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FFmpeg/libavformat/rtsp.c
Martin Storsjö b56fc18b20 sdp: Add an option for sending RTCP packets to the source of the last packets 
An SDP description normally only contains the target IP address
and port for the packets. This means that we don't really have
any clue where to send the RTCP RR packets - previously they're
sent to the destination IP written in the SDP (at the same port),
which rarely is the actual peer. And if the source for the packets
is on a different port than the destination, it's never correct.

With a new option, we can choose to send the packets to the
address that the latest packet on each socket arrived from.
---
Some may even argue that this should be the default - perhaps,
but I'd rather keep it optional at first. Additionally, I'm not
sure if sending RTCP RR directly back to the source is
desireable for e.g. multicast.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-08-14 11:21:33 +03:00

2346 lines
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/*
* RTSP/SDP client
* Copyright (c) 2002 Fabrice Bellard
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/base64.h"
#include "libavutil/avstring.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/mathematics.h"
#include "libavutil/parseutils.h"
#include "libavutil/random_seed.h"
#include "libavutil/dict.h"
#include "libavutil/opt.h"
#include "libavutil/time.h"
#include "avformat.h"
#include "avio_internal.h"
#if HAVE_POLL_H
#include <poll.h>
#endif
#include "internal.h"
#include "network.h"
#include "os_support.h"
#include "http.h"
#include "rtsp.h"
#include "rtpdec.h"
#include "rtpproto.h"
#include "rdt.h"
#include "rtpdec_formats.h"
#include "rtpenc_chain.h"
#include "url.h"
#include "rtpenc.h"
#include "mpegts.h"
/* Timeout values for socket poll, in ms,
* and read_packet(), in seconds */
#define POLL_TIMEOUT_MS 100
#define READ_PACKET_TIMEOUT_S 10
#define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
#define SDP_MAX_SIZE 16384
#define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
#define DEFAULT_REORDERING_DELAY 100000
#define OFFSET(x) offsetof(RTSPState, x)
#define DEC AV_OPT_FLAG_DECODING_PARAM
#define ENC AV_OPT_FLAG_ENCODING_PARAM
#define RTSP_FLAG_OPTS(name, longname) \
{ name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
{ "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
#define RTSP_MEDIATYPE_OPTS(name, longname) \
{ name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
{ "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
{ "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
{ "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
#define RTSP_REORDERING_OPTS() \
{ "reorder_queue_size", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }
const AVOption ff_rtsp_options[] = {
{ "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
{ "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
{ "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
{ "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
{ "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
{ "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
{ "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
{ "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
{ "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
{ "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
RTSP_REORDERING_OPTS(),
{ NULL },
};
static const AVOption sdp_options[] = {
RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
{ "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
{ "rtcp_to_source", "Send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" },
RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
RTSP_REORDERING_OPTS(),
{ NULL },
};
static const AVOption rtp_options[] = {
RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
RTSP_REORDERING_OPTS(),
{ NULL },
};
static void get_word_until_chars(char *buf, int buf_size,
const char *sep, const char **pp)
{
const char *p;
char *q;
p = *pp;
p += strspn(p, SPACE_CHARS);
q = buf;
while (!strchr(sep, *p) && *p != '\0') {
if ((q - buf) < buf_size - 1)
*q++ = *p;
p++;
}
if (buf_size > 0)
*q = '\0';
*pp = p;
}
static void get_word_sep(char *buf, int buf_size, const char *sep,
const char **pp)
{
if (**pp == '/') (*pp)++;
get_word_until_chars(buf, buf_size, sep, pp);
}
static void get_word(char *buf, int buf_size, const char **pp)
{
get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
}
/** Parse a string p in the form of Range:npt=xx-xx, and determine the start
* and end time.
* Used for seeking in the rtp stream.
*/
static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
{
char buf[256];
p += strspn(p, SPACE_CHARS);
if (!av_stristart(p, "npt=", &p))
return;
*start = AV_NOPTS_VALUE;
*end = AV_NOPTS_VALUE;
get_word_sep(buf, sizeof(buf), "-", &p);
av_parse_time(start, buf, 1);
if (*p == '-') {
p++;
get_word_sep(buf, sizeof(buf), "-", &p);
av_parse_time(end, buf, 1);
}
}
static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
{
struct addrinfo hints = { 0 }, *ai = NULL;
hints.ai_flags = AI_NUMERICHOST;
if (getaddrinfo(buf, NULL, &hints, &ai))
return -1;
memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
freeaddrinfo(ai);
return 0;
}
#if CONFIG_RTPDEC
static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
RTSPStream *rtsp_st, AVCodecContext *codec)
{
if (!handler)
return;
if (codec)
codec->codec_id = handler->codec_id;
rtsp_st->dynamic_handler = handler;
if (handler->alloc) {
rtsp_st->dynamic_protocol_context = handler->alloc();
if (!rtsp_st->dynamic_protocol_context)
rtsp_st->dynamic_handler = NULL;
}
}
/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
static int sdp_parse_rtpmap(AVFormatContext *s,
AVStream *st, RTSPStream *rtsp_st,
int payload_type, const char *p)
{
AVCodecContext *codec = st->codec;
char buf[256];
int i;
AVCodec *c;
const char *c_name;
/* See if we can handle this kind of payload.
* The space should normally not be there but some Real streams or
* particular servers ("RealServer Version 6.1.3.970", see issue 1658)
* have a trailing space. */
get_word_sep(buf, sizeof(buf), "/ ", &p);
if (payload_type < RTP_PT_PRIVATE) {
/* We are in a standard case
* (from http://www.iana.org/assignments/rtp-parameters). */
codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
}
if (codec->codec_id == AV_CODEC_ID_NONE) {
RTPDynamicProtocolHandler *handler =
ff_rtp_handler_find_by_name(buf, codec->codec_type);
init_rtp_handler(handler, rtsp_st, codec);
/* If no dynamic handler was found, check with the list of standard
* allocated types, if such a stream for some reason happens to
* use a private payload type. This isn't handled in rtpdec.c, since
* the format name from the rtpmap line never is passed into rtpdec. */
if (!rtsp_st->dynamic_handler)
codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
}
c = avcodec_find_decoder(codec->codec_id);
if (c && c->name)
c_name = c->name;
else
c_name = "(null)";
get_word_sep(buf, sizeof(buf), "/", &p);
i = atoi(buf);
switch (codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:
av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
if (i > 0) {
codec->sample_rate = i;
avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
get_word_sep(buf, sizeof(buf), "/", &p);
i = atoi(buf);
if (i > 0)
codec->channels = i;
}
av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
