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ffmpeg/libavformat/rtpdec.c

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/*
* RTP input format
* Copyright (c) 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg 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 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/* needed for gethostname() */
#define _XOPEN_SOURCE 600
#include "libavcodec/get_bits.h"
#include "avformat.h"
#include "mpegts.h"
#include <unistd.h>
#include "network.h"
#include "rtpdec.h"
#include "rtpdec_amr.h"
#include "rtpdec_asf.h"
#include "rtpdec_h263.h"
#include "rtpdec_h264.h"
#include "rtpdec_mpeg4.h"
#include "rtpdec_xiph.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')
*/
/* statistics functions */
RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
{
handler->next= RTPFirstDynamicPayloadHandler;
RTPFirstDynamicPayloadHandler= handler;
}
void av_register_rtp_dynamic_payload_handlers(void)
{
ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
}
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;
dprintf(s->ic, "sending %d bytes of RR\n", len);
result= url_write(s->rtp_ctx, buf, len);
dprintf(s->ic, "result from url_write: %d\n", result);
av_free(buf);
}
return 0;
}
void rtp_send_punch_packets(URLContext* rtp_handle)
{
ByteIOContext *pb;
uint8_t *buf;
int len;
/* Send a small RTP packet */
if (url_open_dyn_buf(&pb) < 0)
return;
put_byte(pb, (RTP_VERSION << 6));
put_byte(pb, 0); /* Payload type */
put_be16(pb, 0); /* Seq */
put_be32(pb, 0); /* Timestamp */
put_be32(pb, 0); /* SSRC */
put_flush_packet(pb);
len = url_close_dyn_buf(pb, &buf);
if ((len > 0) && buf)
url_write(rtp_handle, buf, len);
av_free(buf);
/* Send a minimal RTCP RR */
if (url_open_dyn_buf(&pb) < 0)
return;
put_byte(pb, (RTP_VERSION << 6));
put_byte(pb, 201); /* receiver report */
put_be16(pb, 1); /* length in words - 1 */
put_be32(pb, 0); /* our own SSRC */
put_flush_packet(pb);
len = url_close_dyn_buf(pb, &buf);
if ((len > 0) && buf)
url_write(rtp_handle, buf, len);
av_free(buf);
}
/**
* 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, URLContext *rtpc, 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;
rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
s->ts = ff_mpegts_parse_open(s->ic);
if (s->ts == NULL) {
av_free(s);
return NULL;
}
} else {
av_set_pts_info(st, 32, 1, 90000);
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_H263:
case CODEC_ID_H264:
st->need_parsing = AVSTREAM_PARSE_FULL;
break;
default:
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
av_set_pts_info(st, 32, 1, st->codec->sample_rate);
}
break;
}
}
// needed to send back RTCP RR in RTSP sessions
s->rtp_ctx = rtpc;
gethostname(s->hostname, sizeof(s->hostname));
return s;
}
void
rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
RTPDynamicProtocolHandler *handler)
{
s->dynamic_protocol_context = ctx;
s->parse_packet = handler->parse_packet;
}
/**
* 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)
{
if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
int64_t addend;
int delta_timestamp;
/* compute pts from timestamp with received ntp_time */
delta_timestamp = timestamp - s->last_rtcp_timestamp;
/* convert to the PTS timebase */
addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);
pkt->pts = s->range_start_offset + addend + delta_timestamp;
}
}
/**
* 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)
{
unsigned int ssrc, h;
int payload_type, seq, ret, flags = 0;
AVStream *st;
uint32_t timestamp;
int rv= 0;
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->ic, s->dynamic_protocol_context,
s->st, pkt, &timestamp, NULL, 0, flags);
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 = ff_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(s, buf, len);
return -1;
}
payload_type = buf[1] & 0x7f;
if (buf[1] & 0x80)
flags |= RTP_FLAG_MARKER;
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 = ff_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;
}
return 0;
} else if (s->parse_packet) {
rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
s->st, pkt, &timestamp, buf, len, flags);
} 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:
case CODEC_ID_MP3:
/* 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:
case CODEC_ID_MPEG2VIDEO:
/* 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;
default:
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
}
pkt->stream_index = st->index;
}
// now perform timestamp things....
finalize_packet(s, pkt, timestamp);
return rv;
}
void rtp_parse_close(RTPDemuxContext *s)
{
if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
ff_mpegts_parse_close(s->ts);
}
av_free(s);
}
int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
int (*parse_fmtp)(AVStream *stream,
PayloadContext *data,
char *attr, char *value))
{
char attr[256];
char value[4096];
int res;
// remove protocol identifier
while (*p && *p == ' ') p++; // strip spaces
while (*p && *p != ' ') p++; // eat protocol identifier
while (*p && *p == ' ') p++; // strip trailing spaces
while (ff_rtsp_next_attr_and_value(&p,
attr, sizeof(attr),
value, sizeof(value))) {
res = parse_fmtp(stream, data, attr, value);
if (res < 0)
return res;
}
return 0;
}
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