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628461b4d1
Check if codec_data was obtained before trying to use it to prevent a segfault
1227 lines
32 KiB
C
1227 lines
32 KiB
C
/* GStreamer
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* Copyright (C) 2012 Smart TV Alliance
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* Author: Thiago Sousa Santos <thiago.sousa.santos@collabora.com>, Collabora Ltd.
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*
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* gstmssmanifest.c:
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <glib.h>
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#include <string.h>
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#include <stdio.h>
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#include <ctype.h>
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#include <libxml/parser.h>
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#include <libxml/tree.h>
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/* for parsing h264 codec data */
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#include <gst/codecparsers/gsth264parser.h>
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#include "gstmssmanifest.h"
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GST_DEBUG_CATEGORY_EXTERN (mssdemux_debug);
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#define GST_CAT_DEFAULT mssdemux_debug
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#define DEFAULT_TIMESCALE 10000000
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#define MSS_NODE_STREAM_FRAGMENT "c"
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#define MSS_NODE_STREAM_QUALITY "QualityLevel"
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#define MSS_PROP_BITRATE "Bitrate"
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#define MSS_PROP_DURATION "d"
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#define MSS_PROP_LANGUAGE "Language"
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#define MSS_PROP_NUMBER "n"
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#define MSS_PROP_STREAM_DURATION "Duration"
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#define MSS_PROP_TIME "t"
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#define MSS_PROP_TIMESCALE "TimeScale"
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#define MSS_PROP_URL "Url"
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typedef struct _GstMssStreamFragment
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{
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guint number;
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guint64 time;
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guint64 duration;
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} GstMssStreamFragment;
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typedef struct _GstMssStreamQuality
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{
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xmlNodePtr xmlnode;
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gchar *bitrate_str;
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guint64 bitrate;
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} GstMssStreamQuality;
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struct _GstMssStream
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{
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xmlNodePtr xmlnode;
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gboolean active; /* if the stream is currently being used */
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gint selectedQualityIndex;
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GList *fragments;
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GList *qualities;
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gchar *url;
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gchar *lang;
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GList *current_fragment;
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GList *current_quality;
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/* TODO move this to somewhere static */
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GRegex *regex_bitrate;
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GRegex *regex_position;
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};
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struct _GstMssManifest
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{
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xmlDocPtr xml;
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xmlNodePtr xmlrootnode;
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gboolean is_live;
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GSList *streams;
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};
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static GstBuffer *gst_buffer_from_hex_string (const gchar * s);
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static gboolean
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node_has_type (xmlNodePtr node, const gchar * name)
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{
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return strcmp ((gchar *) node->name, name) == 0;
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}
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static GstMssStreamQuality *
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gst_mss_stream_quality_new (xmlNodePtr node)
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{
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GstMssStreamQuality *q = g_slice_new (GstMssStreamQuality);
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q->xmlnode = node;
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q->bitrate_str = (gchar *) xmlGetProp (node, (xmlChar *) MSS_PROP_BITRATE);
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if (q->bitrate_str != NULL)
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q->bitrate = g_ascii_strtoull (q->bitrate_str, NULL, 10);
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else
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q->bitrate = 0;
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return q;
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}
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static void
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gst_mss_stream_quality_free (GstMssStreamQuality * quality)