codec->sample_rate);
av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
codec->channels);
break;
case AVMEDIA_TYPE_VIDEO:
av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
if (i > 0)
avpriv_set_pts_info(st, 32, 1, i);
break;
default:
break;
}
if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
rtsp_st->dynamic_handler->init(s, st->index,
rtsp_st->dynamic_protocol_context);
return 0;
}
/* parse the attribute line from the fmtp a line of an sdp response. This
* is broken out as a function because it is used in rtp_h264.c, which is
* forthcoming. */
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
char *value, int value_size)
{
*p += strspn(*p, SPACE_CHARS);
if (**p) {
get_word_sep(attr, attr_size, "=", p);
if (**p == '=')
(*p)++;
get_word_sep(value, value_size, ";", p);
if (**p == ';')
(*p)++;
return 1;
}
return 0;
}
typedef struct SDPParseState {
/* SDP only */
struct sockaddr_storage default_ip;
int default_ttl;
int skip_media; ///< set if an unknown m= line occurs
int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
} SDPParseState;
static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
struct RTSPSource ***dest, int *dest_count)
{
RTSPSource *rtsp_src, *rtsp_src2;
int i;
for (i = 0; i < count; i++) {
rtsp_src = addrs[i];
rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
if (!rtsp_src2)
continue;
memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
dynarray_add(dest, dest_count, rtsp_src2);
}
}
static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
int letter, const char *buf)
{
RTSPState *rt = s->priv_data;
char buf1[64], st_type[64];
const char *p;
enum AVMediaType codec_type;
int payload_type, i;
AVStream *st;
RTSPStream *rtsp_st;
RTSPSource *rtsp_src;
struct sockaddr_storage sdp_ip;
int ttl;
av_dlog(s, "sdp: %c='%s'\n", letter, buf);
p = buf;
if (s1->skip_media && letter != 'm')
return;
switch (letter) {
case 'c':
get_word(buf1, sizeof(buf1), &p);
if (strcmp(buf1, "IN") != 0)
return;
get_word(buf1, sizeof(buf1), &p);
if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
return;
get_word_sep(buf1, sizeof(buf1), "/", &p);
if (get_sockaddr(buf1, &sdp_ip))
return;
ttl = 16;
if (*p == '/') {
p++;
get_word_sep(buf1, sizeof(buf1), "/", &p);
ttl = atoi(buf1);
}
if (s->nb_streams == 0) {
s1->default_ip = sdp_ip;
s1->default_ttl = ttl;
} else {
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
rtsp_st->sdp_ip = sdp_ip;
rtsp_st->sdp_ttl = ttl;
}
break;
case 's':
av_dict_set(&s->metadata, "title", p, 0);
break;
case 'i':
if (s->nb_streams == 0) {
av_dict_set(&s->metadata, "comment", p, 0);
break;
}
break;
case 'm':
/* new stream */
s1->skip_media = 0;
codec_type = AVMEDIA_TYPE_UNKNOWN;
get_word(st_type, sizeof(st_type), &p);
if (!strcmp(st_type, "audio")) {
codec_type = AVMEDIA_TYPE_AUDIO;
} else if (!strcmp(st_type, "video")) {
codec_type = AVMEDIA_TYPE_VIDEO;
} else if (!strcmp(st_type, "application")) {
codec_type = AVMEDIA_TYPE_DATA;
}
if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
s1->skip_media = 1;
return;
}
rtsp_st = av_mallocz(sizeof(RTSPStream));
if (!rtsp_st)
return;
rtsp_st->stream_index = -1;
dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
rtsp_st->sdp_ip = s1->default_ip;
rtsp_st->sdp_ttl = s1->default_ttl;
copy_default_source_addrs(s1->default_include_source_addrs,
s1->nb_default_include_source_addrs,
&rtsp_st->include_source_addrs,
&rtsp_st->nb_include_source_addrs);
copy_default_source_addrs(s1->default_exclude_source_addrs,
s1->nb_default_exclude_source_addrs,
&rtsp_st->exclude_source_addrs,
&rtsp_st->nb_exclude_source_addrs);
get_word(buf1, sizeof(buf1), &p); /* port */
rtsp_st->sdp_port = atoi(buf1);
get_word(buf1, sizeof(buf1), &p); /* protocol */
if (!strcmp(buf1, "udp"))
rt->transport = RTSP_TRANSPORT_RAW;
else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
rtsp_st->feedback = 1;
/* XXX: handle list of formats */
get_word(buf1, sizeof(buf1), &p); /* format list */
rtsp_st->sdp_payload_type = atoi(buf1);
if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
/* no corresponding stream */
if (rt->transport == RTSP_TRANSPORT_RAW) {
if (!rt->ts && CONFIG_RTPDEC)
rt->ts = ff_mpegts_parse_open(s);
} else {
RTPDynamicProtocolHandler *handler;
handler = ff_rtp_handler_find_by_id(
rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
init_rtp_handler(handler, rtsp_st, NULL);
if (handler && handler->init)
handler->init(s, -1, rtsp_st->dynamic_protocol_context);
}
} else if (rt->server_type == RTSP_SERVER_WMS &&
codec_type == AVMEDIA_TYPE_DATA) {
/* RTX stream, a stream that carries all the other actual
* audio/video streams. Don't expose this to the callers. */
} else {
st = avformat_new_stream(s, NULL);
if (!st)
return;
st->id = rt->nb_rtsp_streams - 1;
rtsp_st->stream_index = st->index;
st->codec->codec_type = codec_type;
if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
RTPDynamicProtocolHandler *handler;
/* if standard payload type, we can find the codec right now */
ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
st->codec->sample_rate > 0)
avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
/* Even static payload types may need a custom depacketizer */
handler = ff_rtp_handler_find_by_id(
rtsp_st->sdp_payload_type, st->codec->codec_type);
init_rtp_handler(handler, rtsp_st, st->codec);
if (handler && handler->init)
handler->init(s, st->index,
rtsp_st->dynamic_protocol_context);
}
}
/* put a default control url */
av_strlcpy(rtsp_st->control_url, rt->control_uri,
sizeof(rtsp_st->control_url));
break;
case 'a':
if (av_strstart(p, "control:", &p)) {
if (s->nb_streams == 0) {
if (!strncmp(p, "rtsp://", 7))
av_strlcpy(rt->control_uri, p,
sizeof(rt->control_uri));
} else {
char proto[32];
/* get the control url */
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
/* XXX: may need to add full url resolution */
av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
NULL, NULL, 0, p);
if (proto[0] == '\0') {
/* relative control URL */
if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
av_strlcat(rtsp_st->control_url, "/",
sizeof(rtsp_st->control_url));
av_strlcat(rtsp_st->control_url, p,
sizeof(rtsp_st->control_url));
} else
av_strlcpy(rtsp_st->control_url, p,
sizeof(rtsp_st->control_url));
}
} else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
/* NOTE: rtpmap is only supported AFTER the 'm=' tag */
get_word(buf1, sizeof(buf1), &p);
payload_type = atoi(buf1);
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
if (rtsp_st->stream_index >= 0) {
st = s->streams[rtsp_st->stream_index];
sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
}
} else if (av_strstart(p, "fmtp:", &p) ||
av_strstart(p, "framesize:", &p)) {
/* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
// let dynamic protocol handlers have a stab at the line.
get_word(buf1, sizeof(buf1), &p);
payload_type = atoi(buf1);
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
if (rtsp_st->sdp_payload_type == payload_type &&
rtsp_st->dynamic_handler &&
rtsp_st->dynamic_handler->parse_sdp_a_line)
rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
rtsp_st->dynamic_protocol_context, buf);
}
} else if (av_strstart(p, "range:", &p)) {
int64_t start, end;
// this is so that seeking on a streamed file can work.
rtsp_parse_range_npt(p, &start, &end);
s->start_time = start;
/* AV_NOPTS_VALUE means live broadcast (and can't seek) */
s->duration = (end == AV_NOPTS_VALUE) ?
