ffmpeg source update for sep 09

source/patched-ffmpeg-mt/libavformat/rdt.c

 /*
  * Realmedia RTSP protocol (RDT) support.
  * Copyright (c) 2007 Ronald S. Bultje
  *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
@@ skipping 68 matching lines @@
     int i;
 
     for (i = 1; i < s->n_streams; i++)
         s->streams[i]->priv_data = NULL;
 
     av_free(s);
 }
 
 struct PayloadContext {
     AVFormatContext *rmctx;
-    RMStream *rmst[MAX_STREAMS];
+    int nb_rmst;
+    RMStream **rmst;
     uint8_t *mlti_data;
     unsigned int mlti_data_size;
     char buffer[RTP_MAX_PACKET_LENGTH + FF_INPUT_BUFFER_PADDING_SIZE];
     int audio_pkt_cnt; /**< remaining audio packets in rmdec */
 };
 
 void
 ff_rdt_calc_response_and_checksum(char response[41], char chksum[9],
                                   const char *challenge)
 {
@@ skipping 314 matching lines @@
 
     if (av_strstart(p, "OpaqueData:buffer;", &p)) {
         rdt->mlti_data = rdt_parse_b64buf(&rdt->mlti_data_size, p);
     } else if (av_strstart(p, "StartTime:integer;", &p))
         stream->first_dts = atoi(p);
     else if (av_strstart(p, "ASMRuleBook:string;", &p)) {
         int n, first = -1;
 
         for (n = 0; n < s->nb_streams; n++)
             if (s->streams[n]->priv_data == stream->priv_data) {
+                int count = s->streams[n]->index + 1;
                 if (first == -1) first = n;
+                if (rdt->nb_rmst < count) {
+                    RMStream **rmst= av_realloc(rdt->rmst, count*sizeof(*rmst));
+                    if (!rmst)
+                        return AVERROR(ENOMEM);
+                    memset(rmst + rdt->nb_rmst, 0,
+                           (count - rdt->nb_rmst) * sizeof(*rmst));
+                    rdt->rmst    = rmst;
+                    rdt->nb_rmst = count;
+                }
                 rdt->rmst[s->streams[n]->index] = ff_rm_alloc_rmstream();
                 rdt_load_mdpr(rdt, s->streams[n], (n - first) * 2);
 
                 if (s->streams[n]->codec->codec_id == CODEC_ID_AAC)
                     s->streams[n]->codec->frame_size = 1; // FIXME
            }
     }
 
     return 0;
 }
@@ skipping 23 matching lines @@
     st->first_dts         = orig_st->first_dts;
 
     return st;
 }
 
 static void
 real_parse_asm_rulebook(AVFormatContext *s, AVStream *orig_st,
                         const char *p)
 {
     const char *end;
-    int n_rules, odd = 0;
+    int n_rules = 0, odd = 0;
     AVStream *st;
 
     /**
      * The ASMRuleBook contains a list of comma-separated strings per rule,
      * and each rule is separated by a ;. The last one also has a ; at the
      * end so we can use it as delimiter.
      * Every rule occurs twice, once for when the RTSP packet header marker
      * is set and once for if it isn't. We only read the first because we
      * don't care much (that's what the "odd" variable is for).
      * Each rule contains a set of one or more statements, optionally
      * preceeded by a single condition. If there's a condition, the rule
      * starts with a '#'. Multiple conditions are merged between brackets,
      * so there are never multiple conditions spread out over separate
      * statements. Generally, these conditions are bitrate limits (min/max)
      * for multi-bitrate streams.
      */
     if (*p == '\"') p++;
-    for (n_rules = 0; s->nb_streams < MAX_STREAMS;) {
+    while (1) {
         if (!(end = strchr(p, ';')))
             break;
         if (!odd && end != p) {
             if (n_rules > 0)
                 st = add_dstream(s, orig_st);
             else
                 st = orig_st;
+            if (!st)
+                break;
             real_parse_asm_rule(st, p, end);
             n_rules++;
         }
         p = end + 1;
         odd ^= 1;
     }
 }
 
 void
 ff_real_parse_sdp_a_line (AVFormatContext *s, int stream_index,
@@ skipping 13 matching lines @@
     av_open_input_stream(&rdt->rmctx, NULL, "", &rdt_demuxer, NULL);
 
     return rdt;
 }
 
 static void
 rdt_free_context (PayloadContext *rdt)
 {
     int i;
 
-    for (i = 0; i < MAX_STREAMS; i++)
+    for (i = 0; i < rdt->nb_rmst; i++)
         if (rdt->rmst[i]) {
             ff_rm_free_rmstream(rdt->rmst[i]);
             av_freep(&rdt->rmst[i]);
         }
     if (rdt->rmctx)
         av_close_input_stream(rdt->rmctx);
     av_freep(&rdt->mlti_data);
+    av_freep(&rdt->rmst);
     av_free(rdt);
 }
 
 #define RDT_HANDLER(n, s, t) \
 static RTPDynamicProtocolHandler ff_rdt_ ## n ## _handler = { \
     .enc_name         = s, \
     .codec_type       = t, \
     .codec_id         = CODEC_ID_NONE, \
     .parse_sdp_a_line = rdt_parse_sdp_line, \
     .open             = rdt_new_context, \
     .close            = rdt_free_context, \
     .parse_packet     = rdt_parse_packet \
 };
 
 RDT_HANDLER(live_video, "x-pn-multirate-realvideo-live", AVMEDIA_TYPE_VIDEO);
 RDT_HANDLER(live_audio, "x-pn-multirate-realaudio-live", AVMEDIA_TYPE_AUDIO);
 RDT_HANDLER(video,      "x-pn-realvideo",                AVMEDIA_TYPE_VIDEO);
 RDT_HANDLER(audio,      "x-pn-realaudio",                AVMEDIA_TYPE_AUDIO);
 
 void av_register_rdt_dynamic_payload_handlers(void)
 {
     ff_register_dynamic_payload_handler(&ff_rdt_video_handler);
     ff_register_dynamic_payload_handler(&ff_rdt_audio_handler);
     ff_register_dynamic_payload_handler(&ff_rdt_live_video_handler);
     ff_register_dynamic_payload_handler(&ff_rdt_live_audio_handler);
 }