55 #define POLL_TIMEOUT_MS 100
56 #define READ_PACKET_TIMEOUT_S 10
57 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
58 #define SDP_MAX_SIZE 16384
59 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define DEFAULT_REORDERING_DELAY 100000
62 #define OFFSET(x) offsetof(RTSPState, x)
63 #define DEC AV_OPT_FLAG_DECODING_PARAM
64 #define ENC AV_OPT_FLAG_ENCODING_PARAM
66 #define RTSP_FLAG_OPTS(name, longname) \
67 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
68 { "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" }, \
69 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
71 #define RTSP_MEDIATYPE_OPTS(name, longname) \
72 { 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" }, \
73 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
75 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
77 #define RTSP_REORDERING_OPTS() \
78 { "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 }
81 {
"initial_pause",
"Don't start playing the stream immediately",
OFFSET(initial_pause),
AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1,
DEC },
83 {
"rtsp_transport",
"RTSP transport protocols",
OFFSET(lower_transport_mask),
AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX,
DEC|
ENC,
"rtsp_transport" }, \
92 {
"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 },
112 const char *sep,
const char **pp)
120 while (!strchr(sep, *p) && *p !=
'\0') {
121 if ((q - buf) < buf_size - 1)
133 if (**pp ==
'/') (*pp)++;
137 static void get_word(
char *buf,
int buf_size,
const char **pp)
172 memcpy(sock, ai->ai_addr,
FFMIN(
sizeof(*sock), ai->ai_addrlen));
185 if (handler->
alloc) {
195 int payload_type,
const char *p)
219 init_rtp_handler(handler, rtsp_st, codec);
276 char *value,
int value_size)
291 typedef struct SDPParseState {
299 int letter,
const char *buf)
302 char buf1[64], st_type[64];
311 av_dlog(s,
"sdp: %c='%s'\n", letter, buf);
314 if (s1->skip_media && letter !=
'm')
319 if (strcmp(buf1,
"IN") != 0)
322 if (strcmp(buf1,
"IP4") && strcmp(buf1,
"IP6"))
334 s1->default_ip = sdp_ip;
335 s1->default_ttl = ttl;
355 get_word(st_type,
sizeof(st_type), &p);
356 if (!strcmp(st_type,
"audio")) {
358 }
else if (!strcmp(st_type,
"video")) {
360 }
else if (!strcmp(st_type,
"application")) {
373 rtsp_st->
sdp_ip = s1->default_ip;
374 rtsp_st->
sdp_ttl = s1->default_ttl;
380 if (!strcmp(buf1,
"udp"))
412 init_rtp_handler(handler, rtsp_st, st->
codec);
413 if (handler && handler->
init)
425 if (!strncmp(p,
"rtsp://", 7))
436 if (proto[0] ==
'\0') {
450 payload_type = atoi(buf1);
454 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
461 payload_type = atoi(buf1);
479 }
else if (
av_strstart(p,
"IsRealDataType:integer;",&p)) {
482 }
else if (
av_strstart(p,
"SampleRate:integer;", &p) &&
520 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
534 while (*p !=
'\n' && *p !=
'\r' && *p !=
'\0') {
535 if ((q - buf) <
sizeof(buf) - 1)
540 sdp_parse_line(s, s1, letter, buf);
542 while (*p !=
'\n' && *p !=
'\0')
618 if (reordering_queue_size < 0) {
620 reordering_queue_size = 0;
649 reordering_queue_size);
664 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
665 static void rtsp_parse_range(
int *min_ptr,
int *max_ptr,
const char **pp)
673 v = strtol(q, &p, 10);
677 v = strtol(p, &p, 10);
689 char transport_protocol[16];
691 char lower_transport[16];
705 get_word_sep(transport_protocol,
sizeof(transport_protocol),
709 lower_transport[0] =
'\0';
716 }
else if (!
