    if (!gst_musepack_stream_init (musepackdec))
      goto pause_task;

    gst_musepackdec_send_newsegment (musepackdec);
    samplerate = g_atomic_int_get (&musepackdec->rate);
  }

  bitspersample = g_atomic_int_get (&musepackdec->bps);

  flow = gst_pad_alloc_buffer_and_set_caps (musepackdec->srcpad, -1,
      MPC_DECODER_BUFFER_LENGTH * 4, GST_PAD_CAPS (musepackdec->srcpad), &out);

  if (flow != GST_FLOW_OK) {
    GST_DEBUG_OBJECT (musepackdec, "Flow: %s", gst_flow_get_name (flow));
    goto pause_task;
  }
#ifdef MPC_IS_OLD_API
  num_samples = mpc_decoder_decode (musepackdec->d,
      (MPC_SAMPLE_FORMAT *) GST_BUFFER_DATA (out), &update_acc, &update_bits);

  if (num_samples < 0) {
    GST_ERROR_OBJECT (musepackdec, "Failed to decode sample");
    GST_ELEMENT_ERROR (musepackdec, STREAM, DECODE, (NULL), (NULL));
    goto pause_task;
  } else if (num_samples == 0) {
    goto eos_and_pause;
  }
#else
  frame.buffer = (MPC_SAMPLE_FORMAT *) GST_BUFFER_DATA (out);
  err = mpc_demux_decode (musepackdec->d, &frame);

  if (err != MPC_STATUS_OK) {
    GST_ERROR_OBJECT (musepackdec, "Failed to decode sample");
    GST_ELEMENT_ERROR (musepackdec, STREAM, DECODE, (NULL), (NULL));
    goto pause_task;
  } else if (frame.bits == -1) {
    goto eos_and_pause;
  }

  num_samples = frame.samples;
#endif

  GST_BUFFER_SIZE (out) = num_samples * bitspersample;

  GST_BUFFER_OFFSET (out) = musepackdec->segment.last_stop;
  GST_BUFFER_TIMESTAMP (out) =
      gst_util_uint64_scale_int (musepackdec->segment.last_stop,
      GST_SECOND, samplerate);
  GST_BUFFER_DURATION (out) =
      gst_util_uint64_scale_int (num_samples, GST_SECOND, samplerate);

  musepackdec->segment.last_stop += num_samples;

  GST_LOG_OBJECT (musepackdec, "Pushing buffer, timestamp %" GST_TIME_FORMAT,
      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out)));

  flow = gst_pad_push (musepackdec->srcpad, out);
  if (flow != GST_FLOW_OK) {
    GST_DEBUG_OBJECT (musepackdec, "Flow: %s", gst_flow_get_name (flow));
    goto pause_task;
  }

  /* check if we're at the end of a configured segment */
  if (musepackdec->segment.stop != -1 &&
      musepackdec->segment.last_stop >= musepackdec->segment.stop) {
    gint64 stop_time;

    GST_DEBUG_OBJECT (musepackdec, "Reached end of configured segment");

    if ((musepackdec->segment.flags & GST_SEEK_FLAG_SEGMENT) == 0)
      goto eos_and_pause;

    GST_DEBUG_OBJECT (musepackdec, "Posting SEGMENT_DONE message");

    stop_time = gst_util_uint64_scale_int (musepackdec->segment.stop,
        GST_SECOND, samplerate);

    gst_element_post_message (GST_ELEMENT (musepackdec),
        gst_message_new_segment_done (GST_OBJECT (musepackdec),
            GST_FORMAT_TIME, stop_time));

    goto pause_task;
  }

  return;

eos_and_pause:
  {
    GST_DEBUG_OBJECT (musepackdec, "sending EOS event");
    gst_pad_push_event (musepackdec->srcpad, gst_event_new_eos ());
    /* fall through to pause */
  }

pause_task:
  {
    GST_DEBUG_OBJECT (musepackdec, "Pausing task");
    gst_pad_pause_task (sinkpad);
    return;
  }
}
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);
    }
  }
}
Beispiel #25
0
static void
gst_raw_parse_loop (GstElement * element)
{
  GstRawParse *rp = GST_RAW_PARSE (element);
  GstRawParseClass *rp_class = GST_RAW_PARSE_GET_CLASS (rp);
  GstFlowReturn ret;
  GstBuffer *buffer;
  gint size;

  if (!gst_raw_parse_set_src_caps (rp))
    goto no_caps;

  if (rp->close_segment) {
    GST_DEBUG_OBJECT (rp, "sending close segment");
    gst_pad_push_event (rp->srcpad, rp->close_segment);
    rp->close_segment = NULL;
  }
  if (rp->start_segment) {
    GST_DEBUG_OBJECT (rp, "sending start segment");
    gst_pad_push_event (rp->srcpad, rp->start_segment);
    rp->start_segment = NULL;
  }

