static GstFlowReturn
gst_real_audio_demux_chain (GstPad * pad, GstBuffer * buf)
{
GstRealAudioDemux *demux;
demux = GST_REAL_AUDIO_DEMUX (GST_PAD_PARENT (pad));
return gst_real_audio_demux_handle_buffer (demux, buf);
}
static GstFlowReturn
gst_real_audio_demux_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstRealAudioDemux *demux;
demux = GST_REAL_AUDIO_DEMUX (parent);
return gst_real_audio_demux_handle_buffer (demux, buf);
}
static void
gst_real_audio_demux_loop (GstRealAudioDemux * demux)
{
GstFlowReturn ret;
GstBuffer *buf;
guint bytes_needed;
/* check how much data we need */
switch (demux->state) {
case REAL_AUDIO_DEMUX_STATE_MARKER:
bytes_needed = 6 + 16; /* 16 are beginning of header */
break;
case REAL_AUDIO_DEMUX_STATE_HEADER:
if (!gst_real_audio_demux_get_data_offset_from_header (demux))
goto parse_header_error;
bytes_needed = demux->data_offset - (6 + 16);
break;
case REAL_AUDIO_DEMUX_STATE_DATA:
if (demux->packet_size > 0) {
/* TODO: should probably take into account width/height as well? */
bytes_needed = demux->packet_size;
} else {
bytes_needed = 1024;
}
break;
default:
g_return_if_reached ();
}
/* now get the data */
GST_LOG_OBJECT (demux, "getting data: %5u bytes @ %8" G_GINT64_MODIFIER "u",
bytes_needed, demux->offset);
if (demux->upstream_size > 0 && demux->offset >= demux->upstream_size)
goto eos;
ret = gst_pad_pull_range (demux->sinkpad, demux->offset, bytes_needed, &buf);
if (ret != GST_FLOW_OK)
goto pull_range_error;
if (GST_BUFFER_SIZE (buf) != bytes_needed)
goto pull_range_short_read;
ret = gst_real_audio_demux_handle_buffer (demux, buf);
if (ret != GST_FLOW_OK)
goto handle_flow_error;
/* TODO: increase this in chain function too (for timestamps)? */
demux->offset += bytes_needed;
/* check for the end of the segment */
if (demux->segment.stop != -1 && demux->segment.last_stop != -1 &&
demux->segment.last_stop > demux->segment.stop) {
GST_DEBUG_OBJECT (demux, "reached end of segment");
goto eos;
}
return;
/* ERRORS */
parse_header_error:
{
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL), (NULL));
goto pause_task;
}
handle_flow_error:
{
GST_WARNING_OBJECT (demux, "handle_buf flow: %s", gst_flow_get_name (ret));
goto pause_task;
}
pull_range_error:
{
GST_WARNING_OBJECT (demux, "pull range flow: %s", gst_flow_get_name (ret));
goto pause_task;
}
pull_range_short_read:
{
GST_WARNING_OBJECT (demux, "pull range short read: wanted %u bytes, but "
"got only %u bytes", bytes_needed, GST_BUFFER_SIZE (buf));
gst_buffer_unref (buf);
goto eos;
}
eos:
{
if (demux->state != REAL_AUDIO_DEMUX_STATE_DATA) {
GST_WARNING_OBJECT (demux, "reached EOS before finished parsing header");
goto parse_header_error;
}
GST_INFO_OBJECT (demux, "EOS");
if ((demux->segment.flags & GST_SEEK_FLAG_SEGMENT) != 0) {
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 = demux->segment.stop) == -1)
stop = demux->segment.duration;
GST_DEBUG_OBJECT (demux, "sending segment done, at end of segment");
gst_element_post_message (GST_ELEMENT (demux),
gst_message_new_segment_done (GST_OBJECT (demux), GST_FORMAT_TIME,
stop));
} else {
/* normal playback, send EOS event downstream */
GST_DEBUG_OBJECT (demux, "sending EOS event, at end of stream");
gst_pad_push_event (demux->srcpad, gst_event_new_eos ());
}
goto pause_task;
}
pause_task:
{
demux->segment_running = FALSE;
gst_pad_pause_task (demux->sinkpad);
GST_DEBUG_OBJECT (demux, "pausing task");
return;
}
}
