added rtp support (not activated yet)

git-svn-id: file:///var/local/repositories/ffmpeg/trunk@718 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

libav/rtp.c

+/*
+ * RTP input/output format
+ * Copyright (c) 2002 Fabrice Bellard.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+#include "avformat.h"
+
+#include <unistd.h>
+#include <sys/types.h>
+#include <sys/socket.h>
+#include <netinet/in.h>
+#include <arpa/inet.h>
+#include <netdb.h>
+
+//#define DEBUG
+
+
+/* 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') 
+*/
+
+#define RTP_VERSION 2
+
+#define RTP_MAX_SDES 256   /* maximum text length for SDES */
+
+/* RTCP paquets use 0.5 % of the bandwidth */
+#define RTCP_TX_RATIO_NUM 5
+#define RTCP_TX_RATIO_DEN 1000
+
+typedef enum {
+  RTCP_SR   = 200,
+  RTCP_RR   = 201,
+  RTCP_SDES = 202,
+  RTCP_BYE  = 203,
+  RTCP_APP  = 204
+} rtcp_type_t;
+
+typedef enum {
+  RTCP_SDES_END    =  0,
+  RTCP_SDES_CNAME  =  1,
+  RTCP_SDES_NAME   =  2,
+  RTCP_SDES_EMAIL  =  3,
+  RTCP_SDES_PHONE  =  4,
+  RTCP_SDES_LOC    =  5,
+  RTCP_SDES_TOOL   =  6,
+  RTCP_SDES_NOTE   =  7,
+  RTCP_SDES_PRIV   =  8, 
+  RTCP_SDES_IMG    =  9,
+  RTCP_SDES_DOOR   = 10,
+  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 */
+};
+
+typedef struct RTPContext {
+    int payload_type;
+    UINT32 ssrc;
+    UINT16 seq;
+    UINT32 timestamp;
+    UINT32 base_timestamp;
+    UINT32 cur_timestamp;
+    int max_payload_size;
+    /* rtcp sender statistics receive */
+    INT64 last_rtcp_ntp_time;
+    UINT32 last_rtcp_timestamp;
+    /* rtcp sender statistics */
+    unsigned int packet_count;
+    unsigned int octet_count;
+    unsigned int last_octet_count;
+    int first_packet;
+    /* buffer for output */
+    UINT8 buf[RTP_MAX_PACKET_LENGTH];
+    UINT8 *buf_ptr;
+} RTPContext;
+
+int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
+{
+    switch(payload_type) {
+    case RTP_PT_ULAW:
+        codec->codec_id = CODEC_ID_PCM_MULAW;
+        codec->channels = 1;
+        codec->sample_rate = 8000;
+        break;
+    case RTP_PT_ALAW:
+        codec->codec_id = CODEC_ID_PCM_ALAW;
+        codec->channels = 1;
+        codec->sample_rate = 8000;
+        break;
+    case RTP_PT_S16BE_STEREO:
+        codec->codec_id = CODEC_ID_PCM_S16BE;
+        codec->channels = 2;
+        codec->sample_rate = 44100;
+        break;
+    case RTP_PT_S16BE_MONO:
+        codec->codec_id = CODEC_ID_PCM_S16BE;
+        codec->channels = 1;
+        codec->sample_rate = 44100;
+        break;
+    case RTP_PT_MPEGAUDIO:
+        codec->codec_id = CODEC_ID_MP2;
+        break;
+    case RTP_PT_JPEG:
+        codec->codec_id = CODEC_ID_MJPEG;
+        break;
+    case RTP_PT_MPEGVIDEO:
+        codec->codec_id = CODEC_ID_MPEG1VIDEO;
+        break;
+    default:
+        return -1;
+    }
+    return 0;
+}
+
+/* return < 0 if unknown payload type */
+int rtp_get_payload_type(AVCodecContext *codec)
+{
+    int payload_type;
+
+    /* compute the payload type */
+    payload_type = -1;
+    switch(codec->codec_id) {
+    case CODEC_ID_PCM_MULAW:
+        payload_type = RTP_PT_ULAW;
+        break;
+    case CODEC_ID_PCM_ALAW:
+        payload_type = RTP_PT_ALAW;
+        break;
+    case CODEC_ID_PCM_S16BE:
+        if (codec->channels == 1) {
+            payload_type = RTP_PT_S16BE_MONO;
+        } else if (codec->channels == 2) {
+            payload_type = RTP_PT_S16BE_STEREO;
+        }
+        break;
+    case CODEC_ID_MP2:
+    case CODEC_ID_MP3LAME:
+        payload_type = RTP_PT_MPEGAUDIO;
+        break;
+    case CODEC_ID_MJPEG:
+        payload_type = RTP_PT_JPEG;
+        break;
+    case CODEC_ID_MPEG1VIDEO:
+        payload_type = RTP_PT_MPEGVIDEO;
+        break;
+    default:
+        break;
+    }
+    return payload_type;
+}
+
+static inline UINT32 decode_be32(const UINT8 *p)
+{
+    return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
+}
+
+static inline UINT32 decode_be64(const UINT8 *p)
+{
+    return ((UINT64)decode_be32(p) << 32) | decode_be32(p + 4);