  if (rp_class->multiple_frames_per_buffer && rp->framesize < 4096)
    size = 4096 - (4096 % rp->framesize);
  else
    size = rp->framesize;

  if (rp->segment.rate >= 0) {
    if (rp->offset + size > rp->upstream_length) {
      GstFormat fmt = GST_FORMAT_BYTES;

      if (!gst_pad_query_peer_duration (rp->sinkpad, &fmt,
              &rp->upstream_length)) {
        GST_WARNING_OBJECT (rp,
            "Could not get upstream duration, trying to pull frame by frame");
        size = rp->framesize;
      } else if (rp->upstream_length < rp->offset + rp->framesize) {
        ret = GST_FLOW_UNEXPECTED;
        goto pause;
      } else if (rp->offset + size > rp->upstream_length) {
        size = rp->upstream_length - rp->offset;
        size -= size % rp->framesize;
      }
    }
  } else {
    if (rp->offset == 0) {
      ret = GST_FLOW_UNEXPECTED;
      goto pause;
    } else if (rp->offset < size) {
      size -= rp->offset;
    }
    rp->offset -= size;
  }

  ret = gst_pad_pull_range (rp->sinkpad, rp->offset, size, &buffer);

  if (ret != GST_FLOW_OK) {
    GST_DEBUG_OBJECT (rp, "pull_range (%" G_GINT64_FORMAT ", %u) "
        "failed, flow: %s", rp->offset, size, gst_flow_get_name (ret));
    buffer = NULL;
    goto pause;
  }

  if (GST_BUFFER_SIZE (buffer) < size) {
    GST_DEBUG_OBJECT (rp, "Short read at offset %" G_GINT64_FORMAT
        ", got only %u of %u bytes", rp->offset, GST_BUFFER_SIZE (buffer),
        size);

    if (size > rp->framesize) {
      GST_BUFFER_SIZE (buffer) -= GST_BUFFER_SIZE (buffer) % rp->framesize;
    } else {
      gst_buffer_unref (buffer);
      buffer = NULL;
      ret = GST_FLOW_UNEXPECTED;
      goto pause;
    }
  }

  ret = gst_raw_parse_push_buffer (rp, buffer);
  if (ret != GST_FLOW_OK)
    goto pause;

  return;

  /* ERRORS */
no_caps:
  {
    GST_ERROR_OBJECT (rp, "could not negotiate caps");
    ret = GST_FLOW_NOT_NEGOTIATED;
    goto pause;
  }
pause:
  {
    const gchar *reason = gst_flow_get_name (ret);

    GST_LOG_OBJECT (rp, "pausing task, reason %s", reason);
    gst_pad_pause_task (rp->sinkpad);

    if (ret == GST_FLOW_UNEXPECTED) {
      if (rp->segment.flags & GST_SEEK_FLAG_SEGMENT) {
        GstClockTime stop;

        GST_LOG_OBJECT (rp, "Sending segment done");

        if ((stop = rp->segment.stop) == -1)
          stop = rp->segment.duration;

        gst_element_post_message (GST_ELEMENT_CAST (rp),
            gst_message_new_segment_done (GST_OBJECT_CAST (rp),
                rp->segment.format, stop));
      } else {
        GST_LOG_OBJECT (rp, "Sending EOS, at end of stream");
        gst_pad_push_event (rp->srcpad, gst_event_new_eos ());
      }
    } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_UNEXPECTED) {
      GST_ELEMENT_ERROR (rp, STREAM, FAILED,
          ("Internal data stream error."),
          ("stream stopped, reason %s", reason));
      gst_pad_push_event (rp->srcpad, gst_event_new_eos ());
    }
    return;
  }
}
static void
gst_amc_audio_dec_loop (GstAmcAudioDec * self)
{
  GstFlowReturn flow_ret = GST_FLOW_OK;
  gboolean is_eos;
  GstAmcBuffer *buf;
  GstAmcBufferInfo buffer_info;
  gint idx;
  GError *err = NULL;

  GST_AUDIO_DECODER_STREAM_LOCK (self);

retry:
  /*if (self->input_caps_changed) {
     idx = INFO_OUTPUT_FORMAT_CHANGED;
     } else { */
  GST_DEBUG_OBJECT (self, "Waiting for available output buffer");
  GST_AUDIO_DECODER_STREAM_UNLOCK (self);
  /* Wait at most 100ms here, some codecs don't fail dequeueing if
   * the codec is flushing, causing deadlocks during shutdown */
  idx =
      gst_amc_codec_dequeue_output_buffer (self->codec, &buffer_info, 100000,
      &err);
  GST_AUDIO_DECODER_STREAM_LOCK (self);
  /*} */

  if (idx < 0) {
    if (self->flushing) {
      g_clear_error (&err);
      goto flushing;
    }

    switch (idx) {
      case INFO_OUTPUT_BUFFERS_CHANGED:
        /* Handled internally */
        g_assert_not_reached ();
        break;
      case INFO_OUTPUT_FORMAT_CHANGED:{
        GstAmcFormat *format;
        gchar *format_string;

