# HG changeset patch
# User bellard
# Date 1067437527 0
# Node ID 6091b76cfc2ae20f9722e7848c788bbb70f21925
# Parent  62cec412a1866718ff9f756126febf7b9fcb32f3
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API

diff -r 62cec412a186 -r 6091b76cfc2a rtp.c
--- a/rtp.c	Wed Oct 29 14:20:56 2003 +0000
+++ b/rtp.c	Wed Oct 29 14:25:27 2003 +0000
@@ -17,6 +17,7 @@
  * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
  */
 #include "avformat.h"
+#include "mpegts.h"
 
 #include <unistd.h>
 #include <sys/types.h>
@@ -72,23 +73,9 @@
   RTCP_SDES_SOURCE = 11
 } rtcp_sdes_type_t;
 
-enum RTPPayloadType {
-    RTP_PT_ULAW = 0,
-    RTP_PT_GSM = 3,
-    RTP_PT_G723 = 4,
-    RTP_PT_ALAW = 8,
-    RTP_PT_S16BE_STEREO = 10,
-    RTP_PT_S16BE_MONO = 11,
-    RTP_PT_MPEGAUDIO = 14,
-    RTP_PT_JPEG = 26,
-    RTP_PT_H261 = 31,
-    RTP_PT_MPEGVIDEO = 32,
-    RTP_PT_MPEG2TS = 33,
-    RTP_PT_H263 = 34, /* old H263 encapsulation */
-    RTP_PT_PRIVATE = 96,
-};
-
-typedef struct RTPContext {
+struct RTPDemuxContext {
+    AVFormatContext *ic;
+    AVStream *st;
     int payload_type;
     uint32_t ssrc;
     uint16_t seq;
@@ -96,6 +83,10 @@
     uint32_t base_timestamp;
     uint32_t cur_timestamp;
     int max_payload_size;
+    MpegTSContext *ts; /* only used for RTP_PT_MPEG2TS payloads */
+    int read_buf_index;
+    int read_buf_size;
+    
     /* rtcp sender statistics receive */
     int64_t last_rtcp_ntp_time;
     int64_t first_rtcp_ntp_time;
@@ -108,40 +99,51 @@
     /* buffer for output */
     uint8_t buf[RTP_MAX_PACKET_LENGTH];
     uint8_t *buf_ptr;
-} RTPContext;
+};
 
 int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
 {
     switch(payload_type) {
     case RTP_PT_ULAW:
+        codec->codec_type = CODEC_TYPE_AUDIO;
         codec->codec_id = CODEC_ID_PCM_MULAW;
         codec->channels = 1;
         codec->sample_rate = 8000;
         break;
     case RTP_PT_ALAW:
+        codec->codec_type = CODEC_TYPE_AUDIO;
         codec->codec_id = CODEC_ID_PCM_ALAW;
         codec->channels = 1;
         codec->sample_rate = 8000;
         break;
     case RTP_PT_S16BE_STEREO:
+        codec->codec_type = CODEC_TYPE_AUDIO;
         codec->codec_id = CODEC_ID_PCM_S16BE;
         codec->channels = 2;
         codec->sample_rate = 44100;
         break;
     case RTP_PT_S16BE_MONO:
+        codec->codec_type = CODEC_TYPE_AUDIO;
         codec->codec_id = CODEC_ID_PCM_S16BE;
         codec->channels = 1;
         codec->sample_rate = 44100;
         break;
     case RTP_PT_MPEGAUDIO:
+        codec->codec_type = CODEC_TYPE_AUDIO;
         codec->codec_id = CODEC_ID_MP2;
         break;
     case RTP_PT_JPEG:
+        codec->codec_type = CODEC_TYPE_VIDEO;
         codec->codec_id = CODEC_ID_MJPEG;
         break;
     case RTP_PT_MPEGVIDEO:
+        codec->codec_type = CODEC_TYPE_VIDEO;
         codec->codec_id = CODEC_ID_MPEG1VIDEO;
         break;
+    case RTP_PT_MPEG2TS:
+        codec->codec_type = CODEC_TYPE_DATA;
+        codec->codec_id = CODEC_ID_MPEG2TS;
+        break;
     default:
         return -1;
     }
@@ -179,6 +181,9 @@
     case CODEC_ID_MPEG1VIDEO:
         payload_type = RTP_PT_MPEGVIDEO;
         break;
+    case CODEC_ID_MPEG2TS:
+        payload_type = RTP_PT_MPEG2TS;
+        break;
     default:
         break;
     }
@@ -195,10 +200,8 @@
     return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4);
 }
 
-static int rtcp_parse_packet(AVFormatContext *s1, const unsigned char *buf, int len)
+static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
 {
-    RTPContext *s = s1->priv_data;
-
     if (buf[1] != 200)
         return -1;
     s->last_rtcp_ntp_time = decode_be64(buf + 8);
@@ -209,30 +212,71 @@
 }
 
