0,0 → 1,2344 |
/* |
* 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", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC } |
|
const AVOption ff_rtsp_options[] = { |
{ "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC }, |
FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags), |
{ "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \ |
{ "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \ |
{ "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \ |
{ "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" }, |
{ "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" }, |
RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"), |
{ "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }, |
RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"), |
{ "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC }, |
{ "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC }, |
{ "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC }, |
{ "stimeout", "timeout (in micro seconds) of socket 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 IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" }, |
{ "rtcp_to_source", "Send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" }, |
RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"), |
RTSP_REORDERING_OPTS(), |
{ NULL }, |
}; |
|
static const AVOption rtp_options[] = { |
RTSP_FLAG_OPTS("rtp_flags", "RTP flags"), |
RTSP_REORDERING_OPTS(), |
{ NULL }, |
}; |
|
static void get_word_until_chars(char *buf, int buf_size, |
const char *sep, const char **pp) |
{ |
const char *p; |
char *q; |
|
p = *pp; |
p += strspn(p, SPACE_CHARS); |
q = buf; |
while (!strchr(sep, *p) && *p != '\0') { |
if ((q - buf) < buf_size - 1) |
*q++ = *p; |
p++; |
} |
if (buf_size > 0) |
*q = '\0'; |
*pp = p; |
} |
|
static void get_word_sep(char *buf, int buf_size, const char *sep, |
const char **pp) |
{ |
if (**pp == '/') (*pp)++; |
get_word_until_chars(buf, buf_size, sep, pp); |
} |
|
static void get_word(char *buf, int buf_size, const char **pp) |
{ |
get_word_until_chars(buf, buf_size, SPACE_CHARS, pp); |
} |
|
/** Parse a string p in the form of Range:npt=xx-xx, and determine the start |
* and end time. |
* Used for seeking in the rtp stream. |
*/ |
static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end) |
{ |
char buf[256]; |
|
p += strspn(p, SPACE_CHARS); |
if (!av_stristart(p, "npt=", &p)) |
return; |
|
*start = AV_NOPTS_VALUE; |
*end = AV_NOPTS_VALUE; |
|
get_word_sep(buf, sizeof(buf), "-", &p); |
av_parse_time(start, buf, 1); |
if (*p == '-') { |
p++; |
get_word_sep(buf, sizeof(buf), "-", &p); |
av_parse_time(end, buf, 1); |
} |
} |
|
static int get_sockaddr(const char *buf, struct sockaddr_storage *sock) |
{ |
struct addrinfo hints = { 0 }, *ai = NULL; |
hints.ai_flags = AI_NUMERICHOST; |
if (getaddrinfo(buf, NULL, &hints, &ai)) |
return -1; |
memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen)); |
freeaddrinfo(ai); |
return 0; |
} |
|
#if CONFIG_RTPDEC |
static void init_rtp_handler(RTPDynamicProtocolHandler *handler, |
RTSPStream *rtsp_st, AVCodecContext *codec) |
{ |
if (!handler) |
return; |
if (codec) |
codec->codec_id = handler->codec_id; |
rtsp_st->dynamic_handler = handler; |
if (handler->alloc) { |
rtsp_st->dynamic_protocol_context = handler->alloc(); |
if (!rtsp_st->dynamic_protocol_context) |
rtsp_st->dynamic_handler = NULL; |
} |
} |
|
/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */ |
static int sdp_parse_rtpmap(AVFormatContext *s, |
AVStream *st, RTSPStream *rtsp_st, |
int payload_type, const char *p) |
{ |
AVCodecContext *codec = st->codec; |
char buf[256]; |
int i; |
AVCodec *c; |
const char *c_name; |
|
/* See if we can handle this kind of payload. |
* The space should normally not be there but some Real streams or |
* particular servers ("RealServer Version 6.1.3.970", see issue 1658) |
* have a trailing space. */ |
get_word_sep(buf, sizeof(buf), "/ ", &p); |
if (payload_type < RTP_PT_PRIVATE) { |
/* We are in a standard case |
* (from http://www.iana.org/assignments/rtp-parameters). */ |
codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type); |
} |
|
if (codec->codec_id == AV_CODEC_ID_NONE) { |
RTPDynamicProtocolHandler *handler = |
ff_rtp_handler_find_by_name(buf, codec->codec_type); |
init_rtp_handler(handler, rtsp_st, codec); |
/* If no dynamic handler was found, check with the list of standard |
* allocated types, if such a stream for some reason happens to |
