• 设为首页
  • 点击收藏
  • 手机版
    手机扫一扫访问
    迪恩网络手机版
  • 关注官方公众号
    微信扫一扫关注
    迪恩网络公众号

C++ GST_BUFFER_OFFSET_END函数代码示例

原作者: [db:作者] 来自: [db:来源] 收藏 邀请

本文整理汇总了C++中GST_BUFFER_OFFSET_END函数的典型用法代码示例。如果您正苦于以下问题:C++ GST_BUFFER_OFFSET_END函数的具体用法?C++ GST_BUFFER_OFFSET_END怎么用?C++ GST_BUFFER_OFFSET_END使用的例子?那么恭喜您, 这里精选的函数代码示例或许可以为您提供帮助。



在下文中一共展示了GST_BUFFER_OFFSET_END函数的20个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于我们的系统推荐出更棒的C++代码示例。

示例1: gst_ogg_avi_parse_push_packet

static GstFlowReturn
gst_ogg_avi_parse_push_packet (GstOggAviParse * ogg, ogg_packet * packet)
{
  GstBuffer *buffer;
  GstFlowReturn result;

  /* allocate space for header and body */
  buffer = gst_buffer_new_and_alloc (packet->bytes);
  memcpy (GST_BUFFER_DATA (buffer), packet->packet, packet->bytes);

  GST_LOG_OBJECT (ogg, "created buffer %p from page", buffer);

  GST_BUFFER_OFFSET_END (buffer) = packet->granulepos;

  if (ogg->discont) {
    GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
    ogg->discont = FALSE;
  }

  result = gst_pad_push (ogg->srcpad, buffer);

  return result;
}
开发者ID:prajnashi,项目名称:gst-plugins-base,代码行数:23,代码来源:gstoggaviparse.c


示例2: gst_identity_update_last_message_for_buffer

static void
gst_identity_update_last_message_for_buffer (GstIdentity * identity,
    const gchar * action, GstBuffer * buf)
{
  gchar ts_str[64], dur_str[64];

  GST_OBJECT_LOCK (identity);

  g_free (identity->last_message);
  identity->last_message = g_strdup_printf ("%s   ******* (%s:%s)i "
      "(%u bytes, timestamp: %s, duration: %s, offset: %" G_GINT64_FORMAT ", "
      "offset_end: % " G_GINT64_FORMAT ", flags: %d) %p", action,
      GST_DEBUG_PAD_NAME (GST_BASE_TRANSFORM_CAST (identity)->sinkpad),
      GST_BUFFER_SIZE (buf),
      print_pretty_time (ts_str, sizeof (ts_str), GST_BUFFER_TIMESTAMP (buf)),
      print_pretty_time (dur_str, sizeof (dur_str), GST_BUFFER_DURATION (buf)),
      GST_BUFFER_OFFSET (buf), GST_BUFFER_OFFSET_END (buf),
      GST_BUFFER_FLAGS (buf), buf);

  GST_OBJECT_UNLOCK (identity);

  gst_identity_notify_last_message (identity);
}
开发者ID:166MMX,项目名称:openjdk.java.net-openjfx-8u40-rt,代码行数:23,代码来源:gstidentity.c


示例3: gst_kate_parse_push_buffer

static GstFlowReturn
gst_kate_parse_push_buffer (GstKateParse * parse, GstBuffer * buf,
                            gint64 granulepos)
{
    GST_LOG_OBJECT (parse, "granulepos %16" G_GINT64_MODIFIER "x", granulepos);
    if (granulepos < 0) {
        /* packets coming not from Ogg won't have a granpos in the offset end,
           so we have to synthesize one here - only problem is we don't know
           the backlink - pretend there's none for now */
        GST_INFO_OBJECT (parse, "No granulepos on buffer, synthesizing one");
        granulepos =
            kate_duration_granule (&parse->ki,
                                   GST_BUFFER_TIMESTAMP (buf) /
                                   (double) GST_SECOND) << kate_granule_shift (&parse->ki);
    }
    GST_BUFFER_OFFSET (buf) =
        kate_granule_time (&parse->ki, granulepos) * GST_SECOND;
    GST_BUFFER_OFFSET_END (buf) = granulepos;
    GST_BUFFER_TIMESTAMP (buf) = GST_BUFFER_OFFSET (buf);

    gst_buffer_set_caps (buf, GST_PAD_CAPS (parse->srcpad));

    return gst_pad_push (parse->srcpad, buf);
}
开发者ID:neduartes,项目名称:gst-plugins-bad,代码行数:24,代码来源:gstkateparse.c


示例4: send_buffers_before

static GstFlowReturn
send_buffers_before(AudioTrim *filter, gint64 before)
{
  GstFlowReturn ret = GST_FLOW_OK;
  GList *b = filter->buffers;
  while(b) {
    GstBuffer *buf = b->data;
    b->data = NULL;
    if ((gint64)GST_BUFFER_OFFSET(buf) >= before) {
      gst_buffer_unref(buf);
    } else {
      if ((gint64)GST_BUFFER_OFFSET_END(buf) > before) {
	GstBuffer *head = buffer_head(filter, buf, before);
	gst_buffer_unref(buf);
	buf = head;
      }
      ret = gst_pad_push(filter->srcpad, buf);
      if (ret != GST_FLOW_OK) break;
    }
    b = g_list_next(b);
  }
  release_buffers(filter);
  return ret;
}
开发者ID:fluffware,项目名称:subrec,代码行数:24,代码来源:audiotrim.c


示例5: send_buffers_after

static GstFlowReturn
send_buffers_after(AudioTrim *filter, gint64 after)
{
  GstFlowReturn ret = GST_FLOW_OK;
  GList *b = filter->buffers;
  while(b) {
    GstBuffer *buf = b->data;
    b->data = NULL;
    if ((gint64)GST_BUFFER_OFFSET_END(buf) <= after) {
      gst_buffer_unref(buf);
    } else {
      if ((gint64)GST_BUFFER_OFFSET(buf) < after) {
	GstBuffer *tail = buffer_tail(filter, buf, after);
	gst_buffer_unref(buf);
	buf = tail;
      }
      ret = gst_pad_push(filter->srcpad, buf);
      if (ret != GST_FLOW_OK) break;
    }
    b = g_list_next(b);
  }
  release_buffers(filter);
  return ret;
}
开发者ID:fluffware,项目名称:subrec,代码行数:24,代码来源:audiotrim.c