av_strcasecmp (transport_protocol,
"x-pn-tng") ||
719 get_word_sep(lower_transport,
sizeof(lower_transport),
"/;,", &p);
724 lower_transport[0] =
'\0';
740 while (*p !=
'\0' && *p !=
',') {
742 if (!strcmp(parameter,
"port")) {
747 }
else if (!strcmp(parameter,
"client_port")) {
753 }
else if (!strcmp(parameter,
"server_port")) {
759 }
else if (!strcmp(parameter,
"interleaved")) {
765 }
else if (!strcmp(parameter,
"multicast")) {
768 }
else if (!strcmp(parameter,
"ttl")) {
772 th->
ttl = strtol(p, &end, 10);
775 }
else if (!strcmp(parameter,
"destination")) {
781 }
else if (!strcmp(parameter,
"source")) {
787 }
else if (!strcmp(parameter,
"mode")) {
791 if (!strcmp(buf,
"record") ||
792 !strcmp(buf,
"receive"))
797 while (*p !=
';' && *p !=
'\0' && *p !=
',')
809 static void handle_rtp_info(
RTSPState *rt,
const char *url,
810 uint32_t seq, uint32_t rtptime)
813 if (!rtptime || !url[0])
829 static void rtsp_parse_rtp_info(
RTSPState *rt,
const char *p)
832 char key[20], value[1024], url[1024] =
"";
833 uint32_t seq = 0, rtptime = 0;
845 if (!strcmp(key,
"url"))
847 else if (!strcmp(key,
"seq"))
848 seq = strtoul(value,
NULL, 10);
849 else if (!strcmp(key,
"rtptime"))
850 rtptime = strtoul(value,
NULL, 10);
852 handle_rtp_info(rt, url, seq, rtptime);
861 handle_rtp_info(rt, url, seq, rtptime);
875 (t = strtol(p,
NULL, 10)) > 0) {
881 rtsp_parse_transport(reply, p);
883 reply->
seq = strtol(p,
NULL, 10);
898 }
else if (
av_stristart(p,
"WWW-Authenticate:", &p) && rt) {
901 }
else if (
av_stristart(p,
"Authentication-Info:", &p) && rt) {
904 }
else if (
av_stristart(p,
"Content-Base:", &p) && rt) {
906 if (method && !strcmp(method,
"DESCRIBE"))
910 if (method && !strcmp(method,
"PLAY"))
911 rtsp_parse_rtp_info(rt, p);
913 if (strstr(p,
"GET_PARAMETER") &&
914 method && !strcmp(method,
"OPTIONS"))
916 }
else if (
av_stristart(p,
"x-Accept-Dynamic-Rate:", &p) && rt) {
937 av_dlog(s,
"skipping RTP packet len=%d\n", len);
942 if (len1 >
sizeof(buf))
952 unsigned char **content_ptr,
953 int return_on_interleaved_data,
const char *method)
956 char buf[4096], buf1[1024], *q;
959 int ret, content_length, line_count = 0, request = 0;
960 unsigned char *content =
NULL;
966 memset(reply, 0,
sizeof(*reply));
974 av_dlog(s,
"ret=%d c=%02x [%c]\n", ret, ch, ch);
981 if (return_on_interleaved_data) {
985 }
else if (ch !=
'\r') {
986 if ((q - buf) <
sizeof(buf) - 1)
992 av_dlog(s,
"line='%s'\n", buf);
998 if (line_count == 0) {
1001 if (!strncmp(buf1,
"RTSP/", 5)) {
1022 if (content_length > 0) {
1024 content =
av_malloc(content_length + 1);
1026 content[content_length] =
'\0';
1029 *content_ptr = content;
1036 const char* ptr = buf;
1038 if (!strcmp(reply->
reason,
"OPTIONS")) {
1039 snprintf(buf,
sizeof(buf),
"RTSP/1.0 200 OK\r\n");
1046 snprintf(buf,
sizeof(buf),
"RTSP/1.0 501 Not Implemented\r\n");
1071 if (rt->
seq != reply->
seq) {
1077 if (reply->
notice == 2101 ||
1079 reply->
notice == 2306 ) {
1081 }
else if (reply->
notice >= 4400 && reply->
notice < 5500) {
1083 }
else if (reply->
notice == 2401 ||
1104 const char *method,
const char *url,
1105 const char *headers,
1106 const unsigned char *send_content,
1107 int send_content_length)
1110 char buf[4096], *out_buf;
1116 snprintf(buf,
sizeof(buf),
"%s %s RTSP/1.0\r\n", method, url);
1120 if (rt->
session_id[0] !=
'\0' && (!headers ||
1121 !strstr(headers,
"\nIf-Match:"))) {
1126 rt->
auth, url, method);
1131 if (send_content_length > 0 && send_content)
1132 av_strlcatf(buf,
sizeof(buf),
"Content-Length: %d\r\n", send_content_length);