        GST_DEBUG_OBJECT (self, "Output format has changed");

        format = gst_amc_codec_get_output_format (self->codec, &err);
        if (!format)
          goto format_error;

        format_string = gst_amc_format_to_string (format, &err);
        if (err) {
          gst_amc_format_free (format);
          goto format_error;
        }
        GST_DEBUG_OBJECT (self, "Got new output format: %s", format_string);
        g_free (format_string);

        if (!gst_amc_audio_dec_set_src_caps (self, format)) {
          gst_amc_format_free (format);
          goto format_error;
        }
        gst_amc_format_free (format);

        goto retry;

      }
      case INFO_TRY_AGAIN_LATER:
        GST_DEBUG_OBJECT (self, "Dequeueing output buffer timed out");
        goto retry;

      case G_MININT:
        GST_ERROR_OBJECT (self, "Failure dequeueing output buffer");
        goto dequeue_error;

      default:
        g_assert_not_reached ();
        break;
    }

    goto retry;
  }

  GST_DEBUG_OBJECT (self,
      "Got output buffer at index %d: offset %d size %d time %" G_GINT64_FORMAT
      " flags 0x%08x", idx, buffer_info.offset, buffer_info.size,
      buffer_info.presentation_time_us, buffer_info.flags);

  is_eos = ! !(buffer_info.flags & BUFFER_FLAG_END_OF_STREAM);

  buf = gst_amc_codec_get_output_buffer (self->codec, idx, &err);
  if (err)
    goto failed_to_get_output_buffer;
  else if (!buf)
    goto got_null_output_buffer;

  if (buffer_info.size > 0) {
    GstBuffer *outbuf;
    GstMapInfo minfo;

    /* This sometimes happens at EOS or if the input is not properly framed,
     * let's handle it gracefully by allocating a new buffer for the current
     * caps and filling it
     */

    if (buffer_info.size % self->info.bpf != 0)
      goto invalid_buffer_size;

    outbuf =
        gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (self),
        buffer_info.size);
    if (!outbuf)
      goto failed_allocate;

    gst_buffer_map (outbuf, &minfo, GST_MAP_WRITE);
    if (self->needs_reorder) {
      gint i, n_samples, c, n_channels;
      gint *reorder_map = self->reorder_map;
      gint16 *dest, *source;

      dest = (gint16 *) minfo.data;
      source = (gint16 *) (buf->data + buffer_info.offset);
      n_samples = buffer_info.size / self->info.bpf;
      n_channels = self->info.channels;

      for (i = 0; i < n_samples; i++) {
        for (c = 0; c < n_channels; c++) {
          dest[i * n_channels + reorder_map[c]] = source[i * n_channels + c];
        }
      }
    } else {
      orc_memcpy (minfo.data, buf->data + buffer_info.offset, buffer_info.size);
    }
    gst_buffer_unmap (outbuf, &minfo);

    if (self->spf != -1) {
      gst_adapter_push (self->output_adapter, outbuf);
    } else {
      flow_ret =
          gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf, 1);
    }
  }

  gst_amc_buffer_free (buf);
  buf = NULL;

  if (self->spf != -1) {
    GstBuffer *outbuf;
    guint avail = gst_adapter_available (self->output_adapter);
    guint nframes;

    /* On EOS we take the complete adapter content, no matter
     * if it is a multiple of the codec frame size or not.
     * Otherwise we take a multiple of codec frames and push
     * them downstream
     */
    avail /= self->info.bpf;
    if (!is_eos) {
      nframes = avail / self->spf;
      avail = nframes * self->spf;
    } else {
      nframes = (avail + self->spf - 1) / self->spf;
    }
    avail *= self->info.bpf;

    if (avail > 0) {
      outbuf = gst_adapter_take_buffer (self->output_adapter, avail);
      flow_ret =
          gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf,
          nframes);
    }
  }

  if (!gst_amc_codec_release_output_buffer (self->codec, idx, FALSE, &err)) {
    if (self->flushing) {
      g_clear_error (&err);
      goto flushing;
    }
    goto failed_release;
  }