示例6: getTimeStamp

void ofxGstRTPServer::newOscMsg(ofxOscMessage & msg, GstClockTime timestamp){
	if(!appSrcOsc) return;

	GstClockTime now = timestamp;
	if(!oscAutoTimestamp){
		if(now==GST_CLOCK_TIME_NONE){
			now = getTimeStamp();
		}

		if(firstOscFrame){
			prevTimestampOsc = now;
			firstOscFrame = false;
			return;
		}
	}

	PooledOscPacket * pooledOscPkg = oscPacketPool.newBuffer();
	appendMessage(msg,pooledOscPkg->packet);

	GstBuffer * buffer = gst_buffer_new_wrapped_full(GST_MEMORY_FLAG_READONLY,(void*)pooledOscPkg->compressedData(),pooledOscPkg->compressedSize(),0,pooledOscPkg->compressedSize(),pooledOscPkg,(GDestroyNotify)&ofxOscPacketPool::relaseBuffer);


	if(!oscAutoTimestamp){
		GST_BUFFER_OFFSET(buffer) = numFrameOsc++;
		GST_BUFFER_OFFSET_END(buffer) = numFrameOsc;
		GST_BUFFER_DTS (buffer) = now;
		GST_BUFFER_PTS (buffer) = now;
		GST_BUFFER_DURATION(buffer) = now-prevTimestampOsc;
		prevTimestampOsc = now;
	}

	GstFlowReturn flow_return = gst_app_src_push_buffer((GstAppSrc*)appSrcOsc, buffer);
	if (flow_return != GST_FLOW_OK) {
		ofLogError() << "error pushing osc buffer: flow_return was " << flow_return;
	}
}
开发者ID:anchowee,项目名称:ofxGstRTP,代码行数:36,代码来源:ofxGstRTPServer.cpp


示例7: gst_audio_fx_base_fir_filter_push_residue

void
gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
{
  GstBuffer *outbuf;
  GstFlowReturn res;
  gint rate = GST_AUDIO_FILTER_RATE (self);
  gint channels = GST_AUDIO_FILTER_CHANNELS (self);
  gint bps = GST_AUDIO_FILTER_BPS (self);
  gint outsize, outsamples;
  GstMapInfo map;
  guint8 *in, *out;

  if (channels == 0 || rate == 0 || self->nsamples_in == 0) {
    self->buffer_fill = 0;
    g_free (self->buffer);
    self->buffer = NULL;
    return;
  }

  /* Calculate the number of samples and their memory size that
   * should be pushed from the residue */
  outsamples = self->nsamples_in - (self->nsamples_out - self->latency);
  if (outsamples <= 0) {
    self->buffer_fill = 0;
    g_free (self->buffer);
    self->buffer = NULL;
    return;
  }
  outsize = outsamples * channels * bps;

  if (!self->fft || self->low_latency) {
    gint64 diffsize, diffsamples;

    /* Process the difference between latency and residue length samples
     * to start at the actual data instead of starting at the zeros before
     * when we only got one buffer smaller than latency */
    diffsamples =
        ((gint64) self->latency) - ((gint64) self->buffer_fill) / channels;
    if (diffsamples > 0) {
      diffsize = diffsamples * channels * bps;
      in = g_new0 (guint8, diffsize);
      out = g_new0 (guint8, diffsize);
      self->nsamples_out += self->process (self, in, out, diffsamples);
      g_free (in);
      g_free (out);
    }

    outbuf = gst_buffer_new_and_alloc (outsize);

    /* Convolve the residue with zeros to get the actual remaining data */
    in = g_new0 (guint8, outsize);
    gst_buffer_map (outbuf, &map, GST_MAP_READWRITE);
    self->nsamples_out += self->process (self, in, map.data, outsamples);
    gst_buffer_unmap (outbuf, &map);

    g_free (in);
  } else {
    guint gensamples = 0;

    outbuf = gst_buffer_new_and_alloc (outsize);
    gst_buffer_map (outbuf, &map, GST_MAP_READWRITE);

    while (gensamples < outsamples) {
      guint step_insamples = self->block_length - self->buffer_fill;
      guint8 *zeroes = g_new0 (guint8, step_insamples * channels * bps);
      guint8 *out = g_new (guint8, self->block_length * channels * bps);
      guint step_gensamples;

      step_gensamples = self->process (self, zeroes, out, step_insamples);
      g_free (zeroes);

      memcpy (map.data + gensamples * bps, out, MIN (step_gensamples,
              outsamples - gensamples) * bps);
      gensamples += MIN (step_gensamples, outsamples - gensamples);

      g_free (out);
    }
    self->nsamples_out += gensamples;

    gst_buffer_unmap (outbuf, &map);
  }

  /* Set timestamp, offset, etc from the values we
   * saved when processing the regular buffers */
  if (GST_CLOCK_TIME_IS_VALID (self->start_ts))
    GST_BUFFER_TIMESTAMP (outbuf) = self->start_ts;
  else
    GST_BUFFER_TIMESTAMP (outbuf) = 0;
  GST_BUFFER_TIMESTAMP (outbuf) +=
      gst_util_uint64_scale_int (self->nsamples_out - outsamples -
      self->latency, GST_SECOND, rate);

  GST_BUFFER_DURATION (outbuf) =
      gst_util_uint64_scale_int (outsamples, GST_SECOND, rate);

  if (self->start_off != GST_BUFFER_OFFSET_NONE) {
    GST_BUFFER_OFFSET (outbuf) =
        self->start_off + self->nsamples_out - outsamples - self->latency;
    GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET (outbuf) + outsamples;
  }
//.........这里部分代码省略.........
开发者ID:felipemogollon,项目名称:gst-plugins-good,代码行数:101,代码来源:audiofxbasefirfilter.c


