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gst/rtp/gstrtpmp4gdepay.*: Handle interleaved streams by reordering AU in a queue.
Original commit message from CVS: * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_init), (gst_rtp_mp4g_depay_finalize), (gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_clear_queue), (gst_rtp_mp4g_depay_flush_queue), (gst_rtp_mp4g_depay_queue), (gst_rtp_mp4g_depay_process), (gst_rtp_mp4g_depay_change_state): * gst/rtp/gstrtpmp4gdepay.h: Handle interleaved streams by reordering AU in a queue.
This commit is contained in:
parent
fe9b449678
commit
1c6a371d0c
3 changed files with 223 additions and 13 deletions
10
ChangeLog
10
ChangeLog
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@ -1,3 +1,13 @@
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2008-09-15 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_init),
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(gst_rtp_mp4g_depay_finalize), (gst_rtp_mp4g_depay_setcaps),
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(gst_rtp_mp4g_depay_clear_queue), (gst_rtp_mp4g_depay_flush_queue),
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(gst_rtp_mp4g_depay_queue), (gst_rtp_mp4g_depay_process),
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(gst_rtp_mp4g_depay_change_state):
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* gst/rtp/gstrtpmp4gdepay.h:
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Handle interleaved streams by reordering AU in a queue.
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2008-09-15 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtp/gstrtpmp4gdepay.c: (gst_bs_parse_init),
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@ -188,6 +188,7 @@ gst_rtp_mp4g_depay_init (GstRtpMP4GDepay * rtpmp4gdepay,
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GstRtpMP4GDepayClass * klass)
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{
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rtpmp4gdepay->adapter = gst_adapter_new ();
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rtpmp4gdepay->packets = g_queue_new ();
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}
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static void
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@ -199,6 +200,8 @@ gst_rtp_mp4g_depay_finalize (GObject * object)
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g_object_unref (rtpmp4gdepay->adapter);
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rtpmp4gdepay->adapter = NULL;
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g_queue_free (rtpmp4gdepay->packets);
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rtpmp4gdepay->packets = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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@ -241,10 +244,14 @@ gst_rtp_mp4g_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
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if (strcmp (str, "audio") == 0) {
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srccaps = gst_caps_new_simple ("audio/mpeg",
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"mpegversion", G_TYPE_INT, 4, NULL);
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/* AAC always has a default constant duration of 1024 but it can be
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* overriden below. */
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rtpmp4gdepay->constantDuration = 1024;
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} else if (strcmp (str, "video") == 0) {
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srccaps = gst_caps_new_simple ("video/mpeg",
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"mpegversion", G_TYPE_INT, 4,
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"systemstream", G_TYPE_BOOLEAN, FALSE, NULL);
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rtpmp4gdepay->constantDuration = 0;
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}
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}
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if (srccaps == NULL)
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@ -268,6 +275,12 @@ gst_rtp_mp4g_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
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gst_rtp_mp4g_depay_parse_int (structure, "streamstateindication", 0);
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rtpmp4gdepay->auxiliarydatasizelength =
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gst_rtp_mp4g_depay_parse_int (structure, "auxiliarydatasizelength", 0);
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rtpmp4gdepay->constantSize =
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gst_rtp_mp4g_depay_parse_int (structure, "constantsize", 0);
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rtpmp4gdepay->constantDuration =
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gst_rtp_mp4g_depay_parse_int (structure, "constantduration",
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rtpmp4gdepay->constantDuration);
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/* get config string */
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if ((str = gst_structure_get_string (structure, "config"))) {
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@ -299,11 +312,115 @@ unknown_media:
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}
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}
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static void
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gst_rtp_mp4g_depay_clear_queue (GstRtpMP4GDepay * rtpmp4gdepay)
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{
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GstBuffer *outbuf;
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while ((outbuf = g_queue_pop_head (rtpmp4gdepay->packets)))
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gst_buffer_unref (outbuf);
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}
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static void
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gst_rtp_mp4g_depay_flush_queue (GstRtpMP4GDepay * rtpmp4gdepay)
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{
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GstBuffer *outbuf;
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gboolean discont = FALSE;
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guint AU_index;
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while ((outbuf = g_queue_pop_head (rtpmp4gdepay->packets))) {
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AU_index = GST_BUFFER_OFFSET (outbuf);
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GST_DEBUG_OBJECT (rtpmp4gdepay, "next available AU_index %u", AU_index);
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if (rtpmp4gdepay->next_AU_index != AU_index) {
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GST_DEBUG_OBJECT (rtpmp4gdepay, "discont, expected AU_index %u",
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rtpmp4gdepay->next_AU_index);
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discont = TRUE;
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}
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if (discont) {
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
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discont = FALSE;
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}
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GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing AU_index %u", AU_index);
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gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpmp4gdepay), outbuf);
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rtpmp4gdepay->next_AU_index = AU_index + 1;
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}
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}
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static void
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gst_rtp_mp4g_depay_queue (GstRtpMP4GDepay * rtpmp4gdepay, GstBuffer * outbuf)
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{
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guint AU_index = GST_BUFFER_OFFSET (outbuf);
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if (rtpmp4gdepay->next_AU_index == -1) {
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GST_DEBUG_OBJECT (rtpmp4gdepay, "Init AU counter %u", AU_index);
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rtpmp4gdepay->next_AU_index = AU_index;
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}
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if (rtpmp4gdepay->next_AU_index == AU_index) {
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GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing expected AU_index %u", AU_index);
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/* we received the expected packet, push it and flush as much as we can from