  if (is_eos || flow_ret == GST_FLOW_EOS) {
    GST_AUDIO_DECODER_STREAM_UNLOCK (self);
    g_mutex_lock (&self->drain_lock);
    if (self->draining) {
      GST_DEBUG_OBJECT (self, "Drained");
      self->draining = FALSE;
      g_cond_broadcast (&self->drain_cond);
    } else if (flow_ret == GST_FLOW_OK) {
      GST_DEBUG_OBJECT (self, "Component signalled EOS");
      flow_ret = GST_FLOW_EOS;
    }
    g_mutex_unlock (&self->drain_lock);
    GST_AUDIO_DECODER_STREAM_LOCK (self);
  } else {
    GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
  }

  self->downstream_flow_ret = flow_ret;

  if (flow_ret != GST_FLOW_OK)
    goto flow_error;

  GST_AUDIO_DECODER_STREAM_UNLOCK (self);

  return;

dequeue_error:
  {
    GST_ELEMENT_ERROR_FROM_ERROR (self, err);
    gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
    gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
    self->downstream_flow_ret = GST_FLOW_ERROR;
    GST_AUDIO_DECODER_STREAM_UNLOCK (self);
    g_mutex_lock (&self->drain_lock);
    self->draining = FALSE;
    g_cond_broadcast (&self->drain_cond);
    g_mutex_unlock (&self->drain_lock);
    return;
  }

format_error:
  {
    if (err)
      GST_ELEMENT_ERROR_FROM_ERROR (self, err);
    else
      GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
          ("Failed to handle format"));
    gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
    gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
    self->downstream_flow_ret = GST_FLOW_ERROR;
    GST_AUDIO_DECODER_STREAM_UNLOCK (self);
    g_mutex_lock (&self->drain_lock);
    self->draining = FALSE;
    g_cond_broadcast (&self->drain_cond);
    g_mutex_unlock (&self->drain_lock);
    return;
  }
failed_release:
  {
    GST_AUDIO_DECODER_ERROR_FROM_ERROR (self, err);
    gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
    gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
    self->downstream_flow_ret = GST_FLOW_ERROR;
    GST_AUDIO_DECODER_STREAM_UNLOCK (self);
    g_mutex_lock (&self->drain_lock);
    self->draining = FALSE;
    g_cond_broadcast (&self->drain_cond);
    g_mutex_unlock (&self->drain_lock);
    return;
  }
flushing:
  {
    GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
    gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
    self->downstream_flow_ret = GST_FLOW_FLUSHING;
    GST_AUDIO_DECODER_STREAM_UNLOCK (self);
    return;
  }

flow_error:
  {
    if (flow_ret == GST_FLOW_EOS) {
      GST_DEBUG_OBJECT (self, "EOS");
      gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self),
          gst_event_new_eos ());
      gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
    } else if (flow_ret < GST_FLOW_EOS) {
      GST_ELEMENT_ERROR (self, STREAM, FAILED,
          ("Internal data stream error."), ("stream stopped, reason %s",
              gst_flow_get_name (flow_ret)));
      gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self),
          gst_event_new_eos ());
      gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
    } else if (flow_ret == GST_FLOW_FLUSHING) {
      GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
      gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
    }
    GST_AUDIO_DECODER_STREAM_UNLOCK (self);
    g_mutex_lock (&self->drain_lock);
    self->draining = FALSE;
    g_cond_broadcast (&self->drain_cond);
    g_mutex_unlock (&self->drain_lock);
    return;
  }

failed_to_get_output_buffer:
  {
    GST_AUDIO_DECODER_ERROR_FROM_ERROR (self, err);
    gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
    gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
    self->downstream_flow_ret = GST_FLOW_ERROR;
    GST_AUDIO_DECODER_STREAM_UNLOCK (self);
    g_mutex_lock (&self->drain_lock);
    self->draining = FALSE;
    g_cond_broadcast (&self->drain_cond);
    g_mutex_unlock (&self->drain_lock);
    return;
  }

got_null_output_buffer:
  {
    GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
        ("Got no output buffer"));
    gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
    gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
    self->downstream_flow_ret = GST_FLOW_ERROR;
    GST_AUDIO_DECODER_STREAM_UNLOCK (self);
    g_mutex_lock (&self->drain_lock);
    self->draining = FALSE;
    g_cond_broadcast (&self->drain_cond);
    g_mutex_unlock (&self->drain_lock);
    return;
  }

invalid_buffer_size:
  {
    GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
        ("Invalid buffer size %u (bfp %d)", buffer_info.size, self->info.bpf));
    gst_amc_codec_release_output_buffer (self->codec, idx, FALSE, &err);
    if (err && !self->flushing)
      GST_ELEMENT_WARNING_FROM_ERROR (self, err);
    g_clear_error (&err);
    gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
    gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
    self->downstream_flow_ret = GST_FLOW_ERROR;
    GST_AUDIO_DECODER_STREAM_UNLOCK (self);
    g_mutex_lock (&self->drain_lock);
    self->draining = FALSE;
    g_cond_broadcast (&self->drain_cond);
    g_mutex_unlock (&self->drain_lock);
    return;
  }