+}
+
+static int rtcp_parse_packet(AVFormatContext *s1, 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);
+    s->last_rtcp_timestamp = decode_be32(buf + 16);
+    return 0;
+}
+
+/**
+ * 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
+ * @param pkt returned packet
+ * @param buf input buffer
+ * @param len buffer len
+ * @return zero if no error.
+ */
+int rtp_parse_packet(AVFormatContext *s1, AVPacket *pkt, 
+                     const unsigned char *buf, int len)
+{
+    RTPContext *s = s1->priv_data;
+    unsigned int ssrc, h;
+    int payload_type, seq, delta_timestamp;
+    AVStream *st;
+    UINT32 timestamp;
+    
+    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);
+        return -1;
+    }
+    payload_type = buf[1] & 0x7f;
+    seq  = (buf[2] << 8) | buf[3];
+    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;
+#if defined(DEBUG) || 1
+    if (seq != ((s->seq + 1) & 0xffff)) {
+        printf("RTP: PT=%02x: bad cseq %04x expected=%04x\n", 
+               payload_type, seq, ((s->seq + 1) & 0xffff));
+    }
+    s->seq = seq;
+#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)
+            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;
+    }
+
+    if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
+        /* compute pts from timestamp with received ntp_time */
+        delta_timestamp = timestamp - s->last_rtcp_timestamp;
+        /* XXX: do conversion, but not needed for mpeg at 90 KhZ */
+        pkt->pts = s->last_rtcp_ntp_time + delta_timestamp;
+    }
+    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;
+    return 0;
+}
+
+static int rtp_read_packet(AVFormatContext *s1, AVPacket *pkt)
+{
+    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;
+    }
+    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;
+}
+
+/* rtp output */
+
+static int rtp_write_header(AVFormatContext *s1)
+{
+    RTPContext *s = s1->priv_data;
+    int payload_type, max_packet_size;
+    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)
+        return -1;
+    s->payload_type = payload_type;
+
+    s->base_timestamp = random();
+    s->timestamp = s->base_timestamp;
+    s->ssrc = random();
+    s->first_packet = 1;
+
+    max_packet_size = url_fget_max_packet_size(&s1->pb);
+    if (max_packet_size <= 12)
+        return AVERROR_IO;
+    s->max_payload_size = max_packet_size - 12;
+
+    switch(st->codec.codec_id) {
+    case CODEC_ID_MP2:
+    case CODEC_ID_MP3LAME:
+        s->buf_ptr = s->buf + 4;
+        s->cur_timestamp = 0;
+        break;
+    case CODEC_ID_MPEG1VIDEO:
+        s->cur_timestamp = 0;
+        break;
+    default:
+        s->buf_ptr = s->buf;
+        break;
+    }
+
+    return 0;
+}
+
+/* send an rtcp sender report packet */
+static void rtcp_send_sr(AVFormatContext *s1, INT64 ntp_time)
+{
+    RTPContext *s = s1->priv_data;
+#if defined(DEBUG)
+    printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
+#endif
+    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_be64(&s1->pb, ntp_time);
+    put_be32(&s1->pb, s->timestamp);
+    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 */
+static void rtp_send_data(AVFormatContext *s1, UINT8 *buf1, int len)
+{
+    RTPContext *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);
+    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,
+                             UINT8 *buf1, int size, int sample_size)
+{
+    RTPContext *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();
+    while (size > 0) {
+        len = (max_packet_size - (s->buf_ptr - s->buf));
+        if (len > size)
+            len = size;
+
+        /* copy data */
+        memcpy(s->buf_ptr, buf1, len);
+        s->buf_ptr += len;
+        buf1 += len;
+        size -= len;
+        n = (s->buf_ptr - s->buf);
+        /* if buffer full, then send it */
+        if (n >= max_packet_size) {
+            rtp_send_data(s1, s->buf, n);
+            s->buf_ptr = s->buf;
+            /* update timestamp */
+            s->timestamp += n / sample_size;