AV_NOPTS_VALUE : end - start;
} else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
if (atoi(p) == 1)
rt->transport = RTSP_TRANSPORT_RDT;
} else if (av_strstart(p, "SampleRate:integer;", &p) &&
s->nb_streams > 0) {
st = s->streams[s->nb_streams - 1];
st->codec->sample_rate = atoi(p);
} else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
// RFC 4568
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
get_word(buf1, sizeof(buf1), &p); // ignore tag
get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
p += strspn(p, SPACE_CHARS);
if (av_strstart(p, "inline:", &p))
get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
} else if (av_strstart(p, "source-filter:", &p)) {
int exclude = 0;
get_word(buf1, sizeof(buf1), &p);
if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
return;
exclude = !strcmp(buf1, "excl");
get_word(buf1, sizeof(buf1), &p);
if (strcmp(buf1, "IN") != 0)
return;
get_word(buf1, sizeof(buf1), &p);
if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
return;
// not checking that the destination address actually matches or is wildcard
get_word(buf1, sizeof(buf1), &p);
while (*p != '\0') {
rtsp_src = av_mallocz(sizeof(*rtsp_src));
if (!rtsp_src)
return;
get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
if (exclude) {
if (s->nb_streams == 0) {
dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
} else {
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
}
} else {
if (s->nb_streams == 0) {
dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
} else {
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
}
}
}
} else {
if (rt->server_type == RTSP_SERVER_WMS)
ff_wms_parse_sdp_a_line(s, p);
if (s->nb_streams > 0) {
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
if (rt->server_type == RTSP_SERVER_REAL)
ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
if (rtsp_st->dynamic_handler &&
rtsp_st->dynamic_handler->parse_sdp_a_line)
rtsp_st->dynamic_handler->parse_sdp_a_line(s,
rtsp_st->stream_index,
rtsp_st->dynamic_protocol_context, buf);
}
}
break;
}
}
int ff_sdp_parse(AVFormatContext *s, const char *content)
{
RTSPState *rt = s->priv_data;
const char *p;
int letter, i;
/* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
* contain long SDP lines containing complete ASF Headers (several
* kB) or arrays of MDPR (RM stream descriptor) headers plus
* "rulebooks" describing their properties. Therefore, the SDP line
* buffer is large.
*
* The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
* in rtpdec_xiph.c. */
char buf[16384], *q;
SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
p = content;
for (;;) {
p += strspn(p, SPACE_CHARS);
letter = *p;
if (letter == '\0')
break;
p++;
if (*p != '=')
goto next_line;
p++;
/* get the content */
q = buf;
while (*p != '\n' && *p != '\r' && *p != '\0') {
if ((q - buf) < sizeof(buf) - 1)
*q++ = *p;
p++;
}
*q = '\0';
sdp_parse_line(s, s1, letter, buf);
next_line:
while (*p != '\n' && *p != '\0')
p++;
if (*p == '\n')
p++;
}
for (i = 0; i < s1->nb_default_include_source_addrs; i++)
av_free(s1->default_include_source_addrs[i]);
av_freep(&s1->default_include_source_addrs);
for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
av_free(s1->default_exclude_source_addrs[i]);
av_freep(&s1->default_exclude_source_addrs);
rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
if (!rt->p) return AVERROR(ENOMEM);
return 0;
}
#endif /* CONFIG_RTPDEC */
void ff_rtsp_undo_setup(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
int i;
for (i = 0; i < rt->nb_rtsp_streams; i++) {
RTSPStream *rtsp_st = rt->rtsp_streams[i];
if (!rtsp_st)
continue;
if (rtsp_st->transport_priv) {
if (s->oformat) {
AVFormatContext *rtpctx = rtsp_st->transport_priv;
av_write_trailer(rtpctx);
if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
uint8_t *ptr;
avio_close_dyn_buf(rtpctx->pb, &ptr);
av_free(ptr);
} else {
avio_close(rtpctx->pb);
}
avformat_free_context(rtpctx);
} else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
ff_rdt_parse_close(rtsp_st->transport_priv);
else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
ff_rtp_parse_close(rtsp_st->transport_priv);
}
rtsp_st->transport_priv = NULL;
if (rtsp_st->rtp_handle)
ffurl_close(rtsp_st->rtp_handle);
rtsp_st->rtp_handle = NULL;
}
}
/* close and free RTSP streams */
void ff_rtsp_close_streams(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
int i, j;
RTSPStream *rtsp_st;
ff_rtsp_undo_setup(s);
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
if (rtsp_st) {
if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
rtsp_st->dynamic_handler->free(
rtsp_st->dynamic_protocol_context);
for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
av_free(rtsp_st->include_source_addrs[j]);
av_freep(&rtsp_st->include_source_addrs);
for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
av_free(rtsp_st->exclude_source_addrs[j]);
av_freep(&rtsp_st->exclude_source_addrs);
av_free(rtsp_st);
}
}
av_free(rt->rtsp_streams);
if (rt->asf_ctx) {
avformat_close_input(&rt->asf_ctx);
}
if (rt->ts && CONFIG_RTPDEC)
ff_mpegts_parse_close(rt->ts);
av_free(rt->p);
av_free(rt->recvbuf);
}
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
{
RTSPState *rt = s->priv_data;
AVStream *st = NULL;
int reordering_queue_size = rt->reordering_queue_size;
if (reordering_queue_size < 0) {
if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
reordering_queue_size = 0;
else
reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
}
/* open the RTP context */
if (rtsp_st->stream_index >= 0)
st = s->streams[rtsp_st->stream_index];
if (!st)
s->ctx_flags |= AVFMTCTX_NOHEADER;
if (s->oformat && CONFIG_RTSP_MUXER) {
int ret = ff_rtp_chain_mux_open(&rtsp_st->transport_priv, s, st,
rtsp_st->rtp_handle,
RTSP_TCP_MAX_PACKET_SIZE,
rtsp_st->stream_index);
/* Ownership of rtp_handle is passed to the rtp mux context */
rtsp_st->rtp_handle = NULL;
if (ret < 0)
return ret;
} else if (rt->transport == RTSP_TRANSPORT_RAW) {
return 0; // Don't need to open any parser here
} else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
rtsp_st->dynamic_protocol_context,
rtsp_st->dynamic_handler);
else if (CONFIG_RTPDEC)
rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
rtsp_st->sdp_payload_type,
reordering_queue_size);
if (!rtsp_st->transport_priv) {
return AVERROR(ENOMEM);
} else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
if (rtsp_st->dynamic_handler) {
ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
rtsp_st->dynamic_protocol_context,
rtsp_st->dynamic_handler);
}
if (rtsp_st->crypto_suite[0])
ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
rtsp_st->crypto_suite,
rtsp_st->crypto_params);
}
return 0;
}
#if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
{
const char *q;
char *p;
int v;
q = *pp;
q += strspn(q, SPACE_CHARS);
v = strtol(q, &p, 10);
if (*p == '-') {
p++;
*min_ptr = v;
v = strtol(p, &p, 10);
*max_ptr = v;
} else {
*min_ptr = v;
*max_ptr = v;
}
*pp = p;
}
/* XXX: only one transport specification is parsed */
static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
{
char transport_protocol[16];
char profile[16];
char lower_transport[16];
char parameter[16];
RTSPTransportField *th;
char buf[256];
reply->nb_transports = 0;
for (;;) {
p += strspn(p, SPACE_CHARS);
if (*p == '\0')
break;
th = &reply->transports[reply->nb_transports];
get_word_sep(transport_protocol, sizeof(transport_protocol),
"/", &p);
if (!av_strcasecmp (transport_protocol, "rtp")) {
get_word_sep(profile, sizeof(profile), "/;,", &p);
lower_transport[0] = '\0';
/* rtp/avp/<protocol> */
if (*p == '/') {
get_word_sep(lower_transport, sizeof(lower_transport),
";,", &p);
}
th->transport = RTSP_TRANSPORT_RTP;
} else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
!av_strcasecmp (transport_protocol, "x-real-rdt")) {
/* x-pn-tng/<protocol> */
get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
profile[0] = '\0';
th->transport = RTSP_TRANSPORT_RDT;
} else if (!av_strcasecmp(transport_protocol, "raw")) {
get_word_sep(profile, sizeof(profile), "/;,", &p);
lower_transport[0] = '\0';
/* raw/raw/<protocol> */
if (*p == '/') {
get_word_sep(lower_transport, sizeof(lower_transport),
";,", &p);
}
th->transport = RTSP_TRANSPORT_RAW;
}
if (!av_strcasecmp(lower_transport, "TCP"))
th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
else
th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
if (*p == ';')
p++;
/* get each parameter */
while (*p != '\0' && *p != ',') {
get_word_sep(parameter, sizeof(parameter), "=;,", &p);
if (!strcmp(parameter, "port")) {
if (*p == '=') {
p++;
rtsp_parse_range(&th->port_min, &th->port_max, &p);
}
} else if (!strcmp(parameter, "client_port")) {
if (*p == '=') {
p++;
rtsp_parse_range(&th->client_port_min,