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{
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g_return_if_fail (quality != NULL);
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xmlFree (quality->bitrate_str);
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g_slice_free (GstMssStreamQuality, quality);
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}
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static gint
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compare_bitrate (GstMssStreamQuality * a, GstMssStreamQuality * b)
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{
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if (a->bitrate > b->bitrate)
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return 1;
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if (a->bitrate < b->bitrate)
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return -1;
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return 0;
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}
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static void
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_gst_mss_stream_init (GstMssStream * stream, xmlNodePtr node)
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{
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xmlNodePtr iter;
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GstMssStreamFragment *previous_fragment = NULL;
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guint fragment_number = 0;
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guint64 fragment_time_accum = 0;
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stream->xmlnode = node;
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/* get the base url path generator */
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stream->url = (gchar *) xmlGetProp (node, (xmlChar *) MSS_PROP_URL);
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stream->lang = (gchar *) xmlGetProp (node, (xmlChar *) MSS_PROP_LANGUAGE);
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for (iter = node->children; iter; iter = iter->next) {
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if (node_has_type (iter, MSS_NODE_STREAM_FRAGMENT)) {
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gchar *duration_str;
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gchar *time_str;
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gchar *seqnum_str;
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GstMssStreamFragment *fragment = g_new (GstMssStreamFragment, 1);
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duration_str = (gchar *) xmlGetProp (iter, (xmlChar *) MSS_PROP_DURATION);
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time_str = (gchar *) xmlGetProp (iter, (xmlChar *) MSS_PROP_TIME);
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seqnum_str = (gchar *) xmlGetProp (iter, (xmlChar *) MSS_PROP_NUMBER);
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/* use the node's seq number or use the previous + 1 */
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if (seqnum_str) {
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fragment->number = g_ascii_strtoull (seqnum_str, NULL, 10);
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xmlFree (seqnum_str);
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fragment_number = fragment->number;
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} else {
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fragment->number = fragment_number;
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}
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fragment_number = fragment->number + 1;
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if (time_str) {
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fragment->time = g_ascii_strtoull (time_str, NULL, 10);
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xmlFree (time_str);
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fragment_time_accum = fragment->time;
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} else {
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fragment->time = fragment_time_accum;
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}
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/* if we have a previous fragment, means we need to set its duration */
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if (previous_fragment)
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previous_fragment->duration = fragment->time - previous_fragment->time;
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if (duration_str) {
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fragment->duration = g_ascii_strtoull (duration_str, NULL, 10);
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previous_fragment = NULL;
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fragment_time_accum += fragment->duration;
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xmlFree (duration_str);
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} else {
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/* store to set the duration at the next iteration */
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previous_fragment = fragment;
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}
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/* we reverse it later */
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stream->fragments = g_list_prepend (stream->fragments, fragment);
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} else if (node_has_type (iter, MSS_NODE_STREAM_QUALITY)) {
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GstMssStreamQuality *quality = gst_mss_stream_quality_new (iter);
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stream->qualities = g_list_prepend (stream->qualities, quality);
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} else {
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/* TODO gst log this */
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}
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}
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stream->fragments = g_list_reverse (stream->fragments);
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/* order them from smaller to bigger based on bitrates */
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stream->qualities =
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g_list_sort (stream->qualities, (GCompareFunc) compare_bitrate);
