* RTP input/output format
* Copyright (c) 2002 Fabrice Bellard.
*
- * This library is free software; you can redistribute it and/or
+ * 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 of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library is distributed in the hope that it will be useful,
+ * 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 this library; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avformat.h"
#include "bitstream.h"
#include <unistd.h>
-#include <sys/types.h>
-#include <sys/socket.h>
-#include <netinet/in.h>
-#ifndef __BEOS__
-# include <arpa/inet.h>
-#else
-# include "barpainet.h"
-#endif
-#include <netdb.h>
+#include "network.h"
+
+#include "rtp_internal.h"
+#include "rtp_h264.h"
//#define DEBUG
{9, "G722", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
{10, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 2},
{11, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 1},
- {12, "QCELP", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
+ {12, "QCELP", CODEC_TYPE_AUDIO, CODEC_ID_QCELP, 8000, 1},
{13, "CN", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
{14, "MPA", CODEC_TYPE_AUDIO, CODEC_ID_MP2, 90000, -1},
{15, "G728", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
{-1, "", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}
};
-AVRtpDynamicPayloadType_t AVRtpDynamicPayloadTypes[]=
+/* 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)
{
- {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4},
- {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_MPEG4AAC},
- {"", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE}
-};
+ handler->next= RTPFirstDynamicPayloadHandler;
+ RTPFirstDynamicPayloadHandler= handler;
+}
-struct RTPDemuxContext {
- AVFormatContext *ic;
- AVStream *st;
- int payload_type;
- uint32_t ssrc;
- uint16_t seq;
- uint32_t timestamp;
- uint32_t base_timestamp;
- uint32_t cur_timestamp;
- int max_payload_size;
- MpegTSContext *ts; /* only used for MP2T payloads */
- int read_buf_index;
- int read_buf_size;
-
- /* rtcp sender statistics receive */
- int64_t last_rtcp_ntp_time;
- int64_t first_rtcp_ntp_time;
- uint32_t 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_t buf[RTP_MAX_PACKET_LENGTH];
- uint8_t *buf_ptr;
- /* special infos for au headers parsing */
- rtp_payload_data_t *rtp_payload_data;
-};
+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_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)
return -1;
}
-/* return < 0 if unknown payload type */
int rtp_get_payload_type(AVCodecContext *codec)
{
int i, payload_type;
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, int payload_type, rtp_payload_data_t *rtp_payload_data)
+RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
{
RTPDemuxContext *s;
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) {
case CODEC_ID_MP2:
case CODEC_ID_MP3:
case CODEC_ID_MPEG4:
- st->need_parsing = 1;
+ 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;
}
/* decode the first 2 bytes where are stored the AUHeader sections
length in bits */
- au_headers_length = BE_16(buf);
+ au_headers_length = AV_RB16(buf);
if (au_headers_length > RTP_MAX_PACKET_LENGTH)
return -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.
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->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;
+ 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, ×tamp, 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;
+ }
}
if (len < 12)
seq = (buf[2] << 8) | buf[3];
timestamp = decode_be32(buf + 4);
ssrc = decode_be32(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;
-#if defined(DEBUG) || 1
- if (seq != ((s->seq + 1) & 0xffff)) {
+ // 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;
}
-#endif
+
s->seq = seq;
len -= 12;
buf += 12;
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 */
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
- default:
- av_new_packet(pkt, len);
- memcpy(pkt->data, buf, len);
- 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_MPEG4:
- pkt->pts = timestamp;
- break;
- case CODEC_ID_MPEG4AAC:
+ // 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;
+ {
+ 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;
- s->buf_ptr = (char *)buf;
- pkt->stream_index = s->st->index;
- return 0;
+ s->buf_ptr = buf;
+ rv= 0;
+ break;
default:
- /* no timestamp info yet */
+ if(s->parse_packet) {
+ rv= s->parse_packet(s, pkt, ×tamp, buf, len);
+ } else {
+ av_new_packet(pkt, len);
+ memcpy(pkt->data, buf, len);
+ }
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)
{
+ // TODO: fold this into the protocol specific data fields.
if (!strcmp(AVRtpPayloadTypes[s->payload_type].enc_name, "MP2T")) {
mpegts_parse_close(s->ts);
}
payload_type = RTP_PT_PRIVATE; /* private payload type */
s->payload_type = payload_type;
- s->base_timestamp = random();
+// 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->ssrc = random();
+ s->ssrc = 0; /* FIXME: was random(), what should this be? */
s->first_packet = 1;
max_packet_size = url_fget_max_packet_size(&s1->pb);
{
RTPDemuxContext *s = s1->priv_data;
#if defined(DEBUG)
- printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
+ printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
#endif
put_byte(&s1->pb, (RTP_VERSION << 6));
put_byte(&s1->pb, 200);
return 0;
}
-AVOutputFormat rtp_mux = {
+AVOutputFormat rtp_muxer = {
"rtp",
"RTP output format",
NULL,
rtp_write_packet,
rtp_write_trailer,
};
-
-int rtp_init(void)
-{
- av_register_output_format(&rtp_mux);
- return 0;
-}
projects / ffmpeg / blobdiff