changeset 1431:2d8a17631520 libavformat

fix more dynamic protocol stuff, needed by the forthcoming h264 streaming patch. (Minor additions to give more information to the dynamic protocol handlers, and a slight rearrangement of code.) Patch by Ryan Martell %rdm4 A martellventures P com% Original thread: Date: Oct 29, 2006 2:30 AM Subject: Re: [Ffmpeg-devel] RTP patches & RFC

--- a/rtp.c	Sat Oct 28 18:39:16 2006 +0000
+++ b/rtp.c	Sun Oct 29 10:58:51 2006 +0000
@@ -328,6 +328,7 @@
  * 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)
+ * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
  */
 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
 {
@@ -419,6 +420,39 @@
 }
 
 /**
+ * This was the second switch in rtp_parse packet.  Normalizes time, if required, sets stream_index, etc.
+ */
+static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
+{
+    switch(s->st->codec->codec_id) {
+        case CODEC_ID_MP2:
+        case CODEC_ID_MPEG1VIDEO:
+            if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
+                int64_t addend;
+
+                int delta_timestamp;
+                /* 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;
+        case CODEC_ID_MPEG4AAC:
+        case CODEC_ID_H264:
+        case CODEC_ID_MPEG4:
+            pkt->pts = timestamp;
+            break;
+        default:
+            /* no timestamp info yet */
+            break;
+    }
+    pkt->stream_index = s->st->index;
+}
+
+/**
  * Parse an RTP or RTCP packet directly sent as a buffer.
  * @param s RTP parse context.
  * @param pkt returned packet
@@ -431,15 +465,20 @@
                      const uint8_t *buf, int len)
 {
     unsigned int ssrc, h;
-    int payload_type, seq, delta_timestamp, ret;
+    int payload_type, seq, ret;
     AVStream *st;
     uint32_t timestamp;
+    int rv= 0;
 
     if (!buf) {
         /* return the next packets, if any */
         if(s->st && s->parse_packet) {
-            return s->parse_packet(s, pkt, 0, NULL, 0);
+            timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
+            rv= s->parse_packet(s, pkt, &timestamp, NULL, 0);
+            finalize_packet(s, pkt, timestamp);
+            return rv;
         } else {
+            // TODO: Move to a dynamic packet handler (like above)
             if (s->read_buf_index >= s->read_buf_size)
                 return -1;
             ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
@@ -548,12 +587,11 @@
             }
             s->read_buf_size = len;
             s->buf_ptr = buf;
-            pkt->stream_index = s->st->index;
-            return 0; ///< Temporary return.
+            rv= 0;
             break;
         default:
             if(s->parse_packet) {
-                return s->parse_packet(s, pkt, timestamp, buf, len);
+                rv= s->parse_packet(s, pkt, &timestamp, buf, len);
             } else {
                 av_new_packet(pkt, len);
                 memcpy(pkt->data, buf, len);
@@ -561,32 +599,10 @@
             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;
-            }
-            break;
-        case CODEC_ID_MPEG4AAC:
-        case CODEC_ID_H264:
-        case CODEC_ID_MPEG4:
-            pkt->pts = timestamp;
-            break;
-        default:
-            /* no timestamp info yet */
-            break;
-        }
-        pkt->stream_index = s->st->index;
+        // now perform timestamp things....
+        finalize_packet(s, pkt, timestamp);
     }
-    return 0;
+    return rv;
 }
 
 void rtp_parse_close(RTPDemuxContext *s)

--- a/rtp_internal.h	Sat Oct 28 18:39:16 2006 +0000
+++ b/rtp_internal.h	Sun Oct 29 10:58:51 2006 +0000
@@ -25,7 +25,7 @@
 
 typedef int (*DynamicPayloadPacketHandlerProc) (struct RTPDemuxContext * s,
                                                 AVPacket * pkt,
-                                                uint32_t timestamp,
+                                                uint32_t *timestamp,
                                                 const uint8_t * buf,
                                                 int len);
 

--- a/rtsp.c	Sat Oct 28 18:39:16 2006 +0000
+++ b/rtsp.c	Sun Oct 29 10:58:51 2006 +0000
@@ -200,6 +200,8 @@
                     i = atoi(buf);
                     if (i > 0)
                         codec->channels = i;
+                    // TODO: there is a bug here; if it is a mono stream, and less than 22000Hz, faad upconverts to stereo and twice the
+                    //  frequency.  No problem, but the sample rate is being set here by the sdp line.  Upcoming patch forthcoming. (rdm)
                 }
                 av_log(codec, AV_LOG_DEBUG, " audio samplerate set to : %i\n", codec->sample_rate);
                 av_log(codec, AV_LOG_DEBUG, " audio channels set to : %i\n", codec->channels);
@@ -287,6 +289,25 @@
     {NULL, -1, -1},
 };
 
+/** parse the attribute line from the fmtp a line of an sdp resonse.  This is broken out as a function
+* because it is used in rtp_h264.c, which is forthcoming.
+*/
+int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
+{
+    skip_spaces(p);
+    if(**p)
+    {
+        get_word_sep(attr, attr_size, "=", p);
+        if (**p == '=')
+            (*p)++;
+        get_word_sep(value, value_size, ";", p);
+        if (**p == ';')
+            (*p)++;
+        return 1;
+    }
+    return 0;
+}
+
 /* parse a SDP line and save stream attributes */
 static void sdp_parse_fmtp(AVStream *st, const char *p)
 {
@@ -298,6 +319,7 @@
     AVCodecContext *codec = st->codec;
     rtp_payload_data_t *rtp_payload_data = &rtsp_st->rtp_payload_data;
 
+    // TODO (Replace with rtsp_next_attr_and_value)
     /* loop on each attribute */
     for(;;) {
         skip_spaces(&p);
@@ -471,6 +493,19 @@
                     }
                 }
             }
+        } else if(strstart(p, "framesize:", &p)) {
+            // let dynamic protocol handlers have a stab at the line.
+            get_word(buf1, sizeof(buf1), &p);
+            payload_type = atoi(buf1);
+            for(i = 0; i < s->nb_streams;i++) {
+                st = s->streams[i];
+                rtsp_st = st->priv_data;
+                if (rtsp_st->sdp_payload_type == payload_type) {
+                    if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) {
+                        rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf);
+                    }
+                }
+            }
         }
         break;
     }