1138 out_buf = base64buf;
1141 av_dlog(s,
"Sending:\n%s--\n", buf);
1144 if (send_content_length > 0 && send_content) {
1147 "with content data not supported\n");
1158 const char *url,
const char *headers)
1160 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers,
NULL, 0);
1165 unsigned char **content_ptr)
1168 content_ptr,
NULL, 0);
1172 const char *method,
const char *url,
1175 unsigned char **content_ptr,
1176 const unsigned char *send_content,
1177 int send_content_length)
1181 int ret, attempts = 0;
1185 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1187 send_content_length)))
1211 int lower_transport,
const char *real_challenge)
1214 int rtx = 0, j, i, err, interleave = 0, port_off;
1218 const char *trans_pref;
1221 trans_pref =
"x-pn-tng";
1223 trans_pref =
"RAW/RAW";
1225 trans_pref =
"RTP/AVP";
1239 port_off -= port_off & 0x01;
1241 for (j = rt->
rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1242 char transport[2048];
1278 while (j <= rt->rtp_port_max) {
1280 "?localport=%d", j);
1295 snprintf(transport,
sizeof(transport) - 1,
1296 "%s/UDP;", trans_pref);
1298 av_strlcat(transport,
"unicast;",
sizeof(transport));
1300 "client_port=%d", port);
1303 av_strlcatf(transport,
sizeof(transport),
"-%d", port + 1);
1316 snprintf(transport,
sizeof(transport) - 1,
1317 "%s/TCP;", trans_pref);
1319 av_strlcat(transport,
"unicast;",
sizeof(transport));
1321 "interleaved=%d-%d",
1322 interleave, interleave + 1);
1327 snprintf(transport,
sizeof(transport) - 1,
1328 "%s/UDP;multicast", trans_pref);
1331 av_strlcat(transport,
";mode=record",
sizeof(transport));
1334 av_strlcat(transport,
";mode=play",
sizeof(transport));
1335 snprintf(cmd,
sizeof(cmd),
1336 "Transport: %s\r\n",
1339 av_strlcat(cmd,
"x-Dynamic-Rate: 0\r\n",
sizeof(cmd));
1341 char real_res[41], real_csum[9];
1346 "RealChallenge2: %s, sd=%s\r\n",
1386 char url[1024],
options[30] =
"";
1389 av_strlcpy(options,
"?connect=1",
sizeof(options));
1414 char url[1024], namebuf[50], optbuf[20] =
"";
1428 snprintf(optbuf,
sizeof(optbuf),
"?ttl=%d", ttl);
1429 getnameinfo((
struct sockaddr*) &addr,
sizeof(addr),
1432 port,
"%s", optbuf);
1470 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1471 int port, err, tcp_fd;
1473 int lower_transport_mask = 0;
1474 char real_challenge[64] =
"";
1476 socklen_t peer_len =
sizeof(peer);
1503 host,
sizeof(host), &port, path,
sizeof(path), s->
filename);
1510 if (!lower_transport_mask)
1519 "only UDP and TCP are supported for output.\n");
1529 host, port,
"%s", path);
1533 char httpname[1024];
1534 char sessioncookie[17];
1537 ff_url_join(httpname,
sizeof(httpname),
"http", auth, host, port,
"%s", path);
1538 snprintf(sessioncookie,
sizeof(sessioncookie),
"%08x%08x",
1549 snprintf(headers,
sizeof(headers),
1550 "x-sessioncookie: %s\r\n"
1551 "Accept: application/x-rtsp-tunnelled\r\n"
1552 "Pragma: no-cache\r\n"
1553 "Cache-Control: no-cache\r\n",
1571 snprintf(headers,
sizeof(headers),
1572 "x-sessioncookie: %s\r\n"
1573 "Content-Type: application/x-rtsp-tunnelled\r\n"
1574 "Pragma: no-cache\r\n"
1575 "Cache-Control: no-cache\r\n"
1576 "Content-Length: 32767\r\n"
1577 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1618 if (!getpeername(tcp_fd, (
struct sockaddr*) &peer, &peer_len)) {
1619 getnameinfo((
struct sockaddr*) &peer, peer_len, host,
sizeof(host),
1638 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1639 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1640 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1641 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1668 int lower_transport =
