cosmetics: read_metadata -> read
[ffmpeg.git] / libavformat / rtp.c
index 3aa7eec..6b038c8 100644 (file)
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 #include "avformat.h"
-#include "mpegts.h"
 #include "bitstream.h"
 
 #include <unistd.h>
 #include "network.h"
 
 #include "rtp_internal.h"
-#include "rtp_h264.h"
-#include "rtp_mpv.h"
-#include "rtp_aac.h"
 
 //#define DEBUG
 
-#define RTCP_SR_SIZE 28
-
-/* TODO: - add RTCP statistics reporting (should be optional).
-
-         - add support for h263/mpeg4 packetized output : IDEA: send a
-         buffer to 'rtp_write_packet' contains all the packets for ONE
-         frame. Each packet should have a four byte header containing
-         the length in big endian format (same trick as
-         'url_open_dyn_packet_buf')
-*/
-
 /* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
-AVRtpPayloadType_t AVRtpPayloadTypes[]=
+static const struct
+{
+    int pt;
+    const char enc_name[50]; /* XXX: why 50 ? */
+    enum CodecType codec_type;
+    enum CodecID codec_id;
+    int clock_rate;
+    int audio_channels;
+} AVRtpPayloadTypes[]=
 {
   {0, "PCMU",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_MULAW, 8000, 1},
   {1, "Reserved",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
@@ -79,6 +72,7 @@ AVRtpPayloadType_t AVRtpPayloadTypes[]=
   {30, "unassigned", CODEC_TYPE_VIDEO,   CODEC_ID_NONE, -1, -1},
   {31, "H261",       CODEC_TYPE_VIDEO,   CODEC_ID_H261, 90000, -1},
   {32, "MPV",        CODEC_TYPE_VIDEO,   CODEC_ID_MPEG1VIDEO, 90000, -1},
+  {32, "MPV",        CODEC_TYPE_VIDEO,   CODEC_ID_MPEG2VIDEO, 90000, -1},
   {33, "MP2T",       CODEC_TYPE_DATA,    CODEC_ID_MPEG2TS, 90000, -1},
   {34, "H263",       CODEC_TYPE_VIDEO,   CODEC_ID_H263, 90000, -1},
   {35, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
@@ -177,36 +171,22 @@ AVRtpPayloadType_t AVRtpPayloadTypes[]=
   {-1, "",           CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}
 };
 
-/* statistics functions */
-RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
-
-static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
-static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
-
-static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
-{
-    handler->next= RTPFirstDynamicPayloadHandler;
-    RTPFirstDynamicPayloadHandler= handler;
-}
-
-void av_register_rtp_dynamic_payload_handlers(void)
-{
-    register_dynamic_payload_handler(&mp4v_es_handler);
-    register_dynamic_payload_handler(&mpeg4_generic_handler);
-    register_dynamic_payload_handler(&ff_h264_dynamic_handler);
-}
-
 int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
 {
-    if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) {
-        codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type;
-        codec->codec_id = AVRtpPayloadTypes[payload_type].codec_id;
-        if (AVRtpPayloadTypes[payload_type].audio_channels > 0)
-            codec->channels = AVRtpPayloadTypes[payload_type].audio_channels;
-        if (AVRtpPayloadTypes[payload_type].clock_rate > 0)
-            codec->sample_rate = AVRtpPayloadTypes[payload_type].clock_rate;
-        return 0;
-    }
+    int i = 0;
+
+    for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
+        if (AVRtpPayloadTypes[i].pt == payload_type) {
+            if (AVRtpPayloadTypes[i].codec_id != CODEC_ID_NONE) {
+                codec->codec_type = AVRtpPayloadTypes[i].codec_type;
+                codec->codec_id = AVRtpPayloadTypes[i].codec_id;
+                if (AVRtpPayloadTypes[i].audio_channels > 0)
+                    codec->channels = AVRtpPayloadTypes[i].audio_channels;
+                if (AVRtpPayloadTypes[i].clock_rate > 0)
+                    codec->sample_rate = AVRtpPayloadTypes[i].clock_rate;
+                return 0;
+            }
+        }
     return -1;
 }
 
@@ -225,814 +205,26 @@ int rtp_get_payload_type(AVCodecContext *codec)
     return payload_type;
 }
 