示例8: gst_aiff_parse_stream_data

static GstFlowReturn
gst_aiff_parse_stream_data (GstAiffParse * aiff)
{
  GstBuffer *buf = NULL;
  GstFlowReturn res = GST_FLOW_OK;
  guint64 desired, obtained;
  GstClockTime timestamp, next_timestamp, duration;
  guint64 pos, nextpos;

iterate_adapter:
  GST_LOG_OBJECT (aiff,
      "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
      G_GINT64_FORMAT, aiff->offset, aiff->end_offset, aiff->dataleft);

  /* Get the next n bytes and output them */
  if (aiff->dataleft == 0 || aiff->dataleft < aiff->bytes_per_sample)
    goto found_eos;

  /* scale the amount of data by the segment rate so we get equal
   * amounts of data regardless of the playback rate */
  desired =
      MIN (gst_guint64_to_gdouble (aiff->dataleft),
      MAX_BUFFER_SIZE * aiff->segment.abs_rate);

  if (desired >= aiff->bytes_per_sample && aiff->bytes_per_sample > 0)
    desired -= (desired % aiff->bytes_per_sample);

  GST_LOG_OBJECT (aiff, "Fetching %" G_GINT64_FORMAT " bytes of data "
      "from the sinkpad", desired);

  if (aiff->streaming) {
    guint avail = gst_adapter_available (aiff->adapter);

    if (avail < desired) {
      GST_LOG_OBJECT (aiff, "Got only %d bytes of data from the sinkpad",
          avail);
      return GST_FLOW_OK;
    }

    buf = gst_adapter_take_buffer (aiff->adapter, desired);
  } else {
    if ((res = gst_pad_pull_range (aiff->sinkpad, aiff->offset,
                desired, &buf)) != GST_FLOW_OK)
      goto pull_error;
  }

  /* If we have a pending close/start segment, send it now. */
  if (G_UNLIKELY (aiff->close_segment != NULL)) {
    gst_pad_push_event (aiff->srcpad, aiff->close_segment);
    aiff->close_segment = NULL;
  }
  if (G_UNLIKELY (aiff->start_segment != NULL)) {
    gst_pad_push_event (aiff->srcpad, aiff->start_segment);
    aiff->start_segment = NULL;
  }

  obtained = GST_BUFFER_SIZE (buf);

  /* our positions in bytes */
  pos = aiff->offset - aiff->datastart;
  nextpos = pos + obtained;

  /* update offsets, does not overflow. */
  GST_BUFFER_OFFSET (buf) = pos / aiff->bytes_per_sample;
  GST_BUFFER_OFFSET_END (buf) = nextpos / aiff->bytes_per_sample;

  if (aiff->bps > 0) {
    /* and timestamps if we have a bitrate, be careful for overflows */
    timestamp = uint64_ceiling_scale (pos, GST_SECOND, (guint64) aiff->bps);
    next_timestamp =
        uint64_ceiling_scale (nextpos, GST_SECOND, (guint64) aiff->bps);
    duration = next_timestamp - timestamp;

    /* update current running segment position */
    gst_segment_set_last_stop (&aiff->segment, GST_FORMAT_TIME, next_timestamp);
  } else {
    /* no bitrate, all we know is that the first sample has timestamp 0, all
     * other positions and durations have unknown timestamp. */
    if (pos == 0)
      timestamp = 0;
    else
      timestamp = GST_CLOCK_TIME_NONE;
    duration = GST_CLOCK_TIME_NONE;
    /* update current running segment position with byte offset */
    gst_segment_set_last_stop (&aiff->segment, GST_FORMAT_BYTES, nextpos);
  }
  if (aiff->discont) {
    GST_DEBUG_OBJECT (aiff, "marking DISCONT");
    GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
    aiff->discont = FALSE;
  }

  GST_BUFFER_TIMESTAMP (buf) = timestamp;
  GST_BUFFER_DURATION (buf) = duration;
  gst_buffer_set_caps (buf, aiff->caps);

  GST_LOG_OBJECT (aiff,
      "Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
      ", size:%u", GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration),
      GST_BUFFER_SIZE (buf));
//.........这里部分代码省略.........
开发者ID:PeterXu,项目名称:gst-mobile,代码行数:101,代码来源:aiffparse.c


示例9: gst_split_file_src_create

static GstFlowReturn
gst_split_file_src_create (GstBaseSrc * basesrc, guint64 offset, guint size,
    GstBuffer ** buffer)
{
  GstSplitFileSrc *src = GST_SPLIT_FILE_SRC (basesrc);
  GstFilePart cur_part;
  GInputStream *stream;
  GCancellable *cancel;
  GSeekable *seekable;
  GstBuffer *buf;
  GError *err = NULL;
  guint64 read_offset;
  guint8 *data;
  guint to_read;

  cur_part = src->parts[src->cur_part];
  if (offset < cur_part.start || offset > cur_part.stop) {
    if (!gst_split_file_src_find_part_for_offset (src, offset, &src->cur_part))
      return GST_FLOW_UNEXPECTED;
    cur_part = src->parts[src->cur_part];
  }

  GST_LOG_OBJECT (src, "current part: %u (%" G_GUINT64_FORMAT " - "
      "%" G_GUINT64_FORMAT ", %s)", src->cur_part, cur_part.start,
      cur_part.stop, cur_part.path);

  buf = gst_buffer_new_and_alloc (size);

  GST_BUFFER_OFFSET (buf) = offset;

  data = GST_BUFFER_DATA (buf);

  cancel = src->cancellable;

  while (size > 0) {
    guint64 bytes_to_end_of_part;
    gsize read = 0;

    /* we want the offset into the file part */
    read_offset = offset - cur_part.start;

    GST_LOG ("Reading part %03u from offset %" G_GUINT64_FORMAT " (%s)",
        src->cur_part, read_offset, cur_part.path);

