avformat/rtsp: Use avio_closep() to avoid leaving stale pointers in memory

[ffmpeg] / libavformat / rtpenc.c

diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index c608dfb46af27e06ca1daebd582e16dcda7cd1d4..c4c4d3e41614fd894a8a85551d41e4522c61e259 100644 (file)
--- a/libavformat/rtpenc.c
+++ b/libavformat/rtpenc.c
@@ -89,7 +89,7 @@ static int is_supported(enum AVCodecID id)
 static int rtp_write_header(AVFormatContext *s1)
 {
     RTPMuxContext *s = s1->priv_data;
-    int n;
+    int n, ret = AVERROR(EINVAL);
     AVStream *st;
 
     if (s1->nb_streams != 1) {
@@ -195,6 +195,17 @@ static int rtp_write_header(AVFormatContext *s1)
         s->max_payload_size = n * TS_PACKET_SIZE;
         s->buf_ptr = s->buf;
         break;
+    case AV_CODEC_ID_H261:
+        if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
+            av_log(s, AV_LOG_ERROR,
+                   "Packetizing H261 is experimental and produces incorrect "
+                   "packetization for cases where GOBs don't fit into packets "
+                   "(even though most receivers may handle it just fine). "
+                   "Please set -f_strict experimental in order to enable it.\n");
+            ret = AVERROR_EXPERIMENTAL;
+            goto fail;
+        }
+        break;
     case AV_CODEC_ID_H264:
         /* check for H.264 MP4 syntax */
         if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
@@ -259,8 +270,11 @@ static int rtp_write_header(AVFormatContext *s1)
             av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
             goto fail;
         }
+        s->num_frames = 0;
+        goto defaultcase;
     case AV_CODEC_ID_AAC:
         s->num_frames = 0;
+        goto defaultcase;
     default:
 defaultcase:
         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
@@ -274,7 +288,7 @@ defaultcase:
 
 fail:
     av_freep(&s->buf);
-    return AVERROR(EINVAL);
+    return ret;
 }
 
 /* send an rtcp sender report packet */
@@ -456,6 +470,7 @@ static void rtp_send_mpegts_raw(AVFormatContext *s1,
     RTPMuxContext *s = s1->priv_data;
     int len, out_len;
 
+    s->timestamp = s->cur_timestamp;
     while (size >= TS_PACKET_SIZE) {
         len = s->max_payload_size - (s->buf_ptr - s->buf);
         if (len > size)
@@ -642,4 +657,5 @@ AVOutputFormat ff_rtp_muxer = {
     .write_packet      = rtp_write_packet,
     .write_trailer     = rtp_write_trailer,
     .priv_class        = &rtp_muxer_class,
+    .flags             = AVFMT_TS_NONSTRICT,
 };