ff_log2_tab[lower_transport_mask &
1669 ~(lower_transport_mask - 1)];
1673 real_challenge :
NULL);
1676 lower_transport_mask &= ~(1 << lower_transport);
1677 if (lower_transport_mask == 0 && err == 1) {
1678 err =
AVERROR(EPROTONOSUPPORT);
1705 uint8_t *buf,
int buf_size, int64_t wait_end)
1709 int n, i, ret, tcp_fd, timeout_cnt = 0;
1711 struct pollfd *p = rt->
p;
1712 int *fds =
NULL, fdsnum, fdsidx;
1722 p[max_p].fd = tcp_fd;
1723 p[max_p++].events = POLLIN;
1737 "Number of fds %d not supported\n", fdsnum);
1740 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1741 p[max_p].fd = fds[fdsidx];
1742 p[max_p++].events = POLLIN;
1749 int j = 1 - (tcp_fd == -1);
1754 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1757 *prtsp_st = rtsp_st;
1764 #if CONFIG_RTSP_DEMUXER
1765 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1772 "Unable to answer to TEARDOWN\n");
1788 }
else if (n < 0 && errno != EINTR)
1820 "Unable to pick stream for packet - SSRC not known for "
1842 int64_t wait_end = 0;
1864 }
else if (ret == 1) {
1872 int64_t first_queue_time = 0;
1879 if (queue_time && (queue_time - first_queue_time < 0 ||
1880 !first_queue_time)) {
1881 first_queue_time = queue_time;
1885 if (first_queue_time)
1886 wait_end = first_queue_time + s->
max_delay;
1899 #if CONFIG_RTSP_DEMUXER
1912 len = pick_stream(s, &rtsp_st, rt->
recvbuf, len);
1917 if (len ==
AVERROR(EAGAIN) && first_queue_st &&
1919 rtsp_st = first_queue_st;
1949 if (rtpctx2 && st && st2 &&
1994 #if CONFIG_SDP_DEMUXER
2000 while (p < p_end && *p !=
'\0') {
2001 if (p +
sizeof(
"c=IN IP") - 1 < p_end &&
2005 while (p < p_end - 1 && *p !=
'\n') p++;
2038 content[
size] =
'\0';
2054 "?localport=%d&ttl=%d&connect=%d", rtsp_st->
sdp_port,
2080 static const AVClass sdp_demuxer_class = {
2095 .priv_class = &sdp_demuxer_class,
2099 #if CONFIG_RTP_DEMUXER
2110 char host[500], sdp[500];
2117 socklen_t addrlen =
sizeof(addr);
2129 ret =
ffurl_read(in, recvbuf,
sizeof(recvbuf));
2139 if ((recvbuf[0] & 0xc0) != 0x80) {
2148 payload_type = recvbuf[1] & 0x7f;
2157 "without an SDP file describing it\n",
2163 "properly you need an SDP file "
2170 snprintf(sdp,
sizeof(sdp),
2171 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2172 addr.ss_family == AF_INET ? 4 : 6, host,
2175 port, payload_type);
2186 ret = sdp_read_header(s);
2197 static const AVClass rtp_demuxer_class = {
2213 .priv_class = &rtp_demuxer_class,
char auth[128]
plaintext authorization line (username:password)
int interleaved_min
interleave ids, if TCP transport; each TCP/RTSP data packet starts with a '$', stream length and stre...
void av_url_split(char *proto, int proto_size, char *authorization, int authorization_size, char *hostname, int hostname_size, int *port_ptr, char *path, int path_size, const char *url)
Split a URL string into components.
void ff_rtsp_skip_packet(AVFormatContext *s)
Skip a RTP/TCP interleaved packet.
int rtp_port_min
Minimum and maximum local UDP ports.
int ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
Parse a Windows Media Server-specific SDP line.
void * av_malloc(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Realmedia Data Transport.
void ff_rtsp_undo_setup(AVFormatContext *s)
Undo the effect of ff_rtsp_make_setup_request, close the transport_priv and rtp_handle fields...
void ff_rtp_send_punch_packets(URLContext *rtp_handle)
Send a dummy packet on both port pairs to set up the connection state in potential NAT routers...
