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added rtp support (not activated yet)

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Originally committed as revision 718 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Fabrice Bellard 2002-07-04 10:38:01 +00:00
parent 171bbb03ad
commit e309128f84
2 changed files with 694 additions and 0 deletions
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654
libav/rtp.c Normal file
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/*
* 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;
}

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/*
* 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 */