54 #define POLL_TIMEOUT_MS 100 55 #define READ_PACKET_TIMEOUT_S 10 56 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS 57 #define SDP_MAX_SIZE 16384 58 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH 59 #define DEFAULT_REORDERING_DELAY 100000 61 #define OFFSET(x) offsetof(RTSPState, x) 62 #define DEC AV_OPT_FLAG_DECODING_PARAM 63 #define ENC AV_OPT_FLAG_ENCODING_PARAM 65 #define RTSP_FLAG_OPTS(name, longname) \ 66 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \ 67 { "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 #define RTSP_MEDIATYPE_OPTS(name, longname) \ 70 { 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" }, \ 71 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \ 72 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \ 73 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" } 75 #define COMMON_OPTS() \ 76 { "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 }, \ 77 { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC } \ 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" }, \
93 {
"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 },
126 const char *sep,
const char **pp)
134 while (!strchr(sep, *p) && *p !=
'\0') {
135 if ((q - buf) < buf_size - 1)
147 if (**pp ==
'/') (*pp)++;
151 static void get_word(
char *buf,
int buf_size,
const char **pp)
194 memcpy(sock, ai->ai_addr,
FFMIN(
sizeof(*sock), ai->ai_addrlen));
240 int payload_type,
const char *p)
262 init_rtp_handler(handler, rtsp_st, st);
272 if (desc && desc->
name)
305 finalize_rtp_handler_init(s, rtsp_st, st);
313 char *value,
int value_size)
328 typedef struct SDPParseState {
333 int nb_default_include_source_addrs;
334 struct RTSPSource **default_include_source_addrs;
335 int nb_default_exclude_source_addrs;
336 struct RTSPSource **default_exclude_source_addrs;
339 char delayed_fmtp[2048];
342 static void copy_default_source_addrs(
struct RTSPSource **addrs,
int count,
347 for (i = 0; i < count; i++) {
349 rtsp_src2 =
av_malloc(
sizeof(*rtsp_src2));
352 memcpy(rtsp_src2, rtsp_src,
sizeof(*rtsp_src));
358 int payload_type,
const char *
line)
374 int letter,
const char *buf)
377 char buf1[64], st_type[64];
390 if (s1->skip_media && letter !=
'm')
395 if (strcmp(buf1,
"IN") != 0)
398 if (strcmp(buf1,
"IP4") && strcmp(buf1,
"IP6"))
410 s1->default_ip = sdp_ip;
411 s1->default_ttl = ttl;
433 get_word(st_type,
sizeof(st_type), &p);
434 if (!strcmp(st_type,
"audio")) {
436 }
else if (!strcmp(st_type,
"video")) {
438 }
else if (!strcmp(st_type,
"application") || !strcmp(st_type,
"text")) {
451 rtsp_st->
sdp_ip = s1->default_ip;
452 rtsp_st->
sdp_ttl = s1->default_ttl;
454 copy_default_source_addrs(s1->default_include_source_addrs,
455 s1->nb_default_include_source_addrs,
458 copy_default_source_addrs(s1->default_exclude_source_addrs,
459 s1->nb_default_exclude_source_addrs,
467 if (!strcmp(buf1,
"udp"))
469 else if (strstr(buf1,
"/AVPF") || strstr(buf1,
"/SAVPF"))
485 init_rtp_handler(handler, rtsp_st,
NULL);
486 finalize_rtp_handler_init(s, rtsp_st,
NULL);
509 init_rtp_handler(handler, rtsp_st, st);
510 finalize_rtp_handler_init(s, rtsp_st, st);
522 if (!strncmp(p,
"rtsp://", 7))
533 if (proto[0] ==
'\0') {
547 payload_type = atoi(buf1);
551 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
555 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
561 payload_type = atoi(buf1);
562 if (s1->seen_rtpmap) {
566 av_strlcpy(s1->delayed_fmtp, buf,
sizeof(s1->delayed_fmtp));
571 rtsp_st->
ssrc = strtoll(buf1,
NULL, 10);
591 }
else if (
av_strstart(p,
"IsRealDataType:integer;",&p)) {
594 }
else if (
av_strstart(p,
"SampleRate:integer;", &p) &&
609 if (strcmp(buf1,
"incl") && strcmp(buf1,
"excl"))
611 exclude = !strcmp(buf1,
"excl");
614 if (strcmp(buf1,
"IN") != 0)
617 if (strcmp(buf1,
"IP4") && strcmp(buf1,
"IP6") && strcmp(buf1,
"*"))
629 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
636 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
677 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
691 while (*p !=
'\n' && *p !=
'\r' && *p !=
'\0') {
692 if ((q - buf) <
sizeof(buf) - 1)
697 sdp_parse_line(s, s1, letter, buf);
699 while (*p !=
'\n' && *p !=
'\0')
705 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
706 av_free(s1->default_include_source_addrs[i]);
707 av_freep(&s1->default_include_source_addrs);
708 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
709 av_free(s1->default_exclude_source_addrs[i]);
710 av_freep(&s1->default_exclude_source_addrs);
794 if (reordering_queue_size < 0) {
796 reordering_queue_size = 0;
826 reordering_queue_size);
848 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER 849 static void rtsp_parse_range(
int *min_ptr,
int *max_ptr,
const char **pp)
857 v = strtol(q, &p, 10);
861 v = strtol(p, &p, 10);
874 char transport_protocol[16];
876 char lower_transport[16];
890 get_word_sep(transport_protocol,
sizeof(transport_protocol),
894 lower_transport[0] =
'\0';
901 }
else if (!