failed_allocate:
  {
    GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
        ("Failed to allocate output buffer"));
    gst_amc_codec_release_output_buffer (self->codec, idx, FALSE, &err);
    if (err && !self->flushing)
      GST_ELEMENT_WARNING_FROM_ERROR (self, err);
    g_clear_error (&err);
    gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
    gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
    self->downstream_flow_ret = GST_FLOW_ERROR;
    GST_AUDIO_DECODER_STREAM_UNLOCK (self);
    g_mutex_lock (&self->drain_lock);
    self->draining = FALSE;
    g_cond_broadcast (&self->drain_cond);
    g_mutex_unlock (&self->drain_lock);
    return;
  }
}
Beispiel #27
0
static gboolean
gst_raw_parse_handle_seek_pull (GstRawParse * rp, GstEvent * event)
{
  gdouble rate;
  GstFormat format;
  GstSeekFlags flags;
  GstSeekType start_type, stop_type;
  gint64 start, stop;
  gint64 last_stop;
  gboolean ret = FALSE;
  gboolean flush;
  GstSegment seeksegment;

  if (event) {
    gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start,
        &stop_type, &stop);

    /* convert input offsets to time */
    ret = gst_raw_parse_convert (rp, format, start, GST_FORMAT_TIME, &start);
    ret &= gst_raw_parse_convert (rp, format, stop, GST_FORMAT_TIME, &stop);
    if (!ret)
      goto convert_failed;

    GST_DEBUG_OBJECT (rp, "converted start - stop to time");

    format = GST_FORMAT_TIME;

    gst_event_unref (event);
  } else {
    format = GST_FORMAT_TIME;
    flags = 0;
  }

  flush = ((flags & GST_SEEK_FLAG_FLUSH) != 0);

  /* start flushing up and downstream so that the loop function pauses and we
   * can acquire the STREAM_LOCK. */
  if (flush) {
    GST_LOG_OBJECT (rp, "flushing");
    gst_pad_push_event (rp->sinkpad, gst_event_new_flush_start ());
    gst_pad_push_event (rp->srcpad, gst_event_new_flush_start ());
  } else {
    GST_LOG_OBJECT (rp, "pause task");
    gst_pad_pause_task (rp->sinkpad);
  }

  GST_PAD_STREAM_LOCK (rp->sinkpad);

  memcpy (&seeksegment, &rp->segment, sizeof (GstSegment));

  if (event) {
    /* configure the seek values */
    gst_segment_set_seek (&seeksegment, rate, format, flags,
        start_type, start, stop_type, stop, NULL);
  }

  /* get the desired position */
  last_stop = seeksegment.last_stop;

  GST_LOG_OBJECT (rp, "seeking to %" GST_TIME_FORMAT,
      GST_TIME_ARGS (last_stop));

  /* convert the desired position to bytes */
  ret =
      gst_raw_parse_convert (rp, format, last_stop, GST_FORMAT_BYTES,
      &last_stop);

  /* prepare for streaming */
  if (flush) {
    GST_LOG_OBJECT (rp, "stop flush");
    gst_pad_push_event (rp->sinkpad, gst_event_new_flush_stop ());
    gst_pad_push_event (rp->srcpad, gst_event_new_flush_stop ());
  } else if (ret && rp->running) {
    /* we are running the current segment and doing a non-flushing seek, 
     * close the segment first based on the last_stop. */
    GST_DEBUG_OBJECT (rp, "prepare close segment %" G_GINT64_FORMAT
        " to %" G_GINT64_FORMAT, rp->segment.start, rp->segment.last_stop);

    /* queue the segment for sending in the stream thread */
    if (rp->close_segment)
      gst_event_unref (rp->close_segment);
    rp->close_segment =
        gst_event_new_new_segment_full (TRUE,
        rp->segment.rate, rp->segment.applied_rate, rp->segment.format,
        rp->segment.start, rp->segment.last_stop, rp->segment.time);
  }

  if (ret) {
    /* seek done */

    /* Seek on a frame boundary */
    last_stop -= last_stop % rp->framesize;

    rp->offset = last_stop;
    rp->n_frames = last_stop / rp->framesize;

