Replace separate packet parsing for "old_format" .ra files by a call to

ff_rm_parse_packet(). See "[PATCH] Make RM demuxer behave better with -an
option" thread, which sort-of turned into an aggregate of unrelated rmdec.c
cleanups.



git-svn-id: file:///var/local/repositories/ffmpeg/trunk@17909 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

libavformat/rmdec.c

@@ -691,9 +691,8 @@ ff_rm_retrieve_cache (AVFormatContext *s, ByteIOContext *pb,
 static int rm_read_packet(AVFormatContext *s, AVPacket *pkt)
 {
     RMDemuxContext *rm = s->priv_data;
-    ByteIOContext *pb = s->pb;
     AVStream *st;
-    int i, len;
+    int i, len, seq = 1;
     int64_t timestamp, pos;
     int flags;
 
@@ -702,37 +701,19 @@ static int rm_read_packet(AVFormatContext *s, AVPacket *pkt)
         st = s->streams[rm->audio_stream_num];
         ff_rm_retrieve_cache(s, s->pb, st, st->priv_data, pkt);
     } else if (rm->old_format) {
-        RMStream *ast;
-
-        st = s->streams[0];
-        ast = st->priv_data;
-        if (st->codec->codec_id == CODEC_ID_RA_288) {
-            int x, y;
-
-            for (y = 0; y < ast->sub_packet_h; y++)
-                for (x = 0; x < ast->sub_packet_h/2; x++)
-                    if (get_buffer(pb, ast->pkt.data+x*2*ast->audio_framesize+y*ast->coded_framesize, ast->coded_framesize) <= 0)
-                        return AVERROR(EIO);
-            rm->audio_stream_num = 0;
-            rm->audio_pkt_cnt = ast->sub_packet_h * ast->audio_framesize / st->codec->block_align - 1;
-            // Release first audio packet
-            av_new_packet(pkt, st->codec->block_align);
-            memcpy(pkt->data, ast->pkt.data, st->codec->block_align); //FIXME avoid this
-            pkt->flags |= PKT_FLAG_KEY; // Mark first packet as keyframe
-            pkt->stream_index = 0;
-        } else {
-            /* just read raw bytes */
-            len = RAW_PACKET_SIZE;
-            len= av_get_packet(pb, pkt, len);
-            pkt->stream_index = 0;
-            if (len <= 0) {
-                return AVERROR(EIO);
-            }
-            pkt->size = len;
+        RMStream *ast = s->streams[0]->priv_data;
+        int res, y, h = ast->audio_framesize ? ast->sub_packet_h : 1;
+
+        timestamp = AV_NOPTS_VALUE;
+        len = ast->audio_framesize ? ast->coded_framesize * h / 2 : RAW_PACKET_SIZE;
+        for (y = 0; y < h; y++) {
+            flags = !y ? 2 : 0;
+            res = ff_rm_parse_packet(s, s->pb, s->streams[0], ast, len, pkt,
+                                     &seq, &flags, &timestamp);
         }
-        rm_ac3_swap_bytes(st, pkt);
+        if (res < 0)
+            return res;
     } else {
-        int seq=1;
 resync:
         len=sync(s, &timestamp, &flags, &i, &pos);
         if(len<0)