* use a private payload type. This isn't handled in rtpdec.c, since |
* the format name from the rtpmap line never is passed into rtpdec. */ |
if (!rtsp_st->dynamic_handler) |
codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type); |
} |
|
c = avcodec_find_decoder(codec->codec_id); |
if (c && c->name) |
c_name = c->name; |
else |
c_name = "(null)"; |
|
get_word_sep(buf, sizeof(buf), "/", &p); |
i = atoi(buf); |
switch (codec->codec_type) { |
case AVMEDIA_TYPE_AUDIO: |
av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name); |
codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE; |
codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS; |
if (i > 0) { |
codec->sample_rate = i; |
avpriv_set_pts_info(st, 32, 1, codec->sample_rate); |
get_word_sep(buf, sizeof(buf), "/", &p); |
i = atoi(buf); |
if (i > 0) |
codec->channels = i; |
} |
av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n", |
codec->sample_rate); |
av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n", |
codec->channels); |
break; |
case AVMEDIA_TYPE_VIDEO: |
av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name); |
if (i > 0) |
avpriv_set_pts_info(st, 32, 1, i); |
break; |
default: |
break; |
} |
if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) |
rtsp_st->dynamic_handler->init(s, st->index, |
rtsp_st->dynamic_protocol_context); |
return 0; |
} |
|
/* parse the attribute line from the fmtp a line of an sdp response. This |
* is broken out as a function because it is used in rtp_h264.c, which is |
* forthcoming. */ |
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, |
char *value, int value_size) |
{ |
*p += strspn(*p, SPACE_CHARS); |
if (**p) { |
get_word_sep(attr, attr_size, "=", p); |
if (**p == '=') |
(*p)++; |
get_word_sep(value, value_size, ";", p); |
if (**p == ';') |
(*p)++; |
return 1; |
} |
return 0; |
} |
|
typedef struct SDPParseState { |
/* SDP only */ |
struct sockaddr_storage default_ip; |
int default_ttl; |
int skip_media; ///< set if an unknown m= line occurs |
int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */ |
struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */ |
int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */ |
struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */ |
} SDPParseState; |
|
static void copy_default_source_addrs(struct RTSPSource **addrs, int count, |
struct RTSPSource ***dest, int *dest_count) |
{ |
RTSPSource *rtsp_src, *rtsp_src2; |
int i; |
for (i = 0; i < count; i++) { |
rtsp_src = addrs[i]; |
rtsp_src2 = av_malloc(sizeof(*rtsp_src2)); |
if (!rtsp_src2) |
continue; |
memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src)); |
dynarray_add(dest, dest_count, rtsp_src2); |
} |
} |
|
static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1, |
int letter, const char *buf) |
{ |
RTSPState *rt = s->priv_data; |
char buf1[64], st_type[64]; |
const char *p; |
enum AVMediaType codec_type; |
int payload_type, i; |
AVStream *st; |
RTSPStream *rtsp_st; |
RTSPSource *rtsp_src; |
struct sockaddr_storage sdp_ip; |
int ttl; |
|
av_dlog(s, "sdp: %c='%s'\n", letter, buf); |
|
p = buf; |
if (s1->skip_media && letter != 'm') |
return; |
switch (letter) { |
case 'c': |
get_word(buf1, sizeof(buf1), &p); |
if (strcmp(buf1, "IN") != 0) |
return; |
get_word(buf1, sizeof(buf1), &p); |
if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6")) |
return; |
get_word_sep(buf1, sizeof(buf1), "/", &p); |
if (get_sockaddr(buf1, &sdp_ip)) |
return; |
ttl = 16; |
if (*p == '/') { |
p++; |
get_word_sep(buf1, sizeof(buf1), "/", &p); |
ttl = atoi(buf1); |
} |
if (s->nb_streams == 0) { |
s1->default_ip = sdp_ip; |
s1->default_ttl = ttl; |
} else { |
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; |
rtsp_st->sdp_ip = sdp_ip; |
rtsp_st->sdp_ttl = ttl; |
} |
break; |
case 's': |
av_dict_set(&s->metadata, "title", p, 0); |
break; |
case 'i': |
if (s->nb_streams == 0) { |
av_dict_set(&s->metadata, "comment", p, 0); |
break; |
} |
break; |
case 'm': |
/* new stream */ |
s1->skip_media = 0; |
codec_type = AVMEDIA_TYPE_UNKNOWN; |
get_word(st_type, sizeof(st_type), &p); |
if (!strcmp(st_type, "audio")) { |
codec_type = AVMEDIA_TYPE_AUDIO; |
} else if (!strcmp(st_type, "video")) { |
codec_type = AVMEDIA_TYPE_VIDEO; |
} else if (!strcmp(st_type, "application")) { |
codec_type = AVMEDIA_TYPE_DATA; |
} |
if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) { |
s1->skip_media = 1; |
return; |
} |
rtsp_st = av_mallocz(sizeof(RTSPStream)); |
