GstRtpMP4GDepayClass * klass)
{
rtpmp4gdepay->adapter = gst_adapter_new ();
+ rtpmp4gdepay->packets = g_queue_new ();
}
static void
g_object_unref (rtpmp4gdepay->adapter);
rtpmp4gdepay->adapter = NULL;
+ g_queue_free (rtpmp4gdepay->packets);
+ rtpmp4gdepay->packets = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
if (strcmp (str, "audio") == 0) {
srccaps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 4, NULL);
+ /* AAC always has a default constant duration of 1024 but it can be
+ * overriden below. */
+ rtpmp4gdepay->constantDuration = 1024;
} else if (strcmp (str, "video") == 0) {
srccaps = gst_caps_new_simple ("video/mpeg",
"mpegversion", G_TYPE_INT, 4,
"systemstream", G_TYPE_BOOLEAN, FALSE, NULL);
+ rtpmp4gdepay->constantDuration = 0;
}
}
if (srccaps == NULL)
gst_rtp_mp4g_depay_parse_int (structure, "streamstateindication", 0);
rtpmp4gdepay->auxiliarydatasizelength =
gst_rtp_mp4g_depay_parse_int (structure, "auxiliarydatasizelength", 0);
+ rtpmp4gdepay->constantSize =
+ gst_rtp_mp4g_depay_parse_int (structure, "constantsize", 0);
+ rtpmp4gdepay->constantDuration =
+ gst_rtp_mp4g_depay_parse_int (structure, "constantduration",
+ rtpmp4gdepay->constantDuration);
+
/* get config string */
if ((str = gst_structure_get_string (structure, "config"))) {
}
}
+static void
+gst_rtp_mp4g_depay_clear_queue (GstRtpMP4GDepay * rtpmp4gdepay)
+{
+ GstBuffer *outbuf;
+
+ while ((outbuf = g_queue_pop_head (rtpmp4gdepay->packets)))
+ gst_buffer_unref (outbuf);
+}
+
+static void
+gst_rtp_mp4g_depay_flush_queue (GstRtpMP4GDepay * rtpmp4gdepay)
+{
+ GstBuffer *outbuf;
+ gboolean discont = FALSE;
+ guint AU_index;
+
+ while ((outbuf = g_queue_pop_head (rtpmp4gdepay->packets))) {
+ AU_index = GST_BUFFER_OFFSET (outbuf);
+
+ GST_DEBUG_OBJECT (rtpmp4gdepay, "next available AU_index %u", AU_index);
+
+ if (rtpmp4gdepay->next_AU_index != AU_index) {
+ GST_DEBUG_OBJECT (rtpmp4gdepay, "discont, expected AU_index %u",
+ rtpmp4gdepay->next_AU_index);
+ discont = TRUE;
+ }
+
+ if (discont) {
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ discont = FALSE;
+ }
+
+ GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing AU_index %u", AU_index);
+ gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpmp4gdepay), outbuf);
+ rtpmp4gdepay->next_AU_index = AU_index + 1;
+ }
+}
+
+static void
+gst_rtp_mp4g_depay_queue (GstRtpMP4GDepay * rtpmp4gdepay, GstBuffer * outbuf)
+{
+ guint AU_index = GST_BUFFER_OFFSET (outbuf);
+
+ if (rtpmp4gdepay->next_AU_index == -1) {
+ GST_DEBUG_OBJECT (rtpmp4gdepay, "Init AU counter %u", AU_index);
+ rtpmp4gdepay->next_AU_index = AU_index;
+ }
+
+ if (rtpmp4gdepay->next_AU_index == AU_index) {
+ GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing expected AU_index %u", AU_index);
+
+ /* we received the expected packet, push it and flush as much as we can from
+ * the queue */
+ gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpmp4gdepay), outbuf);
+ rtpmp4gdepay->next_AU_index++;
+
+ while ((outbuf = g_queue_peek_head (rtpmp4gdepay->packets))) {
+ AU_index = GST_BUFFER_OFFSET (outbuf);
+
+ GST_DEBUG_OBJECT (rtpmp4gdepay, "next available AU_index %u", AU_index);
+
+ if (rtpmp4gdepay->next_AU_index == AU_index) {
+ GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing expected AU_index %u",