AVIOInterruptCB interrupt_callback
Custom interrupt callbacks for the I/O layer.
int avio_close_dyn_buf(AVIOContext *s, uint8_t **pbuffer)
Return the written size and a pointer to the buffer.
char source[INET6_ADDRSTRLEN+1]
source IP address
HTTPAuthType
Authentication types, ordered from weakest to strongest.
static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
Parse a string p in the form of Range:npt=xx-xx, and determine the start and end time.
char control_uri[1024]
some MS RTSP streams contain a URL in the SDP that we need to use for all subsequent RTSP requests...
int av_parse_time(int64_t *timeval, const char *timestr, int duration)
Parse timestr and return in *time a corresponding number of microseconds.
int ffurl_write(URLContext *h, const unsigned char *buf, int size)
Write size bytes from buf to the resource accessed by h.
#define RTSP_DEFAULT_PORT
struct pollfd * p
Polling array for udp.
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
Open RTSP transport context.
int ffurl_connect(URLContext *uc, AVDictionary **options)
Connect an URLContext that has been allocated by ffurl_alloc.
int ff_rdt_parse_packet(RDTDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
Parse RDT-style packet data (header + media data).
int index
stream index in AVFormatContext
int mode_record
transport set to record data
enum AVMediaType codec_type
int av_strncasecmp(const char *a, const char *b, size_t n)
Locale-independent case-insensitive compare.
void ff_network_close(void)
initialized and sending/receiving data
char real_challenge[64]
the "RealChallenge1:" field from the server
av_dlog(ac->avr,"%d samples - audio_convert: %s to %s (%s)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt), use_generic?ac->func_descr_generic:ac->func_descr)
#define RTSP_RTP_PORT_MAX
int ctx_flags
Format-specific flags, see AVFMTCTX_xx.
#define RTSP_FLAG_LISTEN
Wait for incoming connections.
char session_id[512]
copy of RTSPMessageHeader->session_id, i.e.
int64_t seek_timestamp
the seek value requested when calling av_seek_frame().
const char * ff_rtp_enc_name(int payload_type)
Return the encoding name (as defined in http://www.iana.org/assignments/rtp-parameters) for a given p...
int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, int lower_transport, const char *real_challenge)
Do the SETUP requests for each stream for the chosen lower transport mode.
enum RTSPLowerTransport lower_transport
network layer transport protocol; e.g.
RTPDemuxContext * ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, int queue_size)
open a new RTP parse context for stream 'st'.
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
enum AVCodecID ff_rtp_codec_id(const char *buf, enum AVMediaType codec_type)
Return the codec id for the given encoding name and codec type.
int ff_rtsp_connect(AVFormatContext *s)
Connect to the RTSP server and set up the individual media streams.
Standards-compliant RTP-server.
int reordering_queue_size
Size of RTP packet reordering queue.
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
int av_stristart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str independent of case.
int get_parameter_supported
Whether the server supports the GET_PARAMETER method.
Opaque data information usually continuous.
int ttl
time-to-live value (required for multicast); the amount of HOPs that packets will be allowed to make ...
int(* init)(AVFormatContext *s, int st_index, PayloadContext *priv_data)
Initialize dynamic protocol handler, called after the full rtpmap line is parsed, may be null...
int ff_network_init(void)
miscellaneous OS support macros and functions.
PayloadContext *(* alloc)(void)
Allocate any data needed by the rtp parsing for this dynamic data.
int id
Format-specific stream ID.
#define DEFAULT_REORDERING_DELAY
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method)
void(* free)(PayloadContext *protocol_data)
Free any data needed by the rtp parsing for this dynamic data.
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
int accept_dynamic_rate
Whether the server accepts the x-Dynamic-Rate header.
URLContext * rtsp_hd_out
Additional output handle, used when input and output are done separately, eg for HTTP tunneling...