if (!rtsp_st) |
return; |
rtsp_st->stream_index = -1; |
dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st); |
|
rtsp_st->sdp_ip = s1->default_ip; |
rtsp_st->sdp_ttl = s1->default_ttl; |
|
copy_default_source_addrs(s1->default_include_source_addrs, |
s1->nb_default_include_source_addrs, |
&rtsp_st->include_source_addrs, |
&rtsp_st->nb_include_source_addrs); |
copy_default_source_addrs(s1->default_exclude_source_addrs, |
s1->nb_default_exclude_source_addrs, |
&rtsp_st->exclude_source_addrs, |
&rtsp_st->nb_exclude_source_addrs); |
|
get_word(buf1, sizeof(buf1), &p); /* port */ |
rtsp_st->sdp_port = atoi(buf1); |
|
get_word(buf1, sizeof(buf1), &p); /* protocol */ |
if (!strcmp(buf1, "udp")) |
rt->transport = RTSP_TRANSPORT_RAW; |
else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF")) |
rtsp_st->feedback = 1; |
|
/* XXX: handle list of formats */ |
get_word(buf1, sizeof(buf1), &p); /* format list */ |
rtsp_st->sdp_payload_type = atoi(buf1); |
|
if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) { |
/* no corresponding stream */ |
if (rt->transport == RTSP_TRANSPORT_RAW) { |
if (!rt->ts && CONFIG_RTPDEC) |
rt->ts = ff_mpegts_parse_open(s); |
} else { |
RTPDynamicProtocolHandler *handler; |
handler = ff_rtp_handler_find_by_id( |
rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA); |
init_rtp_handler(handler, rtsp_st, NULL); |
if (handler && handler->init) |
handler->init(s, -1, rtsp_st->dynamic_protocol_context); |
} |
} else if (rt->server_type == RTSP_SERVER_WMS && |
codec_type == AVMEDIA_TYPE_DATA) { |
/* RTX stream, a stream that carries all the other actual |
* audio/video streams. Don't expose this to the callers. */ |
} else { |
st = avformat_new_stream(s, NULL); |
if (!st) |
return; |
st->id = rt->nb_rtsp_streams - 1; |
rtsp_st->stream_index = st->index; |
st->codec->codec_type = codec_type; |
if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) { |
RTPDynamicProtocolHandler *handler; |
/* if standard payload type, we can find the codec right now */ |
ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type); |
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO && |
st->codec->sample_rate > 0) |
avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate); |
/* Even static payload types may need a custom depacketizer */ |
handler = ff_rtp_handler_find_by_id( |
rtsp_st->sdp_payload_type, st->codec->codec_type); |
init_rtp_handler(handler, rtsp_st, st->codec); |
if (handler && handler->init) |
handler->init(s, st->index, |
rtsp_st->dynamic_protocol_context); |
} |
} |
/* put a default control url */ |
av_strlcpy(rtsp_st->control_url, rt->control_uri, |
sizeof(rtsp_st->control_url)); |
break; |
case 'a': |
if (av_strstart(p, "control:", &p)) { |
if (s->nb_streams == 0) { |
if (!strncmp(p, "rtsp://", 7)) |
av_strlcpy(rt->control_uri, p, |
sizeof(rt->control_uri)); |
} else { |
char proto[32]; |
/* get the control url */ |
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; |
|
/* XXX: may need to add full url resolution */ |
av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0, |
NULL, NULL, 0, p); |
if (proto[0] == '\0') { |
/* relative control URL */ |
if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/') |
av_strlcat(rtsp_st->control_url, "/", |
sizeof(rtsp_st->control_url)); |
av_strlcat(rtsp_st->control_url, p, |
sizeof(rtsp_st->control_url)); |
} else |
av_strlcpy(rtsp_st->control_url, p, |
sizeof(rtsp_st->control_url)); |
} |
} else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) { |
/* NOTE: rtpmap is only supported AFTER the 'm=' tag */ |
get_word(buf1, sizeof(buf1), &p); |
payload_type = atoi(buf1); |
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; |
if (rtsp_st->stream_index >= 0) { |
st = s->streams[rtsp_st->stream_index]; |
sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p); |
} |
} else if (av_strstart(p, "fmtp:", &p) || |
av_strstart(p, "framesize:", &p)) { |
/* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */ |
// let dynamic protocol handlers have a stab at the line. |
get_word(buf1, sizeof(buf1), &p); |
payload_type = atoi(buf1); |
for (i = 0; i < rt->nb_rtsp_streams; i++) { |
rtsp_st = rt->rtsp_streams[i]; |
if (rtsp_st->sdp_payload_type == payload_type && |
rtsp_st->dynamic_handler && |
rtsp_st->dynamic_handler->parse_sdp_a_line) |
rtsp_st->dynamic_handler->parse_sdp_a_line(s, i, |
rtsp_st->dynamic_protocol_context, buf); |
} |
} else if (av_strstart(p, "range:", &p)) { |
int64_t start, end; |