-static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
-{
-    if (buf[1] != 200)
-        return -1;
-    s->last_rtcp_ntp_time = AV_RB64(buf + 8);
-    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
-        s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
-    s->last_rtcp_timestamp = AV_RB32(buf + 16);
-    return 0;
-}
-
-#define RTP_SEQ_MOD (1<<16)
-
-/**
-* called on parse open packet
-*/
-static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
-{
-    memset(s, 0, sizeof(RTPStatistics));
-    s->max_seq= base_sequence;
-    s->probation= 1;
-}
-
-/**
-* called whenever there is a large jump in sequence numbers, or when they get out of probation...
-*/
-static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
-{
-    s->max_seq= seq;
-    s->cycles= 0;
-    s->base_seq= seq -1;
-    s->bad_seq= RTP_SEQ_MOD + 1;
-    s->received= 0;
-    s->expected_prior= 0;
-    s->received_prior= 0;
-    s->jitter= 0;
-    s->transit= 0;
-}
-
-/**
-* returns 1 if we should handle this packet.
-*/
-static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
-{
-    uint16_t udelta= seq - s->max_seq;
-    const int MAX_DROPOUT= 3000;
-    const int MAX_MISORDER = 100;
-    const int MIN_SEQUENTIAL = 2;
-
-    /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
-    if(s->probation)
-    {
-        if(seq==s->max_seq + 1) {
-            s->probation--;
-            s->max_seq= seq;
-            if(s->probation==0) {
-                rtp_init_sequence(s, seq);
-                s->received++;
-                return 1;
-            }
-        } else {
-            s->probation= MIN_SEQUENTIAL - 1;
-            s->max_seq = seq;
-        }
-    } else if (udelta < MAX_DROPOUT) {
-        // in order, with permissible gap
-        if(seq < s->max_seq) {
-            //sequence number wrapped; count antother 64k cycles
-            s->cycles += RTP_SEQ_MOD;
-        }
-        s->max_seq= seq;
-    } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
-        // sequence made a large jump...
-        if(seq==s->bad_seq) {
-            // two sequential packets-- assume that the other side restarted without telling us; just resync.
-            rtp_init_sequence(s, seq);
-        } else {
-            s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
-            return 0;
-        }
-    } else {
-        // duplicate or reordered packet...
-    }
-    s->received++;
-    return 1;
-}
-
-#if 0
-/**
-* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
-* difference between the arrival and sent timestamp.  As a result, the jitter and transit statistics values
-* never change.  I left this in in case someone else can see a way. (rdm)
-*/
-static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
-{
-    uint32_t transit= arrival_timestamp - sent_timestamp;
-    int d;
-    s->transit= transit;
-    d= FFABS(transit - s->transit);
-    s->jitter += d - ((s->jitter + 8)>>4);
-}
-#endif
-
-int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
-{
-    ByteIOContext pb;
-    uint8_t *buf;
-    int len;
-    int rtcp_bytes;
-    RTPStatistics *stats= &s->statistics;
-    uint32_t lost;
-    uint32_t extended_max;
-    uint32_t expected_interval;
-    uint32_t received_interval;
-    uint32_t lost_interval;
-    uint32_t expected;
-    uint32_t fraction;
-    uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
-
-    if (!s->rtp_ctx || (count < 1))
-        return -1;
-
-    /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
-    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
-    s->octet_count += count;
-    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
-        RTCP_TX_RATIO_DEN;
-    rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
-    if (rtcp_bytes < 28)
-        return -1;
-    s->last_octet_count = s->octet_count;
-
-    if (url_open_dyn_buf(&pb) < 0)
-        return -1;
-
-    // Receiver Report
-    put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
-    put_byte(&pb, 201);
-    put_be16(&pb, 7); /* length in words - 1 */
-    put_be32(&pb, s->ssrc); // our own SSRC
-    put_be32(&pb, s->ssrc); // XXX: should be the server's here!
-    // some placeholders we should really fill...
-    // RFC 1889/p64
-    extended_max= stats->cycles + stats->max_seq;
-    expected= extended_max - stats->base_seq + 1;
-    lost= expected - stats->received;
-    lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
-    expected_interval= expected - stats->expected_prior;
-    stats->expected_prior= expected;
-    received_interval= stats->received - stats->received_prior;
-    stats->received_prior= stats->received;