 /**
- * Parse an RTP packet directly sent as raw data. Can only be used if
- * 'raw' is given as input file
- * @param s1 media file context
+ * open a new RTP parse context for stream 'st'. 'st' can be NULL for
+ * MPEG2TS streams to indicate that they should be demuxed inside the
+ * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) 
+ */
+RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type)
+{
+    RTPDemuxContext *s;
+
+    s = av_mallocz(sizeof(RTPDemuxContext));
+    if (!s)
+        return NULL;
+    s->payload_type = payload_type;
+    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
+    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
+    s->ic = s1;
+    s->st = st;
+    if (payload_type == RTP_PT_MPEG2TS) {
+        s->ts = mpegts_parse_open(s->ic);
+        if (s->ts == NULL) {
+            av_free(s);
+            return NULL;
+        }
+    }
+    return s;
+}
+
+/**
+ * Parse an RTP or RTCP packet directly sent as a buffer. 
+ * @param s RTP parse context.
  * @param pkt returned packet
- * @param buf input buffer
+ * @param buf input buffer or NULL to read the next packets
  * @param len buffer len
- * @return zero if no error.
+ * @return 0 if a packet is returned, 1 if a packet is returned and more can follow 
+ * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
  */
-int rtp_parse_packet(AVFormatContext *s1, AVPacket *pkt, 
-                     const unsigned char *buf, int len)
+int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, 
+                     const uint8_t *buf, int len)
 {
-    RTPContext *s = s1->priv_data;
     unsigned int ssrc, h;
-    int payload_type, seq, delta_timestamp;
+    int payload_type, seq, delta_timestamp, ret;
     AVStream *st;
     uint32_t timestamp;
     
+    if (!buf) {
+        /* return the next packets, if any */
+        if (s->read_buf_index >= s->read_buf_size)
+            return -1;
+        ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, 
+                                  s->read_buf_size - s->read_buf_index);
+        if (ret < 0)
+            return -1;
+        s->read_buf_index += ret;
+        if (s->read_buf_index < s->read_buf_size)
+            return 1;
+        else
+            return 0;
+    }
+
     if (len < 12)
         return -1;
 
     if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
         return -1;
     if (buf[1] >= 200 && buf[1] <= 204) {
-        rtcp_parse_packet(s1, buf, len);
+        rtcp_parse_packet(s, buf, len);
         return -1;
     }
     payload_type = buf[1] & 0x7f;
@@ -240,20 +284,6 @@
     timestamp = decode_be32(buf + 4);
     ssrc = decode_be32(buf + 8);
     
-    if (s->payload_type < 0) {
-        s->payload_type = payload_type;
-        
-        if (payload_type == RTP_PT_MPEG2TS) {
-            /* XXX: special case : not a single codec but a whole stream */
-            return -1;
-        } else {
-            st = av_new_stream(s1, 0);
-            if (!st)
-                return -1;
-            rtp_get_codec_info(&st->codec, payload_type);
-        }
-    }
-
     /* NOTE: we can handle only one payload type */
     if (s->payload_type != payload_type)
         return -1;
@@ -266,107 +296,91 @@
 #endif
     len -= 12;
     buf += 12;
-    st = s1->streams[0];
-    switch(st->codec.codec_id) {
-    case CODEC_ID_MP2:
-        /* better than nothing: skip mpeg audio RTP header */
-        if (len <= 4)
+
+    st = s->st;
+    if (!st) {
+        /* specific MPEG2TS demux support */
+        ret = mpegts_parse_packet(s->ts, pkt, buf, len);
+        if (ret < 0)
             return -1;
-        h = decode_be32(buf);
-        len -= 4;
-        buf += 4;
-        av_new_packet(pkt, len);
-        memcpy(pkt->data, buf, len);
-        break;
-    case CODEC_ID_MPEG1VIDEO:
-        /* better than nothing: skip mpeg audio RTP header */
-        if (len <= 4)
-            return -1;
-        h = decode_be32(buf);
-        buf += 4;
-        len -= 4;
-        if (h & (1 << 26)) {
-            /* mpeg2 */
+        if (ret < len) {
+            s->read_buf_size = len - ret;
+            memcpy(s->buf, buf + ret, s->read_buf_size);
+            s->read_buf_index = 0;
+            return 1;
+        }
+    } else {
+        switch(st->codec.codec_id) {
+        case CODEC_ID_MP2:
+            /* better than nothing: skip mpeg audio RTP header */
             if (len <= 4)
                 return -1;
+            h = decode_be32(buf);
+            len -= 4;
+            buf += 4;
+            av_new_packet(pkt, len);
+            memcpy(pkt->data, buf, len);
+            break;
+        case CODEC_ID_MPEG1VIDEO:
+            /* better than nothing: skip mpeg audio RTP header */
+            if (len <= 4)
+                return -1;
+            h = decode_be32(buf);
             buf += 4;
             len -= 4;
+            if (h & (1 << 26)) {
+                /* mpeg2 */
+                if (len <= 4)
+                    return -1;
+                buf += 4;
+                len -= 4;
+            }
+            av_new_packet(pkt, len);
+            memcpy(pkt->data, buf, len);
+            break;
+        default:
+            av_new_packet(pkt, len);
+            memcpy(pkt->data, buf, len);
+            break;
         }
-        av_new_packet(pkt, len);
-        memcpy(pkt->data, buf, len);
-        break;
-    default:
-        av_new_packet(pkt, len);
-        memcpy(pkt->data, buf, len);
-        break;
-    }
-
-    switch(st->codec.codec_id) {
-    case CODEC_ID_MP2:
-    case CODEC_ID_MPEG1VIDEO:
-        if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
-            int64_t addend;
-            /* XXX: is it really necessary to unify the timestamp base ? */
-            /* compute pts from timestamp with received ntp_time */
-            delta_timestamp = timestamp - s->last_rtcp_timestamp;
-            /* convert to 90 kHz without overflow */
-            addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
-            addend = (addend * 5625) >> 14;
-            pkt->pts = addend + delta_timestamp;
+        
+        switch(st->codec.codec_id) {
+        case CODEC_ID_MP2:
+        case CODEC_ID_MPEG1VIDEO:
+            if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
+                int64_t addend;
+                /* XXX: is it really necessary to unify the timestamp base ? */
+                /* compute pts from timestamp with received ntp_time */
+                delta_timestamp = timestamp - s->last_rtcp_timestamp;
+                /* convert to 90 kHz without overflow */
+                addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
+                addend = (addend * 5625) >> 14;
+                pkt->pts = addend + delta_timestamp;
+            }
+            break;
+        default:
+            /* no timestamp info yet */
+            break;
         }
-        break;
-    default:
-        /* no timestamp info yet */
-        break;
+        pkt->stream_index = s->st->index;
     }
     return 0;
 }
 