    /* FIXME: only seek when needed (hopefully gio is smart) */
    seekable = G_SEEKABLE (cur_part.stream);
    if (!g_seekable_seek (seekable, read_offset, G_SEEK_SET, cancel, &err))
      goto seek_failed;

    GST_LOG_OBJECT (src, "now: %" G_GUINT64_FORMAT, g_seekable_tell (seekable));

    bytes_to_end_of_part = (cur_part.stop - cur_part.start) + 1 - read_offset;
    to_read = MIN (size, bytes_to_end_of_part);

    GST_LOG_OBJECT (src, "reading %u bytes from part %u (bytes to end of "
        "part: %u)", to_read, src->cur_part, (guint) bytes_to_end_of_part);

    stream = G_INPUT_STREAM (cur_part.stream);

    /* NB: we won't try to read beyond EOF */
    if (!g_input_stream_read_all (stream, data, to_read, &read, cancel, &err))
      goto read_failed;

    GST_LOG_OBJECT (src, "read %u bytes", (guint) read);

    data += read;
    size -= read;
    offset += read;

    /* are we done? */
    if (size == 0)
      break;

    GST_LOG_OBJECT (src, "%u bytes left to read for this chunk", size);

    /* corner case, this should never really happen (assuming basesrc clips
     * requests beyond the file size) */
    if (read < to_read) {
      if (src->cur_part == src->num_parts - 1) {
        /* last file part, stop reading and truncate buffer */
        GST_BUFFER_SIZE (buf) = offset - GST_BUFFER_OFFSET (buf);
        break;
      } else {
        goto file_part_changed;
      }
    }

    ++src->cur_part;
    cur_part = src->parts[src->cur_part];
  }

  GST_BUFFER_OFFSET_END (buf) = offset;

  *buffer = buf;
  GST_LOG_OBJECT (src, "read %u bytes into buf %p", GST_BUFFER_SIZE (buf), buf);
  return GST_FLOW_OK;

/* ERRORS */
seek_failed:
  {
//.........这里部分代码省略.........
开发者ID:TheBigW,项目名称:gst-plugins-good,代码行数:101,代码来源:gstsplitfilesrc.c


示例10: gst_identity_transform_ip

static GstFlowReturn
gst_identity_transform_ip (GstBaseTransform * trans, GstBuffer * buf)
{
  GstFlowReturn ret = GST_FLOW_OK;
  GstIdentity *identity = GST_IDENTITY (trans);
  GstClockTime runtimestamp = G_GINT64_CONSTANT (0);
  gsize size;

  size = gst_buffer_get_size (buf);

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

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

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

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

  if (identity->dump) {
    GstMapInfo info;

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

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

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

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

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

  if (trans->segment.format == GST_FORMAT_TIME)
    runtimestamp = gst_segment_to_running_time (&trans->segment,
        GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (buf));

  if ((identity->sync) && (trans->segment.format == GST_FORMAT_TIME)) {
    GstClock *clock;

    GST_OBJECT_LOCK (identity);
    if ((clock = GST_ELEMENT (identity)->clock)) {
      GstClockReturn cret;
      GstClockTime timestamp;

      timestamp = runtimestamp + GST_ELEMENT (identity)->base_time;

      /* save id if we need to unlock */
      identity->clock_id = gst_clock_new_single_shot_id (clock, timestamp);
      GST_OBJECT_UNLOCK (identity);

      cret = gst_clock_id_wait (identity->clock_id, NULL);

      GST_OBJECT_LOCK (identity);
      if (identity->clock_id) {
        gst_clock_id_unref (identity->clock_id);
        identity->clock_id = NULL;
      }
      if (cret == GST_CLOCK_UNSCHEDULED)
        ret = GST_FLOW_EOS;
    }
    GST_OBJECT_UNLOCK (identity);
  }

  identity->offset += size;

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

  if (identity->single_segment && (trans->segment.format == GST_FORMAT_TIME)
      && (ret == GST_FLOW_OK)) {
    GST_BUFFER_TIMESTAMP (buf) = runtimestamp;
    GST_BUFFER_OFFSET (buf) = GST_CLOCK_TIME_NONE;
    GST_BUFFER_OFFSET_END (buf) = GST_CLOCK_TIME_NONE;
  }

  return ret;

//.........这里部分代码省略.........
开发者ID:PeterXu,项目名称:gst-mobile,代码行数:101,代码来源:gstidentity.c


示例11: gst_ks_video_src_timestamp_buffer

static gboolean
gst_ks_video_src_timestamp_buffer (GstKsVideoSrc * self, GstBuffer * buf,
    GstClockTime presentation_time)
{
  GstKsVideoSrcPrivate *priv = GST_KS_VIDEO_SRC_GET_PRIVATE (self);
  GstClockTime duration;
  GstClock *clock;
  GstClockTime timestamp;

  duration = gst_ks_video_device_get_duration (priv->device);

  GST_OBJECT_LOCK (self);
  clock = GST_ELEMENT_CLOCK (self);
  if (clock != NULL) {
    gst_object_ref (clock);
    timestamp = GST_ELEMENT (self)->base_time;

    if (GST_CLOCK_TIME_IS_VALID (presentation_time)) {
      if (presentation_time > GST_ELEMENT (self)->base_time)
        presentation_time -= GST_ELEMENT (self)->base_time;
      else
        presentation_time = 0;
    }
  } else {
    timestamp = GST_CLOCK_TIME_NONE;
  }
  GST_OBJECT_UNLOCK (self);

  if (clock != NULL) {

    /* The time according to the current clock */
    timestamp = gst_clock_get_time (clock) - timestamp;
    if (timestamp > duration)
      timestamp -= duration;
    else
      timestamp = 0;

    if (GST_CLOCK_TIME_IS_VALID (presentation_time)) {
      /*
       * We don't use this for anything yet, need to ponder how to deal
       * with pins that use an internal clock and timestamp from 0.
       */
      GstClockTimeDiff diff = GST_CLOCK_DIFF (presentation_time, timestamp);
      GST_DEBUG_OBJECT (self, "diff between gst and driver timestamp: %"
          G_GINT64_FORMAT, diff);
    }

    gst_object_unref (clock);
    clock = NULL;