&th->client_port_max, &p);
}
} else if (!strcmp(parameter, "server_port")) {
if (*p == '=') {
p++;
rtsp_parse_range(&th->server_port_min,
&th->server_port_max, &p);
}
} else if (!strcmp(parameter, "interleaved")) {
if (*p == '=') {
p++;
rtsp_parse_range(&th->interleaved_min,
&th->interleaved_max, &p);
}
} else if (!strcmp(parameter, "multicast")) {
if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
} else if (!strcmp(parameter, "ttl")) {
if (*p == '=') {
char *end;
p++;
th->ttl = strtol(p, &end, 10);
p = end;
}
} else if (!strcmp(parameter, "destination")) {
if (*p == '=') {
p++;
get_word_sep(buf, sizeof(buf), ";,", &p);
get_sockaddr(buf, &th->destination);
}
} else if (!strcmp(parameter, "source")) {
if (*p == '=') {
p++;
get_word_sep(buf, sizeof(buf), ";,", &p);
av_strlcpy(th->source, buf, sizeof(th->source));
}
} else if (!strcmp(parameter, "mode")) {
if (*p == '=') {
p++;
get_word_sep(buf, sizeof(buf), ";, ", &p);
if (!strcmp(buf, "record") ||
!strcmp(buf, "receive"))
th->mode_record = 1;
}
}
while (*p != ';' && *p != '\0' && *p != ',')
p++;
if (*p == ';')
p++;
}
if (*p == ',')
p++;
reply->nb_transports++;
}
}
static void handle_rtp_info(RTSPState *rt, const char *url,
uint32_t seq, uint32_t rtptime)
{
int i;
if (!rtptime || !url[0])
return;
if (rt->transport != RTSP_TRANSPORT_RTP)
return;
for (i = 0; i < rt->nb_rtsp_streams; i++) {
RTSPStream *rtsp_st = rt->rtsp_streams[i];
RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
if (!rtpctx)
continue;
if (!strcmp(rtsp_st->control_url, url)) {
rtpctx->base_timestamp = rtptime;
break;
}
}
}
static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
{
int read = 0;
char key[20], value[1024], url[1024] = "";
uint32_t seq = 0, rtptime = 0;
for (;;) {
p += strspn(p, SPACE_CHARS);
if (!*p)
break;
get_word_sep(key, sizeof(key), "=", &p);
if (*p != '=')
break;
p++;
get_word_sep(value, sizeof(value), ";, ", &p);
read++;
if (!strcmp(key, "url"))
av_strlcpy(url, value, sizeof(url));
else if (!strcmp(key, "seq"))
seq = strtoul(value, NULL, 10);
else if (!strcmp(key, "rtptime"))
rtptime = strtoul(value, NULL, 10);
if (*p == ',') {
handle_rtp_info(rt, url, seq, rtptime);
url[0] = '\0';
seq = rtptime = 0;
read = 0;
}
if (*p)
p++;
}
if (read > 0)
handle_rtp_info(rt, url, seq, rtptime);
}
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
RTSPState *rt, const char *method)
{
const char *p;
/* NOTE: we do case independent match for broken servers */
p = buf;
if (av_stristart(p, "Session:", &p)) {
int t;
get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
if (av_stristart(p, ";timeout=", &p) &&
(t = strtol(p, NULL, 10)) > 0) {
reply->timeout = t;
}
} else if (av_stristart(p, "Content-Length:", &p)) {
reply->content_length = strtol(p, NULL, 10);
} else if (av_stristart(p, "Transport:", &p)) {
rtsp_parse_transport(reply, p);
} else if (av_stristart(p, "CSeq:", &p)) {
reply->seq = strtol(p, NULL, 10);
} else if (av_stristart(p, "Range:", &p)) {
rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
} else if (av_stristart(p, "RealChallenge1:", &p)) {
p += strspn(p, SPACE_CHARS);
av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
} else if (av_stristart(p, "Server:", &p)) {
p += strspn(p, SPACE_CHARS);
av_strlcpy(reply->server, p, sizeof(reply->server));
} else if (av_stristart(p, "Notice:", &p) ||
av_stristart(p, "X-Notice:", &p)) {
reply->notice = strtol(p, NULL, 10);
} else if (av_stristart(p, "Location:", &p)) {
p += strspn(p, SPACE_CHARS);
av_strlcpy(reply->location, p , sizeof(reply->location));
} else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
p += strspn(p, SPACE_CHARS);
ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
} else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
p += strspn(p, SPACE_CHARS);
ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
} else if (av_stristart(p, "Content-Base:", &p) && rt) {
p += strspn(p, SPACE_CHARS);
if (method && !strcmp(method, "DESCRIBE"))
av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
} else if (av_stristart(p, "RTP-Info:", &p) && rt) {
p += strspn(p, SPACE_CHARS);
if (method && !strcmp(method, "PLAY"))
rtsp_parse_rtp_info(rt, p);
} else if (av_stristart(p, "Public:", &p) && rt) {
if (strstr(p, "GET_PARAMETER") &&
method && !strcmp(method, "OPTIONS"))
rt->get_parameter_supported = 1;
} else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
p += strspn(p, SPACE_CHARS);
rt->accept_dynamic_rate = atoi(p);
} else if (av_stristart(p, "Content-Type:", &p)) {
p += strspn(p, SPACE_CHARS);
av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
}
}
/* skip a RTP/TCP interleaved packet */
void ff_rtsp_skip_packet(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
int ret, len, len1;
uint8_t buf[1024];
ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
if (ret != 3)
return;
len = AV_RB16(buf + 1);
av_dlog(s, "skipping RTP packet len=%d\n", len);
/* skip payload */
while (len > 0) {
len1 = len;
if (len1 > sizeof(buf))
len1 = sizeof(buf);
ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
if (ret != len1)
return;
len -= len1;
}
}
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
unsigned char **content_ptr,
int return_on_interleaved_data, const char *method)
{
RTSPState *rt = s->priv_data;
char buf[4096], buf1[1024], *q;
unsigned char ch;
const char *p;
int ret, content_length, line_count = 0, request = 0;
unsigned char *content = NULL;
start:
line_count = 0;
request = 0;
content = NULL;
memset(reply, 0, sizeof(*reply));
/* parse reply (XXX: use buffers) */
rt->last_reply[0] = '\0';
for (;;) {
q = buf;
for (;;) {
ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
if (ret != 1)
return AVERROR_EOF;
if (ch == '\n')
break;
if (ch == '$') {
/* XXX: only parse it if first char on line ? */
if (return_on_interleaved_data) {
return 1;
} else
ff_rtsp_skip_packet(s);
} else if (ch != '\r') {
if ((q - buf) < sizeof(buf) - 1)
*q++ = ch;
}
}
*q = '\0';
av_dlog(s, "line='%s'\n", buf);
/* test if last line */
if (buf[0] == '\0')
break;
p = buf;
if (line_count == 0) {
/* get reply code */
get_word(buf1, sizeof(buf1), &p);
if (!strncmp(buf1, "RTSP/", 5)) {
get_word(buf1, sizeof(buf1), &p);
reply->status_code = atoi(buf1);
av_strlcpy(reply->reason, p, sizeof(reply->reason));
} else {
av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
get_word(buf1, sizeof(buf1), &p); // object
request = 1;
}
} else {
ff_rtsp_parse_line(reply, p, rt, method);
av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
}
line_count++;
}
if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
content_length = reply->content_length;
if (content_length > 0) {
/* leave some room for a trailing '\0' (useful for simple parsing) */
content = av_malloc(content_length + 1);
ffurl_read_complete(rt->rtsp_hd, content, content_length);
content[content_length] = '\0';
}
if (content_ptr)
*content_ptr = content;
else
av_free(content);
if (request) {
char buf[1024];
char base64buf[AV_BASE64_SIZE(sizeof(buf))];
const char* ptr = buf;
if (!strcmp(reply->reason, "OPTIONS")) {
snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
if (reply->seq)
av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
if (reply->session_id[0])
av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
reply->session_id);
} else {
snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
}
av_strlcat(buf, "\r\n", sizeof(buf));
if (rt->control_transport == RTSP_MODE_TUNNEL) {
av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
ptr = base64buf;
}
ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
rt->last_cmd_time = av_gettime();
/* Even if the request from the server had data, it is not the data
* that the caller wants or expects. The memory could also be leaked
* if the actual following reply has content data. */
if (content_ptr)
av_freep(content_ptr);
/* If method is set, this is called from ff_rtsp_send_cmd,
* where a reply to exactly this request is awaited. For
* callers from within packet receiving, we just want to
* return to the caller and go back to receiving packets. */
if (method)
goto start;
return 0;
}
if (rt->seq != reply->seq) {
av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
rt->seq, reply->seq);
}
/* EOS */
if (reply->notice == 2101 /* End-of-Stream Reached */ ||
reply->notice == 2104 /* Start-of-Stream Reached */ ||
reply->notice == 2306 /* Continuous Feed Terminated */) {
rt->state = RTSP_STATE_IDLE;
} else if (reply->notice >= 4400 && reply->notice < 5500) {
return AVERROR(EIO); /* data or server error */
} else if (reply->notice == 2401 /* Ticket Expired */ ||
(reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
return AVERROR(EPERM);
return 0;
}
/**
* Send a command to the RTSP server without waiting for the reply.