-static int rtp_read_header(AVFormatContext *s1,
-                           AVFormatParameters *ap)
-{
-    RTPContext *s = s1->priv_data;
-    s->payload_type = -1;
-    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
-    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
-    return 0;
-}
-
-static int rtp_read_packet(AVFormatContext *s1, AVPacket *pkt)
+void rtp_parse_close(RTPDemuxContext *s)
 {
-    char buf[RTP_MAX_PACKET_LENGTH];
-    int ret;
-
-    /* XXX: needs a better API for packet handling ? */
-    for(;;) {
-        ret = url_read(url_fileno(&s1->pb), buf, sizeof(buf));
-        if (ret < 0)
-            return AVERROR_IO;
-        if (rtp_parse_packet(s1, pkt, buf, ret) == 0)
-            break;
+    if (s->payload_type == RTP_PT_MPEG2TS) {
+        mpegts_parse_close(s->ts);
     }
-    return 0;
-}
-
-static int rtp_read_close(AVFormatContext *s1)
-{
-    //    RTPContext *s = s1->priv_data;
-    return 0;
-}
-
-static int rtp_probe(AVProbeData *p)
-{
-    if (strstart(p->filename, "rtp://", NULL))
-        return AVPROBE_SCORE_MAX;
-    return 0;
+    av_free(s);
 }
 
 /* rtp output */
 
 static int rtp_write_header(AVFormatContext *s1)
 {
-    RTPContext *s = s1->priv_data;
-    int payload_type, max_packet_size;
+    RTPDemuxContext *s = s1->priv_data;
+    int payload_type, max_packet_size, n;
     AVStream *st;
 
     if (s1->nb_streams != 1)
@@ -397,6 +411,13 @@
     case CODEC_ID_MPEG1VIDEO:
         s->cur_timestamp = 0;
         break;
+    case CODEC_ID_MPEG2TS:
+        n = s->max_payload_size / TS_PACKET_SIZE;
+        if (n < 1)
+            n = 1;
+        s->max_payload_size = n * TS_PACKET_SIZE;
+        s->buf_ptr = s->buf;
+        break;
     default:
         s->buf_ptr = s->buf;
         break;
@@ -408,7 +429,7 @@
 /* send an rtcp sender report packet */
 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
 {
-    RTPContext *s = s1->priv_data;
+    RTPDemuxContext *s = s1->priv_data;
 #if defined(DEBUG)
     printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
 #endif
@@ -427,7 +448,7 @@
    must update the timestamp itself */
 static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len)
 {
-    RTPContext *s = s1->priv_data;
+    RTPDemuxContext *s = s1->priv_data;
 