-    lost_interval= expected_interval - received_interval;
-    if (expected_interval==0 || lost_interval<=0) fraction= 0;
-    else fraction = (lost_interval<<8)/expected_interval;
-
-    fraction= (fraction<<24) | lost;
-
-    put_be32(&pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
-    put_be32(&pb, extended_max); /* max sequence received */
-    put_be32(&pb, stats->jitter>>4); /* jitter */
-
-    if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
-    {
-        put_be32(&pb, 0); /* last SR timestamp */
-        put_be32(&pb, 0); /* delay since last SR */
-    } else {
-        uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
-        uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
-
-        put_be32(&pb, middle_32_bits); /* last SR timestamp */
-        put_be32(&pb, delay_since_last); /* delay since last SR */
-    }
-
-    // CNAME
-    put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
-    put_byte(&pb, 202);
-    len = strlen(s->hostname);
-    put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */
-    put_be32(&pb, s->ssrc);
-    put_byte(&pb, 0x01);
-    put_byte(&pb, len);
-    put_buffer(&pb, s->hostname, len);
-    // padding
-    for (len = (6 + len) % 4; len % 4; len++) {
-        put_byte(&pb, 0);
-    }
-
-    put_flush_packet(&pb);
-    len = url_close_dyn_buf(&pb, &buf);
-    if ((len > 0) && buf) {
-        int result;
-#if defined(DEBUG)
-        printf("sending %d bytes of RR\n", len);
-#endif
-        result= url_write(s->rtp_ctx, buf, len);
-#if defined(DEBUG)
-        printf("result from url_write: %d\n", result);
-#endif
-        av_free(buf);
-    }
-    return 0;
-}
-
-/**
- * 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)
-{
-    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;
-    s->rtp_payload_data = rtp_payload_data;
-    rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
-    if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) {
-        s->ts = mpegts_parse_open(s->ic);
-        if (s->ts == NULL) {
-            av_free(s);
-            return NULL;
-        }
-    } else {
-        switch(st->codec->codec_id) {
-        case CODEC_ID_MPEG1VIDEO:
-        case CODEC_ID_MPEG2VIDEO:
-        case CODEC_ID_MP2:
-        case CODEC_ID_MP3:
-        case CODEC_ID_MPEG4:
-        case CODEC_ID_H264:
-            st->need_parsing = AVSTREAM_PARSE_FULL;
-            break;
-        default:
-            break;
-        }
-    }
-    // needed to send back RTCP RR in RTSP sessions
-    s->rtp_ctx = rtpc;
-    gethostname(s->hostname, sizeof(s->hostname));
-    return s;
-}
-
-static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
-{
-    int au_headers_length, au_header_size, i;
-    GetBitContext getbitcontext;
-    rtp_payload_data_t *infos;
-
-    infos = s->rtp_payload_data;
-
-    if (infos == NULL)
-        return -1;
-
-    /* decode the first 2 bytes where are stored the AUHeader sections
-       length in bits */
-    au_headers_length = AV_RB16(buf);
-
-    if (au_headers_length > RTP_MAX_PACKET_LENGTH)
-      return -1;
-
-    infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
-
-    /* skip AU headers length section (2 bytes) */
-    buf += 2;
-
-    init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
-
-    /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
-    au_header_size = infos->sizelength + infos->indexlength;
-    if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
-        return -1;
-
-    infos->nb_au_headers = au_headers_length / au_header_size;
-    infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
-
-    /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
-       In my test, the FAAD decoder does not behave correctly when sending each AU one by one
-       but does when sending the whole as one big packet...  */
-    infos->au_headers[0].size = 0;
-    infos->au_headers[0].index = 0;
-    for (i = 0; i < infos->nb_au_headers; ++i) {
-        infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
-        infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
-    }
-
-    infos->nb_au_headers = 1;
-
-    return 0;
-}
-
-/**
- * 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_AAC:
-        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
- * @param buf input buffer or NULL to read the next packets
- * @param len buffer len
- * @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(RTPDemuxContext *s, AVPacket *pkt,
-                     const uint8_t *buf, int len)
+const char *ff_rtp_enc_name(int payload_type)
 {
-    unsigned int ssrc, h;
-    int payload_type, seq, ret;
-    AVStream *st;
-    uint32_t timestamp;
-    int rv= 0;
+    int i;
 