av_strcasecmp (transport_protocol,
"x-pn-tng") ||
904 get_word_sep(lower_transport,
sizeof(lower_transport),
"/;,", &p);
909 lower_transport[0] =
'\0';
925 while (*p !=
'\0' && *p !=
',') {
927 if (!strcmp(parameter,
"port")) {
932 }
else if (!strcmp(parameter,
"client_port")) {
938 }
else if (!strcmp(parameter,
"server_port")) {
944 }
else if (!strcmp(parameter,
"interleaved")) {
950 }
else if (!strcmp(parameter,
"multicast")) {
953 }
else if (!strcmp(parameter,
"ttl")) {
957 th->
ttl = strtol(p, &end, 10);
960 }
else if (!strcmp(parameter,
"destination")) {
966 }
else if (!strcmp(parameter,
"source")) {
972 }
else if (!strcmp(parameter,
"mode")) {
976 if (!strcmp(buf,
"record") ||
977 !strcmp(buf,
"receive"))
982 while (*p !=
';' && *p !=
'\0' && *p !=
',')
996 static void handle_rtp_info(
RTSPState *rt,
const char *url,
997 uint32_t seq, uint32_t rtptime)
1000 if (!rtptime || !url[0])
1016 static void rtsp_parse_rtp_info(
RTSPState *rt,
const char *p)
1019 char key[20], value[1024], url[1024] =
"";
1020 uint32_t seq = 0, rtptime = 0;
1032 if (!strcmp(key,
"url"))
1034 else if (!strcmp(key,
"seq"))
1035 seq = strtoul(value,
NULL, 10);
1036 else if (!strcmp(key,
"rtptime"))
1037 rtptime = strtoul(value,
NULL, 10);
1039 handle_rtp_info(rt, url, seq, rtptime);
1048 handle_rtp_info(rt, url, seq, rtptime);
1063 (t = strtol(p,
NULL, 10)) > 0) {
1069 rtsp_parse_transport(s, reply, p);
1071 reply->
seq = strtol(p,
NULL, 10);
1086 }
else if (
av_stristart(p,
"WWW-Authenticate:", &p) && rt) {
1089 }
else if (
av_stristart(p,
"Authentication-Info:", &p) && rt) {
1092 }
else if (
av_stristart(p,
"Content-Base:", &p) && rt) {
1094 if (method && !strcmp(method,
"DESCRIBE"))
1098 if (method && !strcmp(method,
"PLAY"))
1099 rtsp_parse_rtp_info(rt, p);
1101 if (strstr(p,
"GET_PARAMETER") &&
1102 method && !strcmp(method,
"OPTIONS"))
1104 }
else if (
av_stristart(p,
"x-Accept-Dynamic-Rate:", &p) && rt) {
1130 if (len1 >
sizeof(buf))
1140 unsigned char **content_ptr,
1141 int return_on_interleaved_data,
const char *method)
1144 char buf[4096], buf1[1024], *q;
1147 int ret, content_length, line_count = 0, request = 0;
1148 unsigned char *content =
NULL;
1154 memset(reply, 0,
sizeof(*reply));
1167 if (ch ==
'$' && q == buf) {
1168 if (return_on_interleaved_data) {
1172 }
else if (ch !=
'\r') {
1173 if ((q - buf) <
sizeof(buf) - 1)
1185 if (line_count == 0) {
1188 if (!strncmp(buf1,
"RTSP/", 5)) {
1209 if (content_length > 0) {
1211 content =
av_malloc(content_length + 1);
1215 content[content_length] =
'\0';
1218 *content_ptr = content;
1225 const char* ptr = buf;
1227 if (!strcmp(reply->
reason,
"OPTIONS")) {
1228 snprintf(buf,
sizeof(buf),
"RTSP/1.0 200 OK\r\n");
1235 snprintf(buf,
sizeof(buf),
"RTSP/1.0 501 Not Implemented\r\n");
1260 if (rt->
seq != reply->
seq) {
1266 if (reply->
notice == 2101 ||
1268 reply->
notice == 2306 ) {
1270 }
else if (reply->
notice >= 4400 && reply->
notice < 5500) {
1272 }
else if (reply->
notice == 2401 ||
1293 const char *method,
const char *url,
1294 const char *headers,
1295 const unsigned char *send_content,
1296 int send_content_length)
1299 char buf[4096], *out_buf;
1305 snprintf(buf,
sizeof(buf),
"%s %s RTSP/1.0\r\n", method, url);
1310 if (rt->
session_id[0] !=
'\0' && (!headers ||
1311 !strstr(headers,
"\nIf-Match:"))) {
1316 rt->
auth, url, method);
1321 if (send_content_length > 0 && send_content)
