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FFmpeg/libavformat/rtsp.c
Jonathan Baudanza 6b3f9c2e92 avformat/rtpdec: fix integer overflow in start_time_realtime calculation 
I encountered this problem with NTP timestamps that are extremely old,
like from January, 1990.

Although RFC3550 suggests that the timestamps in the RTCP packets use
the actual wallclock, some implementations use other clocks, such as
the CLOCK_MONOTONIC on linux.

I'm my case, I'm dealing with packets from mediasoup.

Without this patch, start_time_realtime shows up in the distance future
instead of around Jan 1900.

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2024-09-23 17:08:33 +02:00

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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 "config_components.h"
#include "libavutil/avassert.h"
#include "libavutil/base64.h"
#include "libavutil/bprint.h"
#include "libavutil/avstring.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/mathematics.h"
#include "libavutil/mem.h"
#include "libavutil/parseutils.h"
#include "libavutil/random_seed.h"
#include "libavutil/dict.h"
#include "libavutil/opt.h"
#include "libavutil/time.h"
#include "libavcodec/codec_desc.h"
#include "avformat.h"
#include "avio_internal.h"
#include "demux.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"
#include "version.h"
/* Default timeout values for read packet in seconds */
#define READ_PACKET_TIMEOUT_S 10
#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, .unit = "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, .unit = "rtsp_flags" }
#define RTSP_MEDIATYPE_OPTS(name, longname) \
{ name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1 }, INT_MIN, INT_MAX, DEC, .unit = "allowed_media_types" }, \
{ "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, .unit = "allowed_media_types" }, \
{ "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, .unit = "allowed_media_types" }, \
{ "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, .unit = "allowed_media_types" }, \
{ "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, .unit = "allowed_media_types" }
#define COMMON_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 }, \
{ "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC }, \
{ "pkt_size", "Underlying protocol send packet size", OFFSET(pkt_size), AV_OPT_TYPE_INT, { .i64 = 1472 }, -1, INT_MAX, ENC } \
const AVOption ff_rtsp_options[] = {
{ "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_BOOL, {.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, .unit = "rtsp_transport" }, \
{ "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, .unit = "rtsp_transport" }, \
{ "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, .unit = "rtsp_transport" }, \
{ "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, .unit = "rtsp_transport" },
{ "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, .unit = "rtsp_transport" },
{ "https", "HTTPS tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTPS )}, 0, 0, DEC, .unit = "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, .unit = "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, .unit = "rtsp_flags" },
{ "satip_raw", "export raw MPEG-TS stream instead of demuxing", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_SATIP_RAW}, 0, 0, DEC, .unit = "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 },
{ "listen_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 },
{ "timeout", "set timeout (in microseconds) of socket I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT64, {.i64 = 0}, INT_MIN, INT64_MAX, DEC },
COMMON_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, .unit = "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, .unit = "rtsp_flags" },
{ "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(stimeout), AV_OPT_TYPE_DURATION, {.i64 = READ_PACKET_TIMEOUT_S*1000000}, INT_MIN, INT64_MAX, DEC },
{ "localaddr", "local address", OFFSET(localaddr),AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, DEC }, \
RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
COMMON_OPTS(),
{ NULL },
};
static const AVOption rtp_options[] = {
RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