-    if (!buf) {
-        /* return the next packets, if any */
-        if(s->st && s->parse_packet) {
-            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,
-                                      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;
+    for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
+        if (AVRtpPayloadTypes[i].pt == payload_type) {
+            return AVRtpPayloadTypes[i].enc_name;
         }
-    }
-
-    if (len < 12)
-        return -1;
 
-    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
-        return -1;
-    if (buf[1] >= 200 && buf[1] <= 204) {
-        rtcp_parse_packet(s, buf, len);
-        return -1;
-    }
-    payload_type = buf[1] & 0x7f;
-    seq  = AV_RB16(buf + 2);
-    timestamp = AV_RB32(buf + 4);
-    ssrc = AV_RB32(buf + 8);
-    /* store the ssrc in the RTPDemuxContext */
-    s->ssrc = ssrc;
-
-    /* NOTE: we can handle only one payload type */
-    if (s->payload_type != payload_type)
-        return -1;
-
-    st = s->st;
-    // only do something with this if all the rtp checks pass...
-    if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
-    {
-        av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
-               payload_type, seq, ((s->seq + 1) & 0xffff));
-        return -1;
-    }
-
-    s->seq = seq;
-    len -= 12;
-    buf += 12;
-
-    if (!st) {
-        /* specific MPEG2TS demux support */
-        ret = mpegts_parse_packet(s->ts, pkt, buf, len);
-        if (ret < 0)
-            return -1;
-        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 {
-        // at this point, the RTP header has been stripped;  This is ASSUMING that there is only 1 CSRC, which in't wise.
-        switch(st->codec->codec_id) {
-        case CODEC_ID_MP2:
-            /* better than nothing: skip mpeg audio RTP header */
-            if (len <= 4)
-                return -1;
-            h = AV_RB32(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 video RTP header */
-            if (len <= 4)
-                return -1;
-            h = AV_RB32(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;
-            // moved from below, verbatim.  this is because this section handles packets, and the lower switch handles
-            // timestamps.
-            // TODO: Put this into a dynamic packet handler...
-        case CODEC_ID_AAC:
-            if (rtp_parse_mp4_au(s, buf))
-                return -1;
-            {
-                rtp_payload_data_t *infos = s->rtp_payload_data;
-                if (infos == NULL)
-                    return -1;
-                buf += infos->au_headers_length_bytes + 2;
-                len -= infos->au_headers_length_bytes + 2;
-
-                /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
-                    one au_header */
-                av_new_packet(pkt, infos->au_headers[0].size);
-                memcpy(pkt->data, buf, infos->au_headers[0].size);
-                buf += infos->au_headers[0].size;
-                len -= infos->au_headers[0].size;
-            }
-            s->read_buf_size = len;
-            rv= 0;
-            break;
-        default:
-            if(s->parse_packet) {
-                rv= s->parse_packet(s, pkt, &timestamp, buf, len);
-            } else {
-                av_new_packet(pkt, len);
-                memcpy(pkt->data, buf, len);
-            }
-            break;
-        }
-
-        // now perform timestamp things....
-        finalize_packet(s, pkt, timestamp);
-    }
-    return rv;
-}
-
-void rtp_parse_close(RTPDemuxContext *s)
-{
-    // TODO: fold this into the protocol specific data fields.
-    if (!strcmp(AVRtpPayloadTypes[s->payload_type].enc_name, "MP2T")) {
-        mpegts_parse_close(s->ts);
-    }
-    av_free(s);
+    return "";
 }
 
-/* rtp output */
-
-static int rtp_write_header(AVFormatContext *s1)
+enum CodecID ff_rtp_codec_id(const char *buf, enum CodecType codec_type)
 {
-    RTPDemuxContext *s = s1->priv_data;
-    int payload_type, max_packet_size, n;
-    AVStream *st;
-
-    if (s1->nb_streams != 1)
-        return -1;
-    st = s1->streams[0];
-
-    payload_type = rtp_get_payload_type(st->codec);
-    if (payload_type < 0)
-        payload_type = RTP_PT_PRIVATE; /* private payload type */
-    s->payload_type = payload_type;
-
-// following 2 FIXMies could be set based on the current time, theres normaly no info leak, as rtp will likely be transmitted immedeatly
-    s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
-    s->timestamp = s->base_timestamp;
-    s->cur_timestamp = 0;
-    s->ssrc = 0; /* FIXME: was random(), what should this be? */
-    s->first_packet = 1;
-    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
-
-    max_packet_size = url_fget_max_packet_size(&s1->pb);
-    if (max_packet_size <= 12)
-        return AVERROR(EIO);
-    s->max_payload_size = max_packet_size - 12;
+    int i;
 