|
// this is so that seeking on a streamed file can work. |
rtsp_parse_range_npt(p, &start, &end); |
s->start_time = start; |
/* AV_NOPTS_VALUE means live broadcast (and can't seek) */ |
s->duration = (end == AV_NOPTS_VALUE) ? |
AV_NOPTS_VALUE : end - start; |
} else if (av_strstart(p, "IsRealDataType:integer;",&p)) { |
if (atoi(p) == 1) |
rt->transport = RTSP_TRANSPORT_RDT; |
} else if (av_strstart(p, "SampleRate:integer;", &p) && |
s->nb_streams > 0) { |
st = s->streams[s->nb_streams - 1]; |
st->codec->sample_rate = atoi(p); |
} else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) { |
// RFC 4568 |
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; |
get_word(buf1, sizeof(buf1), &p); // ignore tag |
get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p); |
p += strspn(p, SPACE_CHARS); |
if (av_strstart(p, "inline:", &p)) |
get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p); |
} else if (av_strstart(p, "source-filter:", &p)) { |
int exclude = 0; |
get_word(buf1, sizeof(buf1), &p); |
if (strcmp(buf1, "incl") && strcmp(buf1, "excl")) |
return; |
exclude = !strcmp(buf1, "excl"); |
|
get_word(buf1, sizeof(buf1), &p); |
if (strcmp(buf1, "IN") != 0) |
return; |
get_word(buf1, sizeof(buf1), &p); |
if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*")) |
return; |
// not checking that the destination address actually matches or is wildcard |
get_word(buf1, sizeof(buf1), &p); |
|
while (*p != '\0') { |
rtsp_src = av_mallocz(sizeof(*rtsp_src)); |
if (!rtsp_src) |
return; |
get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p); |
if (exclude) { |
if (s->nb_streams == 0) { |
dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src); |
} else { |
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; |
dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src); |
} |
} else { |
if (s->nb_streams == 0) { |
dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src); |
} else { |
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; |
dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src); |
} |
} |
} |
} else { |
if (rt->server_type == RTSP_SERVER_WMS) |
ff_wms_parse_sdp_a_line(s, p); |
if (s->nb_streams > 0) { |
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; |
|
if (rt->server_type == RTSP_SERVER_REAL) |
ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p); |
|
if (rtsp_st->dynamic_handler && |
rtsp_st->dynamic_handler->parse_sdp_a_line) |
rtsp_st->dynamic_handler->parse_sdp_a_line(s, |
rtsp_st->stream_index, |
rtsp_st->dynamic_protocol_context, buf); |
} |
} |
break; |
} |
} |
|
int ff_sdp_parse(AVFormatContext *s, const char *content) |
{ |
RTSPState *rt = s->priv_data; |
const char *p; |
int letter, i; |
/* Some SDP lines, particularly for Realmedia or ASF RTSP streams, |
* contain long SDP lines containing complete ASF Headers (several |
* kB) or arrays of MDPR (RM stream descriptor) headers plus |
* "rulebooks" describing their properties. Therefore, the SDP line |
* buffer is large. |
* |
* The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line |
* in rtpdec_xiph.c. */ |
char buf[16384], *q; |
SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state; |
|
p = content; |
for (;;) { |
p += strspn(p, SPACE_CHARS); |
letter = *p; |
if (letter == '\0') |
break; |
p++; |
if (*p != '=') |
goto next_line; |
p++; |
/* get the content */ |
q = buf; |
while (*p != '\n' && *p != '\r' && *p != '\0') { |
if ((q - buf) < sizeof(buf) - 1) |
*q++ = *p; |
p++; |
} |
*q = '\0'; |
sdp_parse_line(s, s1, letter, buf); |
next_line: |
while (*p != '\n' && *p != '\0') |
p++; |
if (*p == '\n') |
p++; |
} |
|
for (i = 0; i < s1->nb_default_include_source_addrs; i++) |
av_free(s1->default_include_source_addrs[i]); |
av_freep(&s1->default_include_source_addrs); |
for (i = 0; i < s1->nb_default_exclude_source_addrs; i++) |
av_free(s1->default_exclude_source_addrs[i]); |
av_freep(&s1->default_exclude_source_addrs); |
|
rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1)); |
if (!rt->p) return AVERROR(ENOMEM); |
return 0; |
} |
#endif /* CONFIG_RTPDEC */ |
|
void ff_rtsp_undo_setup(AVFormatContext *s) |
{ |
RTSPState *rt = s->priv_data; |
int i; |
|
for (i = 0; i < rt->nb_rtsp_streams; i++) { |
RTSPStream *rtsp_st = rt->rtsp_streams[i]; |
if (!rtsp_st) |
continue; |
if (rtsp_st->transport_priv) { |
if (s->oformat) { |
AVFormatContext *rtpctx = rtsp_st->transport_priv; |
av_write_trailer(rtpctx); |
if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) { |