1322 av_strlcatf(buf,
sizeof(buf),
"Content-Length: %d\r\n", send_content_length);
1328 out_buf = base64buf;
1334 if (send_content_length > 0 && send_content) {
1337 "with content data not supported\n");
1348 const char *url,
const char *headers)
1350 return rtsp_send_cmd_with_content_async(s, method, url, headers,
NULL, 0);
1355 unsigned char **content_ptr)
1358 content_ptr,
NULL, 0);
1362 const char *method,
const char *url,
1365 unsigned char **content_ptr,
1366 const unsigned char *send_content,
1367 int send_content_length)
1371 int ret, attempts = 0;
1375 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1377 send_content_length)))
1401 int lower_transport,
const char *real_challenge)
1404 int rtx = 0, j, i, err, interleave = 0, port_off;
1408 const char *trans_pref;
1411 trans_pref =
"x-pn-tng";
1413 trans_pref =
"RAW/RAW";
1415 trans_pref =
"RTP/AVP";
1429 port_off -= port_off & 0x01;
1431 for (j = rt->
rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1432 char transport[2048];
1468 while (j <= rt->rtp_port_max) {
1472 "?localport=%d", j);
1491 snprintf(transport,
sizeof(transport) - 1,
1492 "%s/UDP;", trans_pref);
1494 av_strlcat(transport,
"unicast;",
sizeof(transport));
1496 "client_port=%d", port);
1499 av_strlcatf(transport,
sizeof(transport),
"-%d", port + 1);
1512 snprintf(transport,
sizeof(transport) - 1,
1513 "%s/TCP;", trans_pref);
1515 av_strlcat(transport,
"unicast;",
sizeof(transport));
1517 "interleaved=%d-%d",
1518 interleave, interleave + 1);
1523 snprintf(transport,
sizeof(transport) - 1,
1524 "%s/UDP;multicast", trans_pref);
1527 av_strlcat(transport,
";mode=record",
sizeof(transport));
1530 av_strlcat(transport,
";mode=play",
sizeof(transport));
1531 snprintf(cmd,
sizeof(cmd),
1532 "Transport: %s\r\n",
1535 av_strlcat(cmd,
"x-Dynamic-Rate: 0\r\n",
sizeof(cmd));
1537 char real_res[41], real_csum[9];
1542 "RealChallenge2: %s, sd=%s\r\n",
1582 char url[1024],
options[30] =
"";
1583 const char *peer = host;
1586 av_strlcpy(options,
"?connect=1",
sizeof(options));
1600 char url[1024], namebuf[50], optbuf[20] =
"";
1614 snprintf(optbuf,
sizeof(optbuf),
"?ttl=%d", ttl);
1615 getnameinfo((
struct sockaddr*) &addr,
sizeof(addr),
1618 port,
"%s", optbuf);
1656 char proto[128], host[1024], path[1024];
1657 char tcpname[1024], cmd[2048], auth[128];
1658 const char *lower_rtsp_proto =
"tcp";
1659 int port, err, tcp_fd;
1661 int lower_transport_mask = 0;
1663 char real_challenge[64] =
"";
1665 socklen_t peer_len =
sizeof(peer);
1698 host,
sizeof(host), &port, path,
sizeof(path), s->
filename);
1700 if (!strcmp(proto,
"rtsps")) {
1701 lower_rtsp_proto =
"tls";
1710 port = default_port;
1714 if (!lower_transport_mask)
1723 "only UDP and TCP are supported for output.\n");
1733 host, port,
"%s", path);
1737 char httpname[1024];
1738 char sessioncookie[17];
1741 ff_url_join(httpname,
sizeof(httpname),
"http", auth, host, port,
"%s", path);
1742 snprintf(sessioncookie,
sizeof(sessioncookie),
"%08x%08x",
1753 snprintf(headers,
sizeof(headers),
1754 "x-sessioncookie: %s\r\n" 1755 "Accept: application/x-rtsp-tunnelled\r\n" 1756 "Pragma: no-cache\r\n" 1757 "Cache-Control: no-cache\r\n",
1775 snprintf(headers,
sizeof(headers),
1776 "x-sessioncookie: %s\r\n" 1777 "Content-Type: application/x-rtsp-tunnelled\r\n" 1778 "Pragma: no-cache\r\n" 1779 "Cache-Control: no-cache\r\n" 1780 "Content-Length: 32767\r\n" 1781 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1827 if (!getpeername(tcp_fd, (