 #ifdef DEBUG
     printf("rtp_send_data size=%d\n", len);
@@ -453,7 +474,7 @@
 static void rtp_send_samples(AVFormatContext *s1,
                              const uint8_t *buf1, int size, int sample_size)
 {
-    RTPContext *s = s1->priv_data;
+    RTPDemuxContext *s = s1->priv_data;
     int len, max_packet_size, n;
 
     max_packet_size = (s->max_payload_size / sample_size) * sample_size;
@@ -486,7 +507,7 @@
 static void rtp_send_mpegaudio(AVFormatContext *s1,
                                const uint8_t *buf1, int size)
 {
-    RTPContext *s = s1->priv_data;
+    RTPDemuxContext *s = s1->priv_data;
     AVStream *st = s1->streams[0];
     int len, count, max_packet_size;
 
@@ -542,7 +563,7 @@
 static void rtp_send_mpegvideo(AVFormatContext *s1,
                                const uint8_t *buf1, int size)
 {
-    RTPContext *s = s1->priv_data;
+    RTPDemuxContext *s = s1->priv_data;
     AVStream *st = s1->streams[0];
     int len, h, max_packet_size;
     uint8_t *q;
@@ -589,7 +610,7 @@
 static void rtp_send_raw(AVFormatContext *s1,
                          const uint8_t *buf1, int size)
 {
-    RTPContext *s = s1->priv_data;
+    RTPDemuxContext *s = s1->priv_data;
     AVStream *st = s1->streams[0];
     int len, max_packet_size;
 
@@ -611,11 +632,35 @@
     s->cur_timestamp++;
 }
 
+/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
+static void rtp_send_mpegts_raw(AVFormatContext *s1,
+                                const uint8_t *buf1, int size)
+{
+    RTPDemuxContext *s = s1->priv_data;
+    int len, out_len;
+
+    while (size >= TS_PACKET_SIZE) {
+        len = s->max_payload_size - (s->buf_ptr - s->buf);
+        if (len > size)
+            len = size;
+        memcpy(s->buf_ptr, buf1, len);
+        buf1 += len;
+        size -= len;
+        s->buf_ptr += len;
+        
+        out_len = s->buf_ptr - s->buf;
+        if (out_len >= s->max_payload_size) {
+            rtp_send_data(s1, s->buf, out_len);
+            s->buf_ptr = s->buf;
+        }
+    }
+}
+
 /* write an RTP packet. 'buf1' must contain a single specific frame. */
 static int rtp_write_packet(AVFormatContext *s1, int stream_index,
                             const uint8_t *buf1, int size, int64_t pts)
 {
-    RTPContext *s = s1->priv_data;
+    RTPDemuxContext *s = s1->priv_data;
     AVStream *st = s1->streams[0];
     int rtcp_bytes;
     int64_t ntp_time;
@@ -656,6 +701,9 @@
     case CODEC_ID_MPEG1VIDEO:
         rtp_send_mpegvideo(s1, buf1, size);
         break;
+    case CODEC_ID_MPEG2TS:
+        rtp_send_mpegts_raw(s1, buf1, size);
+        break;
     default:
         /* better than nothing : send the codec raw data */
         rtp_send_raw(s1, buf1, size);
@@ -666,27 +714,16 @@
 
 static int rtp_write_trailer(AVFormatContext *s1)
 {
-    //    RTPContext *s = s1->priv_data;
+    //    RTPDemuxContext *s = s1->priv_data;
     return 0;
 }
 
-AVInputFormat rtp_demux = {
-    "rtp",
-    "RTP input format",
-    sizeof(RTPContext),    
-    rtp_probe,
-    rtp_read_header,
-    rtp_read_packet,
-    rtp_read_close,
-    .flags = AVFMT_NOHEADER,
-};
-
 AVOutputFormat rtp_mux = {
     "rtp",
     "RTP output format",
     NULL,
     NULL,
-    sizeof(RTPContext),
+    sizeof(RTPDemuxContext),
     CODEC_ID_PCM_MULAW,
     CODEC_ID_NONE,
     rtp_write_header,
@@ -697,6 +734,5 @@
 int rtp_init(void)
 {
     av_register_output_format(&rtp_mux);
-    av_register_input_format(&rtp_demux);
     return 0;
 }
diff -r 62cec412a186 -r 6091b76cfc2a rtp.h
--- a/rtp.h	Wed Oct 29 14:20:56 2003 +0000
+++ b/rtp.h	Wed Oct 29 14:25:27 2003 +0000
@@ -19,14 +19,35 @@
 #ifndef RTP_H
 #define RTP_H
 