    /* Unless it's the first frame, align the current timestamp on a multiple
     * of duration since the previous */
    if (GST_CLOCK_TIME_IS_VALID (priv->prev_ts)) {
      GstClockTime delta;
      guint delta_remainder, delta_offset;

      /* REVISIT: I've seen this happen with the GstSystemClock on Windows,
       *          scary... */
      if (timestamp < priv->prev_ts) {
        GST_INFO_OBJECT (self, "clock is ticking backwards");
        return FALSE;
      }

      /* Round to a duration boundary */
      delta = timestamp - priv->prev_ts;
      delta_remainder = delta % duration;

      if (delta_remainder < duration / 3)
        timestamp -= delta_remainder;
      else
        timestamp += duration - delta_remainder;

      /* How many frames are we off then? */
      delta = timestamp - priv->prev_ts;
      delta_offset = delta / duration;

      if (delta_offset == 1)    /* perfect */
        GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
      else if (delta_offset > 1) {
        guint lost = delta_offset - 1;
        GST_INFO_OBJECT (self, "lost %d frame%s, setting discont flag",
            lost, (lost > 1) ? "s" : "");
        GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
      } else if (delta_offset == 0) {   /* overproduction, skip this frame */
        GST_INFO_OBJECT (self, "skipping frame");
        return FALSE;
      }

      priv->offset += delta_offset;
    }

    priv->prev_ts = timestamp;
  }

  GST_BUFFER_OFFSET (buf) = priv->offset;
  GST_BUFFER_OFFSET_END (buf) = GST_BUFFER_OFFSET (buf) + 1;
  GST_BUFFER_TIMESTAMP (buf) = timestamp;
  GST_BUFFER_DURATION (buf) = duration;

  return TRUE;
//.........这里部分代码省略.........
开发者ID:dylansong77,项目名称:gstreamer,代码行数:101,代码来源:gstksvideosrc.c


示例12: gst_base_video_decoder_finish_frame


//.........这里部分代码省略.........
        gst_base_video_decoder_get_field_duration (base_video_decoder,
        frame->n_fields);
  }

  if (GST_CLOCK_TIME_IS_VALID (base_video_decoder->last_timestamp)) {
    if (frame->presentation_timestamp < base_video_decoder->last_timestamp) {
      GST_WARNING ("decreasing timestamp (%" GST_TIME_FORMAT " < %"
          GST_TIME_FORMAT ")", GST_TIME_ARGS (frame->presentation_timestamp),
          GST_TIME_ARGS (base_video_decoder->last_timestamp));
    }
  }
  base_video_decoder->last_timestamp = frame->presentation_timestamp;

  src_buffer = frame->src_buffer;

  GST_BUFFER_FLAG_UNSET (src_buffer, GST_BUFFER_FLAG_DELTA_UNIT);
  if (base_video_decoder->state.interlaced) {
#ifndef GST_VIDEO_BUFFER_TFF
#define GST_VIDEO_BUFFER_TFF (GST_MINI_OBJECT_FLAG_LAST << 5)
#endif
#ifndef GST_VIDEO_BUFFER_RFF
#define GST_VIDEO_BUFFER_RFF (GST_MINI_OBJECT_FLAG_LAST << 6)
#endif
#ifndef GST_VIDEO_BUFFER_ONEFIELD
#define GST_VIDEO_BUFFER_ONEFIELD (GST_MINI_OBJECT_FLAG_LAST << 7)
#endif

    if (GST_VIDEO_FRAME_FLAG_IS_SET (frame, GST_VIDEO_FRAME_FLAG_TFF)) {
      GST_BUFFER_FLAG_SET (src_buffer, GST_VIDEO_BUFFER_TFF);
    } else {
      GST_BUFFER_FLAG_UNSET (src_buffer, GST_VIDEO_BUFFER_TFF);
    }
    GST_BUFFER_FLAG_UNSET (src_buffer, GST_VIDEO_BUFFER_RFF);
    GST_BUFFER_FLAG_UNSET (src_buffer, GST_VIDEO_BUFFER_ONEFIELD);
    if (frame->n_fields == 3) {
      GST_BUFFER_FLAG_SET (src_buffer, GST_VIDEO_BUFFER_RFF);
    } else if (frame->n_fields == 1) {
      GST_BUFFER_FLAG_SET (src_buffer, GST_VIDEO_BUFFER_ONEFIELD);
    }
  }
  if (base_video_decoder->discont) {
    GST_BUFFER_FLAG_UNSET (src_buffer, GST_BUFFER_FLAG_DISCONT);
    base_video_decoder->discont = FALSE;
  }

  GST_BUFFER_TIMESTAMP (src_buffer) = frame->presentation_timestamp;
  GST_BUFFER_DURATION (src_buffer) = frame->presentation_duration;
  GST_BUFFER_OFFSET (src_buffer) = GST_BUFFER_OFFSET_NONE;
  GST_BUFFER_OFFSET_END (src_buffer) = GST_BUFFER_OFFSET_NONE;