    GST_LOG_OBJECT (rp, "seeking to bytes %" G_GINT64_FORMAT, last_stop);

    memcpy (&rp->segment, &seeksegment, sizeof (GstSegment));

    if (rp->segment.flags & GST_SEEK_FLAG_SEGMENT) {
      gst_element_post_message (GST_ELEMENT_CAST (rp),
          gst_message_new_segment_start (GST_OBJECT_CAST (rp),
              rp->segment.format, rp->segment.last_stop));
    }

    /* for deriving a stop position for the playback segment from the seek
     * segment, we must take the duration when the stop is not set */
    if ((stop = rp->segment.stop) == -1)
      stop = rp->segment.duration;

    GST_DEBUG_OBJECT (rp, "preparing newsegment from %" G_GINT64_FORMAT
        " to %" G_GINT64_FORMAT, rp->segment.start, stop);

    /* now replace the old segment so that we send it in the stream thread the
     * next time it is scheduled. */
    if (rp->start_segment)
      gst_event_unref (rp->start_segment);

    if (rp->segment.rate >= 0.0) {
      /* forward, we send data from last_stop to stop */
      rp->start_segment =
          gst_event_new_new_segment_full (FALSE,
          rp->segment.rate, rp->segment.applied_rate, rp->segment.format,
          rp->segment.last_stop, stop, rp->segment.time);
    } else {
      /* reverse, we send data from last_stop to start */
      rp->start_segment =
          gst_event_new_new_segment_full (FALSE,
          rp->segment.rate, rp->segment.applied_rate, rp->segment.format,
          rp->segment.start, rp->segment.last_stop, rp->segment.time);
    }
  }
  rp->discont = TRUE;

  GST_LOG_OBJECT (rp, "start streaming");
  rp->running = TRUE;
  gst_pad_start_task (rp->sinkpad, (GstTaskFunction) gst_raw_parse_loop, rp);

  GST_PAD_STREAM_UNLOCK (rp->sinkpad);

  return ret;

  /* ERRORS */
convert_failed:
  {
    GST_DEBUG_OBJECT (rp, "Seek failed: couldn't convert to byte positions");
    return FALSE;
  }
}
Beispiel #28
0
/* chain function
 * this function does the actual processing
 */
static GstFlowReturn
gst_ivf_parse_chain (GstPad * pad, GstBuffer * buf)
{
  GstIvfParse *ivf = GST_IVF_PARSE (GST_OBJECT_PARENT (pad));
  gboolean res;

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

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

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

  res = GST_FLOW_OK;

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

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

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

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

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

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

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

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

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

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

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

      /* fall through */

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

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

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

          gst_adapter_flush (ivf->adapter, 12);

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

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

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

    default:
      g_return_val_if_reached (GST_FLOW_ERROR);
  }

  return res;
}
Beispiel #29
0
static gboolean
gst_base_video_codec_src_event (GstPad * pad, GstEvent * event)
{
  GstBaseVideoCodec *base_video_codec;
  gboolean res = FALSE;

  base_video_codec = GST_BASE_VIDEO_CODEC (gst_pad_get_parent (pad));

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_SEEK:
    {
      GstFormat format, tformat;
      gdouble rate;
      GstEvent *real_seek;
      GstSeekFlags flags;
      GstSeekType cur_type, stop_type;
      gint64 cur, stop;
      gint64 tcur, tstop;

      gst_event_parse_seek (event, &rate, &format, &flags, &cur_type,
          &cur, &stop_type, &stop);
      gst_event_unref (event);

      tformat = GST_FORMAT_TIME;
      res = gst_base_video_encoded_video_convert (&base_video_codec->state,
          format, cur, &tformat, &tcur);
      if (!res)
        goto convert_error;
      res = gst_base_video_encoded_video_convert (&base_video_codec->state,
          format, stop, &tformat, &tstop);
      if (!res)
        goto convert_error;

      real_seek = gst_event_new_seek (rate, GST_FORMAT_TIME,
          flags, cur_type, tcur, stop_type, tstop);

      res = gst_pad_push_event (base_video_codec->sinkpad, real_seek);

      break;
    }
#if 0
    case GST_EVENT_QOS:
    {
      gdouble proportion;
      GstClockTimeDiff diff;
      GstClockTime timestamp;

      gst_event_parse_qos (event, &proportion, &diff, &timestamp);

      GST_OBJECT_LOCK (base_video_codec);
      base_video_codec->proportion = proportion;
      base_video_codec->earliest_time = timestamp + diff;
      GST_OBJECT_UNLOCK (base_video_codec);