+enum RTPPayloadType {
+    RTP_PT_ULAW = 0,
+    RTP_PT_GSM = 3,
+    RTP_PT_G723 = 4,
+    RTP_PT_ALAW = 8,
+    RTP_PT_S16BE_STEREO = 10,
+    RTP_PT_S16BE_MONO = 11,
+    RTP_PT_MPEGAUDIO = 14,
+    RTP_PT_JPEG = 26,
+    RTP_PT_H261 = 31,
+    RTP_PT_MPEGVIDEO = 32,
+    RTP_PT_MPEG2TS = 33,
+    RTP_PT_H263 = 34, /* old H263 encapsulation */
+    RTP_PT_PRIVATE = 96,
+};
+
 #define RTP_MIN_PACKET_LENGTH 12
 #define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
 
 int rtp_init(void);
 int rtp_get_codec_info(AVCodecContext *codec, int payload_type);
 int rtp_get_payload_type(AVCodecContext *codec);
-int rtp_parse_packet(AVFormatContext *s1, AVPacket *pkt, 
-                     const unsigned char *buf, int len);
+
+typedef struct RTPDemuxContext RTPDemuxContext;
+
+RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type);
+int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, 
+                     const uint8_t *buf, int len);
+void rtp_parse_close(RTPDemuxContext *s);
 
 extern AVOutputFormat rtp_mux;
 extern AVInputFormat rtp_demux;
diff -r 62cec412a186 -r 6091b76cfc2a rtsp.c
--- a/rtsp.c	Wed Oct 29 14:20:56 2003 +0000
+++ b/rtsp.c	Wed Oct 29 14:25:27 2003 +0000
@@ -33,16 +33,23 @@
 
 typedef struct RTSPState {
     URLContext *rtsp_hd; /* RTSP TCP connexion handle */
+    int nb_rtsp_streams;
+    struct RTSPStream **rtsp_streams;
+
     /* XXX: currently we use unbuffered input */
     //    ByteIOContext rtsp_gb;
     int seq;        /* RTSP command sequence number */
     char session_id[512];
     enum RTSPProtocol protocol;
     char last_reply[2048]; /* XXX: allocate ? */
+    RTPDemuxContext *cur_rtp;
 } RTSPState;
 
 typedef struct RTSPStream {
-    AVFormatContext *ic;
+    URLContext *rtp_handle; /* RTP stream handle */
+    RTPDemuxContext *rtp_ctx; /* RTP parse context */
+    
+    int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */
     int interleaved_min, interleaved_max;  /* interleave ids, if TCP transport */
     char control_url[1024]; /* url for this stream (from SDP) */
 
@@ -218,6 +225,7 @@
 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
                            int letter, const char *buf)
 {
+    RTSPState *rt = s->priv_data;
     char buf1[64], st_type[64];
     const char *p;
     int codec_type, payload_type, i;
@@ -280,16 +288,12 @@
         rtsp_st = av_mallocz(sizeof(RTSPStream));
         if (!rtsp_st)
             return;
-        st = av_new_stream(s, s->nb_streams);
-        if (!st) 
-            return;
-        st->priv_data = rtsp_st;
+        rtsp_st->stream_index = -1;
+        dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
 
         rtsp_st->sdp_ip = s1->default_ip;
         rtsp_st->sdp_ttl = s1->default_ttl;
 
-        st->codec.codec_type = codec_type;
-
         get_word(buf1, sizeof(buf1), &p); /* port */
         rtsp_st->sdp_port = atoi(buf1);
 
@@ -298,11 +302,21 @@
         /* XXX: handle list of formats */
         get_word(buf1, sizeof(buf1), &p); /* format list */
         rtsp_st->sdp_payload_type = atoi(buf1);
-        if (rtsp_st->sdp_payload_type < 96) {
-            /* if standard payload type, we can find the codec right now */
-            rtp_get_codec_info(&st->codec, rtsp_st->sdp_payload_type);
+
+        if (rtsp_st->sdp_payload_type == RTP_PT_MPEG2TS) {
+            /* no corresponding stream */
+        } else {
+            st = av_new_stream(s, 0);
+            if (!st)
+                return;
+            st->priv_data = rtsp_st;
+            rtsp_st->stream_index = st->index;
+            st->codec.codec_type = codec_type;
+            if (rtsp_st->sdp_payload_type < 96) {
+                /* if standard payload type, we can find the codec right now */
+                rtp_get_codec_info(&st->codec, rtsp_st->sdp_payload_type);
+            }
         }
-
         /* put a default control url */
         pstrcpy(rtsp_st->control_url, sizeof(rtsp_st->control_url), s->filename);
         break;
@@ -629,6 +643,25 @@
 }
 
 
+/* close and free RTSP streams */
+static void rtsp_close_streams(RTSPState *rt)
+{
+    int i;
+    RTSPStream *rtsp_st;
+
+    for(i=0;i<rt->nb_rtsp_streams;i++) {
+        rtsp_st = rt->rtsp_streams[i];
+        if (rtsp_st) {
+            if (rtsp_st->rtp_ctx)
+                rtp_parse_close(rtsp_st->rtp_ctx);
+            if (rtsp_st->rtp_handle)
+                url_close(rtsp_st->rtp_handle);
+        }
+        av_free(rtsp_st);
+    }
+    av_free(rt->rtsp_streams);
+}
+
 static int rtsp_read_header(AVFormatContext *s,
                             AVFormatParameters *ap)
 {
@@ -638,9 +671,9 @@
     int port, i, ret, err;
     RTSPHeader reply1, *reply = &reply1;
     unsigned char *content = NULL;
-    AVStream *st;
     RTSPStream *rtsp_st;
     int protocol_mask;
+    AVStream *st;
 