  GST_DEBUG ("pushing frame %" GST_TIME_FORMAT,
      GST_TIME_ARGS (frame->presentation_timestamp));

  gst_base_video_decoder_set_src_caps (base_video_decoder);

  if (base_video_decoder->sink_clipping) {
    gint64 start = GST_BUFFER_TIMESTAMP (src_buffer);
    gint64 stop = GST_BUFFER_TIMESTAMP (src_buffer) +
        GST_BUFFER_DURATION (src_buffer);

    if (gst_segment_clip (&base_video_decoder->segment, GST_FORMAT_TIME,
            start, stop, &start, &stop)) {
      GST_BUFFER_TIMESTAMP (src_buffer) = start;
      GST_BUFFER_DURATION (src_buffer) = stop - start;
      GST_DEBUG ("accepting buffer inside segment: %" GST_TIME_FORMAT
          " %" GST_TIME_FORMAT
          " seg %" GST_TIME_FORMAT " to %" GST_TIME_FORMAT
          " time %" GST_TIME_FORMAT,
          GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (src_buffer)),
          GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (src_buffer) +
              GST_BUFFER_DURATION (src_buffer)),
          GST_TIME_ARGS (base_video_decoder->segment.start),
          GST_TIME_ARGS (base_video_decoder->segment.stop),
          GST_TIME_ARGS (base_video_decoder->segment.time));
    } else {
      GST_DEBUG ("dropping buffer outside segment: %" GST_TIME_FORMAT
          " %" GST_TIME_FORMAT
          " seg %" GST_TIME_FORMAT " to %" GST_TIME_FORMAT
          " time %" GST_TIME_FORMAT,
          GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (src_buffer)),
          GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (src_buffer) +
              GST_BUFFER_DURATION (src_buffer)),
          GST_TIME_ARGS (base_video_decoder->segment.start),
          GST_TIME_ARGS (base_video_decoder->segment.stop),
          GST_TIME_ARGS (base_video_decoder->segment.time));
      gst_video_frame_unref (frame);
      return GST_FLOW_OK;
    }
  }

  gst_buffer_ref (src_buffer);
  gst_video_frame_unref (frame);

  if (base_video_decoder_class->shape_output)
    return base_video_decoder_class->shape_output (base_video_decoder,
        src_buffer);

  return gst_pad_push (GST_BASE_VIDEO_DECODER_SRC_PAD (base_video_decoder),
      src_buffer);
}
开发者ID:spunktsch,项目名称:svtplayer,代码行数:101,代码来源:gstbasevideodecoder.c


示例13: gst_file_src_create_read

static GstFlowReturn
gst_file_src_create_read (GstFileSrc * src, guint64 offset, guint length,
    GstBuffer ** buffer)
{
  int ret;
  GstBuffer *buf;

  if (G_UNLIKELY (src->read_position != offset)) {
    off_t res;

    res = lseek (src->fd, offset, SEEK_SET);
    if (G_UNLIKELY (res < 0 || res != offset))
      goto seek_failed;

    src->read_position = offset;
  }

  buf = gst_buffer_try_new_and_alloc (length);
  if (G_UNLIKELY (buf == NULL && length > 0)) {
    GST_ERROR_OBJECT (src, "Failed to allocate %u bytes", length);
    return GST_FLOW_ERROR;
  }

  /* No need to read anything if length is 0 */
  if (length > 0) {
    GST_LOG_OBJECT (src, "Reading %d bytes at offset 0x%" G_GINT64_MODIFIER "x",
        length, offset);
    ret = read (src->fd, GST_BUFFER_DATA (buf), length);
    if (G_UNLIKELY (ret < 0))
      goto could_not_read;

    /* seekable regular files should have given us what we expected */
    if (G_UNLIKELY ((guint) ret < length && src->seekable))
      goto unexpected_eos;

    /* other files should eos if they read 0 and more was requested */
    if (G_UNLIKELY (ret == 0 && length > 0))
      goto eos;

    length = ret;
    GST_BUFFER_SIZE (buf) = length;
    GST_BUFFER_OFFSET (buf) = offset;
    GST_BUFFER_OFFSET_END (buf) = offset + length;

    src->read_position += length;
  }

  *buffer = buf;

  return GST_FLOW_OK;

  /* ERROR */
seek_failed:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), GST_ERROR_SYSTEM);
    return GST_FLOW_ERROR;
  }
could_not_read:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), GST_ERROR_SYSTEM);
    gst_buffer_unref (buf);
    return GST_FLOW_ERROR;
  }
unexpected_eos:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
        ("unexpected end of file."));
    gst_buffer_unref (buf);
    return GST_FLOW_ERROR;
  }
eos:
  {
    GST_DEBUG ("non-regular file hits EOS");
    gst_buffer_unref (buf);
    return GST_FLOW_UNEXPECTED;
  }
}
开发者ID:puce77,项目名称:openjfx-8u-dev-rt,代码行数:77,代码来源:gstfilesrc.c


示例14: gst_rtp_dtmf_depay_process


//.........这里部分代码省略.........

  /* clip to whole units of unit_time */
  if (rtpdtmfdepay->unit_time) {
    guint unit_time_clock =
        (rtpdtmfdepay->unit_time * depayload->clock_rate) / 1000;
    if (dtmf_payload.duration % unit_time_clock) {
      /* Make sure we don't overflow the duration */
      if (dtmf_payload.duration < G_MAXUINT16 - unit_time_clock)
        dtmf_payload.duration += unit_time_clock -
            (dtmf_payload.duration % unit_time_clock);
      else
        dtmf_payload.duration -= dtmf_payload.duration % unit_time_clock;
    }
  }

  /* clip to max duration */
  if (rtpdtmfdepay->max_duration) {
    guint max_duration_clock =
        (rtpdtmfdepay->max_duration * depayload->clock_rate) / 1000;

    if (max_duration_clock < G_MAXUINT16 &&
        dtmf_payload.duration > max_duration_clock)
      dtmf_payload.duration = max_duration_clock;
  }

  GST_DEBUG_OBJECT (depayload, "Received new RTP DTMF packet : "
      "marker=%d - timestamp=%u - event=%d - duration=%d",
      marker, timestamp, dtmf_payload.event, dtmf_payload.duration);

  GST_DEBUG_OBJECT (depayload,
      "Previous information : timestamp=%u - duration=%d",
      rtpdtmfdepay->previous_ts, rtpdtmfdepay->previous_duration);