{ "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(stimeout), AV_OPT_TYPE_DURATION, {.i64 = READ_PACKET_TIMEOUT_S*1000000}, INT_MIN, INT64_MAX, DEC },
{ "localaddr", "local address", OFFSET(localaddr),AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, DEC }, \
RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
COMMON_OPTS(),
{ NULL },
};
static AVDictionary *map_to_opts(RTSPState *rt)
{
AVDictionary *opts = NULL;
av_dict_set_int(&opts, "buffer_size", rt->buffer_size, 0);
av_dict_set_int(&opts, "pkt_size", rt->pkt_size, 0);
if (rt->localaddr && rt->localaddr[0])
av_dict_set(&opts, "localaddr", rt->localaddr, 0);
return opts;
}
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);
if (av_parse_time(start, buf, 1) < 0)
return;
if (*p == '-') {
p++;
get_word_sep(buf, sizeof(buf), "-", &p);
if (av_parse_time(end, buf, 1) < 0)
av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
}
}
static int get_sockaddr(AVFormatContext *s,
const char *buf, struct sockaddr_storage *sock)
{
struct addrinfo hints = { 0 }, *ai = NULL;
int ret;
hints.ai_flags = AI_NUMERICHOST;
if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
buf,
gai_strerror(ret));
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(const RTPDynamicProtocolHandler *handler,
RTSPStream *rtsp_st, AVStream *st)
{
AVCodecParameters *par = st ? st->codecpar : NULL;
if (!handler)
return;
if (par)
par->codec_id = handler->codec_id;
rtsp_st->dynamic_handler = handler;
if (st)
ffstream(st)->need_parsing = handler->need_parsing;
if (handler->priv_data_size) {
rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
if (!rtsp_st->dynamic_protocol_context)
rtsp_st->dynamic_handler = NULL;
}
}
static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
AVStream *st)
{
if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
rtsp_st->dynamic_protocol_context);
if (ret < 0) {
if (rtsp_st->dynamic_protocol_context) {
if (rtsp_st->dynamic_handler->close)
rtsp_st->dynamic_handler->close(
rtsp_st->dynamic_protocol_context);
av_free(rtsp_st->dynamic_protocol_context);
}
rtsp_st->dynamic_protocol_context = NULL;
rtsp_st->dynamic_handler = NULL;
}
}
}
#if CONFIG_RTSP_DEMUXER
static int init_satip_stream(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
RTSPStream *rtsp_st = av_mallocz(sizeof(RTSPStream));
if (!rtsp_st)
return AVERROR(ENOMEM);
dynarray_add(&rt->rtsp_streams,
&rt->nb_rtsp_streams, rtsp_st);
rtsp_st->sdp_payload_type = 33; // MP2T
av_strlcpy(rtsp_st->control_url,
rt->control_uri, sizeof(rtsp_st->control_url));
if (rt->rtsp_flags & RTSP_FLAG_SATIP_RAW) {
AVStream *st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->id = rt->nb_rtsp_streams - 1;
rtsp_st->stream_index = st->index;
st->codecpar->codec_type = AVMEDIA_TYPE_DATA;
st->codecpar->codec_id = AV_CODEC_ID_MPEG2TS;
} else {
rtsp_st->stream_index = -1;
init_rtp_handler(&ff_mpegts_dynamic_handler, rtsp_st, NULL);
finalize_rtp_handler_init(s, rtsp_st, NULL);
}
return 0;
}
#endif
/* 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)
{
AVCodecParameters *par = st->codecpar;
char buf[256];
int i;
const AVCodecDescriptor *desc;
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). */
par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
}
if (par->codec_id == AV_CODEC_ID_NONE) {
const RTPDynamicProtocolHandler *handler =
ff_rtp_handler_find_by_name(buf, par->codec_type);
init_rtp_handler(handler, rtsp_st, st);
/* 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)
par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
}
desc = avcodec_descriptor_get(par->codec_id);
if (desc && desc->name)
c_name = desc->name;
else
c_name = "(null)";
get_word_sep(buf, sizeof(buf), "/", &p);
i = atoi(buf);
switch (par->codec_type) {
case AVMEDIA_TYPE_AUDIO:
av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
par->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
par->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
if (i > 0) {
par->sample_rate = i;
avpriv_set_pts_info(st, 32, 1, par->sample_rate);
get_word_sep(buf, sizeof(buf), "/", &p);
i = atoi(buf);
if (i > 0)
av_channel_layout_default(&par->ch_layout, i);
}
av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
par->sample_rate);
av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