     /* extract hostname and port */
     url_split(NULL, 0,
@@ -683,12 +716,10 @@
     /* for each stream, make the setup request */
     /* XXX: we assume the same server is used for the control of each
        RTSP stream */
-    for(i=0;i<s->nb_streams;i++) {
+    for(i=0;i<rt->nb_rtsp_streams;i++) {
         char transport[2048];
-        AVInputFormat *fmt;
 
-        st = s->streams[i];
-        rtsp_st = st->priv_data;
+        rtsp_st = rt->rtsp_streams[i];
 
         /* compute available transports */
         transport[0] = '\0';
@@ -702,21 +733,19 @@
             if (rtsp_rtp_port_min != 0) {
                 for(j=rtsp_rtp_port_min;j<=rtsp_rtp_port_max;j++) {
                     snprintf(buf, sizeof(buf), "rtp://?localport=%d", j);
-                    if (!av_open_input_file(&rtsp_st->ic, buf, 
-                                            &rtp_demux, 0, NULL))
+                    if (url_open(&rtsp_st->rtp_handle, buf, URL_RDONLY) == 0)
                         goto rtp_opened;
                 }
             }
 
             /* then try on any port */
-            if (av_open_input_file(&rtsp_st->ic, "rtp://", 
-                                       &rtp_demux, 0, NULL) < 0) {
-                    err = AVERROR_INVALIDDATA;
-                    goto fail;
+            if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
+                err = AVERROR_INVALIDDATA;
+                goto fail;
             }
 
         rtp_opened:
-            port = rtp_get_local_port(url_fileno(&rtsp_st->ic->pb));
+            port = rtp_get_local_port(rtsp_st->rtp_handle);
             if (transport[0] != '\0')
                 pstrcat(transport, sizeof(transport), ",");
             snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1,
@@ -763,17 +792,12 @@
         /* close RTP connection if not choosen */
         if (reply->transports[0].protocol != RTSP_PROTOCOL_RTP_UDP &&
             (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP))) {
-            av_close_input_file(rtsp_st->ic);
-            rtsp_st->ic = NULL;
+            url_close(rtsp_st->rtp_handle);
+            rtsp_st->rtp_handle = NULL;
         }
 
         switch(reply->transports[0].protocol) {
         case RTSP_PROTOCOL_RTP_TCP:
-            fmt = &rtp_demux;
-            if (av_open_input_file(&rtsp_st->ic, "null", fmt, 0, NULL) < 0) {
-                err = AVERROR_INVALIDDATA;
-                goto fail;
-            }
             rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
             rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
             break;
@@ -785,7 +809,7 @@
                 /* XXX: also use address if specified */
                 snprintf(url, sizeof(url), "rtp://%s:%d", 
                          host, reply->transports[0].server_port_min);
-                if (rtp_set_remote_url(url_fileno(&rtsp_st->ic->pb), url) < 0) {
+                if (rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
                     err = AVERROR_INVALIDDATA;
                     goto fail;
                 }
@@ -796,7 +820,6 @@
                 char url[1024];
                 int ttl;
 
-                fmt = &rtp_demux;
                 ttl = reply->transports[0].ttl;
                 if (!ttl)
                     ttl = 16;
@@ -804,13 +827,24 @@
                          host, 
                          reply->transports[0].server_port_min,
                          ttl);
-                if (av_open_input_file(&rtsp_st->ic, url, fmt, 0, NULL) < 0) {
+                if (url_open(&rtsp_st->rtp_handle, url, URL_RDONLY) < 0) {
                     err = AVERROR_INVALIDDATA;
                     goto fail;
                 }
             }
             break;
         }
+        /* open the RTP context */
+        st = NULL;
+        if (rtsp_st->stream_index >= 0)
+            st = s->streams[rtsp_st->stream_index];
+        if (!st)
+            s->ctx_flags |= AVFMTCTX_NOHEADER;
+        rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type);
+        if (!rtsp_st->rtp_ctx) {
+            err = AVERROR_NOMEM;
+            goto fail;
+        }
     }
 
     /* use callback if available to extend setup */
@@ -845,28 +879,18 @@
 
     return 0;
  fail:
-    for(i=0;i<s->nb_streams;i++) {
-        st = s->streams[i];
-        rtsp_st = st->priv_data;
-        if (rtsp_st) {
-            if (rtsp_st->ic)
-                av_close_input_file(rtsp_st->ic);
-        }
-        av_free(rtsp_st);
-    }
+    rtsp_close_streams(rt);
     av_freep(&content);
     url_close(rt->rtsp_hd);
     return err;
 }
 