uint8_t *ptr; |
avio_close_dyn_buf(rtpctx->pb, &ptr); |
av_free(ptr); |
} else { |
avio_close(rtpctx->pb); |
} |
avformat_free_context(rtpctx); |
} else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) |
ff_rdt_parse_close(rtsp_st->transport_priv); |
else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) |
ff_rtp_parse_close(rtsp_st->transport_priv); |
} |
rtsp_st->transport_priv = NULL; |
if (rtsp_st->rtp_handle) |
ffurl_close(rtsp_st->rtp_handle); |
rtsp_st->rtp_handle = NULL; |
} |
} |
|
/* close and free RTSP streams */ |
void ff_rtsp_close_streams(AVFormatContext *s) |
{ |
RTSPState *rt = s->priv_data; |
int i, j; |
RTSPStream *rtsp_st; |
|
ff_rtsp_undo_setup(s); |
for (i = 0; i < rt->nb_rtsp_streams; i++) { |
rtsp_st = rt->rtsp_streams[i]; |
if (rtsp_st) { |
if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) |
rtsp_st->dynamic_handler->free( |
rtsp_st->dynamic_protocol_context); |
for (j = 0; j < rtsp_st->nb_include_source_addrs; j++) |
av_free(rtsp_st->include_source_addrs[j]); |
av_freep(&rtsp_st->include_source_addrs); |
for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++) |
av_free(rtsp_st->exclude_source_addrs[j]); |
av_freep(&rtsp_st->exclude_source_addrs); |
|
av_free(rtsp_st); |
} |
} |
av_free(rt->rtsp_streams); |
if (rt->asf_ctx) { |
avformat_close_input(&rt->asf_ctx); |
} |
if (rt->ts && CONFIG_RTPDEC) |
ff_mpegts_parse_close(rt->ts); |
av_free(rt->p); |
av_free(rt->recvbuf); |
} |
|
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st) |
{ |
RTSPState *rt = s->priv_data; |
AVStream *st = NULL; |
int reordering_queue_size = rt->reordering_queue_size; |
if (reordering_queue_size < 0) { |
if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay) |
reordering_queue_size = 0; |
else |
reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE; |
} |
|
/* open the RTP context */ |
if (rtsp_st->stream_index >= 0) |
st = s->streams[rtsp_st->stream_index]; |
if (!st) |
s->ctx_flags |= AVFMTCTX_NOHEADER; |
|
if (s->oformat && CONFIG_RTSP_MUXER) { |
int ret = ff_rtp_chain_mux_open((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; |
} 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); |
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)]; |
|
err = ff_rtsp_make_setup_request(s, host, port, lower_transport, |
rt->server_type == RTSP_SERVER_REAL ? |
real_challenge : NULL); |
if (err < 0) |
goto fail; |
lower_transport_mask &= ~(1 << lower_transport); |
if (lower_transport_mask == 0 && err == 1) { |
err = AVERROR(EPROTONOSUPPORT); |
goto fail; |
} |
} while (err); |
|
rt->lower_transport_mask = lower_transport_mask; |
av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge)); |
rt->state = RTSP_STATE_IDLE; |
rt->seek_timestamp = 0; /* default is to start stream at position zero */ |
return 0; |
fail: |
ff_rtsp_close_streams(s); |
ff_rtsp_close_connections(s); |
if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) { |
av_strlcpy(s->filename, reply->location, sizeof(s->filename)); |
av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n", |
reply->status_code, |
s->filename); |
goto redirect; |
} |
ff_network_close(); |
return err; |
} |
#endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */ |
|
#if CONFIG_RTPDEC |
static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, |
uint8_t *buf, int buf_size, int64_t wait_end) |
{ |
RTSPState *rt = s->priv_data; |
RTSPStream *rtsp_st; |
int n, i, ret, tcp_fd, timeout_cnt = 0; |
int max_p = 0; |
struct pollfd *p = rt->p; |
int *fds = NULL, fdsnum, fdsidx; |
|
for (;;) { |
if (ff_check_interrupt(&s->interrupt_callback)) |
return AVERROR_EXIT; |
if (wait_end && wait_end - av_gettime() < 0) |
return AVERROR(EAGAIN); |
max_p = 0; |
if (rt->rtsp_hd) { |
tcp_fd = ffurl_get_file_handle(rt->rtsp_hd); |
p[max_p].fd = tcp_fd; |
p[max_p++].events = POLLIN; |
} else { |
tcp_fd = -1; |
} |
for (i = 0; i < rt->nb_rtsp_streams; i++) { |
rtsp_st = rt->rtsp_streams[i]; |
if (rtsp_st->rtp_handle) { |
if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle, |
&fds, &fdsnum)) { |
av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n"); |
return ret; |
} |
if (fdsnum != 2) { |
av_log(s, AV_LOG_ERROR, |
"Number of fds %d not supported\n", fdsnum); |
return AVERROR_INVALIDDATA; |
} |
for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) { |
p[max_p].fd = fds[fdsidx]; |
p[max_p++].events = POLLIN; |
} |
av_free(fds); |
} |
} |
n = poll(p, max_p, POLL_TIMEOUT_MS); |
if (n > 0) { |