*
* @param s RTSP (de)muxer context
* @param method the method for the request
* @param url the target url for the request
* @param headers extra header lines to include in the request
* @param send_content if non-null, the data to send as request body content
* @param send_content_length the length of the send_content data, or 0 if
* send_content is null
*
* @return zero if success, nonzero otherwise
*/
static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
const char *method, const char *url,
const char *headers,
const unsigned char *send_content,
int send_content_length)
{
RTSPState *rt = s->priv_data;
char buf[4096], *out_buf;
char base64buf[AV_BASE64_SIZE(sizeof(buf))];
/* Add in RTSP headers */
out_buf = buf;
rt->seq++;
snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
if (headers)
av_strlcat(buf, headers, sizeof(buf));
av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", LIBAVFORMAT_IDENT);
if (rt->session_id[0] != '\0' && (!headers ||
!strstr(headers, "\nIf-Match:"))) {
av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
}
if (rt->auth[0]) {
char *str = ff_http_auth_create_response(&rt->auth_state,
rt->auth, url, method);
if (str)
av_strlcat(buf, str, sizeof(buf));
av_free(str);
}
if (send_content_length > 0 && send_content)
av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
av_strlcat(buf, "\r\n", sizeof(buf));
/* base64 encode rtsp if tunneling */
if (rt->control_transport == RTSP_MODE_TUNNEL) {
av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
out_buf = base64buf;
}
av_dlog(s, "Sending:\n%s--\n", buf);
ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
if (send_content_length > 0 && send_content) {
if (rt->control_transport == RTSP_MODE_TUNNEL) {
av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
"with content data not supported\n");
return AVERROR_PATCHWELCOME;
}
ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
}
rt->last_cmd_time = av_gettime();
return 0;
}
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
const char *url, const char *headers)
{
return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
}
int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
const char *headers, RTSPMessageHeader *reply,
unsigned char **content_ptr)
{
return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
content_ptr, NULL, 0);
}
int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
const char *method, const char *url,
const char *header,
RTSPMessageHeader *reply,
unsigned char **content_ptr,
const unsigned char *send_content,
int send_content_length)
{
RTSPState *rt = s->priv_data;
HTTPAuthType cur_auth_type;
int ret, attempts = 0;
retry:
cur_auth_type = rt->auth_state.auth_type;
if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
send_content,
send_content_length)))
return ret;
if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
return ret;
attempts++;
if (reply->status_code == 401 &&
(cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
goto retry;
if (reply->status_code > 400){
av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
method,
reply->status_code,
reply->reason);
av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
}
return 0;
}
int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
int lower_transport, const char *real_challenge)
{
RTSPState *rt = s->priv_data;
int rtx = 0, j, i, err, interleave = 0, port_off;
RTSPStream *rtsp_st;
RTSPMessageHeader reply1, *reply = &reply1;
char cmd[2048];
const char *trans_pref;
if (rt->transport == RTSP_TRANSPORT_RDT)
trans_pref = "x-pn-tng";
else if (rt->transport == RTSP_TRANSPORT_RAW)
trans_pref = "RAW/RAW";
else
trans_pref = "RTP/AVP";
/* default timeout: 1 minute */
rt->timeout = 60;
/* for each stream, make the setup request */
/* XXX: we assume the same server is used for the control of each
* RTSP stream */
/* Choose a random starting offset within the first half of the
* port range, to allow for a number of ports to try even if the offset
* happens to be at the end of the random range. */
port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
/* even random offset */
port_off -= port_off & 0x01;
for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
char transport[2048];
/*
* WMS serves all UDP data over a single connection, the RTX, which
* isn't necessarily the first in the SDP but has to be the first
* to be set up, else the second/third SETUP will fail with a 461.