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stream->current_fragment = stream->fragments;
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stream->current_quality = stream->qualities;
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stream->regex_bitrate = g_regex_new ("\\{[Bb]itrate\\}", 0, 0, NULL);
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stream->regex_position = g_regex_new ("\\{start[ _]time\\}", 0, 0, NULL);
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}
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GstMssManifest *
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gst_mss_manifest_new (GstBuffer * data)
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{
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GstMssManifest *manifest;
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xmlNodePtr root;
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xmlNodePtr nodeiter;
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gchar *live_str;
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GstMapInfo mapinfo;
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if (!gst_buffer_map (data, &mapinfo, GST_MAP_READ)) {
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return NULL;
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}
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manifest = g_malloc0 (sizeof (GstMssManifest));
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manifest->xml = xmlReadMemory ((const gchar *) mapinfo.data,
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mapinfo.size, "manifest", NULL, 0);
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root = manifest->xmlrootnode = xmlDocGetRootElement (manifest->xml);
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live_str = (gchar *) xmlGetProp (root, (xmlChar *) "IsLive");
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if (live_str) {
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manifest->is_live = g_ascii_strcasecmp (live_str, "true") == 0;
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xmlFree (live_str);
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}
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for (nodeiter = root->children; nodeiter; nodeiter = nodeiter->next) {
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if (nodeiter->type == XML_ELEMENT_NODE
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&& (strcmp ((const char *) nodeiter->name, "StreamIndex") == 0)) {
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GstMssStream *stream = g_new0 (GstMssStream, 1);
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manifest->streams = g_slist_append (manifest->streams, stream);
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_gst_mss_stream_init (stream, nodeiter);
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}
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}
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gst_buffer_unmap (data, &mapinfo);
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return manifest;
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}
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static void
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gst_mss_stream_free (GstMssStream * stream)
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{
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g_list_free_full (stream->fragments, g_free);
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g_list_free_full (stream->qualities,
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(GDestroyNotify) gst_mss_stream_quality_free);
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xmlFree (stream->url);
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xmlFree (stream->lang);
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g_regex_unref (stream->regex_position);
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g_regex_unref (stream->regex_bitrate);
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g_free (stream);
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}
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void
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gst_mss_manifest_free (GstMssManifest * manifest)
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{
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g_return_if_fail (manifest != NULL);
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g_slist_free_full (manifest->streams, (GDestroyNotify) gst_mss_stream_free);
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xmlFreeDoc (manifest->xml);
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g_free (manifest);
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}
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GSList *
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gst_mss_manifest_get_streams (GstMssManifest * manifest)
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{
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return manifest->streams;
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}
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GstMssStreamType
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gst_mss_stream_get_type (GstMssStream * stream)
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{
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gchar *prop = (gchar *) xmlGetProp (stream->xmlnode, (xmlChar *) "Type");
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GstMssStreamType ret = MSS_STREAM_TYPE_UNKNOWN;
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if (prop == NULL)
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return MSS_STREAM_TYPE_UNKNOWN;
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if (strcmp (prop, "video") == 0) {
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ret = MSS_STREAM_TYPE_VIDEO;
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} else if (strcmp (prop, "audio") == 0) {
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ret = MSS_STREAM_TYPE_AUDIO;
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}
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xmlFree (prop);
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return ret;
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}
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static GstCaps *
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_gst_mss_stream_video_caps_from_fourcc (gchar * fourcc)
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{
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if (!fourcc)