Describe a single stream, as identified by a single m= line block in the SDP content.
Custom IO - not a public option for lower_transport_mask, but set in the SDP demuxer based on a flag...
void ff_http_init_auth_state(URLContext *dest, const URLContext *src)
Initialize the authentication state based on another HTTP URLContext.
static av_cold int read_close(AVFormatContext *ctx)
const uint8_t ff_log2_tab[256]
int ff_rtsp_parse_streaming_commands(AVFormatContext *s)
Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in listen mode.
int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr)
Send a command to the RTSP server and wait for the reply.
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
int ff_rtp_get_local_rtp_port(URLContext *h)
Return the local rtp port used by the RTP connection.
struct AVOutputFormat * oformat
#define RTSP_DEFAULT_AUDIO_SAMPLERATE
void ff_rdt_parse_close(RDTDemuxContext *s)
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
int ff_sdp_parse(AVFormatContext *s, const char *content)
Parse an SDP description of streams by populating an RTSPState struct within the AVFormatContext; als...
Private data for the RTSP demuxer.
int64_t last_cmd_time
timestamp of the last RTSP command that we sent to the RTSP server.
int ffurl_alloc(URLContext **puc, const char *filename, int flags, const AVIOInterruptCB *int_cb)
Create a URLContext for accessing to the resource indicated by url, but do not initiate the connectio...
void av_free(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc(). ...
int ffurl_get_multi_file_handle(URLContext *h, int **handles, int *numhandles)
Return the file descriptors associated with this URL.
int timeout
copy of RTSPMessageHeader->timeout, i.e.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
int avio_close(AVIOContext *s)
Close the resource accessed by the AVIOContext s and free it.
const AVOption ff_rtsp_options[]
void av_log(void *avcl, int level, const char *fmt,...)
AVStream * avformat_new_stream(AVFormatContext *s, AVCodec *c)
Add a new stream to a media file.
const char * name
Name of the codec implementation.
enum RTSPControlTransport control_transport
RTSP transport mode, such as plain or tunneled.
int ffio_read_partial(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
char * av_base64_encode(char *out, int out_size, const uint8_t *in, int in_size)
Encode data to base64 and null-terminate.
size_t av_strlcpy(char *dst, const char *src, size_t size)
Copy the string src to dst, but no more than size - 1 bytes, and null-terminate dst.
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_id(int id, enum AVMediaType codec_type)
struct RTSPStream ** rtsp_streams
streams in this session
int ff_rtp_get_codec_info(AVCodecContext *codec, int payload_type)
Initialize a codec context based on the payload type.
int stream_index
corresponding stream index, if any.
AVCodecContext * codec
Codec context associated with this stream.
MpegTSContext * ff_mpegts_parse_open(AVFormatContext *s)
int buf_size
Size of buf except extra allocated bytes.
int seq
RTSP command sequence number.
#define CONFIG_RTSP_DEMUXER
unsigned char * buf
Buffer must have AVPROBE_PADDING_SIZE of extra allocated bytes filled with zero.
uint8_t * recvbuf
Reusable buffer for receiving packets.
unsigned int nb_streams
A list of all streams in the file.
#define RTSP_FLAG_CUSTOM_IO
Do all IO via the AVIOContext.
AVFormatContext * asf_ctx
The following are used for RTP/ASF streams.
char filename[1024]
input or output filename
int nb_rtsp_streams
number of items in the 'rtsp_streams' variable
int64_t first_rtcp_ntp_time
#define AV_BASE64_SIZE(x)
Calculate the output size needed to base64-encode x bytes.
void * cur_transport_priv
RTSPStream->transport_priv of the last stream that we read a packet from.
int av_strcasecmp(const char *a, const char *b)
#define RTSP_TCP_MAX_PACKET_SIZE
HTTP tunneled - not a proper transport mode as such, only for use via AVOptions.
This describes a single item in the "Transport:" line of one stream as negotiated by the SETUP RTSP c...