*/
if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
rt->server_type == RTSP_SERVER_WMS) {
if (i == 0) {
/* rtx first */
for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
int len = strlen(rt->rtsp_streams[rtx]->control_url);
if (len >= 4 &&
!strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
"/rtx"))
break;
}
if (rtx == rt->nb_rtsp_streams)
return -1; /* no RTX found */
rtsp_st = rt->rtsp_streams[rtx];
} else
rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
} else
rtsp_st = rt->rtsp_streams[i];
/* RTP/UDP */
if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
char buf[256];
if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
port = reply->transports[0].client_port_min;
goto have_port;
}
/* first try in specified port range */
while (j <= rt->rtp_port_max) {
ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
"?localport=%d", j);
/* we will use two ports per rtp stream (rtp and rtcp) */
j += 2;
if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
&s->interrupt_callback, NULL))
goto rtp_opened;
}
av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
err = AVERROR(EIO);
goto fail;
rtp_opened:
port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
have_port:
snprintf(transport, sizeof(transport) - 1,
"%s/UDP;", trans_pref);
if (rt->server_type != RTSP_SERVER_REAL)
av_strlcat(transport, "unicast;", sizeof(transport));
av_strlcatf(transport, sizeof(transport),
"client_port=%d", port);
if (rt->transport == RTSP_TRANSPORT_RTP &&
!(rt->server_type == RTSP_SERVER_WMS && i > 0))
av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
}
/* RTP/TCP */
else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
/* For WMS streams, the application streams are only used for
* UDP. When trying to set it up for TCP streams, the server
* will return an error. Therefore, we skip those streams. */
if (rt->server_type == RTSP_SERVER_WMS &&
(rtsp_st->stream_index < 0 ||
s->streams[rtsp_st->stream_index]->codec->codec_type ==
AVMEDIA_TYPE_DATA))
continue;
snprintf(transport, sizeof(transport) - 1,
"%s/TCP;", trans_pref);
if (rt->transport != RTSP_TRANSPORT_RDT)
av_strlcat(transport, "unicast;", sizeof(transport));
av_strlcatf(transport, sizeof(transport),
"interleaved=%d-%d",
interleave, interleave + 1);
interleave += 2;
}
else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
snprintf(transport, sizeof(transport) - 1,
"%s/UDP;multicast", trans_pref);
}
if (s->oformat) {
av_strlcat(transport, ";mode=record", sizeof(transport));
} else if (rt->server_type == RTSP_SERVER_REAL ||
rt->server_type == RTSP_SERVER_WMS)
av_strlcat(transport, ";mode=play", sizeof(transport));
snprintf(cmd, sizeof(cmd),
"Transport: %s\r\n",
transport);
if (rt->accept_dynamic_rate)
av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
char real_res[41], real_csum[9];
ff_rdt_calc_response_and_checksum(real_res, real_csum,
real_challenge);
av_strlcatf(cmd, sizeof(cmd),
"If-Match: %s\r\n"
"RealChallenge2: %s, sd=%s\r\n",
rt->session_id, real_res, real_csum);
}
ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
err = 1;
goto fail;
} else if (reply->status_code != RTSP_STATUS_OK ||
reply->nb_transports != 1) {
err = AVERROR_INVALIDDATA;
goto fail;
}
/* XXX: same protocol for all streams is required */
if (i > 0) {
if (reply->transports[0].lower_transport != rt->lower_transport ||
reply->transports[0].transport != rt->transport) {
err = AVERROR_INVALIDDATA;
goto fail;
}
} else {
rt->lower_transport = reply->transports[0].lower_transport;
rt->transport = reply->transports[0].transport;
}
/* Fail if the server responded with another lower transport mode
* than what we requested. */
if (reply->transports[0].lower_transport != lower_transport) {
av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
err = AVERROR_INVALIDDATA;
goto fail;
}
switch(reply->transports[0].lower_transport) {
case RTSP_LOWER_TRANSPORT_TCP:
rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
break;
case RTSP_LOWER_TRANSPORT_UDP: {
char url[1024], options[30] = "";
const char *peer = host;
if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
av_strlcpy(options, "?connect=1", sizeof(options));
/* Use source address if specified */
if (reply->transports[0].source[0])
peer = reply->transports[0].source;
ff_url_join(url, sizeof(url), "rtp", NULL, peer,
reply->transports[0].server_port_min, "%s", options);
if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
/* Try to initialize the connection state in a
* potential NAT router by sending dummy packets.
* RTP/RTCP dummy packets are used for RDT, too.
*/
if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
CONFIG_RTPDEC)
ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
break;
}
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
char url[1024], namebuf[50], optbuf[20] = "";
struct sockaddr_storage addr;
int port, ttl;
if (reply->transports[0].destination.ss_family) {
addr = reply->transports[0].destination;
port = reply->transports[0].port_min;
ttl = reply->transports[0].ttl;
} else {
addr = rtsp_st->sdp_ip;
port = rtsp_st->sdp_port;
ttl = rtsp_st->sdp_ttl;
}
if (ttl > 0)
snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
getnameinfo((struct sockaddr*) &addr, sizeof(addr),
namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
port, "%s", optbuf);
if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
&s->interrupt_callback, NULL) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
break;
}
}
if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
goto fail;
}
if (rt->nb_rtsp_streams && reply->timeout > 0)
rt->timeout = reply->timeout;
if (rt->server_type == RTSP_SERVER_REAL)
rt->need_subscription = 1;
return 0;
fail:
ff_rtsp_undo_setup(s);
return err;
}
void ff_rtsp_close_connections(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
ffurl_close(rt->rtsp_hd);
rt->rtsp_hd = rt->rtsp_hd_out = NULL;
}
int ff_rtsp_connect(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
int port, err, tcp_fd;
RTSPMessageHeader reply1 = {0}, *reply = &reply1;
int lower_transport_mask = 0;
char real_challenge[64] = "";
struct sockaddr_storage peer;
socklen_t peer_len = sizeof(peer);
if (rt->rtp_port_max < rt->rtp_port_min) {
av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
"than min port %d\n", rt->rtp_port_max,
rt->rtp_port_min);
return AVERROR(EINVAL);
}
if (!ff_network_init())
return AVERROR(EIO);
if (s->max_delay < 0) /* Not set by the caller */
s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
rt->control_transport = RTSP_MODE_PLAIN;
if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
rt->control_transport = RTSP_MODE_TUNNEL;
}
/* Only pass through valid flags from here */
rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
redirect:
lower_transport_mask = rt->lower_transport_mask;
/* extract hostname and port */
av_url_split(NULL, 0, auth, sizeof(auth),
host, sizeof(host), &port, path, sizeof(path), s->filename);
if (*auth) {
av_strlcpy(rt->auth, auth, sizeof(rt->auth));
}
if (port < 0)
port = RTSP_DEFAULT_PORT;
if (!lower_transport_mask)
lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
if (s->oformat) {
/* Only UDP or TCP - UDP multicast isn't supported. */
lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
(1 << RTSP_LOWER_TRANSPORT_TCP);
if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
"only UDP and TCP are supported for output.\n");
err = AVERROR(EINVAL);
goto fail;
}
}
/* Construct the URI used in request; this is similar to s->filename,
* but with authentication credentials removed and RTSP specific options
* stripped out. */
ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
host, port, "%s", path);
if (rt->control_transport == RTSP_MODE_TUNNEL) {
/* set up initial handshake for tunneling */
char httpname[1024];
char sessioncookie[17];
char headers[1024];
ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
av_get_random_seed(), av_get_random_seed());
/* GET requests */
if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
&s->interrupt_callback) < 0) {
err = AVERROR(EIO);
goto fail;
}
/* generate GET headers */
snprintf(headers, sizeof(headers),
"x-sessioncookie: %s\r\n"
"Accept: application/x-rtsp-tunnelled\r\n"
"Pragma: no-cache\r\n"
"Cache-Control: no-cache\r\n",
sessioncookie);
av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
/* complete the connection */
if (ffurl_connect(rt->rtsp_hd, NULL)) {
err = AVERROR(EIO);
goto fail;
}
/* POST requests */
if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
&s->interrupt_callback) < 0 ) {
err = AVERROR(EIO);
goto fail;
}
/* generate POST headers */
snprintf(headers, sizeof(headers),
"x-sessioncookie: %s\r\n"
"Content-Type: application/x-rtsp-tunnelled\r\n"
"Pragma: no-cache\r\n"
"Cache-Control: no-cache\r\n"
"Content-Length: 32767\r\n"
"Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
sessioncookie);
av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
/* Initialize the authentication state for the POST session. The HTTP
* protocol implementation doesn't properly handle multi-pass
* authentication for POST requests, since it would require one of
* the following:
* - implementing Expect: 100-continue, which many HTTP servers
* don't support anyway, even less the RTSP servers that do HTTP
* tunneling
* - sending the whole POST data until getting a 401 reply specifying
* what authentication method to use, then resending all that data
* - waiting for potential 401 replies directly after sending the
* POST header (waiting for some unspecified time)
* Therefore, we copy the full auth state, which works for both basic
* and digest. (For digest, we would have to synchronize the nonce
* count variable between the two sessions, if we'd do more requests
* with the original session, though.)
*/
ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
/* complete the connection */
if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
err = AVERROR(EIO);
goto fail;
}
} else {
/* open the tcp connection */
ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
&s->interrupt_callback, NULL) < 0) {
err = AVERROR(EIO);
goto fail;
}
rt->rtsp_hd_out = rt->rtsp_hd;
}
rt->seq = 0;
tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
NULL, 0, NI_NUMERICHOST);
}
/* request options supported by the server; this also detects server
* type */
for (rt->server_type = RTSP_SERVER_RTP;;) {
cmd[0] = 0;
if (rt->server_type == RTSP_SERVER_REAL)
av_strlcat(cmd,
/*
* The following entries are required for proper
* streaming from a Realmedia server. They are
* interdependent in some way although we currently
* don't quite understand how. Values were copied
* from mplayer SVN r23589.