par->ch_layout.nb_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;
}
finalize_rtp_handler_init(s, rtsp_st, st);
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_memdup(rtsp_src, sizeof(*rtsp_src));
if (!rtsp_src2)
continue;
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, rtsp_st->stream_index,
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_log(s, AV_LOG_TRACE, "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(s, 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;
} else if (!strcmp(st_type, "text")) {
codec_type = AVMEDIA_TYPE_SUBTITLE;
}
if (codec_type == AVMEDIA_TYPE_UNKNOWN ||
!(rt->media_type_mask & (1 << codec_type)) ||
rt->nb_rtsp_streams >= s->max_streams
) {
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 (CONFIG_RTPDEC && !rt->ts)
rt->ts = avpriv_mpegts_parse_open(s);
} else {
const RTPDynamicProtocolHandler *handler;
handler = ff_rtp_handler_find_by_id(
rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
init_rtp_handler(handler, rtsp_st, NULL);
finalize_rtp_handler_init(s, rtsp_st, NULL);
}
} 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->codecpar->codec_type = codec_type;
if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
const RTPDynamicProtocolHandler *handler;
/* if standard payload type, we can find the codec right now */
ff_rtp_get_codec_info(st->codecpar, rtsp_st->sdp_payload_type);
if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
st->codecpar->sample_rate > 0)
avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
/* Even static payload types may need a custom depacketizer */
handler = ff_rtp_handler_find_by_id(
rtsp_st->sdp_payload_type, st->codecpar->codec_type);
init_rtp_handler(handler, rtsp_st, st);
finalize_rtp_handler_init(s, rtsp_st, st);
}
if (rt->default_lang[0])
av_dict_set(&st->metadata, "language", rt->default_lang, 0);
}
/* 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 (rt->nb_rtsp_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, "ssrc:", &p) && s->nb_streams > 0) {
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
get_word(buf1, sizeof(buf1), &p);
rtsp_st->ssrc = strtoll(buf1, NULL, 10);
} 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, "lang:", &p)) {
if (s->nb_streams > 0) {
get_word(buf1, sizeof(buf1), &p);
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
if (rtsp_st->stream_index >= 0) {
st = s->streams[rtsp_st->stream_index];
av_dict_set(&st->metadata, "language", buf1, 0);
}
} else
get_word(rt->default_lang, sizeof(rt->default_lang), &p);
} 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->codecpar->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)
{
const char *p;
int letter, i;
char buf[SDP_MAX_SIZE], *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_freep(&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_freep(&s1->default_exclude_source_addrs[i]);
av_freep(&s1->default_exclude_source_addrs);
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) {
if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
ff_rtsp_tcp_write_packet(s, rtsp_st);
ffio_free_dyn_buf(&rtpctx->pb);
} else {
avio_closep(&rtpctx->pb);
}
avformat_free_context(rtpctx);
} else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
ff_rdt_parse_close(rtsp_st->transport_priv);
else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
ff_rtp_parse_close(rtsp_st->transport_priv);
}
rtsp_st->transport_priv = NULL;
ffurl_closep(&rtsp_st->rtp_handle);
}
}
/* 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) {
if (rtsp_st->dynamic_handler->close)
rtsp_st->dynamic_handler->close(
rtsp_st->dynamic_protocol_context);
av_free(rtsp_st->dynamic_protocol_context);
}
for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
av_freep(&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_freep(&rtsp_st->exclude_source_addrs[j]);
av_freep(&rtsp_st->exclude_source_addrs);
av_freep(&rtsp_st);
}
}
av_freep(&rt->rtsp_streams);
if (rt->asf_ctx) {
avformat_close_input(&rt->asf_ctx);
}
if (CONFIG_RTPDEC && rt->ts)
avpriv_mpegts_parse_close(rt->ts);
av_freep(&rt->p);
av_freep(&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 (CONFIG_RTSP_MUXER && s->oformat && st) {
int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