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return NULL;
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if (strcmp (fourcc, "H264") == 0 || strcmp (fourcc, "AVC1") == 0) {
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return gst_caps_new_simple ("video/x-h264", "stream-format", G_TYPE_STRING,
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"avc", NULL);
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} else if (strcmp (fourcc, "WVC1") == 0) {
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return gst_caps_new_simple ("video/x-wmv", "wmvversion", G_TYPE_INT, 3,
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"format", G_TYPE_STRING, "WVC1", NULL);
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}
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return NULL;
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}
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static GstCaps *
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_gst_mss_stream_audio_caps_from_fourcc (gchar * fourcc)
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{
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if (!fourcc)
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return NULL;
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if (strcmp (fourcc, "AACL") == 0) {
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return gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 4,
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NULL);
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} else if (strcmp (fourcc, "WmaPro") == 0 || strcmp (fourcc, "WMAP") == 0) {
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return gst_caps_new_simple ("audio/x-wma", "wmaversion", G_TYPE_INT, 3,
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NULL);
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}
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return NULL;
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}
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static GstCaps *
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_gst_mss_stream_audio_caps_from_audio_tag (gint audiotag)
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{
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switch (audiotag) {
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case 83: /* MP3 */
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return gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 1,
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"layer", G_TYPE_INT, 3, NULL);
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case 255: /* AAC */
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return gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 4,
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NULL);
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default:
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break;
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}
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return NULL;
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}
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/* copied and adapted from h264parse */
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static GstBuffer *
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_make_h264_codec_data (GstBuffer * sps, GstBuffer * pps)
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{
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GstBuffer *buf;
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gint sps_size = 0, pps_size = 0, num_sps = 0, num_pps = 0;
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guint8 profile_idc = 0, profile_comp = 0, level_idc = 0;
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guint8 *data;
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gint nl;
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GstMapInfo spsinfo, ppsinfo, codecdatainfo;
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if (gst_buffer_get_size (sps) < 4)
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return NULL;
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gst_buffer_map (sps, &spsinfo, GST_MAP_READ);
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gst_buffer_map (pps, &ppsinfo, GST_MAP_READ);
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sps_size += spsinfo.size + 2;
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profile_idc = spsinfo.data[1];
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profile_comp = spsinfo.data[2];
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level_idc = spsinfo.data[3];
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num_sps = 1;
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pps_size += ppsinfo.size + 2;
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num_pps = 1;
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buf = gst_buffer_new_and_alloc (5 + 1 + sps_size + 1 + pps_size);
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gst_buffer_map (buf, &codecdatainfo, GST_MAP_WRITE);
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data = codecdatainfo.data;
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nl = 4;
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data[0] = 1; /* AVC Decoder Configuration Record ver. 1 */
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data[1] = profile_idc; /* profile_idc */
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data[2] = profile_comp; /* profile_compability */
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data[3] = level_idc; /* level_idc */
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data[4] = 0xfc | (nl - 1); /* nal_length_size_minus1 */
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data[5] = 0xe0 | num_sps; /* number of SPSs */
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data += 6;
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GST_WRITE_UINT16_BE (data, spsinfo.size);
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memcpy (data + 2, spsinfo.data, spsinfo.size);
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data += 2 + spsinfo.size;
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data[0] = num_pps;
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data++;
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GST_WRITE_UINT16_BE (data, ppsinfo.size);
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memcpy (data + 2, ppsinfo.data, ppsinfo.size);