+ AU_index);
+ outbuf = g_queue_pop_head (rtpmp4gdepay->packets);
+ gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpmp4gdepay),
+ outbuf);
+ rtpmp4gdepay->next_AU_index++;
+ } else {
+ GST_DEBUG_OBJECT (rtpmp4gdepay, "waiting for next AU_index %u",
+ rtpmp4gdepay->next_AU_index);
+ break;
+ }
+ }
+ } else {
+ GList *list;
+
+ GST_DEBUG_OBJECT (rtpmp4gdepay, "queueing AU_index %u", AU_index);
+
+ /* loop the list to skip strictly smaller AU_index buffers */
+ for (list = rtpmp4gdepay->packets->head; list; list = g_list_next (list)) {
+ guint idx;
+ gint gap;
+
+ idx = GST_BUFFER_OFFSET (GST_BUFFER_CAST (list->data));
+
+ /* compare the new seqnum to the one in the buffer */
+ gap = (gint) (idx - AU_index);
+
+ GST_DEBUG_OBJECT (rtpmp4gdepay, "compare with AU_index %u, gap %d", idx,
+ gap);
+
+ /* AU_index <= idx, we can stop looking */
+ if (G_LIKELY (gap > 0))
+ break;
+ }
+ if (G_LIKELY (list))
+ g_queue_insert_before (rtpmp4gdepay->packets, list, outbuf);
+ else
+ g_queue_push_tail (rtpmp4gdepay->packets, outbuf);
+ }
+}
+
static GstBuffer *
gst_rtp_mp4g_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpMP4GDepay *rtpmp4gdepay;
GstBuffer *outbuf;
+ GstClockTime timestamp;
rtpmp4gdepay = GST_RTP_MP4G_DEPAY (depayload);
/* flush remaining data on discont */
if (GST_BUFFER_IS_DISCONT (buf)) {
+ GST_DEBUG_OBJECT (rtpmp4gdepay, "received DISCONT");
gst_adapter_clear (rtpmp4gdepay->adapter);
}
+ timestamp = GST_BUFFER_TIMESTAMP (buf);
+
{
gint payload_len, payload_AU;
guint8 *payload;
- guint32 timestamp;
+ guint32 rtptime;
guint AU_headers_len;
- guint AU_size, AU_index, payload_AU_size;
+ guint AU_size, AU_index, AU_index_delta, payload_AU_size;
gboolean M;
payload_len = gst_rtp_buffer_get_payload_len (buf);
payload = gst_rtp_buffer_get_payload (buf);
- timestamp = gst_rtp_buffer_get_timestamp (buf);
+ rtptime = gst_rtp_buffer_get_timestamp (buf);
M = gst_rtp_buffer_get_marker (buf);
if (rtpmp4gdepay->sizelength > 0) {
/* point the bitstream parser to the first AU header bit */
gst_bs_parse_init (&bs, payload, payload_len);
+ AU_index = AU_index_delta = 0;
for (i = 0; i < num_AU_headers && payload_AU_size > 0; i++) {
/* parse AU header
* +---------------------------------------+
*/
AU_size = gst_bs_parse_read (&bs, rtpmp4gdepay->sizelength);
- if (i == 0)
+
+ /* calculate the AU_index, which is only on the first AU of the packet
+ * and the AU_index_delta on the other AUs. This will be used to
+ * reconstruct the AU ordering when interleaving. */
+ if (i == 0) {
AU_index = gst_bs_parse_read (&bs, rtpmp4gdepay->indexlength);
- else
- AU_index = gst_bs_parse_read (&bs, rtpmp4gdepay->indexdeltalength);
+ if (AU_index == 0 && rtpmp4gdepay->prev_AU_index == 0) {
+ gint diff;
+
+ /* if we see two consecutive packets with AU_index of 0, we can
+ * assume we have constantDuration packets. Since we don't have
+ * the index we must use the AU duration to calculate the
+ * index. Get the diff between the timestamps first, this can be
+ * positive or negative. */
+ if (rtpmp4gdepay->prev_rtptime <= rtptime)
+ diff = rtptime - rtpmp4gdepay->prev_rtptime;
+ else
+ diff = -(rtpmp4gdepay->prev_rtptime - rtptime);
+
+ /* get the number of packets by dividing with the duration */
+ diff /= rtpmp4gdepay->constantDuration;
