[FFmpeg-devel] [PATCH] Make RM demuxer behave better with -an option

[Ronald S. Bultje]

Hi,

On Mon, Mar 2, 2009 at 11:15 AM, Ronald S. Bultje <rsbultje at gmail.com> wrote:
> On Mon, Mar 2, 2009 at 10:07 AM, Kostya <kostya.shishkov at gmail.com> wrote:
>> oh, and I think returning
>> first subpacket from reassembled audio frame is not needed too (cache will
>> handle it).
>
> I'll implement this in a bit.

Attached is a patch doing this. It removes code duplication in
ff_rm_parse_packet() vs. ff_rm_retrieve_cache(). rdt.c needed a small
update to continue working in the same way as it did before. Let me
know if you think the goto-thing is sloppy and I'll get rid of that.

Ronald
-------------- next part --------------
Index: ffmpeg-svn/libavformat/rdt.c
===================================================================
--- ffmpeg-svn.orig/libavformat/rdt.c	2009-03-08 12:51:49.000000000 -0400
+++ ffmpeg-svn/libavformat/rdt.c	2009-03-08 12:55:41.000000000 -0400
@@ -312,14 +312,16 @@
         pos = url_ftell(&pb);
         if (res < 0)
             return res;
-        rdt->audio_pkt_cnt = res;
-        if (rdt->audio_pkt_cnt > 0 &&
-            st->codec->codec_id == CODEC_ID_AAC) {
+        if (res > 0) {
+            if (st->codec->codec_id == CODEC_ID_AAC) {
             memcpy (rdt->buffer, buf + pos, len - pos);
             rdt->rmctx->pb = av_alloc_put_byte (rdt->buffer, len - pos, 0,
                                                 NULL, NULL, NULL, NULL);
+            }
+            goto get_cache;
         }
     } else {
+get_cache:
         rdt->audio_pkt_cnt =
             ff_rm_retrieve_cache (rdt->rmctx, rdt->rmctx->pb,
                                   st, rdt->rmst[st->index], pkt);
Index: ffmpeg-svn/libavformat/rmdec.c
===================================================================
--- ffmpeg-svn.orig/libavformat/rmdec.c	2009-03-08 12:52:05.000000000 -0400
+++ ffmpeg-svn/libavformat/rmdec.c	2009-03-08 12:55:49.000000000 -0400
@@ -614,16 +614,9 @@
 
             if (++(ast->sub_packet_cnt) < h)
                 return -1;
-            else {
                 ast->sub_packet_cnt = 0;
                 rm->audio_stream_num = st->index;
-                rm->audio_pkt_cnt = h * w / 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
-                *timestamp = ast->audiotimestamp;
-                *flags = 2; // Mark first packet as keyframe
-            }
+                rm->audio_pkt_cnt = h * w / st->codec->block_align;
         } else if (st->codec->codec_id == CODEC_ID_AAC) {
             int x;
             rm->audio_stream_num = st->index;
@@ -631,11 +624,10 @@
             if (ast->sub_packet_cnt) {
                 for (x = 0; x < ast->sub_packet_cnt; x++)
                     ast->sub_packet_lengths[x] = get_be16(pb);
-                // Release first audio packet
-                rm->audio_pkt_cnt = ast->sub_packet_cnt - 1;
-                av_get_packet(pb, pkt, ast->sub_packet_lengths[0]);
-                *flags = 2; // Mark first packet as keyframe
-            }
+                rm->audio_pkt_cnt = ast->sub_packet_cnt;
+                ast->audiotimestamp = *timestamp;
+            } else
+                return -1;
         } else {
             av_get_packet(pb, pkt, len);
             rm_ac3_swap_bytes(st, pkt);
@@ -682,6 +674,10 @@
                st->codec->block_align);
     }
     rm->audio_pkt_cnt--;
+    if ((pkt->pts = ast->audiotimestamp) != AV_NOPTS_VALUE) {
+        ast->audiotimestamp = AV_NOPTS_VALUE;
+        pkt->flags = PKT_FLAG_KEY;
+    } else
     pkt->flags = 0;
     pkt->stream_index = st->index;
 
@@ -692,17 +688,18 @@
 {
     RMDemuxContext *rm = s->priv_data;
     AVStream *st;
-    int i, len, seq = 1;
+    int res, i, len, seq = 1;
     int64_t timestamp, pos;
     int flags;
 
     if (rm->audio_pkt_cnt) {
+get_cache:
         // If there are queued audio packet return them first
         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 = s->streams[0]->priv_data;
-        int res, y, h = ast->audio_framesize ? ast->sub_packet_h : 1;
+        int 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;
@@ -713,6 +710,8 @@
         }
         if (res < 0)
             return res;
+        if (res > 0)
+            goto get_cache;
     } else {
 resync:
         len=sync(s, &timestamp, &flags, &i, &pos);
@@ -720,9 +719,11 @@
             return AVERROR(EIO);
         st = s->streams[i];
 
-        if (ff_rm_parse_packet (s, s->pb, st, st->priv_data, len, pkt,
-                                &seq, &flags, &timestamp) < 0)
+        if ((res = ff_rm_parse_packet (s, s->pb, st, st->priv_data, len, pkt,
+                                       &seq, &flags, &timestamp)) < 0)
             goto resync;
+        if (res > 0)
+            goto get_cache;
 
         if(  (st->discard >= AVDISCARD_NONKEY && !(flags&2))
            || st->discard >= AVDISCARD_ALL){