-static int tcp_read_packet(AVFormatContext *s,
-                           AVPacket *pkt)
+static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
+                           uint8_t *buf, int buf_size)
 {
     RTSPState *rt = s->priv_data;
     int id, len, i, ret;
-    AVStream *st;
     RTSPStream *rtsp_st;
-    uint8_t buf[RTP_MAX_PACKET_LENGTH];
 
 #ifdef DEBUG_RTP_TCP
     printf("tcp_read_packet:\n");
@@ -878,84 +902,71 @@
         printf("ret=%d c=%02x [%c]\n", ret, buf[0], buf[0]);
 #endif
         if (ret != 1)
-            return AVERROR_IO;
+            return -1;
         if (buf[0] == '$')
             break;
     }
     ret = url_read(rt->rtsp_hd, buf, 3);
     if (ret != 3)
-        return AVERROR_IO;
+        return -1;
     id = buf[0];
     len = (buf[1] << 8) | buf[2];
 #ifdef DEBUG_RTP_TCP
     printf("id=%d len=%d\n", id, len);
 #endif
-    if (len > RTP_MAX_PACKET_LENGTH || len < 12)
+    if (len > buf_size || len < 12)
         goto redo;
     /* get the data */
     ret = url_read(rt->rtsp_hd, buf, len);
     if (ret != len)
-        return AVERROR_IO;
+        return -1;
         
     /* find the matching stream */
-    for(i = 0; i < s->nb_streams; i++) {
-        st = s->streams[i];
-        rtsp_st = st->priv_data;
+    for(i = 0; i < rt->nb_rtsp_streams; i++) {
+        rtsp_st = rt->rtsp_streams[i];
         if (id >= rtsp_st->interleaved_min && 
             id <= rtsp_st->interleaved_max) 
             goto found;
     }
     goto redo;
  found:
-    ret = rtp_parse_packet(rtsp_st->ic, pkt, buf, len);
-    if (ret < 0)
-        goto redo;
-    pkt->stream_index = i;
-    return ret;
+    *prtsp_st = rtsp_st;
+    return len;
 }
 
-/* NOTE: output one packet at a time. May need to add a small fifo */
-static int udp_read_packet(AVFormatContext *s,
-                           AVPacket *pkt)
+static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, 
+                           uint8_t *buf, int buf_size)
 {
-    AVFormatContext *ic;
-    AVStream *st;
+    RTSPState *rt = s->priv_data;
     RTSPStream *rtsp_st;
     fd_set rfds;
     int fd1, fd2, fd_max, n, i, ret;
-    char buf[RTP_MAX_PACKET_LENGTH];
     struct timeval tv;
 
     for(;;) {
         if (url_interrupt_cb())
-            return -EIO;
+            return -1;
         FD_ZERO(&rfds);
         fd_max = -1;
-        for(i = 0; i < s->nb_streams; i++) {
-            st = s->streams[i];
-            rtsp_st = st->priv_data;
-            ic = rtsp_st->ic;
+        for(i = 0; i < rt->nb_rtsp_streams; i++) {
+            rtsp_st = rt->rtsp_streams[i];
             /* currently, we cannot probe RTCP handle because of blocking restrictions */
-            rtp_get_file_handles(url_fileno(&ic->pb), &fd1, &fd2);
+            rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2);
             if (fd1 > fd_max)
                 fd_max = fd1;
             FD_SET(fd1, &rfds);
         }
-        /* XXX: also add proper API to abort */
         tv.tv_sec = 0;
         tv.tv_usec = 100 * 1000;
         n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
         if (n > 0) {
-            for(i = 0; i < s->nb_streams; i++) {
-                st = s->streams[i];
-                rtsp_st = st->priv_data;
-                ic = rtsp_st->ic;
-                rtp_get_file_handles(url_fileno(&ic->pb), &fd1, &fd2);
+            for(i = 0; i < rt->nb_rtsp_streams; i++) {
+                rtsp_st = rt->rtsp_streams[i];
+                rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2);
                 if (FD_ISSET(fd1, &rfds)) {
-                    ret = url_read(url_fileno(&ic->pb), buf, sizeof(buf));
-                    if (ret >= 0 && 
-                        rtp_parse_packet(ic, pkt, buf, ret) == 0) {
-                        pkt->stream_index = i;
+                    ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
+                    if (ret > 0) {
+                        *prtsp_st = rtsp_st;
                         return ret;
                     }
                 }
@@ -968,18 +979,45 @@
                             AVPacket *pkt)
 {
     RTSPState *rt = s->priv_data;
-    int ret;
+    RTSPStream *rtsp_st;
+    int ret, len;
+    uint8_t buf[RTP_MAX_PACKET_LENGTH];
 