  /* First packet */
  if (marker || rtpdtmfdepay->previous_ts != timestamp) {
    rtpdtmfdepay->sample = 0;
    rtpdtmfdepay->previous_ts = timestamp;
    rtpdtmfdepay->previous_duration = dtmf_payload.duration;
    rtpdtmfdepay->first_gst_ts = GST_BUFFER_TIMESTAMP (buf);

    structure = gst_structure_new ("dtmf-event",
        "number", G_TYPE_INT, dtmf_payload.event,
        "volume", G_TYPE_INT, dtmf_payload.volume,
        "type", G_TYPE_INT, 1, "method", G_TYPE_INT, 1, NULL);
    if (structure) {
      dtmf_message =
          gst_message_new_element (GST_OBJECT (depayload), structure);
      if (dtmf_message) {
        if (!gst_element_post_message (GST_ELEMENT (depayload), dtmf_message)) {
          GST_ERROR_OBJECT (depayload,
              "Unable to send dtmf-event message to bus");
        }
      } else {
        GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event message");
      }
    } else {
      GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event structure");
    }
  } else {
    guint16 duration = dtmf_payload.duration;
    dtmf_payload.duration -= rtpdtmfdepay->previous_duration;
    /* If late buffer, ignore */
    if (duration > rtpdtmfdepay->previous_duration)
      rtpdtmfdepay->previous_duration = duration;
  }

  GST_DEBUG_OBJECT (depayload, "new previous duration : %d - new duration : %d"
      " - diff  : %d - clock rate : %d - timestamp : %llu",
      rtpdtmfdepay->previous_duration, dtmf_payload.duration,
      (rtpdtmfdepay->previous_duration - dtmf_payload.duration),
      depayload->clock_rate, GST_BUFFER_TIMESTAMP (buf));

  /* If late or duplicate packet (like the redundant end packet). Ignore */
  if (dtmf_payload.duration > 0) {
    outbuf = gst_buffer_new ();
    gst_dtmf_src_generate_tone (rtpdtmfdepay, dtmf_payload, outbuf);


    GST_BUFFER_TIMESTAMP (outbuf) = rtpdtmfdepay->first_gst_ts +
        (rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
        GST_SECOND / depayload->clock_rate;
    GST_BUFFER_OFFSET (outbuf) =
        (rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
        GST_SECOND / depayload->clock_rate;
    GST_BUFFER_OFFSET_END (outbuf) = rtpdtmfdepay->previous_duration *
        GST_SECOND / depayload->clock_rate;

    GST_DEBUG_OBJECT (depayload, "timestamp : %llu - time %" GST_TIME_FORMAT,
        GST_BUFFER_TIMESTAMP (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));

  }

  return outbuf;


bad_packet:
  GST_ELEMENT_WARNING (rtpdtmfdepay, STREAM, DECODE,
      ("Packet did not validate"), (NULL));
  return NULL;
}
开发者ID:bilboed,项目名称:gst-plugins-bad,代码行数:101,代码来源:gstrtpdtmfdepay.c


示例15: gst_fake_src_create

static GstFlowReturn
gst_fake_src_create (GstBaseSrc * basesrc, guint64 offset, guint length,
    GstBuffer ** ret)
{
  GstFakeSrc *src;
  GstBuffer *buf;
  GstClockTime time;
  gsize size;

  src = GST_FAKE_SRC (basesrc);

  buf = gst_fake_src_create_buffer (src, &size);
  GST_BUFFER_OFFSET (buf) = offset;

  if (src->datarate > 0) {
    time = (src->bytes_sent * GST_SECOND) / src->datarate;

    GST_BUFFER_DURATION (buf) = size * GST_SECOND / src->datarate;
  } else if (gst_base_src_is_live (basesrc)) {
    GstClock *clock;

    clock = gst_element_get_clock (GST_ELEMENT (src));

    if (clock) {
      time = gst_clock_get_time (clock);
      time -= gst_element_get_base_time (GST_ELEMENT (src));
      gst_object_unref (clock);
    } else {
      /* not an error not to have a clock */
      time = GST_CLOCK_TIME_NONE;
    }
  } else {
    time = GST_CLOCK_TIME_NONE;
  }

  GST_BUFFER_DTS (buf) = time;
  GST_BUFFER_PTS (buf) = time;

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

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

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

    flag_str = gst_buffer_get_flags_string (buf);
    src->last_message =
        g_strdup_printf ("create   ******* (%s:%s) (%u bytes, dts: %s, pts:%s"
        ", duration: %s, offset: %" G_GINT64_FORMAT ", offset_end: %"
        G_GINT64_FORMAT ", flags: %08x %s) %p",
        GST_DEBUG_PAD_NAME (GST_BASE_SRC_CAST (src)->srcpad), (guint) size,
        dts_str, pts_str, dur_str, GST_BUFFER_OFFSET (buf),
        GST_BUFFER_OFFSET_END (buf), GST_MINI_OBJECT_CAST (buf)->flags,
        flag_str, buf);
    g_free (flag_str);
    GST_OBJECT_UNLOCK (src);

    g_object_notify_by_pspec ((GObject *) src, pspec_last_message);
  }

  if (src->signal_handoffs) {
    GST_LOG_OBJECT (src, "pre handoff emit");
    g_signal_emit (src, gst_fake_src_signals[SIGNAL_HANDOFF], 0, buf,
        basesrc->srcpad);
    GST_LOG_OBJECT (src, "post handoff emit");
  }

  src->bytes_sent += size;

  *ret = buf;
  return GST_FLOW_OK;
}
开发者ID:MathieuDuponchelle,项目名称:gstreamer,代码行数:91,代码来源:gstfakesrc.c


示例16: gst_cmml_enc_parse_tag_head

/* encode the CMML head tag and push the CMML headers
 */
static void
gst_cmml_enc_parse_tag_head (GstCmmlEnc * enc, GstCmmlTagHead * head)
{
  GList *headers = NULL;
  GList *walk;
  guchar *head_string;
  GstCaps *caps;
  GstBuffer *ident_buf, *preamble_buf, *head_buf;
  GstBuffer *buffer;

  if (enc->preamble == NULL)
    goto flow_unexpected;