s, st, rtsp_st->rtp_handle,
rt->pkt_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 (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
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 (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP &&
s->iformat) {
RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
rtpctx->ssrc = rtsp_st->ssrc;
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(AVFormatContext *s,
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;
} else {
break;
}
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(s, 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 (!av_strcasecmp(buf, "record") ||
!av_strcasecmp(buf, "receive"))
th->mode_record = 1;
}
}
while (*p != ';' && *p != '\0' && *p != ',')
p++;
if (*p == ';')
p++;
}
if (*p == ',')
p++;
reply->nb_transports++;
if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
break;
}
}
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[MAX_URL_SIZE], url[MAX_URL_SIZE] = "";
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(AVFormatContext *s,
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(s, 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));
} else if (av_stristart(p, "com.ses.streamID:", &p)) {
p += strspn(p, SPACE_CHARS);
av_strlcpy(reply->stream_id, p, sizeof(reply->stream_id));
}
}
/* skip a RTP/TCP interleaved packet */
int ff_rtsp_skip_packet(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
int ret, len, len1;
uint8_t buf[MAX_URL_SIZE];
ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
if (ret != 3)
return ret < 0 ? ret : AVERROR(EIO);
len = AV_RB16(buf + 1);
av_log(s, AV_LOG_TRACE, "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 ret < 0 ? ret : AVERROR(EIO);
len -= len1;
}
return 0;
}
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[MAX_URL_SIZE], buf1[MAX_URL_SIZE], *q;
unsigned char ch;
const char *p;
int ret, content_length, line_count, request;
unsigned char *content;
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_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
if (ret != 1)
return ret < 0 ? ret : AVERROR(EIO);
if (ch == '\n')
break;
if (ch == '$' && q == buf) {
if (return_on_interleaved_data) {
return 1;
} else {
ret = ff_rtsp_skip_packet(s);
if (ret < 0)
return ret;
}
} else if (ch != '\r') {
if ((q - buf) < sizeof(buf) - 1)
*q++ = ch;
}
}
*q = '\0';
av_log(s, AV_LOG_TRACE, "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(s, 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);
if (!content)
return AVERROR(ENOMEM);
if ((ret = ffurl_read_complete(rt->rtsp_hd, content, content_length)) != content_length) {
av_freep(&content);
return ret < 0 ? ret : AVERROR(EIO);
}
content[content_length] = '\0';
}
if (content_ptr)
*content_ptr = content;
else
av_freep(&content);
if (request) {
char buf[MAX_URL_SIZE];
char base64buf[AV_BASE64_SIZE(sizeof(buf))];
const char* ptr = buf;
if (!strcmp(reply->reason, "OPTIONS") ||
!strcmp(reply->reason, "GET_PARAMETER")) {
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_relative();
/* 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[MAX_URL_SIZE], *out_buf;
char base64buf[AV_BASE64_SIZE(sizeof(buf))];
if (!rt->rtsp_hd_out)
return AVERROR(ENOTCONN);
/* 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_log(s, AV_LOG_TRACE, "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) {
avpriv_report_missing_feature(s, "Tunneling of RTSP requests with content data");
return AVERROR_PATCHWELCOME;
}
ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
}
rt->last_cmd_time = av_gettime_relative();
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)) < 0)
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 = 0;
RTSPStream *rtsp_st;
RTSPMessageHeader reply1, *reply = &reply1;
char cmd[MAX_URL_SIZE];
const char *trans_pref;
memset(&reply1, 0, sizeof(reply1));
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. */
if (rt->rtp_port_max - rt->rtp_port_min >= 4) {
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[MAX_URL_SIZE];
/*
* 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 + 1 <= rt->rtp_port_max) {
AVDictionary *opts = map_to_opts(rt);
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;
