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FFmpeg/libavformat/rtsp.c
Moritz Barsnick 10b8481a26 fix several typos
2014-06-20 10:21:31 -08:00

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/*
* RTSP/SDP client
* 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
*/
#include "libavutil/avassert.h"
#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", "set 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", "do not 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", "set 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", "set RTSP flags"),
{ "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
{ "prefer_tcp", "try RTP via TCP first, if available", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_PREFER_TCP}, 0, 0, DEC|ENC, "rtsp_flags" },
RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
{ "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
{ "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
{ "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
{ "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
RTSP_REORDERING_OPTS(),
{ "user-agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
{ NULL },
};
static const AVOption sdp_options[] = {
RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
{ "custom_io", "use custom I/O", 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", "set media types to accept from the server"),
RTSP_REORDERING_OPTS(),
{ NULL },
};
static const AVOption rtp_options[] = {
RTSP_FLAG_OPTS("rtp_flags", "set 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) */
int seen_rtpmap;
int seen_fmtp;
char delayed_fmtp[2048];
} 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 parse_fmtp(AVFormatContext *s, RTSPState *rt,
int payload_type, const char *line)
{
int i;
for (i = 0; i < rt->nb_rtsp_streams; i++) {
RTSPStream *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, line);
}
}
}
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;
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;
s1->seen_fmtp = 0;
s1->seen_rtpmap = 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);
}
s1->seen_rtpmap = 1;
if (s1->seen_fmtp) {
parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
}
} else if (av_strstart(p, "fmtp:", &p) ||
av_strstart(p, "framesize:", &p)) {
// let dynamic protocol handlers have a stab at the line.
get_word(buf1, sizeof(buf1), &p);
payload_type = atoi(buf1);
if (s1->seen_rtpmap) {
parse_fmtp(s, rt, payload_type, buf);
} else {
s1->seen_fmtp = 1;
av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
}
} 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, int send_packets)
{
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;
if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
ff_rtsp_tcp_write_packet(s, rtsp_st);
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, 0);
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((AVFormatContext **)&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;
st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
} 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", rt->user_agent);
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;
/* 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, 0);
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,
"?timeout=%d", rt->stimeout);
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)];
if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
&& (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
lower_transport = RTSP_LOWER_TRANSPORT_TCP;
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);
}
}
// Make real NTP start time available in AVFormatContext
if (s->start_time_realtime == AV_NOPTS_VALUE) {
s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
if (rtpctx->st) {
s->start_time_realtime -=
av_rescale (rtpctx->rtcp_ts_offset,
(uint64_t) rtpctx->st->time_base.num * 1000000,
rtpctx->st->time_base.den);
}
}
}
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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