+    /* get next frames from the same RTP packet */
+    if (rt->cur_rtp) {
+        ret = rtp_parse_packet(rt->cur_rtp, pkt, NULL, 0);
+        if (ret == 0) {
+            rt->cur_rtp = NULL;
+            return 0;
+        } else if (ret == 1) {
+            return 0;
+        } else {
+            rt->cur_rtp = NULL;
+        }
+    }
+
+    /* read next RTP packet */
+ redo:
     switch(rt->protocol) {
     default:
     case RTSP_PROTOCOL_RTP_TCP:
-        ret = tcp_read_packet(s, pkt);
+        len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
         break;
     case RTSP_PROTOCOL_RTP_UDP:
-        ret = udp_read_packet(s, pkt);
+    case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
+        len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
         break;
     }
-    return ret;
+    if (len < 0)
+        return AVERROR_IO;
+    ret = rtp_parse_packet(rtsp_st->rtp_ctx, pkt, buf, len);
+    if (ret < 0)
+        goto redo;
+    if (ret == 1) {
+        /* more packets may follow, so we save the RTP context */
+        rt->cur_rtp = rtsp_st->rtp_ctx;
+    }
+    return 0;
 }
 
 /* pause the stream */
@@ -1031,10 +1069,7 @@
 static int rtsp_read_close(AVFormatContext *s)
 {
     RTSPState *rt = s->priv_data;
-    AVStream *st;
-    RTSPStream *rtsp_st;
     RTSPHeader reply1, *reply = &reply1;
-    int i;
     char cmd[1024];
 
 #if 0
@@ -1053,15 +1088,7 @@
                          NULL, 0, NULL);
     }
 
-    for(i=0;i<s->nb_streams;i++) {
-        st = s->streams[i];
-        rtsp_st = st->priv_data;
-        if (rtsp_st) {
-            if (rtsp_st->ic)
-                av_close_input_file(rtsp_st->ic);
-        }
-        av_free(rtsp_st);
-    }
+    rtsp_close_streams(rt);
     url_close(rt->rtsp_hd);
     return 0;
 }
@@ -1101,11 +1128,12 @@
 static int sdp_read_header(AVFormatContext *s,
                            AVFormatParameters *ap)
 {
-    AVStream *st;
+    RTSPState *rt = s->priv_data;
     RTSPStream *rtsp_st;
     int size, i, err;
     char *content;
     char url[1024];
+    AVStream *st;
 
     /* read the whole sdp file */
     /* XXX: better loading */
@@ -1121,54 +1149,45 @@
     av_free(content);
 
     /* open each RTP stream */
-    for(i=0;i<s->nb_streams;i++) {
-        st = s->streams[i];
-        rtsp_st = st->priv_data;
+    for(i=0;i<rt->nb_rtsp_streams;i++) {
+        rtsp_st = rt->rtsp_streams[i];
         
         snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d", 
                  inet_ntoa(rtsp_st->sdp_ip), 
                  rtsp_st->sdp_port,
                  rtsp_st->sdp_ttl);
-        if (av_open_input_file(&rtsp_st->ic, url, &rtp_demux, 0, NULL) < 0) {
+        if (url_open(&rtsp_st->rtp_handle, url, URL_RDONLY) < 0) {
             err = AVERROR_INVALIDDATA;
             goto fail;
         }
+        /* open the RTP context */
+        st = NULL;
+        if (rtsp_st->stream_index >= 0)
+            st = s->streams[rtsp_st->stream_index];
+        if (!st)
+            s->ctx_flags |= AVFMTCTX_NOHEADER;
+        rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type);
+        if (!rtsp_st->rtp_ctx) {
+            err = AVERROR_NOMEM;
+            goto fail;
+        }
     }
     return 0;
  fail:
-    for(i=0;i<s->nb_streams;i++) {
-        st = s->streams[i];
-        rtsp_st = st->priv_data;
-        if (rtsp_st) {
-            if (rtsp_st->ic)
-                av_close_input_file(rtsp_st->ic);
-        }
-        av_free(rtsp_st);
-    }
+    rtsp_close_streams(rt);
     return err;
 }
 
 static int sdp_read_packet(AVFormatContext *s,
                             AVPacket *pkt)
 {
-    return udp_read_packet(s, pkt);
+    return rtsp_read_packet(s, pkt);
 }
 
 static int sdp_read_close(AVFormatContext *s)
 {
-    AVStream *st;
-    RTSPStream *rtsp_st;
-    int i;
-
-    for(i=0;i<s->nb_streams;i++) {
-        st = s->streams[i];
-        rtsp_st = st->priv_data;
-        if (rtsp_st) {
-            if (rtsp_st->ic)
-                av_close_input_file(rtsp_st->ic);
-        }
-        av_free(rtsp_st);
-    }
+    RTSPState *rt = s->priv_data;
+    rtsp_close_streams(rt);
     return 0;
 }