err = ffurl_open_whitelist(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
&s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
av_dict_free(&opts);
if (!err)
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:
av_strlcpy(transport, trans_pref, sizeof(transport));
av_strlcat(transport,
rt->server_type == RTSP_SERVER_SATIP ? ";" : "/UDP;",
sizeof(transport));
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]->codecpar->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);
} else {
err = AVERROR(EINVAL);
goto fail; // transport would be uninitialized
}
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 (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
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 = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
goto fail;
}
if (rt->server_type == RTSP_SERVER_SATIP && reply->stream_id[0]) {
char proto[128], host[128], path[512], auth[128];
int port;
av_url_split(proto, sizeof(proto), auth, sizeof(auth), host, sizeof(host),
&port, path, sizeof(path), rt->control_uri);
ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL, host,
port, "/stream=%s", reply->stream_id);
}
/* 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[MAX_URL_SIZE], 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;
}
break;
}
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
char url[MAX_URL_SIZE], namebuf[50], optbuf[20] = "";
struct sockaddr_storage addr;
int port, ttl;
AVDictionary *opts = map_to_opts(rt);
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);
err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
&s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
av_dict_free(&opts);
if (err < 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_closep(&rt->rtsp_hd_out);
rt->rtsp_hd_out = NULL;
ffurl_closep(&rt->rtsp_hd);
}
int ff_rtsp_connect(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
char proto[128], host[1024], path[2048];
char tcpname[1024], cmd[MAX_URL_SIZE], auth[128];
const char *lower_rtsp_proto = "tcp";
int port, err, tcp_fd;
RTSPMessageHeader reply1, *reply = &reply1;
int lower_transport_mask = 0;
int default_port = RTSP_DEFAULT_PORT;
int https_tunnel = 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) |
(1 << RTSP_LOWER_TRANSPORT_HTTPS))) {
https_tunnel = !!(rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTPS));
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:
memset(&reply1, 0, sizeof(reply1));
/* extract hostname and port */
av_url_split(proto, sizeof(proto), auth, sizeof(auth),
host, sizeof(host), &port, path, sizeof(path), s->url);
if (!strcmp(proto, "rtsps")) {
lower_rtsp_proto = "tls";
default_port = RTSPS_DEFAULT_PORT;
rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
} else if (!strcmp(proto, "satip")) {
av_strlcpy(proto, "rtsp", sizeof(proto));
rt->server_type = RTSP_SERVER_SATIP;
}
if (*auth) {
av_strlcpy(rt->auth, auth, sizeof(rt->auth));
}
if (port < 0)
port = default_port;
lower_transport_mask = rt->lower_transport_mask;
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->url,
* but with authentication credentials removed and RTSP specific options
* stripped out. */
ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, 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];
AVDictionary *options = NULL;
av_dict_set_int(&options, "timeout", rt->stimeout, 0);
ff_url_join(httpname, sizeof(httpname), https_tunnel ? "https" : "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);
if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
rt->rtsp_hd->protocol_whitelist = av_strdup(s->protocol_whitelist);
if (!rt->rtsp_hd->protocol_whitelist) {
err = AVERROR(ENOMEM);
goto fail;
}
}
/* complete the connection */
if (ffurl_connect(rt->rtsp_hd, &options)) {
av_dict_free(&options);
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);
av_opt_set(rt->rtsp_hd_out->priv_data, "send_expect_100", "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, &options)) {
av_dict_free(&options);
err = AVERROR(EIO);
goto fail;
}
av_dict_free(&options);
} else {
int ret;
/* open the tcp connection */
ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
host, port,
"?timeout=%"PRId64, rt->stimeout);
if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