struct sockaddr*) &peer, &peer_len)) {
1828 getnameinfo((
struct sockaddr*) &peer, peer_len, host,
sizeof(host),
1847 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n" 1848 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n" 1849 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n" 1850 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1877 int lower_transport =
ff_log2_tab[lower_transport_mask &
1878 ~(lower_transport_mask - 1)];
1882 real_challenge :
NULL);
1885 lower_transport_mask &= ~(1 << lower_transport);
1886 if (lower_transport_mask == 0 && err == 1) {
1887 err =
AVERROR(EPROTONOSUPPORT);
1915 uint8_t *buf,
int buf_size, int64_t wait_end)
1919 int n, i, ret, tcp_fd, timeout_cnt = 0;
1921 struct pollfd *p = rt->
p;
1922 int *fds =
NULL, fdsnum, fdsidx;
1932 p[max_p].fd = tcp_fd;
1933 p[max_p++].events = POLLIN;
1947 "Number of fds %d not supported\n", fdsnum);
1950 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1951 p[max_p].fd = fds[fdsidx];
1952 p[max_p++].events = POLLIN;
1959 int j = 1 - (tcp_fd == -1);
1964 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1967 *prtsp_st = rtsp_st;
1974 #if CONFIG_RTSP_DEMUXER 1975 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1982 "Unable to answer to TEARDOWN\n");
1998 }
else if (n < 0 && errno != EINTR)
2030 "Unable to pick stream for packet - SSRC not known for " 2052 int64_t wait_end = 0;
2074 }
else if (ret == 1) {
2083 int64_t first_queue_time = 0;
2090 if (queue_time && (queue_time - first_queue_time < 0 ||
2091 !first_queue_time)) {
2092 first_queue_time = queue_time;
2096 if (first_queue_time) {
2097 wait_end = first_queue_time + s->
max_delay;
2100 first_queue_st =
NULL;
2113 #if CONFIG_RTSP_DEMUXER 2130 len = pick_stream(s, &rtsp_st, rt->
recvbuf, len);
2135 if (len ==
AVERROR(EAGAIN) && first_queue_st &&
2138 "max delay reached. need to consume packet\n");
2139 rtsp_st = first_queue_st;
2175 if (rtpctx2 && st && st2 &&
2220 #if CONFIG_SDP_DEMUXER 2226 while (p < p_end && *p !=
'\0') {
2227 if (p +
sizeof(
"c=IN IP") - 1 < p_end &&
2231 while (p < p_end - 1 && *p !=
'\n') p++;
2240 static void append_source_addrs(
char *buf,
int size,
const char *
name,
2247 for (i = 1; i < count; i++)
2284 content[
size] =
'\0';
2309 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2314 append_source_addrs(url,
sizeof(url),
"sources",
2317 append_source_addrs(url,
sizeof(url),
"block",
2347 static const AVClass sdp_demuxer_class = {
2362 .priv_class = &sdp_demuxer_class,
2366 #if CONFIG_RTP_DEMUXER 2377 char host[500], sdp[500];
2384 socklen_t addrlen =
sizeof(addr);
2403 ret =
ffurl_read(in, recvbuf,
sizeof(recvbuf));
2413 if ((recvbuf[0] & 0xc0) != 0x80) {
2422 payload_type = recvbuf[1] & 0x7f;
2437 "without an SDP file describing it\n",
2443 "properly you need an SDP file " 2450 snprintf(sdp,
sizeof(sdp),
2451 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2452 addr.ss_family == AF_INET ? 4 : 6, host,
2455 port, payload_type);
2467 ret = sdp_read_header(s);
2479 static const AVClass rtp_demuxer_class = {
2495 .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 ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite, const char *params)
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.