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* the queue */
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gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpmp4gdepay), outbuf);
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rtpmp4gdepay->next_AU_index++;
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while ((outbuf = g_queue_peek_head (rtpmp4gdepay->packets))) {
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AU_index = GST_BUFFER_OFFSET (outbuf);
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GST_DEBUG_OBJECT (rtpmp4gdepay, "next available AU_index %u", AU_index);
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if (rtpmp4gdepay->next_AU_index == AU_index) {
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GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing expected AU_index %u",
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AU_index);
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outbuf = g_queue_pop_head (rtpmp4gdepay->packets);
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gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpmp4gdepay),
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outbuf);
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rtpmp4gdepay->next_AU_index++;
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} else {
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GST_DEBUG_OBJECT (rtpmp4gdepay, "waiting for next AU_index %u",
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rtpmp4gdepay->next_AU_index);
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break;
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}
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}
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} else {
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GList *list;
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GST_DEBUG_OBJECT (rtpmp4gdepay, "queueing AU_index %u", AU_index);
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/* loop the list to skip strictly smaller AU_index buffers */
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for (list = rtpmp4gdepay->packets->head; list; list = g_list_next (list)) {
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guint idx;
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gint gap;
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idx = GST_BUFFER_OFFSET (GST_BUFFER_CAST (list->data));
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/* compare the new seqnum to the one in the buffer */
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gap = (gint) (idx - AU_index);
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GST_DEBUG_OBJECT (rtpmp4gdepay, "compare with AU_index %u, gap %d", idx,
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gap);
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/* AU_index <= idx, we can stop looking */
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if (G_LIKELY (gap > 0))
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break;
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}
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if (G_LIKELY (list))
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g_queue_insert_before (rtpmp4gdepay->packets, list, outbuf);
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else
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g_queue_push_tail (rtpmp4gdepay->packets, outbuf);
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}
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}
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static GstBuffer *
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gst_rtp_mp4g_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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{
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GstRtpMP4GDepay *rtpmp4gdepay;
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GstBuffer *outbuf;
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GstClockTime timestamp;
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rtpmp4gdepay = GST_RTP_MP4G_DEPAY (depayload);
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@ -312,21 +429,24 @@ gst_rtp_mp4g_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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/* flush remaining data on discont */
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if (GST_BUFFER_IS_DISCONT (buf)) {
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GST_DEBUG_OBJECT (rtpmp4gdepay, "received DISCONT");
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gst_adapter_clear (rtpmp4gdepay->adapter);
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}
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timestamp = GST_BUFFER_TIMESTAMP (buf);
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{
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gint payload_len, payload_AU;
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guint8 *payload;
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guint32 timestamp;
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guint32 rtptime;
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guint AU_headers_len;
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guint AU_size, AU_index, payload_AU_size;
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guint AU_size, AU_index, AU_index_delta, payload_AU_size;
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gboolean M;
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payload_len = gst_rtp_buffer_get_payload_len (buf);
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payload = gst_rtp_buffer_get_payload (buf);
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timestamp = gst_rtp_buffer_get_timestamp (buf);
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rtptime = gst_rtp_buffer_get_timestamp (buf);
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M = gst_rtp_buffer_get_marker (buf);
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if (rtpmp4gdepay->sizelength > 0) {
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@ -364,6 +484,7 @@ gst_rtp_mp4g_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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/* point the bitstream parser to the first AU header bit */
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gst_bs_parse_init (&bs, payload, payload_len);
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AU_index = AU_index_delta = 0;
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for (i = 0; i < num_AU_headers && payload_AU_size > 0; i++) {
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/* parse AU header
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* +---------------------------------------+
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*/
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AU_size = gst_bs_parse_read (&bs, rtpmp4gdepay->sizelength);
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if (i == 0)
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/* calculate the AU_index, which is only on the first AU of the packet
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* and the AU_index_delta on the other AUs. This will be used to
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* reconstruct the AU ordering when interleaving. */
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if (i == 0) {
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AU_index = gst_bs_parse_read (&bs, rtpmp4gdepay->indexlength);
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else
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AU_index = gst_bs_parse_read (&bs, rtpmp4gdepay->indexdeltalength);
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if (AU_index == 0 && rtpmp4gdepay->prev_AU_index == 0) {
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gint diff;
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/* if we see two consecutive packets with AU_index of 0, we can
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* assume we have constantDuration packets. Since we don't have
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* the index we must use the AU duration to calculate the
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* index. Get the diff between the timestamps first, this can be
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* positive or negative. */
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if (rtpmp4gdepay->prev_rtptime <= rtptime)
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diff = rtptime - rtpmp4gdepay->prev_rtptime;
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else
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diff = -(rtpmp4gdepay->prev_rtptime - rtptime);
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/* get the number of packets by dividing with the duration */
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diff /= rtpmp4gdepay->constantDuration;