RTSP over HTTP (tunneling)
static void get_word_until_chars(char *buf, int buf_size, const char *sep, const char **pp)
static void get_word(char *buf, int buf_size, const char **pp)
Usually treated as AVMEDIA_TYPE_DATA.
int(* parse_sdp_a_line)(AVFormatContext *s, int st_index, PayloadContext *priv_data, const char *line)
Parse the a= line from the sdp field.
int ffurl_get_file_handle(URLContext *h)
Return the file descriptor associated with this URL.
int sdp_port
The following are used only in SDP, not RTSP.
int ff_mpegts_parse_packet(MpegTSContext *ts, AVPacket *pkt, const uint8_t *buf, int len)
struct MpegTSContext * ts
The following are used for parsing raw mpegts in udp.
int stale
Auth ok, but needs to be resent with a new nonce.
int sdp_payload_type
payload type
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, RTPDynamicProtocolHandler *handler)
static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
static int read_header(FFV1Context *f)
int64_t av_gettime(void)
Get the current time in microseconds.
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
struct sockaddr_storage sdp_ip
IP address (from SDP content)
enum AVMediaType codec_type
#define AVIO_FLAG_READ_WRITE
int ff_check_interrupt(AVIOInterruptCB *cb)
Check if the user has requested to interrup a blocking function associated with cb.
int sample_rate
samples per second
AVIOContext * pb
I/O context.
int media_type_mask
Mask of all requested media types.
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
main external API structure.
AVCodec * avcodec_find_decoder(enum AVCodecID id)
Find a registered decoder with a matching codec ID.
#define RTSP_FLAG_OPTS(name, longname)
RDTDemuxContext * ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx, void *priv_data, RTPDynamicProtocolHandler *handler)
Allocate and init the RDT parsing context.
#define RTSP_FLAG_FILTER_SRC
Filter incoming UDP packets - receive packets only from the right source address and port...
enum RTSPTransport transport
the negotiated data/packet transport protocol; e.g.
int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
Announce the stream to the server and set up the RTSPStream child objects for each media stream...
static int read_packet(AVFormatContext *ctx, AVPacket *pkt)
int av_dict_set(AVDictionary **pm, const char *key, const char *value, int flags)
Set the given entry in *pm, overwriting an existing entry.
int rtsp_flags
Various option flags for the RTSP muxer/demuxer.
Describe the class of an AVClass context structure.
void ff_real_parse_sdp_a_line(AVFormatContext *s, int stream_index, const char *line)
Parse a server-related SDP line.
struct RTSPState RTSPState
Private data for the RTSP demuxer.
PayloadContext * dynamic_protocol_context
private data associated with the dynamic protocol
char last_reply[2048]
The last reply of the server to a RTSP command.
enum RTSPTransport transport
data/packet transport protocol; e.g.
size_t av_strlcatf(char *dst, size_t size, const char *fmt,...)
#define RTSP_MEDIATYPE_OPTS(name, longname)
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, uint8_t *buf, int buf_size)
Receive one RTP packet from an TCP interleaved RTSP stream.
void ff_rtsp_close_streams(AVFormatContext *s)
Close and free all streams within the RTSP (de)muxer.
int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio, int count)
some rtp servers assume client is dead if they don't hear from them...
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
This structure contains the data a format has to probe a file.
#define RTSP_DEFAULT_NB_AUDIO_CHANNELS
char * ff_http_auth_create_response(HTTPAuthState *state, const char *auth, const char *path, const char *method)
size_t av_strlcat(char *dst, const char *src, size_t size)
Append the string src to the string dst, but to a total length of no more than size - 1 bytes...
#define RTP_PT_IS_RTCP(x)
enum RTSPServerType server_type
brand of server that we're talking to; e.g.
int ffurl_close(URLContext *h)
Close the resource accessed by the URLContext h, and free the memory used by it.