      GST_DEBUG_OBJECT (base_video_codec,
          "got QoS %" GST_TIME_FORMAT ", %" G_GINT64_FORMAT,
          GST_TIME_ARGS (timestamp), diff);

      res = gst_pad_push_event (base_video_codec->sinkpad, event);
      break;
    }
#endif
    default:
      res = gst_pad_push_event (base_video_codec->sinkpad, event);
      break;
  }
done:
  gst_object_unref (base_video_codec);
  return res;

convert_error:
  GST_DEBUG_OBJECT (base_video_codec, "could not convert format");
  goto done;
}
Beispiel #30
0
static void
gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
{
  GstOMXAudioEncClass *klass;
  GstOMXPort *port = self->enc_out_port;
  GstOMXBuffer *buf = NULL;
  GstFlowReturn flow_ret = GST_FLOW_OK;
  GstOMXAcquireBufferReturn acq_return;
  OMX_ERRORTYPE err;

  klass = GST_OMX_AUDIO_ENC_GET_CLASS (self);

  acq_return = gst_omx_port_acquire_buffer (port, &buf);
  if (acq_return == GST_OMX_ACQUIRE_BUFFER_ERROR) {
    goto component_error;
  } else if (acq_return == GST_OMX_ACQUIRE_BUFFER_FLUSHING) {
    goto flushing;
  } else if (acq_return == GST_OMX_ACQUIRE_BUFFER_EOS) {
    goto eos;
  }

  if (!gst_pad_has_current_caps (GST_AUDIO_ENCODER_SRC_PAD (self))
      || acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
    GstAudioInfo *info =
        gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self));
    GstCaps *caps;

    GST_DEBUG_OBJECT (self, "Port settings have changed, updating caps");

    /* Reallocate all buffers */
    if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
      err = gst_omx_port_set_enabled (port, FALSE);
      if (err != OMX_ErrorNone)
        goto reconfigure_error;

      err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND);
      if (err != OMX_ErrorNone)
        goto reconfigure_error;

      err = gst_omx_port_deallocate_buffers (port);
      if (err != OMX_ErrorNone)
        goto reconfigure_error;

      err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND);
      if (err != OMX_ErrorNone)
        goto reconfigure_error;

    }

    GST_AUDIO_ENCODER_STREAM_LOCK (self);

    caps = klass->get_caps (self, self->enc_out_port, info);
    if (!caps) {
      if (buf)
        gst_omx_port_release_buffer (self->enc_out_port, buf);
      GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
      goto caps_failed;
    }

    GST_DEBUG_OBJECT (self, "Setting output caps: %" GST_PTR_FORMAT, caps);

    if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
      gst_caps_unref (caps);
      if (buf)
        gst_omx_port_release_buffer (self->enc_out_port, buf);
      GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
      goto caps_failed;
    }
    gst_caps_unref (caps);

    GST_AUDIO_ENCODER_STREAM_UNLOCK (self);

    if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
      err = gst_omx_port_set_enabled (port, TRUE);
      if (err != OMX_ErrorNone)
        goto reconfigure_error;

      err = gst_omx_port_allocate_buffers (port);
      if (err != OMX_ErrorNone)
        goto reconfigure_error;

      err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND);
      if (err != OMX_ErrorNone)
        goto reconfigure_error;

      err = gst_omx_port_populate (port);
      if (err != OMX_ErrorNone)
        goto reconfigure_error;

      err = gst_omx_port_mark_reconfigured (port);
      if (err != OMX_ErrorNone)
        goto reconfigure_error;
    }

    /* Now get a buffer */
    if (acq_return != GST_OMX_ACQUIRE_BUFFER_OK) {
      return;
    }
  }

  g_assert (acq_return == GST_OMX_ACQUIRE_BUFFER_OK);
  if (!buf) {
    g_assert ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER));
    GST_AUDIO_ENCODER_STREAM_LOCK (self);
    goto eos;
  }

  GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %" G_GUINT64_FORMAT,
      (guint) buf->omx_buf->nFlags, (guint64) buf->omx_buf->nTimeStamp);

  /* This prevents a deadlock between the srcpad stream
   * lock and the videocodec stream lock, if ::reset()
   * is called at the wrong time
   */
  if (gst_omx_port_is_flushing (self->enc_out_port)) {
    GST_DEBUG_OBJECT (self, "Flushing");
    gst_omx_port_release_buffer (self->enc_out_port, buf);
    goto flushing;
  }

  GST_AUDIO_ENCODER_STREAM_LOCK (self);

  if ((buf->omx_buf->nFlags & OMX_BUFFERFLAG_CODECCONFIG)
      && buf->omx_buf->nFilledLen > 0) {
    GstCaps *caps;
    GstBuffer *codec_data;
    GstMapInfo map = GST_MAP_INFO_INIT;