* ClientChallenge is a 16-byte ID in hex
* CompanyID is a 16-byte ID in base64
*/
"ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
"PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
"CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
"GUID: 00000000-0000-0000-0000-000000000000\r\n",
sizeof(cmd));
ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
err = AVERROR_INVALIDDATA;
goto fail;
}
/* detect server type if not standard-compliant RTP */
if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
rt->server_type = RTSP_SERVER_REAL;
continue;
} else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
rt->server_type = RTSP_SERVER_WMS;
} else if (rt->server_type == RTSP_SERVER_REAL)
strcpy(real_challenge, reply->real_challenge);
break;
}
if (s->iformat && CONFIG_RTSP_DEMUXER)
err = ff_rtsp_setup_input_streams(s, reply);
else if (CONFIG_RTSP_MUXER)
err = ff_rtsp_setup_output_streams(s, host);
if (err)
goto fail;
do {
int lower_transport = ff_log2_tab[lower_transport_mask &
~(lower_transport_mask - 1)];
err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
rt->server_type == RTSP_SERVER_REAL ?
real_challenge : NULL);
if (err < 0)
goto fail;
lower_transport_mask &= ~(1 << lower_transport);
if (lower_transport_mask == 0 && err == 1) {
err = AVERROR(EPROTONOSUPPORT);
goto fail;
}
} while (err);
rt->lower_transport_mask = lower_transport_mask;
av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
rt->state = RTSP_STATE_IDLE;
rt->seek_timestamp = 0; /* default is to start stream at position zero */
return 0;
fail:
ff_rtsp_close_streams(s);
ff_rtsp_close_connections(s);
if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
av_strlcpy(s->filename, reply->location, sizeof(s->filename));
av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
reply->status_code,
s->filename);
goto redirect;
}
ff_network_close();
return err;
}
#endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
#if CONFIG_RTPDEC
static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
uint8_t *buf, int buf_size, int64_t wait_end)
{
RTSPState *rt = s->priv_data;
RTSPStream *rtsp_st;
int n, i, ret, tcp_fd, timeout_cnt = 0;
int max_p = 0;
struct pollfd *p = rt->p;
int *fds = NULL, fdsnum, fdsidx;
for (;;) {
if (ff_check_interrupt(&s->interrupt_callback))
return AVERROR_EXIT;
if (wait_end && wait_end - av_gettime() < 0)
return AVERROR(EAGAIN);
max_p = 0;
if (rt->rtsp_hd) {
tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
p[max_p].fd = tcp_fd;
p[max_p++].events = POLLIN;
} else {
tcp_fd = -1;
}
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
if (rtsp_st->rtp_handle) {
if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
&fds, &fdsnum)) {
av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
return ret;
}
if (fdsnum != 2) {
av_log(s, AV_LOG_ERROR,
"Number of fds %d not supported\n", fdsnum);
return AVERROR_INVALIDDATA;
}
for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
p[max_p].fd = fds[fdsidx];
p[max_p++].events = POLLIN;
}
av_free(fds);
}
}
n = poll(p, max_p, POLL_TIMEOUT_MS);
if (n > 0) {
int j = 1 - (tcp_fd == -1);
timeout_cnt = 0;
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
if (rtsp_st->rtp_handle) {
if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
if (ret > 0) {
*prtsp_st = rtsp_st;
return ret;
}
}
j+=2;
}
}
#if CONFIG_RTSP_DEMUXER
if (tcp_fd != -1 && p[0].revents & POLLIN) {
if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
if (rt->state == RTSP_STATE_STREAMING) {
if (!ff_rtsp_parse_streaming_commands(s))
return AVERROR_EOF;
else
av_log(s, AV_LOG_WARNING,
"Unable to answer to TEARDOWN\n");
} else
return 0;
} else {
RTSPMessageHeader reply;
ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
if (ret < 0)
return ret;
/* XXX: parse message */
if (rt->state != RTSP_STATE_STREAMING)
return 0;
}
}
#endif
} else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
return AVERROR(ETIMEDOUT);
} else if (n < 0 && errno != EINTR)
return AVERROR(errno);
}
}
static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
const uint8_t *buf, int len)
{
RTSPState *rt = s->priv_data;
int i;
if (len < 0)
return len;
if (rt->nb_rtsp_streams == 1) {
*rtsp_st = rt->rtsp_streams[0];
return len;
}
if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
int no_ssrc = 0;
for (i = 0; i < rt->nb_rtsp_streams; i++) {
RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
if (!rtpctx)
continue;
if (rtpctx->ssrc == AV_RB32(&buf[4])) {
*rtsp_st = rt->rtsp_streams[i];
return len;
}
if (!rtpctx->ssrc)
no_ssrc = 1;
}
if (no_ssrc) {
av_log(s, AV_LOG_WARNING,
"Unable to pick stream for packet - SSRC not known for "
"all streams\n");
return AVERROR(EAGAIN);
}
} else {
for (i = 0; i < rt->nb_rtsp_streams; i++) {
if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
*rtsp_st = rt->rtsp_streams[i];
return len;
}
}
}
}
av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
return AVERROR(EAGAIN);
}
int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
{
RTSPState *rt = s->priv_data;
int ret, len;
RTSPStream *rtsp_st, *first_queue_st = NULL;
int64_t wait_end = 0;
if (rt->nb_byes == rt->nb_rtsp_streams)
return AVERROR_EOF;
/* get next frames from the same RTP packet */
if (rt->cur_transport_priv) {
if (rt->transport == RTSP_TRANSPORT_RDT) {
ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
} else if (rt->transport == RTSP_TRANSPORT_RTP) {
ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
} else if (rt->ts && CONFIG_RTPDEC) {
ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
if (ret >= 0) {
rt->recvbuf_pos += ret;
ret = rt->recvbuf_pos < rt->recvbuf_len;
}
} else
ret = -1;
if (ret == 0) {
rt->cur_transport_priv = NULL;
return 0;
} else if (ret == 1) {
return 0;
} else
rt->cur_transport_priv = NULL;
}
redo:
if (rt->transport == RTSP_TRANSPORT_RTP) {
int i;
int64_t first_queue_time = 0;
for (i = 0; i < rt->nb_rtsp_streams; i++) {
RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
int64_t queue_time;
if (!rtpctx)
continue;
queue_time = ff_rtp_queued_packet_time(rtpctx);
if (queue_time && (queue_time - first_queue_time < 0 ||
!first_queue_time)) {
first_queue_time = queue_time;
first_queue_st = rt->rtsp_streams[i];
}
}
if (first_queue_time) {
wait_end = first_queue_time + s->max_delay;
} else {
wait_end = 0;
first_queue_st = NULL;
}
}
/* read next RTP packet */
if (!rt->recvbuf) {
rt->recvbuf = av_malloc(RECVBUF_SIZE);
if (!rt->recvbuf)
return AVERROR(ENOMEM);
}
switch(rt->lower_transport) {
default:
#if CONFIG_RTSP_DEMUXER
case RTSP_LOWER_TRANSPORT_TCP:
len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
break;
#endif
case RTSP_LOWER_TRANSPORT_UDP:
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
break;
case RTSP_LOWER_TRANSPORT_CUSTOM:
if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
wait_end && wait_end < av_gettime())
len = AVERROR(EAGAIN);
else
len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
break;
}
if (len == AVERROR(EAGAIN) && first_queue_st &&
rt->transport == RTSP_TRANSPORT_RTP) {
rtsp_st = first_queue_st;
ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
goto end;
}
if (len < 0)
return len;
if (len == 0)
return AVERROR_EOF;
if (rt->transport == RTSP_TRANSPORT_RDT) {
ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