+        }
+    }
+} 
+
+/* NOTE: we suppose that exactly one frame is given as argument here */
+/* XXX: test it */
+static void rtp_send_mpegaudio(AVFormatContext *s1,
+                               UINT8 *buf1, int size)
+{
+    RTPContext *s = s1->priv_data;
+    AVStream *st = s1->streams[0];
+    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) {
+            rtp_send_data(s1, s->buf, s->buf_ptr - s->buf);
+            s->buf_ptr = s->buf + 4;
+            /* 90 KHz time stamp */
+            s->timestamp = s->base_timestamp + 
+                (s->cur_timestamp * 90000LL) / st->codec.sample_rate;
+        }
+    }
+
+    /* 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);
+            rtp_send_data(s1, s->buf, len + 4);
+            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;
+    }
+    s->cur_timestamp += st->codec.frame_size;
+}
+
+/* NOTE: a single frame must be passed with sequence header if
+   needed. XXX: use slices. */
+static void rtp_send_mpegvideo(AVFormatContext *s1,
+                               UINT8 *buf1, int size)
+{
+    RTPContext *s = s1->priv_data;
+    AVStream *st = s1->streams[0];
+    int len, h, max_packet_size;
+    UINT8 *q;
+
+    max_packet_size = s->max_payload_size;
+
+    while (size > 0) {
+        /* XXX: more correct headers */
+        h = 0;
+        if (st->codec.sub_id == 2)
+            h |= 1 << 26; /* mpeg 2 indicator */
+        q = s->buf;
+        *q++ = h >> 24;
+        *q++ = h >> 16;
+        *q++ = h >> 8;
+        *q++ = h;
+
+        if (st->codec.sub_id == 2) {
+            h = 0;
+            *q++ = h >> 24;
+            *q++ = h >> 16;
+            *q++ = h >> 8;
+            *q++ = h;
+        }
+        
+        len = max_packet_size - (q - s->buf);
+        if (len > size)
+            len = size;
+
+        memcpy(q, buf1, len);
+        q += len;
+
+        /* 90 KHz time stamp */
+        /* XXX: overflow */
+        s->timestamp = s->base_timestamp + 
+            (s->cur_timestamp * 90000LL * FRAME_RATE_BASE) / st->codec.frame_rate;
+        rtp_send_data(s1, s->buf, q - s->buf);
+
+        buf1 += len;
+        size -= len;
+    }
+    s->cur_timestamp++;
+}
+
+/* write an RTP packet. 'buf1' must contain a single specific frame. */
+static int rtp_write_packet(AVFormatContext *s1, int stream_index,
+                            UINT8 *buf1, int size, int force_pts)
+{
+    RTPContext *s = s1->priv_data;
+    AVStream *st = s1->streams[0];
+    int rtcp_bytes;
+    INT64 ntp_time;
+    
+#ifdef DEBUG
+    printf("%d: write len=%d\n", 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 >= 28) {
+        /* compute NTP time */
+        ntp_time = force_pts; // ((INT64)force_pts << 28) / 5625
+        rtcp_send_sr(s1, ntp_time); 
+        s->last_octet_count = s->octet_count;
+        s->first_packet = 0;
+    }
+
+    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_MP3LAME:
+        rtp_send_mpegaudio(s1, buf1, size);
+        break;
+    case CODEC_ID_MPEG1VIDEO:
+        rtp_send_mpegvideo(s1, buf1, size);
+        break;
+    default:
+        return AVERROR_IO;
+    }
+    return 0;
+}
+
+static int rtp_write_trailer(AVFormatContext *s1)
+{
+    //    RTPContext *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),
+    CODEC_ID_PCM_MULAW,
+    CODEC_ID_NONE,
+    rtp_write_header,
+    rtp_write_packet,
+    rtp_write_trailer,
+};
+
+int rtp_init(void)
+{
+    av_register_output_format(&rtp_mux);
+    av_register_input_format(&rtp_demux);
+    return 0;
+}

libav/rtp.h

+/*
+ * RTP definitions
+ * Copyright (c) 2002 Fabrice Bellard.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+#ifndef RTP_H
+#define RTP_H
+
+#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);
+
+extern AVOutputFormat rtp_mux;
+extern AVInputFormat rtp_demux;
+
+int rtp_get_local_port(URLContext *h);
+int rtp_set_remote_url(URLContext *h, const char *uri);
+void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd);
+
+extern URLProtocol rtp_protocol;
+
+#endif /* RTP_H */