+
+ rtpmp4gdepay->last_AU_index += diff;
+ rtpmp4gdepay->prev_AU_index = AU_index;
+
+ AU_index = rtpmp4gdepay->last_AU_index;
+
+ } else {
+ rtpmp4gdepay->prev_AU_index = AU_index;
+ rtpmp4gdepay->last_AU_index = AU_index;
+ }
+
+ /* keep track of the higest AU_index */
+ if (rtpmp4gdepay->max_AU_index != -1
+ && rtpmp4gdepay->max_AU_index <= AU_index) {
+ GST_DEBUG_OBJECT (rtpmp4gdepay, "new interleave group, flushing");
+ /* a new interleave group started, flush */
+ gst_rtp_mp4g_depay_flush_queue (rtpmp4gdepay);
+ }
+ rtpmp4gdepay->prev_rtptime = rtptime;
+ } else {
+ AU_index_delta =
+ gst_bs_parse_read (&bs, rtpmp4gdepay->indexdeltalength);
+ AU_index += AU_index_delta + 1;
+ }
+ /* keep track of highest AU_index */
+ if (rtpmp4gdepay->max_AU_index == -1
+ || AU_index > rtpmp4gdepay->max_AU_index)
+ rtpmp4gdepay->max_AU_index = AU_index;
+
+ /* the presentation time offset, a 2s-complement value, we need this to
+ * calculate the timestamp on the output packet. */
if (rtpmp4gdepay->ctsdeltalength > 0) {
if (gst_bs_parse_read (&bs, 1))
gst_bs_parse_read (&bs, rtpmp4gdepay->ctsdeltalength);
}
+ /* the decoding time offset, a 2s-complement value */
if (rtpmp4gdepay->dtsdeltalength > 0) {
if (gst_bs_parse_read (&bs, 1))
gst_bs_parse_read (&bs, rtpmp4gdepay->dtsdeltalength);
}
+ /* RAP-flag to indicate that the AU contains a keyframe */
if (rtpmp4gdepay->randomaccessindication)
gst_bs_parse_read (&bs, 1);
+ /* stream-state */
if (rtpmp4gdepay->streamstateindication > 0)
gst_bs_parse_read (&bs, rtpmp4gdepay->streamstateindication);
- GST_DEBUG_OBJECT (rtpmp4gdepay, "size %d, index %d", AU_size, AU_index);
+ GST_DEBUG_OBJECT (rtpmp4gdepay, "size %d, index %d, delta %d", AU_size,
+ AU_index, AU_index_delta);
/* fragmented pakets have the AU_size set to the size of the
* unfragmented AU. */
outbuf = gst_adapter_take_buffer (rtpmp4gdepay->adapter, avail);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
- GST_DEBUG ("gst_rtp_mp4g_depay_chain: pushing buffer of size %d",
+ /* copy some of the fields we calculated above on the buffer. We also
+ * copy the AU_index so that we can sort the packets in our queue. */
+ GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
+ GST_BUFFER_OFFSET (outbuf) = AU_index;
+
+ /* make sure we don't use the timestamp again for other AUs in this
+ * RTP packet. */
+ timestamp = -1;
+
+ GST_DEBUG_OBJECT (depayload, "pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
- /* only apply the timestamp for the first buffer */
- if (i == 0)
- gst_base_rtp_depayload_push_ts (depayload, timestamp, outbuf);
- else
- gst_base_rtp_depayload_push (depayload, outbuf);
+ gst_rtp_mp4g_depay_queue (rtpmp4gdepay, outbuf);
+
}
payload_AU += AU_size;
payload_AU_size -= AU_size;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_adapter_clear (rtpmp4gdepay->adapter);
+ rtpmp4gdepay->max_AU_index = -1;
+ rtpmp4gdepay->next_AU_index = -1;
+ rtpmp4gdepay->prev_AU_index = -1;
+ rtpmp4gdepay->prev_rtptime = -1;
+ rtpmp4gdepay->last_AU_index = -1;
break;
default:
break;
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ gst_adapter_clear (rtpmp4gdepay->adapter);
+ gst_rtp_mp4g_depay_clear_queue (rtpmp4gdepay);
+ break;
default:
break;
}
projects / platform / upstream / gst-plugins-good.git / commitdiff