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data += 2 + ppsinfo.size;
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gst_buffer_unmap (sps, &spsinfo);
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gst_buffer_unmap (pps, &ppsinfo);
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gst_buffer_unmap (buf, &codecdatainfo);
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return buf;
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}
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static void
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_gst_mss_stream_add_h264_codec_data (GstCaps * caps, const gchar * codecdatastr)
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{
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GstBuffer *sps;
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GstBuffer *pps;
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GstBuffer *buffer;
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gchar *sps_str;
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gchar *pps_str;
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GstH264NalUnit nalu;
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GstH264SPS sps_struct;
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GstH264ParserResult parseres;
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GstMapInfo spsinfo;
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/* search for the sps start */
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if (g_str_has_prefix (codecdatastr, "00000001")) {
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sps_str = (gchar *) codecdatastr + 8;
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} else {
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return; /* invalid mss codec data */
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}
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/* search for the pps start */
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pps_str = g_strstr_len (sps_str, -1, "00000001");
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if (!pps_str) {
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return; /* invalid mss codec data */
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}
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pps_str[0] = '\0';
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sps = gst_buffer_from_hex_string (sps_str);
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pps_str[0] = '0';
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pps_str = pps_str + 8;
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pps = gst_buffer_from_hex_string (pps_str);
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gst_buffer_map (sps, &spsinfo, GST_MAP_READ);
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nalu.ref_idc = (spsinfo.data[0] & 0x60) >> 5;
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nalu.type = GST_H264_NAL_SPS;
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nalu.size = spsinfo.size;
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nalu.data = spsinfo.data;
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nalu.offset = 0;
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nalu.sc_offset = 0;
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nalu.valid = TRUE;
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parseres = gst_h264_parse_sps (&nalu, &sps_struct, TRUE);
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if (parseres == GST_H264_PARSER_OK) {
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gst_caps_set_simple (caps, "framerate", GST_TYPE_FRACTION,
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sps_struct.fps_num, sps_struct.fps_den, NULL);
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}
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buffer = _make_h264_codec_data (sps, pps);
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gst_buffer_unmap (sps, &spsinfo);
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gst_buffer_unref (sps);
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gst_buffer_unref (pps);
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if (buffer != NULL) {
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gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, buffer, NULL);
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gst_buffer_unref (buffer);
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}
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}
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static GstCaps *
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_gst_mss_stream_video_caps_from_qualitylevel_xml (GstMssStreamQuality * q)
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{
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xmlNodePtr node = q->xmlnode;
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GstCaps *caps;
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GstStructure *structure;
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gchar *fourcc = (gchar *) xmlGetProp (node, (xmlChar *) "FourCC");
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gchar *max_width = (gchar *) xmlGetProp (node, (xmlChar *) "MaxWidth");
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gchar *max_height = (gchar *) xmlGetProp (node, (xmlChar *) "MaxHeight");
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gchar *codec_data =
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(gchar *) xmlGetProp (node, (xmlChar *) "CodecPrivateData");
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if (!max_width)
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max_width = (gchar *) xmlGetProp (node, (xmlChar *) "Width");
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if (!max_height)
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max_height = (gchar *) xmlGetProp (node, (xmlChar *) "Height");
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caps = _gst_mss_stream_video_caps_from_fourcc (fourcc);
|
|
if (!caps)
|
|
goto end;
|
|
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
|
|
if (max_width) {
|
|
gst_structure_set (structure, "width", G_TYPE_INT,
|
|
(int) g_ascii_strtoull (max_width, NULL, 10), NULL);
|
|
}
|
|
if (max_height) {
|
|
gst_structure_set (structure, "height", G_TYPE_INT,
|
|
(int) g_ascii_strtoull (max_height, NULL, 10), NULL);
|
|
}
|
|
|
|
if (codec_data && strlen (codec_data)) {
|
|
if (strcmp (fourcc, "H264") == 0 || strcmp (fourcc, "AVC1") == 0) {
|
|
_gst_mss_stream_add_h264_codec_data (caps, codec_data);