  GST_INFO_OBJECT (enc, "parsing head tag");

  enc->flow_return = gst_cmml_enc_new_ident_header (enc, &ident_buf);
  if (enc->flow_return != GST_FLOW_OK)
    goto alloc_error;
  headers = g_list_append (headers, ident_buf);

  enc->flow_return = gst_cmml_enc_new_buffer (enc,
      enc->preamble, strlen ((gchar *) enc->preamble), &preamble_buf);
  if (enc->flow_return != GST_FLOW_OK)
    goto alloc_error;
  headers = g_list_append (headers, preamble_buf);

  head_string = gst_cmml_parser_tag_head_to_string (enc->parser, head);
  enc->flow_return = gst_cmml_enc_new_buffer (enc,
      head_string, strlen ((gchar *) head_string), &head_buf);
  g_free (head_string);
  if (enc->flow_return != GST_FLOW_OK)
    goto alloc_error;
  headers = g_list_append (headers, head_buf);

  caps = gst_pad_get_caps (enc->srcpad);
  caps = gst_cmml_enc_set_header_on_caps (enc, caps,
      ident_buf, preamble_buf, head_buf);

  while (headers) {
    buffer = GST_BUFFER (headers->data);
    /* set granulepos 0 on headers */
    GST_BUFFER_OFFSET_END (buffer) = 0;
    gst_buffer_set_caps (buffer, caps);

    enc->flow_return = gst_cmml_enc_push (enc, buffer);
    headers = g_list_delete_link (headers, headers);

    if (GST_FLOW_IS_FATAL (enc->flow_return))
      goto push_error;
  }

  gst_caps_unref (caps);

  enc->sent_headers = TRUE;
  return;

flow_unexpected:
  GST_ELEMENT_ERROR (enc, STREAM, ENCODE,
      (NULL), ("got head tag before preamble"));
  enc->flow_return = GST_FLOW_ERROR;
  return;
push_error:
  gst_caps_unref (caps);
  /* fallthrough */
alloc_error:
  for (walk = headers; walk; walk = walk->next)
    gst_buffer_unref (GST_BUFFER (walk->data));
  g_list_free (headers);
  return;
}
开发者ID:rosciio,项目名称:gst-plugins-good,代码行数:72,代码来源:gstcmmlenc.c


示例17: gst_gsmdec_chain

static GstFlowReturn
gst_gsmdec_chain (GstPad * pad, GstBuffer * buf)
{
    GstGSMDec *gsmdec;
    gsm_byte *data;
    GstFlowReturn ret = GST_FLOW_OK;
    GstClockTime timestamp;
    gint needed;

    gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));

    timestamp = GST_BUFFER_TIMESTAMP (buf);

    if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
        gst_adapter_clear (gsmdec->adapter);
        gsmdec->next_ts = GST_CLOCK_TIME_NONE;
        /* FIXME, do some good offset */
        gsmdec->next_of = 0;
    }
    gst_adapter_push (gsmdec->adapter, buf);

    needed = 33;
    /* do we have enough bytes to read a frame */
    while (gst_adapter_available (gsmdec->adapter) >= needed) {
        GstBuffer *outbuf;

        /* always the same amount of output samples */
        outbuf = gst_buffer_new_and_alloc (ENCODED_SAMPLES * sizeof (gsm_signal));

        /* If we are not given any timestamp, interpolate from last seen
         * timestamp (if any). */
        if (timestamp == GST_CLOCK_TIME_NONE)
            timestamp = gsmdec->next_ts;

        GST_BUFFER_TIMESTAMP (outbuf) = timestamp;

        /* interpolate in the next run */
        if (timestamp != GST_CLOCK_TIME_NONE)
            gsmdec->next_ts = timestamp + gsmdec->duration;
        timestamp = GST_CLOCK_TIME_NONE;

        GST_BUFFER_DURATION (outbuf) = gsmdec->duration;
        GST_BUFFER_OFFSET (outbuf) = gsmdec->next_of;
        if (gsmdec->next_of != -1)
            gsmdec->next_of += ENCODED_SAMPLES;
        GST_BUFFER_OFFSET_END (outbuf) = gsmdec->next_of;

        gst_buffer_set_caps (outbuf, GST_PAD_CAPS (gsmdec->srcpad));

        /* now encode frame into the output buffer */
        data = (gsm_byte *) gst_adapter_peek (gsmdec->adapter, needed);
        if (gsm_decode (gsmdec->state, data,
                        (gsm_signal *) GST_BUFFER_DATA (outbuf)) < 0) {
            /* invalid frame */
            GST_WARNING_OBJECT (gsmdec, "tried to decode an invalid frame, skipping");
        }
        gst_adapter_flush (gsmdec->adapter, needed);

        /* WAV49 requires alternating 33 and 32 bytes of input */
        if (gsmdec->use_wav49)
            needed = (needed == 33 ? 32 : 33);

        GST_DEBUG_OBJECT (gsmdec, "Pushing buffer of size %d ts %" GST_TIME_FORMAT,
                          GST_BUFFER_SIZE (outbuf),
                          GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));

        /* push */
        ret = gst_pad_push (gsmdec->srcpad, outbuf);
    }

    gst_object_unref (gsmdec);

    return ret;
}
开发者ID:JJCG,项目名称:gst-plugins-bad,代码行数:74,代码来源:gstgsmdec.c



鲜花

握手

雷人

路过

鸡蛋
该文章已有0人参与评论

请发表评论

全部评论

专题导读
上一篇:
C++ GST_BUFFER_SIZE函数代码示例发布时间:2022-05-30
下一篇:
C++ GST_BUFFER_OFFSET函数代码示例发布时间:2022-05-30
热门推荐
阅读排行榜

扫描微信二维码

查看手机版网站

随时了解更新最新资讯

139-2527-9053

在线客服(服务时间 9:00~18:00)

在线QQ客服
地址:深圳市南山区西丽大学城创智工业园
电邮:jeky_zhao#qq.com
移动电话:139-2527-9053

Powered by 互联科技 X3.4© 2001-2213 极客世界.|Sitemap