int j = 1 - (tcp_fd == -1); |
timeout_cnt = 0; |
for (i = 0; i < rt->nb_rtsp_streams; i++) { |
rtsp_st = rt->rtsp_streams[i]; |
if (rtsp_st->rtp_handle) { |
if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) { |
ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size); |
if (ret > 0) { |
*prtsp_st = rtsp_st; |
return ret; |
} |
} |
j+=2; |
} |
} |
#if CONFIG_RTSP_DEMUXER |
if (tcp_fd != -1 && p[0].revents & POLLIN) { |
if (rt->rtsp_flags & RTSP_FLAG_LISTEN) { |
if (rt->state == RTSP_STATE_STREAMING) { |
if (!ff_rtsp_parse_streaming_commands(s)) |
return AVERROR_EOF; |
else |
av_log(s, AV_LOG_WARNING, |
"Unable to answer to TEARDOWN\n"); |
} else |
return 0; |
} else { |
RTSPMessageHeader reply; |
ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL); |
if (ret < 0) |
return ret; |
/* XXX: parse message */ |
if (rt->state != RTSP_STATE_STREAMING) |
return 0; |
} |
} |
#endif |
} else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) { |
return AVERROR(ETIMEDOUT); |
} else if (n < 0 && errno != EINTR) |
return AVERROR(errno); |
} |
} |
|
static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st, |
const uint8_t *buf, int len) |
{ |
RTSPState *rt = s->priv_data; |
int i; |
if (len < 0) |
return len; |
if (rt->nb_rtsp_streams == 1) { |
*rtsp_st = rt->rtsp_streams[0]; |
return len; |
} |
if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) { |
if (RTP_PT_IS_RTCP(rt->recvbuf[1])) { |
int no_ssrc = 0; |
for (i = 0; i < rt->nb_rtsp_streams; i++) { |
RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv; |
if (!rtpctx) |
continue; |
if (rtpctx->ssrc == AV_RB32(&buf[4])) { |
*rtsp_st = rt->rtsp_streams[i]; |
return len; |
} |
if (!rtpctx->ssrc) |
no_ssrc = 1; |
} |
if (no_ssrc) { |
av_log(s, AV_LOG_WARNING, |
"Unable to pick stream for packet - SSRC not known for " |
"all streams\n"); |
return AVERROR(EAGAIN); |
} |
} else { |
for (i = 0; i < rt->nb_rtsp_streams; i++) { |
if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) { |
*rtsp_st = rt->rtsp_streams[i]; |
return len; |
} |
} |
} |
} |
av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n"); |
return AVERROR(EAGAIN); |
} |
|
int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt) |
{ |
RTSPState *rt = s->priv_data; |
int ret, len; |
RTSPStream *rtsp_st, *first_queue_st = NULL; |
int64_t wait_end = 0; |
|
if (rt->nb_byes == rt->nb_rtsp_streams) |
return AVERROR_EOF; |
|
/* get next frames from the same RTP packet */ |
if (rt->cur_transport_priv) { |
if (rt->transport == RTSP_TRANSPORT_RDT) { |
ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0); |
} else if (rt->transport == RTSP_TRANSPORT_RTP) { |
ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0); |
} else if (rt->ts && CONFIG_RTPDEC) { |
ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos); |
if (ret >= 0) { |
rt->recvbuf_pos += ret; |
ret = rt->recvbuf_pos < rt->recvbuf_len; |
} |
} else |
ret = -1; |
if (ret == 0) { |
rt->cur_transport_priv = NULL; |
return 0; |
} else if (ret == 1) { |
return 0; |
} else |
rt->cur_transport_priv = NULL; |
} |
|
redo: |
if (rt->transport == RTSP_TRANSPORT_RTP) { |
int i; |
int64_t first_queue_time = 0; |
for (i = 0; i < rt->nb_rtsp_streams; i++) { |
RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv; |
int64_t queue_time; |
if (!rtpctx) |
continue; |
queue_time = ff_rtp_queued_packet_time(rtpctx); |
if (queue_time && (queue_time - first_queue_time < 0 || |
!first_queue_time)) { |
first_queue_time = queue_time; |
first_queue_st = rt->rtsp_streams[i]; |
} |
} |
if (first_queue_time) { |
wait_end = first_queue_time + s->max_delay; |
} else { |
wait_end = 0; |
first_queue_st = NULL; |
} |
} |
|
/* read next RTP packet */ |
if (!rt->recvbuf) { |
rt->recvbuf = av_malloc(RECVBUF_SIZE); |
if (!rt->recvbuf) |
return AVERROR(ENOMEM); |
} |
|
switch(rt->lower_transport) { |
default: |
#if CONFIG_RTSP_DEMUXER |
case RTSP_LOWER_TRANSPORT_TCP: |
len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE); |
break; |
#endif |
case RTSP_LOWER_TRANSPORT_UDP: |
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: |
len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end); |
if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP) |
ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len); |
break; |
case RTSP_LOWER_TRANSPORT_CUSTOM: |
if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP && |
wait_end && wait_end < av_gettime()) |
len = AVERROR(EAGAIN); |
else |
len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE); |