|
|
} else {
|
|
GstBuffer *buffer = gst_buffer_from_hex_string ((gchar *) codec_data);
|
|
gst_structure_set (structure, "codec_data", GST_TYPE_BUFFER, buffer,
|
|
NULL);
|
|
gst_buffer_unref (buffer);
|
|
}
|
|
}
|
|
|
|
end:
|
|
xmlFree (fourcc);
|
|
xmlFree (max_width);
|
|
xmlFree (max_height);
|
|
xmlFree (codec_data);
|
|
|
|
return caps;
|
|
}
|
|
|
|
static guint8
|
|
_frequency_index_from_sampling_rate (guint sampling_rate)
|
|
{
|
|
static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000, 44100,
|
|
32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350
|
|
};
|
|
|
|
guint8 i;
|
|
|
|
for (i = 0; i < G_N_ELEMENTS (aac_sample_rates); i++) {
|
|
if (aac_sample_rates[i] == sampling_rate)
|
|
return i;
|
|
}
|
|
return 15;
|
|
}
|
|
|
|
static GstBuffer *
|
|
_make_aacl_codec_data (guint64 sampling_rate, guint64 channels)
|
|
{
|
|
GstBuffer *buf;
|
|
guint8 *data;
|
|
guint8 frequency_index;
|
|
guint8 buf_size;
|
|
GstMapInfo info;
|
|
|
|
buf_size = 2;
|
|
frequency_index = _frequency_index_from_sampling_rate (sampling_rate);
|
|
if (frequency_index == 15)
|
|
buf_size += 3;
|
|
|
|
buf = gst_buffer_new_and_alloc (buf_size);
|
|
gst_buffer_map (buf, &info, GST_MAP_WRITE);
|
|
data = info.data;
|
|
|
|
data[0] = 2 << 3; /* AAC-LC object type is 2 */
|
|
data[0] += frequency_index >> 1;
|
|
data[1] = (frequency_index & 0x01) << 7;
|
|
|
|
/* Sampling rate is not in frequencies table, write manually */
|
|
if (frequency_index == 15) {
|
|
data[1] += sampling_rate >> 17;
|
|
data[2] = (sampling_rate >> 9) & 0xFF;
|
|
data[3] = (sampling_rate >> 1) & 0xFF;
|
|
data[4] = sampling_rate & 0x01;
|
|
data += 3;
|
|
}
|
|
|
|
data[1] += (channels & 0x0F) << 3;
|
|
|
|
gst_buffer_unmap (buf, &info);
|
|
|
|
return buf;
|
|
}
|
|
|
|
static GstCaps *
|
|
_gst_mss_stream_audio_caps_from_qualitylevel_xml (GstMssStreamQuality * q)
|
|
{
|
|
xmlNodePtr node = q->xmlnode;
|
|
GstCaps *caps = NULL;
|
|
GstStructure *structure;
|
|
gchar *fourcc = (gchar *) xmlGetProp (node, (xmlChar *) "FourCC");
|
|
gchar *audiotag = (gchar *) xmlGetProp (node, (xmlChar *) "AudioTag");
|
|
gchar *channels_str = (gchar *) xmlGetProp (node, (xmlChar *) "Channels");
|
|
gchar *rate_str = (gchar *) xmlGetProp (node, (xmlChar *) "SamplingRate");
|
|
gchar *block_align_str =
|
|
(gchar *) xmlGetProp (node, (xmlChar *) "PacketSize");
|
|
gchar *codec_data_str =
|
|
(gchar *) xmlGetProp (node, (xmlChar *) "CodecPrivateData");
|
|
GstBuffer *codec_data = NULL;
|
|
gint block_align = 0;
|
|
gint rate = 0;
|
|
gint channels = 0;
|
|
gint atag = 0;
|
|
|
|
if (!fourcc) /* sometimes the fourcc is omitted, we fallback to the Subtype in the StreamIndex node */
|
|
fourcc = (gchar *) xmlGetProp (node->parent, (xmlChar *) "Subtype");
|
|
|
|
if (fourcc) {
|
|
caps = _gst_mss_stream_audio_caps_from_fourcc (fourcc);
|
|
} else if (audiotag) {
|
|
atag = g_ascii_strtoull (audiotag, NULL, 10);
|
|
caps = _gst_mss_stream_audio_caps_from_audio_tag (atag);
|
|
}
|
|
|
|
if (!caps)
|
|
goto end;
|
|
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
if (codec_data_str && strlen (codec_data_str)) {
|
|
codec_data = gst_buffer_from_hex_string ((gchar *) codec_data_str);
|
|
}
|
|
|
|
if (rate_str)
|
|
rate = (gint) g_ascii_strtoull (rate_str, NULL, 10);
|
|
if (channels_str)
|
|
channels = (int) g_ascii_strtoull (channels_str, NULL, 10);
|
|
if (block_align_str)
|
|
block_align = (int) g_ascii_strtoull (block_align_str, NULL, 10);
|
|
|
|
if (!codec_data) {
|
|
gint codec_data_len;
|
|
codec_data_str = (gchar *) xmlGetProp (node, (xmlChar *) "WaveFormatEx");
|
|
|
|
if (codec_data_str != NULL) {
|
|
codec_data_len = strlen (codec_data_str) / 2;
|
|
|
|
/* a WAVEFORMATEX structure is 18 bytes */
|
|
if (codec_data_str && codec_data_len >= 18) {
|
|
GstMapInfo mapinfo;
|
|
codec_data = gst_buffer_from_hex_string ((gchar *) codec_data_str);
|
|
|
|
/* since this is a WAVEFORMATEX, try to get the block_align and rate */
|
|
gst_buffer_map (codec_data, &mapinfo, GST_MAP_READ);
|
|
if (!channels_str) {
|
|
channels = GST_READ_UINT16_LE (mapinfo.data + 2);
|
|
}
|
|
if (!rate_str) {
|
|
rate = GST_READ_UINT32_LE (mapinfo.data + 4);
|
|
}
|
|
if (!block_align) {
|
|
block_align = GST_READ_UINT16_LE (mapinfo.data + 12);
|
|
}
|
|
gst_buffer_unmap (codec_data, &mapinfo);
|
|
|
|
/* Consume all the WAVEFORMATEX structure, and pass only the rest of
|
|
* the data as the codec private data */
|
|
gst_buffer_resize (codec_data, 18, -1);
|
|
} else {
|
|
GST_WARNING ("Dropping WaveFormatEx: data is %d bytes, "
|
|
"but at least 18 bytes are expected", codec_data_len);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (!codec_data && ((fourcc && strcmp (fourcc, "AACL") == 0) || atag == 255)
|
|
&& rate && channels) {
|
|
codec_data = _make_aacl_codec_data (rate, channels);
|
|
}
|
|
|
|
if (block_align)
|
|
gst_structure_set (structure, "block_align", G_TYPE_INT, block_align, NULL);
|
|
|
|
if (channels)
|
|
gst_structure_set (structure, "channels", G_TYPE_INT, channels, NULL);
|
|
|
|
if (rate)
|
|
gst_structure_set (structure, "rate", G_TYPE_INT, rate, NULL);
|
|
|
|
if (q->bitrate)
|
|
gst_structure_set (structure, "bitrate", G_TYPE_INT, q->bitrate, NULL);
|
|
|
|
if (codec_data)
|
|
gst_structure_set (structure, "codec_data", GST_TYPE_BUFFER, codec_data,
|
|
NULL);
|
|
|
|
end:
|
|
if (codec_data)
|
|
gst_buffer_unref (codec_data);
|
|
xmlFree (fourcc);
|
|
xmlFree (audiotag);
|
|
xmlFree (channels_str);
|
|
xmlFree (rate_str);
|
|
xmlFree (block_align_str);
|
|
xmlFree (codec_data_str);
|
|
|
|
return caps;
|
|
}
|
|
|
|
void
|
|
gst_mss_stream_set_active (GstMssStream * stream, gboolean active)
|
|
{
|
|
stream->active = active;
|
|
}
|
|
|
|
guint64
|
|
gst_mss_stream_get_timescale (GstMssStream * stream)
|
|
{
|
|
gchar *timescale;
|
|
guint64 ts = DEFAULT_TIMESCALE;
|
|
|
|
timescale =
|
|
(gchar *) xmlGetProp (stream->xmlnode, (xmlChar *) MSS_PROP_TIMESCALE);
|
|
if (!timescale) {
|
|
timescale =
|
|
(gchar *) xmlGetProp (stream->xmlnode->parent,
|
|
(xmlChar *) MSS_PROP_TIMESCALE);
|
|
}
|
|
|
|
if (timescale) {
|
|
ts = g_ascii_strtoull (timescale, NULL, 10);
|
|
xmlFree (timescale);
|
|
}
|
|
return ts;
|
|
}
|
|
|
|
guint64
|
|
gst_mss_manifest_get_timescale (GstMssManifest * manifest)
|
|
{
|
|
gchar *timescale;
|
|
guint64 ts = DEFAULT_TIMESCALE;
|
|
|
|
timescale =
|
|
(gchar *) xmlGetProp (manifest->xmlrootnode,
|
|
(xmlChar *) MSS_PROP_TIMESCALE);
|
|
if (timescale) {
|
|
ts = g_ascii_strtoull (timescale, NULL, 10);
|
|
xmlFree (timescale);
|
|
}
|
|