&s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL)) < 0) {
err = ret;
goto fail;
}
rt->rtsp_hd_out = rt->rtsp_hd;
}
rt->seq = 0;
tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
if (tcp_fd < 0) {
err = tcp_fd;
goto fail;
}
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 */
if (rt->server_type != RTSP_SERVER_SATIP)
rt->server_type = RTSP_SERVER_RTP;
for (;;) {
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 = ff_rtsp_averror(reply->status_code, 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 CONFIG_RTSP_DEMUXER
if (s->iformat) {
if (rt->server_type == RTSP_SERVER_SATIP)
err = init_satip_stream(s);
else
err = ff_rtsp_setup_input_streams(s, reply);
} else
#endif
if (CONFIG_RTSP_MUXER)
err = ff_rtsp_setup_output_streams(s, host);
else
av_assert0(0);
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) {
char *new_url = av_strdup(reply->location);
if (!new_url) {
err = AVERROR(ENOMEM);
goto fail2;
}
ff_format_set_url(s, new_url);
rt->session_id[0] = '\0';
av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
reply->status_code,
s->url);
goto redirect;
}
fail2:
ff_network_close();
return err;
}
#endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
#if CONFIG_RTPDEC
#if CONFIG_RTSP_DEMUXER
static int parse_rtsp_message(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
int ret;
if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
if (rt->state == RTSP_STATE_STREAMING) {
return ff_rtsp_parse_streaming_commands(s);
} else
return AVERROR_EOF;
} 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;
}
return 0;
}
#endif
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;
struct pollfd *p = rt->p;
int *fds = NULL, fdsnum, fdsidx;
int64_t runs = rt->stimeout / POLLING_TIME / 1000;
if (!p) {
p = rt->p = av_malloc_array(2 * rt->nb_rtsp_streams + 1, sizeof(*p));
if (!p)
return AVERROR(ENOMEM);
if (rt->rtsp_hd) {
p[rt->max_p].fd = ffurl_get_file_handle(rt->rtsp_hd);
p[rt->max_p++].events = POLLIN;
}
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);
av_freep(&fds);
return AVERROR_INVALIDDATA;
}
for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
p[rt->max_p].fd = fds[fdsidx];
p[rt->max_p++].events = POLLIN;
}
av_freep(&fds);
}
}
}
for (;;) {
if (ff_check_interrupt(&s->interrupt_callback))
return AVERROR_EXIT;
if (wait_end && wait_end - av_gettime_relative() < 0)
return AVERROR(EAGAIN);
n = poll(p, rt->max_p, POLLING_TIME);
if (n > 0) {
int j = rt->rtsp_hd ? 1 : 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 (rt->rtsp_hd && p[0].revents & POLLIN) {
if ((ret = parse_rtsp_message(s)) < 0) {
return ret;
}
}
#endif
} else if (n == 0 && rt->stimeout > 0 && --runs <= 0) {
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);
}
static int read_packet(AVFormatContext *s,
RTSPStream **rtsp_st, RTSPStream *first_queue_st,
int64_t wait_end)
{
RTSPState *rt = s->priv_data;
int len;
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_relative())
len = AVERROR(EAGAIN);
else
len = avio_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 == 0)
return AVERROR_EOF;
return len;
}
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 (CONFIG_RTPDEC && rt->ts) {
ret = avpriv_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);
}
len = read_packet(s, &rtsp_st, first_queue_st, wait_end);
if (len == AVERROR(EAGAIN) && first_queue_st &&
rt->transport == RTSP_TRANSPORT_RTP) {
av_log(s, AV_LOG_WARNING,
"max delay reached. need to consume packet\n");
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 (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, 1000000, 1LL << 32) - NTP_OFFSET_US;
if (rtpctx->st) {
s->start_time_realtime -=
av_rescale_q (rtpctx->rtcp_ts_offset, rtpctx->st->time_base, AV_TIME_BASE_Q);
}
}
}
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 (CONFIG_RTPDEC && rt->ts) {
ret = avpriv_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(const 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 (sizeof("c=IN IP") - 1 < p_end - p &&
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 i, err;
char url[MAX_URL_SIZE];
AVBPrint bp;
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 */
av_bprint_init(&bp, 0, AV_BPRINT_SIZE_UNLIMITED);
err = avio_read_to_bprint(s->pb, &bp, INT_MAX);