    GST_DEBUG_OBJECT (self, "Handling codec data");
    caps =
        gst_caps_copy (gst_pad_get_current_caps (GST_AUDIO_ENCODER_SRC_PAD
            (self)));
    codec_data = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);

    gst_buffer_map (codec_data, &map, GST_MAP_WRITE);
    memcpy (map.data,
        buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
        buf->omx_buf->nFilledLen);
    gst_buffer_unmap (codec_data, &map);

    gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data, NULL);
    if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
      gst_caps_unref (caps);
      if (buf)
        gst_omx_port_release_buffer (self->enc_out_port, buf);
      GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
      goto caps_failed;
    }
    gst_caps_unref (caps);
    flow_ret = GST_FLOW_OK;
  } else if (buf->omx_buf->nFilledLen > 0) {
    GstBuffer *outbuf;
    guint n_samples;

    GST_DEBUG_OBJECT (self, "Handling output data");

    n_samples =
        klass->get_num_samples (self, self->enc_out_port,
        gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self)), buf);

    if (buf->omx_buf->nFilledLen > 0) {
      GstMapInfo map = GST_MAP_INFO_INIT;
      outbuf = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);

      gst_buffer_map (outbuf, &map, GST_MAP_WRITE);

      memcpy (map.data,
          buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
          buf->omx_buf->nFilledLen);
      gst_buffer_unmap (outbuf, &map);

    } else {
      outbuf = gst_buffer_new ();
    }

    GST_BUFFER_TIMESTAMP (outbuf) =
        gst_util_uint64_scale (buf->omx_buf->nTimeStamp, GST_SECOND,
        OMX_TICKS_PER_SECOND);
    if (buf->omx_buf->nTickCount != 0)
      GST_BUFFER_DURATION (outbuf) =
          gst_util_uint64_scale (buf->omx_buf->nTickCount, GST_SECOND,
          OMX_TICKS_PER_SECOND);

    flow_ret =
        gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (self),
        outbuf, n_samples);
  }

  GST_DEBUG_OBJECT (self, "Handled output data");

  GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));

  err = gst_omx_port_release_buffer (port, buf);
  if (err != OMX_ErrorNone)
    goto release_error;

  self->downstream_flow_ret = flow_ret;

  if (flow_ret != GST_FLOW_OK)
    goto flow_error;

  GST_AUDIO_ENCODER_STREAM_UNLOCK (self);

  return;

component_error:
  {
    GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
        ("OpenMAX component in error state %s (0x%08x)",
            gst_omx_component_get_last_error_string (self->enc),
            gst_omx_component_get_last_error (self->enc)));
    gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
    gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
    self->downstream_flow_ret = GST_FLOW_ERROR;
    self->started = FALSE;
    return;
  }
flushing:
  {
    GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
    gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
    self->downstream_flow_ret = GST_FLOW_FLUSHING;
    self->started = FALSE;
    return;
  }
eos:
  {
    g_mutex_lock (&self->drain_lock);
    if (self->draining) {
      GST_DEBUG_OBJECT (self, "Drained");
      self->draining = FALSE;
      g_cond_broadcast (&self->drain_cond);
      flow_ret = GST_FLOW_OK;
      gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
    } else {
      GST_DEBUG_OBJECT (self, "Component signalled EOS");
      flow_ret = GST_FLOW_EOS;
    }
    g_mutex_unlock (&self->drain_lock);

    GST_AUDIO_ENCODER_STREAM_LOCK (self);
    self->downstream_flow_ret = flow_ret;

    /* Here we fallback and pause the task for the EOS case */
    if (flow_ret != GST_FLOW_OK)
      goto flow_error;

    GST_AUDIO_ENCODER_STREAM_UNLOCK (self);

    return;
  }
flow_error:
  {
    if (flow_ret == GST_FLOW_EOS) {
      GST_DEBUG_OBJECT (self, "EOS");

      gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self),
          gst_event_new_eos ());
      gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
    } else if (flow_ret == GST_FLOW_NOT_LINKED || flow_ret < GST_FLOW_EOS) {
      GST_ELEMENT_ERROR (self, STREAM, FAILED, ("Internal data stream error."),
          ("stream stopped, reason %s", gst_flow_get_name (flow_ret)));

      gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self),
          gst_event_new_eos ());
      gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
    }
    self->started = FALSE;
    GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
    return;
  }
reconfigure_error:
  {
    GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
        ("Unable to reconfigure output port"));
    gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
    gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
    self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
    self->started = FALSE;
    return;
  }
caps_failed:
  {
    GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to set caps"));
    gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
    gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
    self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
    self->started = FALSE;
    return;
  }
release_error:
  {
    GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
        ("Failed to relase output buffer to component: %s (0x%08x)",
            gst_omx_error_to_string (err), err));
    gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
    gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
    self->downstream_flow_ret = GST_FLOW_ERROR;
    self->started = FALSE;
    GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
    return;
  }
}