-    s->max_frames_per_packet = 0;
-    if (s1->max_delay) {
-        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
-            if (st->codec->frame_size == 0) {
-                av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
-            } else {
-                s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN);
-            }
-        }
-        if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
-            /* FIXME: We should round down here... */
-            s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base);
+    for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
+        if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec_type == AVRtpPayloadTypes[i].codec_type)){
+            return AVRtpPayloadTypes[i].codec_id;
         }
-    }
 
-    av_set_pts_info(st, 32, 1, 90000);
-    switch(st->codec->codec_id) {
-    case CODEC_ID_MP2:
-    case CODEC_ID_MP3:
-        s->buf_ptr = s->buf + 4;
-        break;
-    case CODEC_ID_MPEG1VIDEO:
-        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;
-    case CODEC_ID_AAC:
-        s->read_buf_index = 0;
-    default:
-        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
-            av_set_pts_info(st, 32, 1, st->codec->sample_rate);
-        }
-        s->buf_ptr = s->buf;
-        break;
-    }
-
-    return 0;
+    return CODEC_ID_NONE;
 }
-
-/* send an rtcp sender report packet */
-static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
-{
-    RTPDemuxContext *s = s1->priv_data;
-    uint32_t rtp_ts;
-
-#if defined(DEBUG)
-    printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
-#endif
-
-    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
-    rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q,
-                          s1->streams[0]->time_base) + s->base_timestamp;
-    put_byte(&s1->pb, (RTP_VERSION << 6));
-    put_byte(&s1->pb, 200);
-    put_be16(&s1->pb, 6); /* length in words - 1 */
-    put_be32(&s1->pb, s->ssrc);
-    put_be32(&s1->pb, ntp_time / 1000000);
-    put_be32(&s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
-    put_be32(&s1->pb, rtp_ts);
-    put_be32(&s1->pb, s->packet_count);
-    put_be32(&s1->pb, s->octet_count);
-    put_flush_packet(&s1->pb);
-}
-
-/* send an rtp packet. sequence number is incremented, but the caller
-   must update the timestamp itself */
-void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
-{
-    RTPDemuxContext *s = s1->priv_data;
-
-#ifdef DEBUG
-    printf("rtp_send_data size=%d\n", len);
-#endif
-
-    /* build the RTP header */
-    put_byte(&s1->pb, (RTP_VERSION << 6));
-    put_byte(&s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
-    put_be16(&s1->pb, s->seq);
-    put_be32(&s1->pb, s->timestamp);
-    put_be32(&s1->pb, s->ssrc);
-
-    put_buffer(&s1->pb, buf1, len);
-    put_flush_packet(&s1->pb);
-
-    s->seq++;
-    s->octet_count += len;
-    s->packet_count++;
-}
-
-/* send an integer number of samples and compute time stamp and fill
-   the rtp send buffer before sending. */
-static void rtp_send_samples(AVFormatContext *s1,
-                             const uint8_t *buf1, int size, int sample_size)
-{
-    RTPDemuxContext *s = s1->priv_data;
-    int len, max_packet_size, n;
-
-    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
-    /* not needed, but who nows */
-    if ((size % sample_size) != 0)
-        av_abort();
-    n = 0;
-    while (size > 0) {
-        s->buf_ptr = s->buf;
-        len = FFMIN(max_packet_size, size);
-
-        /* copy data */
-        memcpy(s->buf_ptr, buf1, len);
-        s->buf_ptr += len;
-        buf1 += len;
-        size -= len;
-        s->timestamp = s->cur_timestamp + n / sample_size;
-        ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
-        n += (s->buf_ptr - s->buf);
-    }
-}
-
-/* NOTE: we suppose that exactly one frame is given as argument here */
-/* XXX: test it */