len = pick_stream(s, &rtsp_st, rt->recvbuf, len); |
if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP) |
ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len); |
break; |
} |
if (len == AVERROR(EAGAIN) && first_queue_st && |
rt->transport == RTSP_TRANSPORT_RTP) { |
rtsp_st = first_queue_st; |
ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0); |
goto end; |
} |
if (len < 0) |
return len; |
if (len == 0) |
return AVERROR_EOF; |
if (rt->transport == RTSP_TRANSPORT_RDT) { |
ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len); |
} else if (rt->transport == RTSP_TRANSPORT_RTP) { |
ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len); |
if (rtsp_st->feedback) { |
AVIOContext *pb = NULL; |
if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM) |
pb = s->pb; |
ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb); |
} |
if (ret < 0) { |
/* Either bad packet, or a RTCP packet. Check if the |
* first_rtcp_ntp_time field was initialized. */ |
RTPDemuxContext *rtpctx = rtsp_st->transport_priv; |
if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) { |
/* first_rtcp_ntp_time has been initialized for this stream, |
* copy the same value to all other uninitialized streams, |
* in order to map their timestamp origin to the same ntp time |
* as this one. */ |
int i; |
AVStream *st = NULL; |
if (rtsp_st->stream_index >= 0) |
st = s->streams[rtsp_st->stream_index]; |
for (i = 0; i < rt->nb_rtsp_streams; i++) { |
RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv; |
AVStream *st2 = NULL; |
if (rt->rtsp_streams[i]->stream_index >= 0) |
st2 = s->streams[rt->rtsp_streams[i]->stream_index]; |
if (rtpctx2 && st && st2 && |
rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) { |
rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time; |
rtpctx2->rtcp_ts_offset = av_rescale_q( |
rtpctx->rtcp_ts_offset, st->time_base, |
st2->time_base); |
} |
} |
} |
if (ret == -RTCP_BYE) { |
rt->nb_byes++; |
|
av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n", |
rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams); |
|
if (rt->nb_byes == rt->nb_rtsp_streams) |
return AVERROR_EOF; |
} |
} |
} else if (rt->ts && CONFIG_RTPDEC) { |
ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len); |
if (ret >= 0) { |
if (ret < len) { |
rt->recvbuf_len = len; |
rt->recvbuf_pos = ret; |
rt->cur_transport_priv = rt->ts; |
return 1; |
} else { |
ret = 0; |
} |
} |
} else { |
return AVERROR_INVALIDDATA; |
} |
end: |
if (ret < 0) |
goto redo; |
if (ret == 1) |
/* more packets may follow, so we save the RTP context */ |
rt->cur_transport_priv = rtsp_st->transport_priv; |
|
return ret; |
} |
#endif /* CONFIG_RTPDEC */ |
|
#if CONFIG_SDP_DEMUXER |
static int sdp_probe(AVProbeData *p1) |
{ |
const char *p = p1->buf, *p_end = p1->buf + p1->buf_size; |
|
/* we look for a line beginning "c=IN IP" */ |
while (p < p_end && *p != '\0') { |
if (p + sizeof("c=IN IP") - 1 < p_end && |
av_strstart(p, "c=IN IP", NULL)) |
return AVPROBE_SCORE_EXTENSION; |
|
while (p < p_end - 1 && *p != '\n') p++; |
if (++p >= p_end) |
break; |
if (*p == '\r') |
p++; |
} |
return 0; |
} |
|
static void append_source_addrs(char *buf, int size, const char *name, |
int count, struct RTSPSource **addrs) |
{ |
int i; |
if (!count) |
return; |
av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr); |
for (i = 1; i < count; i++) |
av_strlcatf(buf, size, ",%s", addrs[i]->addr); |
} |
|
static int sdp_read_header(AVFormatContext *s) |
{ |
RTSPState *rt = s->priv_data; |
RTSPStream *rtsp_st; |
int size, i, err; |
char *content; |
char url[1024]; |
|
if (!ff_network_init()) |
return AVERROR(EIO); |
|
if (s->max_delay < 0) /* Not set by the caller */ |
s->max_delay = DEFAULT_REORDERING_DELAY; |
if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO) |
rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM; |
|
/* read the whole sdp file */ |
/* XXX: better loading */ |
content = av_malloc(SDP_MAX_SIZE); |
size = avio_read(s->pb, content, SDP_MAX_SIZE - 1); |
if (size <= 0) { |
av_free(content); |
return AVERROR_INVALIDDATA; |
} |
content[size] ='\0'; |
|
err = ff_sdp_parse(s, content); |
av_free(content); |
if (err) goto fail; |
|
/* open each RTP stream */ |
for (i = 0; i < rt->nb_rtsp_streams; i++) { |
char namebuf[50]; |
rtsp_st = rt->rtsp_streams[i]; |
|
if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) { |
getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip), |
namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST); |
ff_url_join(url, sizeof(url), "rtp", NULL, |
namebuf, rtsp_st->sdp_port, |
"?localport=%d&ttl=%d&connect=%d&write_to_source=%d", |
rtsp_st->sdp_port, rtsp_st->sdp_ttl, |
rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0, |
rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0); |
|
append_source_addrs(url, sizeof(url), "sources", |
rtsp_st->nb_include_source_addrs, |
rtsp_st->include_source_addrs); |
append_source_addrs(url, sizeof(url), "block", |
rtsp_st->nb_exclude_source_addrs, |
rtsp_st->exclude_source_addrs); |
if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE, |
&s->interrupt_callback, NULL) < 0) { |
err = AVERROR_INVALIDDATA; |
goto fail; |
} |
} |
if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st))) |
goto fail; |
} |
return 0; |
fail: |
ff_rtsp_close_streams(s); |
ff_network_close(); |
return err; |
} |
|
static int sdp_read_close(AVFormatContext *s) |
{ |
ff_rtsp_close_streams(s); |
ff_network_close(); |
return 0; |
} |
|
static const AVClass sdp_demuxer_class = { |
.class_name = "SDP demuxer", |
.item_name = av_default_item_name, |
.option = sdp_options, |
.version = LIBAVUTIL_VERSION_INT, |
}; |
|
AVInputFormat ff_sdp_demuxer = { |
.name = "sdp", |
.long_name = NULL_IF_CONFIG_SMALL("SDP"), |
.priv_data_size = sizeof(RTSPState), |
.read_probe = sdp_probe, |
.read_header = sdp_read_header, |
.read_packet = ff_rtsp_fetch_packet, |
.read_close = sdp_read_close, |
.priv_class = &sdp_demuxer_class, |
}; |
#endif /* CONFIG_SDP_DEMUXER */ |
|
#if CONFIG_RTP_DEMUXER |
static int rtp_probe(AVProbeData *p) |
{ |
if (av_strstart(p->filename, "rtp:", NULL)) |
return AVPROBE_SCORE_MAX; |
return 0; |
} |
|
static int rtp_read_header(AVFormatContext *s) |
{ |
uint8_t recvbuf[RTP_MAX_PACKET_LENGTH]; |
char host[500], sdp[500]; |
int ret, port; |
URLContext* in = NULL; |
int payload_type; |
AVCodecContext codec = { 0 }; |
struct sockaddr_storage addr; |
AVIOContext pb; |
socklen_t addrlen = sizeof(addr); |
RTSPState *rt = s->priv_data; |
|
if (!ff_network_init()) |
return AVERROR(EIO); |
|
ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ, |
&s->interrupt_callback, NULL); |
if (ret) |
goto fail; |
|
while (1) { |
ret = ffurl_read(in, recvbuf, sizeof(recvbuf)); |
if (ret == AVERROR(EAGAIN)) |
continue; |
if (ret < 0) |
goto fail; |
if (ret < 12) { |
av_log(s, AV_LOG_WARNING, "Received too short packet\n"); |
continue; |
} |
|
if ((recvbuf[0] & 0xc0) != 0x80) { |
av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet " |
"received\n"); |
continue; |
} |
|
if (RTP_PT_IS_RTCP(recvbuf[1])) |
continue; |
|
payload_type = recvbuf[1] & 0x7f; |
break; |
} |
getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen); |
ffurl_close(in); |
in = NULL; |
|
if (ff_rtp_get_codec_info(&codec, payload_type)) { |
av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d " |
"without an SDP file describing it\n", |
payload_type); |
goto fail; |
} |
if (codec.codec_type != AVMEDIA_TYPE_DATA) { |
av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received " |
"properly you need an SDP file " |
"describing it\n"); |
} |
|
av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port, |
NULL, 0, s->filename); |
|
snprintf(sdp, sizeof(sdp), |
"v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n", |
addr.ss_family == AF_INET ? 4 : 6, host, |
codec.codec_type == AVMEDIA_TYPE_DATA ? "application" : |
codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio", |
port, payload_type); |
av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp); |
|
ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL); |
s->pb = &pb; |
|
/* sdp_read_header initializes this again */ |
ff_network_close(); |
|
rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1; |
|
ret = sdp_read_header(s); |
s->pb = NULL; |
return ret; |
|
fail: |
if (in) |
ffurl_close(in); |
ff_network_close(); |
return ret; |
} |
|
static const AVClass rtp_demuxer_class = { |
.class_name = "RTP demuxer", |
.item_name = av_default_item_name, |
.option = rtp_options, |
.version = LIBAVUTIL_VERSION_INT, |
}; |
|
AVInputFormat ff_rtp_demuxer = { |
.name = "rtp", |
.long_name = NULL_IF_CONFIG_SMALL("RTP input"), |
.priv_data_size = sizeof(RTSPState), |
.read_probe = rtp_probe, |
.read_header = rtp_read_header, |
.read_packet = ff_rtsp_fetch_packet, |
.read_close = sdp_read_close, |
.flags = AVFMT_NOFILE, |
.priv_class = &rtp_demuxer_class, |
}; |
#endif /* CONFIG_RTP_DEMUXER */ |