return ts;
|
|
}
|
|
|
|
guint64
|
|
gst_mss_manifest_get_duration (GstMssManifest * manifest)
|
|
{
|
|
gchar *duration;
|
|
guint64 dur = -1;
|
|
|
|
duration =
|
|
(gchar *) xmlGetProp (manifest->xmlrootnode,
|
|
(xmlChar *) MSS_PROP_STREAM_DURATION);
|
|
if (duration) {
|
|
dur = g_ascii_strtoull (duration, NULL, 10);
|
|
xmlFree (duration);
|
|
}
|
|
return dur;
|
|
}
|
|
|
|
|
|
/**
|
|
* Gets the duration in nanoseconds
|
|
*/
|
|
GstClockTime
|
|
gst_mss_manifest_get_gst_duration (GstMssManifest * manifest)
|
|
{
|
|
guint64 duration = -1;
|
|
guint64 timescale;
|
|
GstClockTime gstdur = GST_CLOCK_TIME_NONE;
|
|
|
|
duration = gst_mss_manifest_get_duration (manifest);
|
|
timescale = gst_mss_manifest_get_timescale (manifest);
|
|
|
|
if (duration != -1 && timescale != -1)
|
|
gstdur =
|
|
(GstClockTime) gst_util_uint64_scale_round (duration, GST_SECOND,
|
|
timescale);
|
|
|
|
return gstdur;
|
|
}
|
|
|
|
GstCaps *
|
|
gst_mss_stream_get_caps (GstMssStream * stream)
|
|
{
|
|
GstMssStreamType streamtype = gst_mss_stream_get_type (stream);
|
|
GstMssStreamQuality *qualitylevel = stream->current_quality->data;
|
|
GstCaps *caps = NULL;
|
|
|
|
if (streamtype == MSS_STREAM_TYPE_VIDEO)
|
|
caps = _gst_mss_stream_video_caps_from_qualitylevel_xml (qualitylevel);
|
|
else if (streamtype == MSS_STREAM_TYPE_AUDIO)
|
|
caps = _gst_mss_stream_audio_caps_from_qualitylevel_xml (qualitylevel);
|
|
|
|
return caps;
|
|
}
|
|
|
|
GstFlowReturn
|
|
gst_mss_stream_get_fragment_url (GstMssStream * stream, gchar ** url)
|
|
{
|
|
gchar *tmp;
|
|
gchar *start_time_str;
|
|
GstMssStreamFragment *fragment;
|
|
GstMssStreamQuality *quality = stream->current_quality->data;
|
|
|
|
g_return_val_if_fail (stream->active, GST_FLOW_ERROR);
|
|
|
|
if (stream->current_fragment == NULL) /* stream is over */
|
|
return GST_FLOW_EOS;
|
|
|
|
fragment = stream->current_fragment->data;
|
|
|
|
start_time_str = g_strdup_printf ("%" G_GUINT64_FORMAT, fragment->time);
|
|
|
|
tmp = g_regex_replace_literal (stream->regex_bitrate, stream->url,
|
|
strlen (stream->url), 0, quality->bitrate_str, 0, NULL);
|
|
*url = g_regex_replace_literal (stream->regex_position, tmp,
|
|
strlen (tmp), 0, start_time_str, 0, NULL);
|
|
|
|
g_free (tmp);
|
|
g_free (start_time_str);
|
|
|
|
if (*url == NULL)
|
|
return GST_FLOW_ERROR;
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
GstClockTime
|
|
gst_mss_stream_get_fragment_gst_timestamp (GstMssStream * stream)
|
|
{
|
|
guint64 time;
|
|
guint64 timescale;
|
|
GstMssStreamFragment *fragment;
|
|
|
|
g_return_val_if_fail (stream->active, GST_FLOW_ERROR);
|
|
|
|
if (!stream->current_fragment)
|
|
return GST_CLOCK_TIME_NONE;
|
|
|
|
fragment = stream->current_fragment->data;
|
|
|
|
time = fragment->time;
|
|
timescale = gst_mss_stream_get_timescale (stream);
|
|
return (GstClockTime) gst_util_uint64_scale_round (time, GST_SECOND,
|
|
timescale);
|
|
}
|
|
|
|
GstClockTime
|
|
gst_mss_stream_get_fragment_gst_duration (GstMssStream * stream)
|
|
{
|
|
guint64 dur;
|
|
guint64 timescale;
|
|
GstMssStreamFragment *fragment;
|
|
|
|
g_return_val_if_fail (stream->active, GST_FLOW_ERROR);
|
|
|
|
if (!stream->current_fragment)
|
|
return GST_CLOCK_TIME_NONE;
|
|
|
|
fragment = stream->current_fragment->data;
|
|
|
|
dur = fragment->duration;
|
|
timescale = gst_mss_stream_get_timescale (stream);
|
|
return (GstClockTime) gst_util_uint64_scale_round (dur, GST_SECOND,
|
|
timescale);
|
|
}
|
|
|
|
GstFlowReturn
|
|
gst_mss_stream_advance_fragment (GstMssStream * stream)
|
|
{
|
|
g_return_val_if_fail (stream->active, GST_FLOW_ERROR);
|
|
|
|
if (stream->current_fragment == NULL)
|
|
return GST_FLOW_EOS;
|
|
|
|
stream->current_fragment = g_list_next (stream->current_fragment);
|
|
if (stream->current_fragment == NULL)
|
|
return GST_FLOW_EOS;
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
const gchar *
|
|
gst_mss_stream_type_name (GstMssStreamType streamtype)
|
|
{
|
|
switch (streamtype) {
|
|
case MSS_STREAM_TYPE_VIDEO:
|
|
return "video";
|
|
case MSS_STREAM_TYPE_AUDIO:
|
|
return "audio";
|
|
case MSS_STREAM_TYPE_UNKNOWN:
|
|
default:
|
|
return "unknown";
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Seeks all streams to the fragment that contains the set time
|
|
*
|
|
* @time: time in nanoseconds
|
|
*/
|
|
gboolean
|
|
gst_mss_manifest_seek (GstMssManifest * manifest, guint64 time)
|
|
{
|
|
gboolean ret = TRUE;
|
|
GSList *iter;
|
|
|
|
for (iter = manifest->streams; iter; iter = g_slist_next (iter)) {
|
|
ret = gst_mss_stream_seek (iter->data, time) & ret;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* Seeks this stream to the fragment that contains the sample at time
|
|
*
|
|
* @time: time in nanoseconds
|
|
*/
|
|
gboolean
|
|
gst_mss_stream_seek (GstMssStream * stream, guint64 time)
|
|
{
|
|
GList *iter;
|
|
guint64 timescale;
|
|
|
|
timescale = gst_mss_stream_get_timescale (stream);
|
|
time = gst_util_uint64_scale_round (time, timescale, GST_SECOND);
|
|
|
|
for (iter = stream->fragments; iter; iter = g_list_next (iter)) {
|
|
GList *next = g_list_next (iter);
|
|
if (next) {
|
|
GstMssStreamFragment *fragment = next->data;
|
|
|
|
if (fragment->time > time) {
|
|
stream->current_fragment = iter;
|
|
break;
|
|
}
|
|
} else {
|
|
GstMssStreamFragment *fragment = iter->data;
|
|
if (fragment->time + fragment->duration > time) {
|
|
stream->current_fragment = iter;
|
|
} else {
|
|
stream->current_fragment = NULL; /* EOS */
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
guint64
|
|
gst_mss_manifest_get_current_bitrate (GstMssManifest * manifest)
|
|
{
|
|
guint64 bitrate = 0;
|
|
GSList *iter;
|
|
|
|
for (iter = gst_mss_manifest_get_streams (manifest); iter;
|
|
iter = g_slist_next (iter)) {
|
|
GstMssStream *stream = iter->data;
|
|
if (stream->active && stream->current_quality) {
|
|
GstMssStreamQuality *q = stream->current_quality->data;
|
|
|
|
bitrate += q->bitrate;
|
|
}
|
|
}
|
|
|
|
return bitrate;
|
|
}
|
|
|
|
gboolean
|
|
gst_mss_manifest_is_live (GstMssManifest * manifest)
|
|
{
|
|
return manifest->is_live;
|
|
}
|
|
|
|
static void
|
|
gst_mss_stream_reload_fragments (GstMssStream * stream, xmlNodePtr streamIndex)
|
|
{
|
|
xmlNodePtr iter;
|
|
GList *new_fragments = NULL;
|
|
GstMssStreamFragment *previous_fragment = NULL;
|
|
GstMssStreamFragment *current_fragment =
|
|
stream->current_fragment ? stream->current_fragment->data : NULL;
|
|
guint64 current_time = gst_mss_stream_get_fragment_gst_timestamp (stream);
|
|
guint fragment_number = 0;
|
|
guint64 fragment_time_accum = 0;