if (err < 0 ) {
ff_network_close();
av_bprint_finalize(&bp, NULL);
return err;
}
err = ff_sdp_parse(s, bp.str);
av_bprint_finalize(&bp, NULL);
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)) {
AVDictionary *opts = map_to_opts(rt);
char buf[MAX_URL_SIZE];
const char *p;
err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
sizeof(rtsp_st->sdp_ip),
namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
if (err) {
av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
err = AVERROR(EIO);
av_dict_free(&opts);
goto fail;
}
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);
p = strchr(s->url, '?');
if (p && av_find_info_tag(buf, sizeof(buf), "localaddr", p))
av_strlcatf(url, sizeof(url), "&localaddr=%s", buf);
else if (rt->localaddr && rt->localaddr[0])
av_strlcatf(url, sizeof(url), "&localaddr=%s", rt->localaddr);
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);
err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
&s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
av_dict_free(&opts);
if (err < 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,
};
const FFInputFormat ff_sdp_demuxer = {
.p.name = "sdp",
.p.long_name = NULL_IF_CONFIG_SMALL("SDP"),
.p.priv_class = &sdp_demuxer_class,
.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,
};
#endif /* CONFIG_SDP_DEMUXER */
#if CONFIG_RTP_DEMUXER
static int rtp_probe(const 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], filters_buf[1000];
int ret, port;
URLContext* in = NULL;
int payload_type;
AVCodecParameters *par = NULL;
struct sockaddr_storage addr;
FFIOContext pb;
socklen_t addrlen = sizeof(addr);
RTSPState *rt = s->priv_data;
const char *p;
AVBPrint sdp;
AVDictionary *opts = NULL;
if (!ff_network_init())
return AVERROR(EIO);
opts = map_to_opts(rt);
ret = ffurl_open_whitelist(&in, s->url, AVIO_FLAG_READ,
&s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
av_dict_free(&opts);
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_closep(&in);
par = avcodec_parameters_alloc();
if (!par) {
ret = AVERROR(ENOMEM);
goto fail;
}
if (ff_rtp_get_codec_info(par, payload_type)) {
av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
"without an SDP file describing it\n",
payload_type);
ret = AVERROR_INVALIDDATA;
goto fail;
}
if (par->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->url);
av_bprint_init(&sdp, 0, AV_BPRINT_SIZE_UNLIMITED);
av_bprintf(&sdp, "v=0\r\nc=IN IP%d %s\r\n",
addr.ss_family == AF_INET ? 4 : 6, host);
p = strchr(s->url, '?');
if (p) {
static const char filters[][2][8] = { { "sources", "incl" },
{ "block", "excl" } };
int i;
char *q;
for (i = 0; i < FF_ARRAY_ELEMS(filters); i++) {
if (av_find_info_tag(filters_buf, sizeof(filters_buf), filters[i][0], p)) {
q = filters_buf;
while ((q = strchr(q, ',')) != NULL)
*q = ' ';
av_bprintf(&sdp, "a=source-filter:%s IN IP%d %s %s\r\n",
filters[i][1],
addr.ss_family == AF_INET ? 4 : 6, host,
filters_buf);
}
}
}
av_bprintf(&sdp, "m=%s %d RTP/AVP %d\r\n",
par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
port, payload_type);
av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp.str);
if (!av_bprint_is_complete(&sdp))
goto fail_nobuf;
avcodec_parameters_free(&par);
ffio_init_read_context(&pb, sdp.str, sdp.len);
s->pb = &pb.pub;
/* if sdp_read_header() fails then following ff_network_close() cancels out */
/* ff_network_init() at the start of this function. Otherwise it cancels out */
/* ff_network_init() inside sdp_read_header() */
ff_network_close();
rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
ret = sdp_read_header(s);
s->pb = NULL;
av_bprint_finalize(&sdp, NULL);
return ret;
fail_nobuf:
ret = AVERROR(ENOMEM);
av_log(s, AV_LOG_ERROR, "rtp_read_header(): not enough buffer space for sdp-headers\n");
av_bprint_finalize(&sdp, NULL);
fail:
avcodec_parameters_free(&par);
ffurl_closep(&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,
};
const FFInputFormat ff_rtp_demuxer = {
.p.name = "rtp",
.p.long_name = NULL_IF_CONFIG_SMALL("RTP input"),
.p.flags = AVFMT_NOFILE,
.p.priv_class = &rtp_demuxer_class,
.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,
};
#endif /* CONFIG_RTP_DEMUXER */
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