-static void rtp_send_mpegaudio(AVFormatContext *s1,
-                               const uint8_t *buf1, int size)
-{
-    RTPDemuxContext *s = s1->priv_data;
-    int len, count, max_packet_size;
-
-    max_packet_size = s->max_payload_size;
-
-    /* test if we must flush because not enough space */
-    len = (s->buf_ptr - s->buf);
-    if ((len + size) > max_packet_size) {
-        if (len > 4) {
-            ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
-            s->buf_ptr = s->buf + 4;
-        }
-    }
-    if (s->buf_ptr == s->buf + 4) {
-        s->timestamp = s->cur_timestamp;
-    }
-
-    /* add the packet */
-    if (size > max_packet_size) {
-        /* big packet: fragment */
-        count = 0;
-        while (size > 0) {
-            len = max_packet_size - 4;
-            if (len > size)
-                len = size;
-            /* build fragmented packet */
-            s->buf[0] = 0;
-            s->buf[1] = 0;
-            s->buf[2] = count >> 8;
-            s->buf[3] = count;
-            memcpy(s->buf + 4, buf1, len);
-            ff_rtp_send_data(s1, s->buf, len + 4, 0);
-            size -= len;
-            buf1 += len;
-            count += len;
-        }
-    } else {
-        if (s->buf_ptr == s->buf + 4) {
-            /* no fragmentation possible */
-            s->buf[0] = 0;
-            s->buf[1] = 0;
-            s->buf[2] = 0;
-            s->buf[3] = 0;
-        }
-        memcpy(s->buf_ptr, buf1, size);
-        s->buf_ptr += size;
-    }
-}
-
-static void rtp_send_raw(AVFormatContext *s1,
-                         const uint8_t *buf1, int size)
-{
-    RTPDemuxContext *s = s1->priv_data;
-    int len, max_packet_size;
-
-    max_packet_size = s->max_payload_size;
-
-    while (size > 0) {
-        len = max_packet_size;
-        if (len > size)
-            len = size;
-
-        s->timestamp = s->cur_timestamp;
-        ff_rtp_send_data(s1, buf1, len, (len == size));
-
-        buf1 += len;
-        size -= len;
-    }
-}
-
-/* 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) {
-            ff_rtp_send_data(s1, s->buf, out_len, 0);
-            s->buf_ptr = s->buf;
-        }
-    }
-}
-
-/* write an RTP packet. 'buf1' must contain a single specific frame. */
-static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
-{
-    RTPDemuxContext *s = s1->priv_data;
-    AVStream *st = s1->streams[0];
-    int rtcp_bytes;
-    int size= pkt->size;
-    uint8_t *buf1= pkt->data;
-
-#ifdef DEBUG
-    printf("%d: write len=%d\n", pkt->stream_index, size);
-#endif
-
-    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
-    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
-        RTCP_TX_RATIO_DEN;
-    if (s->first_packet || rtcp_bytes >= RTCP_SR_SIZE) {
-        rtcp_send_sr(s1, av_gettime());
-        s->last_octet_count = s->octet_count;
-        s->first_packet = 0;
-    }
-    s->cur_timestamp = s->base_timestamp + pkt->pts;
-
-    switch(st->codec->codec_id) {
-    case CODEC_ID_PCM_MULAW:
-    case CODEC_ID_PCM_ALAW:
-    case CODEC_ID_PCM_U8:
-    case CODEC_ID_PCM_S8:
-        rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
-        break;
-    case CODEC_ID_PCM_U16BE:
-    case CODEC_ID_PCM_U16LE:
-    case CODEC_ID_PCM_S16BE:
-    case CODEC_ID_PCM_S16LE:
-        rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
-        break;
-    case CODEC_ID_MP2:
-    case CODEC_ID_MP3:
-        rtp_send_mpegaudio(s1, buf1, size);
-        break;
-    case CODEC_ID_MPEG1VIDEO:
-        ff_rtp_send_mpegvideo(s1, buf1, size);
-        break;
-    case CODEC_ID_AAC:
-        ff_rtp_send_aac(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);
-        break;
-    }
-    return 0;
-}
-
-AVOutputFormat rtp_muxer = {
-    "rtp",
-    "RTP output format",
-    NULL,
-    NULL,
-    sizeof(RTPDemuxContext),
-    CODEC_ID_PCM_MULAW,
-    CODEC_ID_NONE,
-    rtp_write_header,
-    rtp_write_packet,
-};