} else if (rt->transport == RTSP_TRANSPORT_RTP) {
ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
if (rtsp_st->feedback) {
AVIOContext *pb = NULL;
if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
pb = s->pb;
ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
}
if (ret < 0) {
/* Either bad packet, or a RTCP packet. Check if the
* first_rtcp_ntp_time field was initialized. */
RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
/* first_rtcp_ntp_time has been initialized for this stream,
* copy the same value to all other uninitialized streams,
* in order to map their timestamp origin to the same ntp time
* as this one. */
int i;
AVStream *st = NULL;
if (rtsp_st->stream_index >= 0)
st = s->streams[rtsp_st->stream_index];
for (i = 0; i < rt->nb_rtsp_streams; i++) {
RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
AVStream *st2 = NULL;
if (rt->rtsp_streams[i]->stream_index >= 0)
st2 = s->streams[rt->rtsp_streams[i]->stream_index];
if (rtpctx2 && st && st2 &&
rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
rtpctx2->rtcp_ts_offset = av_rescale_q(
rtpctx->rtcp_ts_offset, st->time_base,
st2->time_base);
}
}
}
if (ret == -RTCP_BYE) {
rt->nb_byes++;
av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
if (rt->nb_byes == rt->nb_rtsp_streams)
return AVERROR_EOF;
}
}
} else if (rt->ts && CONFIG_RTPDEC) {
ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
if (ret >= 0) {
if (ret < len) {
rt->recvbuf_len = len;
rt->recvbuf_pos = ret;
rt->cur_transport_priv = rt->ts;
return 1;
} else {
ret = 0;
}
}
} else {
return AVERROR_INVALIDDATA;
}
end:
if (ret < 0)
goto redo;
if (ret == 1)
/* more packets may follow, so we save the RTP context */
rt->cur_transport_priv = rtsp_st->transport_priv;
return ret;
}
#endif /* CONFIG_RTPDEC */
#if CONFIG_SDP_DEMUXER
static int sdp_probe(AVProbeData *p1)
{
const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
/* we look for a line beginning "c=IN IP" */
while (p < p_end && *p != '\0') {
if (p + sizeof("c=IN IP") - 1 < p_end &&
av_strstart(p, "c=IN IP", NULL))
return AVPROBE_SCORE_EXTENSION;
while (p < p_end - 1 && *p != '\n') p++;
if (++p >= p_end)
break;
if (*p == '\r')
p++;
}
return 0;
}
static void append_source_addrs(char *buf, int size, const char *name,
int count, struct RTSPSource **addrs)
{
int i;
if (!count)
return;
av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
for (i = 1; i < count; i++)
av_strlcatf(buf, size, ",%s", addrs[i]->addr);
}
static int sdp_read_header(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
RTSPStream *rtsp_st;
int size, i, err;
char *content;
char url[1024];
if (!ff_network_init())
return AVERROR(EIO);
if (s->max_delay < 0) /* Not set by the caller */
s->max_delay = DEFAULT_REORDERING_DELAY;
if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
/* read the whole sdp file */
/* XXX: better loading */
content = av_malloc(SDP_MAX_SIZE);
size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
if (size <= 0) {
av_free(content);
return AVERROR_INVALIDDATA;
}
content[size] ='\0';
err = ff_sdp_parse(s, content);
av_free(content);
if (err) goto fail;
/* open each RTP stream */
for (i = 0; i < rt->nb_rtsp_streams; i++) {
char namebuf[50];
rtsp_st = rt->rtsp_streams[i];
if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
ff_url_join(url, sizeof(url), "rtp", NULL,
namebuf, rtsp_st->sdp_port,
"?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
rtsp_st->sdp_port, rtsp_st->sdp_ttl,
rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
append_source_addrs(url, sizeof(url), "sources",
rtsp_st->nb_include_source_addrs,
rtsp_st->include_source_addrs);
append_source_addrs(url, sizeof(url), "block",
rtsp_st->nb_exclude_source_addrs,
rtsp_st->exclude_source_addrs);
if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
&s->interrupt_callback, NULL) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
}
if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
goto fail;
}
return 0;
fail:
ff_rtsp_close_streams(s);
ff_network_close();
return err;
}
static int sdp_read_close(AVFormatContext *s)
{
ff_rtsp_close_streams(s);
ff_network_close();
return 0;
}
static const AVClass sdp_demuxer_class = {
.class_name = "SDP demuxer",
.item_name = av_default_item_name,
.option = sdp_options,
.version = LIBAVUTIL_VERSION_INT,
};
AVInputFormat ff_sdp_demuxer = {
.name = "sdp",
.long_name = NULL_IF_CONFIG_SMALL("SDP"),
.priv_data_size = sizeof(RTSPState),
.read_probe = sdp_probe,
.read_header = sdp_read_header,
.read_packet = ff_rtsp_fetch_packet,
.read_close = sdp_read_close,
.priv_class = &sdp_demuxer_class,
};
#endif /* CONFIG_SDP_DEMUXER */
#if CONFIG_RTP_DEMUXER
static int rtp_probe(AVProbeData *p)
{
if (av_strstart(p->filename, "rtp:", NULL))
return AVPROBE_SCORE_MAX;
return 0;
}
static int rtp_read_header(AVFormatContext *s)
{
uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
char host[500], sdp[500];
int ret, port;
URLContext* in = NULL;
int payload_type;
AVCodecContext codec = { 0 };
struct sockaddr_storage addr;
AVIOContext pb;
socklen_t addrlen = sizeof(addr);
RTSPState *rt = s->priv_data;
if (!ff_network_init())
return AVERROR(EIO);
ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
&s->interrupt_callback, NULL);
if (ret)
goto fail;
while (1) {
ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
if (ret == AVERROR(EAGAIN))
continue;
if (ret < 0)
goto fail;
if (ret < 12) {
av_log(s, AV_LOG_WARNING, "Received too short packet\n");
continue;
}
if ((recvbuf[0] & 0xc0) != 0x80) {
av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
"received\n");
continue;
}
if (RTP_PT_IS_RTCP(recvbuf[1]))
continue;
payload_type = recvbuf[1] & 0x7f;
break;
}
getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
ffurl_close(in);
in = NULL;
if (ff_rtp_get_codec_info(&codec, payload_type)) {
av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
"without an SDP file describing it\n",
payload_type);
goto fail;
}
if (codec.codec_type != AVMEDIA_TYPE_DATA) {
av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
"properly you need an SDP file "
"describing it\n");
}
av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
NULL, 0, s->filename);
snprintf(sdp, sizeof(sdp),
"v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
addr.ss_family == AF_INET ? 4 : 6, host,
codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
port, payload_type);
av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
s->pb = &pb;
/* sdp_read_header initializes this again */
ff_network_close();
rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
ret = sdp_read_header(s);
s->pb = NULL;
return ret;
fail:
if (in)
ffurl_close(in);
ff_network_close();
return ret;
}
static const AVClass rtp_demuxer_class = {
.class_name = "RTP demuxer",
.item_name = av_default_item_name,
.option = rtp_options,
.version = LIBAVUTIL_VERSION_INT,
};
AVInputFormat ff_rtp_demuxer = {
.name = "rtp",
.long_name = NULL_IF_CONFIG_SMALL("RTP input"),
.priv_data_size = sizeof(RTSPState),
.read_probe = rtp_probe,
.read_header = rtp_read_header,
.read_packet = ff_rtsp_fetch_packet,
.read_close = sdp_read_close,
.flags = AVFMT_NOFILE,
.priv_class = &rtp_demuxer_class,
};
#endif /* CONFIG_RTP_DEMUXER */
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