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rtpmp4gdepay->last_AU_index += diff;
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rtpmp4gdepay->prev_AU_index = AU_index;
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AU_index = rtpmp4gdepay->last_AU_index;
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} else {
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rtpmp4gdepay->prev_AU_index = AU_index;
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rtpmp4gdepay->last_AU_index = AU_index;
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}
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/* keep track of the higest AU_index */
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if (rtpmp4gdepay->max_AU_index != -1
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&& rtpmp4gdepay->max_AU_index <= AU_index) {
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GST_DEBUG_OBJECT (rtpmp4gdepay, "new interleave group, flushing");
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/* a new interleave group started, flush */
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gst_rtp_mp4g_depay_flush_queue (rtpmp4gdepay);
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}
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rtpmp4gdepay->prev_rtptime = rtptime;
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} else {
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AU_index_delta =
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gst_bs_parse_read (&bs, rtpmp4gdepay->indexdeltalength);
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AU_index += AU_index_delta + 1;
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}
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/* keep track of highest AU_index */
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if (rtpmp4gdepay->max_AU_index == -1
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|| AU_index > rtpmp4gdepay->max_AU_index)
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rtpmp4gdepay->max_AU_index = AU_index;
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/* the presentation time offset, a 2s-complement value, we need this to
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* calculate the timestamp on the output packet. */
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if (rtpmp4gdepay->ctsdeltalength > 0) {
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if (gst_bs_parse_read (&bs, 1))
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gst_bs_parse_read (&bs, rtpmp4gdepay->ctsdeltalength);
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}
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/* the decoding time offset, a 2s-complement value */
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if (rtpmp4gdepay->dtsdeltalength > 0) {
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if (gst_bs_parse_read (&bs, 1))
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gst_bs_parse_read (&bs, rtpmp4gdepay->dtsdeltalength);
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}
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/* RAP-flag to indicate that the AU contains a keyframe */
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if (rtpmp4gdepay->randomaccessindication)
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gst_bs_parse_read (&bs, 1);
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/* stream-state */
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if (rtpmp4gdepay->streamstateindication > 0)
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gst_bs_parse_read (&bs, rtpmp4gdepay->streamstateindication);
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GST_DEBUG_OBJECT (rtpmp4gdepay, "size %d, index %d", AU_size, AU_index);
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GST_DEBUG_OBJECT (rtpmp4gdepay, "size %d, index %d, delta %d", AU_size,
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AU_index, AU_index_delta);
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/* fragmented pakets have the AU_size set to the size of the
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* unfragmented AU. */
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outbuf = gst_adapter_take_buffer (rtpmp4gdepay->adapter, avail);
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gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
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GST_DEBUG ("gst_rtp_mp4g_depay_chain: pushing buffer of size %d",
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/* copy some of the fields we calculated above on the buffer. We also
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* copy the AU_index so that we can sort the packets in our queue. */
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GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
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GST_BUFFER_OFFSET (outbuf) = AU_index;
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/* make sure we don't use the timestamp again for other AUs in this
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* RTP packet. */
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timestamp = -1;
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GST_DEBUG_OBJECT (depayload, "pushing buffer of size %d",
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GST_BUFFER_SIZE (outbuf));
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/* only apply the timestamp for the first buffer */
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if (i == 0)
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gst_base_rtp_depayload_push_ts (depayload, timestamp, outbuf);
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else
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gst_base_rtp_depayload_push (depayload, outbuf);
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gst_rtp_mp4g_depay_queue (rtpmp4gdepay, outbuf);
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}
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payload_AU += AU_size;
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payload_AU_size -= AU_size;
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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gst_adapter_clear (rtpmp4gdepay->adapter);
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rtpmp4gdepay->max_AU_index = -1;
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rtpmp4gdepay->next_AU_index = -1;
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rtpmp4gdepay->prev_AU_index = -1;
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rtpmp4gdepay->prev_rtptime = -1;
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rtpmp4gdepay->last_AU_index = -1;
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break;
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default:
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break;
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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gst_adapter_clear (rtpmp4gdepay->adapter);
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gst_rtp_mp4g_depay_clear_queue (rtpmp4gdepay);
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break;
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default:
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break;
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}
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@ -46,6 +46,10 @@ struct _GstRtpMP4GDepay
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gint profile_level_id;
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gint streamtype;
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gint constantSize;
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gint constantDuration;
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gint sizelength;
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gint indexlength;
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gint indexdeltalength;
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@ -54,6 +58,14 @@ struct _GstRtpMP4GDepay
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gint randomaccessindication;
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gint streamstateindication;
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gint auxiliarydatasizelength;
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guint max_AU_index;
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guint prev_AU_index;
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guint last_AU_index;
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guint next_AU_index;
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guint32 prev_rtptime;
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GQueue *packets;
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GstAdapter *adapter;
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};
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