|
|
|
|
if (!current_fragment && stream->fragments) {
|
|
current_fragment = g_list_last (stream->fragments)->data;
|
|
} else if (g_list_previous (stream->current_fragment)) {
|
|
/* rewind one as this is the next to be pushed */
|
|
current_fragment = g_list_previous (stream->current_fragment)->data;
|
|
} else {
|
|
current_fragment = NULL;
|
|
}
|
|
|
|
if (current_fragment) {
|
|
current_time = current_fragment->time;
|
|
fragment_number = current_fragment->number;
|
|
fragment_time_accum = current_fragment->time;
|
|
}
|
|
|
|
for (iter = streamIndex->children; iter; iter = iter->next) {
|
|
if (node_has_type (iter, MSS_NODE_STREAM_FRAGMENT)) {
|
|
gchar *duration_str;
|
|
gchar *time_str;
|
|
gchar *seqnum_str;
|
|
GstMssStreamFragment *fragment = g_new (GstMssStreamFragment, 1);
|
|
|
|
duration_str = (gchar *) xmlGetProp (iter, (xmlChar *) MSS_PROP_DURATION);
|
|
time_str = (gchar *) xmlGetProp (iter, (xmlChar *) MSS_PROP_TIME);
|
|
seqnum_str = (gchar *) xmlGetProp (iter, (xmlChar *) MSS_PROP_NUMBER);
|
|
|
|
/* use the node's seq number or use the previous + 1 */
|
|
if (seqnum_str) {
|
|
fragment->number = g_ascii_strtoull (seqnum_str, NULL, 10);
|
|
xmlFree (seqnum_str);
|
|
} else {
|
|
fragment->number = fragment_number;
|
|
}
|
|
fragment_number = fragment->number + 1;
|
|
|
|
if (time_str) {
|
|
fragment->time = g_ascii_strtoull (time_str, NULL, 10);
|
|
xmlFree (time_str);
|
|
fragment_time_accum = fragment->time;
|
|
} else {
|
|
fragment->time = fragment_time_accum;
|
|
}
|
|
|
|
/* if we have a previous fragment, means we need to set its duration */
|
|
if (previous_fragment)
|
|
previous_fragment->duration = fragment->time - previous_fragment->time;
|
|
|
|
if (duration_str) {
|
|
fragment->duration = g_ascii_strtoull (duration_str, NULL, 10);
|
|
|
|
previous_fragment = NULL;
|
|
fragment_time_accum += fragment->duration;
|
|
xmlFree (duration_str);
|
|
} else {
|
|
/* store to set the duration at the next iteration */
|
|
previous_fragment = fragment;
|
|
}
|
|
|
|
if (fragment->time > current_time) {
|
|
new_fragments = g_list_append (new_fragments, fragment);
|
|
} else {
|
|
previous_fragment = NULL;
|
|
g_free (fragment);
|
|
}
|
|
|
|
} else {
|
|
/* TODO gst log this */
|
|
}
|
|
}
|
|
|
|
/* store the new fragments list */
|
|
if (new_fragments) {
|
|
g_list_free_full (stream->fragments, g_free);
|
|
stream->fragments = new_fragments;
|
|
stream->current_fragment = new_fragments;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_mss_manifest_reload_fragments_from_xml (GstMssManifest * manifest,
|
|
xmlNodePtr root)
|
|
{
|
|
xmlNodePtr nodeiter;
|
|
GSList *streams = manifest->streams;
|
|
|
|
/* we assume the server is providing the streams in the same order in
|
|
* every manifest */
|
|
for (nodeiter = root->children; nodeiter && streams;
|
|
nodeiter = nodeiter->next) {
|
|
if (nodeiter->type == XML_ELEMENT_NODE
|
|
&& (strcmp ((const char *) nodeiter->name, "StreamIndex") == 0)) {
|
|
gst_mss_stream_reload_fragments (streams->data, nodeiter);
|
|
streams = g_slist_next (streams);
|
|
}
|
|
}
|
|
}
|
|
|
|
void
|
|
gst_mss_manifest_reload_fragments (GstMssManifest * manifest, GstBuffer * data)
|
|
{
|
|
xmlDocPtr xml;
|
|
xmlNodePtr root;
|
|
GstMapInfo info;
|
|
|
|
g_return_if_fail (manifest->is_live);
|
|
|
|
gst_buffer_map (data, &info, GST_MAP_READ);
|
|
|
|
xml = xmlReadMemory ((const gchar *) info.data,
|
|
info.size, "manifest", NULL, 0);
|
|
root = xmlDocGetRootElement (xml);
|
|
|
|
gst_mss_manifest_reload_fragments_from_xml (manifest, root);
|
|
|
|
xmlFreeDoc (xml);
|
|
|
|
gst_buffer_unmap (data, &info);
|
|
}
|
|
|
|
gboolean
|
|
gst_mss_stream_select_bitrate (GstMssStream * stream, guint64 bitrate)
|
|
{
|
|
GList *iter = stream->current_quality;
|
|
GList *next;
|
|
GstMssStreamQuality *q = iter->data;
|
|
|
|
while (q->bitrate > bitrate) {
|
|
next = g_list_previous (iter);
|
|
if (next) {
|
|
iter = next;
|
|
q = iter->data;
|
|
} else {
|
|
break;
|
|
}
|
|
}
|
|
|
|
while (q->bitrate < bitrate) {
|
|
GstMssStreamQuality *next_q;
|
|
next = g_list_next (iter);
|
|
if (next) {
|
|
next_q = next->data;
|
|
if (next_q->bitrate < bitrate) {
|
|
iter = next;
|
|
q = iter->data;
|
|
} else {
|
|
break;
|
|
}
|
|
} else {
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (iter == stream->current_quality)
|
|
return FALSE;
|
|
stream->current_quality = iter;
|
|
return TRUE;
|
|
}
|
|
|
|
guint64
|
|
gst_mss_stream_get_current_bitrate (GstMssStream * stream)
|
|
{
|
|
GstMssStreamQuality *q;
|
|
if (stream->current_quality == NULL)
|
|
return 0;
|
|
|
|
q = stream->current_quality->data;
|
|
return q->bitrate;
|
|
}
|
|
|
|
/**
|
|
* gst_mss_manifest_change_bitrate:
|
|
* @manifest: the manifest
|
|
* @bitrate: the maximum bitrate to use (bps)
|
|
*
|
|
* Iterates over the active streams and changes their bitrates to the maximum
|
|
* value so that the bitrates of all streams are not larger than
|
|
* @bitrate.
|
|
*
|
|
* Return: %TRUE if any stream changed its bitrate
|
|
*/
|
|
gboolean
|
|
gst_mss_manifest_change_bitrate (GstMssManifest * manifest, guint64 bitrate)
|
|
{
|
|
gboolean ret = FALSE;
|
|
GSList *iter;
|
|
|
|
/* TODO This algorithm currently sets the same bitrate for all streams,
|
|
* it should actually use the sum of all streams bitrates to compare to
|
|
* the target value */
|
|
|
|
if (bitrate == 0) {
|
|
/* use maximum */
|
|
bitrate = G_MAXUINT64;
|
|
}
|
|
|
|
for (iter = gst_mss_manifest_get_streams (manifest); iter;
|
|
iter = g_slist_next (iter)) {
|
|
GstMssStream *stream = iter->data;
|
|
if (stream->active) {
|
|
ret = ret | gst_mss_stream_select_bitrate (stream, bitrate);
|
|
}
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstBuffer *
|
|
gst_buffer_from_hex_string (const gchar * s)
|
|
{
|
|
GstBuffer *buffer = NULL;
|
|
gint len;
|
|
gchar ts[3];
|
|
guint8 *data;
|
|
gint i;
|
|
GstMapInfo info;
|
|
|
|
len = strlen (s);
|
|
if (len & 1)
|
|
return NULL;
|
|
|
|
buffer = gst_buffer_new_and_alloc (len / 2);
|
|
gst_buffer_map (buffer, &info, GST_MAP_WRITE);
|
|
data = info.data;
|
|
for (i = 0; i < len / 2; i++) {
|
|
if (!isxdigit ((int) s[i * 2]) || !isxdigit ((int) s[i * 2 + 1])) {
|
|
gst_buffer_unref (buffer);
|
|
return NULL;
|
|
}
|
|
|
|
ts[0] = s[i * 2 + 0];
|
|
ts[1] = s[i * 2 + 1];
|
|
ts[2] = 0;
|
|
|
|
data[i] = (guint8) strtoul (ts, NULL, 16);
|
|
}
|
|
|
|
gst_buffer_unmap (buffer, &info);
|
|
return buffer;
|
|
}
|
|
|
|
const gchar *
|
|
gst_mss_stream_get_lang (GstMssStream * stream)
|
|
{
|
|
return stream->lang;
|
|
}
|