#define CONFIG_RTSP_MUXER
int64_t start_time
Decoding: position of the first frame of the component, in AV_TIME_BASE fractional seconds...
enum RTSPClientState state
indicator of whether we are currently receiving data from the server.
const OptionDef options[]
int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
Receive one packet from the RTSPStreams set up in the AVFormatContext (which should contain a RTSPSta...
int av_strstart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str.
int ff_rtsp_send_cmd_with_content(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr, const unsigned char *send_content, int send_content_length)
Send a command to the RTSP server and wait for the reply.
static const AVOption sdp_options[]
void ff_mpegts_parse_close(MpegTSContext *ts)
int ff_rtp_chain_mux_open(AVFormatContext **out, AVFormatContext *s, AVStream *st, URLContext *handle, int packet_size, int idx)
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_name(const char *name, enum AVMediaType codec_type)
int ffio_init_context(AVIOContext *s, unsigned char *buffer, int buffer_size, int write_flag, void *opaque, int(*read_packet)(void *opaque, uint8_t *buf, int buf_size), int(*write_packet)(void *opaque, uint8_t *buf, int buf_size), int64_t(*seek)(void *opaque, int64_t offset, int whence))
int ffurl_open(URLContext **puc, const char *filename, int flags, const AVIOInterruptCB *int_cb, AVDictionary **options)
Create an URLContext for accessing to the resource indicated by url, and open it. ...
int need_subscription
The following are used for Real stream selection.
RTPDynamicProtocolHandler * dynamic_handler
The following are used for dynamic protocols (rtpdec_*.c/rdt.c)
#define AVERROR_INVALIDDATA
int ffurl_read_complete(URLContext *h, unsigned char *buf, int size)
Read as many bytes as possible (up to size), calling the read function multiple times if necessary...
void ff_rdt_calc_response_and_checksum(char response[41], char chksum[9], const char *challenge)
Calculate the response (RealChallenge2 in the RTSP header) to the challenge (RealChallenge1 in the RT...
#define AVPROBE_SCORE_MAX
#define RTSP_REORDERING_OPTS()
struct AVInputFormat * iformat
Can only be iformat or oformat, not both at the same time.
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
void ff_http_auth_handle_header(HTTPAuthState *state, const char *key, const char *value)
#define AVERROR_PATCHWELCOME
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, unsigned char **content_ptr, int return_on_interleaved_data, const char *method)
Read a RTSP message from the server, or prepare to read data packets if we're reading data interleave...
#define AVFMTCTX_NOHEADER
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, const char *url, const char *headers)
Send a command to the RTSP server without waiting for the reply.
static void get_word_sep(char *buf, int buf_size, const char *sep, const char **pp)
TCP; interleaved in RTSP.
HTTPAuthState auth_state
authentication state
#define RTSP_RTP_PORT_MIN
int channels
number of audio channels
char control_url[1024]
url for this stream (from SDP)
void * priv_data
Format private data.
int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
Get the description of the stream and set up the RTSPStream child objects.
void ff_rtp_parse_close(RTPDemuxContext *s)
int sdp_ttl
IP Time-To-Live (from SDP content)
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
int64_t duration
Decoding: duration of the stream, in AV_TIME_BASE fractional seconds.
HTTPAuthType auth_type
The currently chosen auth type.
int lower_transport_mask
A mask with all requested transport methods.
unbuffered private I/O API
uint32_t av_get_random_seed(void)
Get random data.
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
int ff_rtp_set_remote_url(URLContext *h, const char *uri)
If no filename is given to av_open_input_file because you want to get the local port first...
int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
Parse an RTP or RTCP packet directly sent as a buffer.
struct sockaddr_storage destination
destination IP address
#define RTP_REORDER_QUEUE_DEFAULT_SIZE
int interleaved_min
interleave IDs; copies of RTSPTransportField->interleaved_min/max for the selected transport...
This structure stores compressed data.
int server_port_min
UDP unicast server port range; the ports to which we should connect to receive unicast UDP RTP/RTCP d...
void ff_rtsp_close_connections(AVFormatContext *s)
Close all connection handles within the RTSP (de)muxer.
int av_opt_set(void *obj, const char *name, const char *val, int search_flags)
static const AVOption rtp_options[]
int ffurl_read(URLContext *h, unsigned char *buf, int size)
Read up to size bytes from the resource accessed by h, and store the read bytes in buf...
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
URLContext * rtp_handle
RTP stream handle (if UDP)
int port_min
UDP multicast port range; the ports to which we should connect to receive multicast UDP data...
void * transport_priv
RTP/RDT parse context if input, RTP AVFormatContext if output.
No authentication specified.
int client_port_min
UDP client ports; these should be the local ports of the UDP RTP (and RTCP) sockets over which we rec...