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
int ff_rtp_get_local_rtp_port(URLContext *h)
Return the local rtp port used by the RTP connection.
#define RTP_MAX_PACKET_LENGTH
AVIOInterruptCB interrupt_callback
Custom interrupt callbacks for the I/O layer.
char source[INET6_ADDRSTRLEN+1]
source IP address
HTTPAuthType
Authentication types, ordered from weakest to strongest.
#define AV_LOG_WARNING
Something somehow does not look correct.
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.
static int parse_fmtp(AVFormatContext *s, AVStream *stream, PayloadContext *data, const char *attr, const char *value)
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
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
av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (%s)\, 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 AVIO_FLAG_READ
read-only
#define AVIO_FLAG_WRITE
write-only
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
#define RTSP_FLAG_RTCP_TO_SOURCE
Send RTCP packets to the source address of received packets.
#define RTSP_RTP_PORT_MAX
int nb_include_source_addrs
Number of source-specific multicast include source IP addresses (from SDP content) ...
int ctx_flags
Flags signalling stream properties.
#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...
This struct describes the properties of an encoded stream.
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.
#define CONFIG_RTSP_DEMUXER
#define RTSP_MAX_TRANSPORTS
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 ...
static int get_sockaddr(AVFormatContext *s, const char *buf, struct sockaddr_storage *sock)
int ff_network_init(void)
AVCodecParameters * avcodec_parameters_alloc(void)
Allocate a new AVCodecParameters and set its fields to default values (unknown/invalid/0).
miscellaneous OS support macros and functions.
int feedback
Enable sending RTCP feedback messages according to RFC 4585.
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
int id
Format-specific stream ID.
enum AVStreamParseType need_parsing
#define DEFAULT_REORDERING_DELAY
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
AVStream ** streams
A list of all streams in the file.
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...
int ff_rtp_get_codec_info(AVCodecParameters *par, int payload_type)
Initialize a codec context based on the payload type.
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...
char * protocol_whitelist
A comma-separated list of protocol names that can be used internally by libavformat.
#define CONFIG_RTSP_MUXER
#define AVERROR_EOF
End of file.
void ff_http_init_auth_state(URLContext *dest, const URLContext *src)
Initialize the authentication state based on another HTTP URLContext.
#define AV_LOG_VERBOSE
Detailed information.
static av_cold int read_close(AVFormatContext *ctx)
const uint8_t ff_log2_tab[256]
int ffurl_open(URLContext **puc, const char *filename, int flags, const AVIOInterruptCB *int_cb, AVDictionary **options, const URLProtocol **protocols, URLContext *parent)
Create an URLContext for accessing to the resource indicated by url, and open it. ...
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.
void ff_rtsp_parse_line(AVFormatContext *s, RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method)
struct AVOutputFormat * oformat
The output container format.
const OptionDef options[]
#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...
struct RTSPSource ** exclude_source_addrs
Source-specific multicast exclude source IP addresses (from SDP content)
Private data for the RTSP demuxer.
int64_t last_cmd_time
timestamp of the last RTSP command that we sent to the RTSP server.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
AVDictionary * metadata
Metadata that applies to the whole file.
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.
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
const AVOption ff_rtsp_options[]
void av_dict_free(AVDictionary **pm)
Free all the memory allocated for an AVDictionary struct and all keys and values. ...
enum AVMediaType codec_type
General type of the encoded data.
enum RTSPControlTransport control_transport
RTSP transport mode, such as plain or tunneled.
struct RTSPSource ** include_source_addrs
Source-specific multicast include source IP addresses (from SDP content)
int ffio_read_partial(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
int ffurl_alloc(URLContext **puc, const char *filename, int flags, const AVIOInterruptCB *int_cb, const URLProtocol **protocols)
Create a URLContext for accessing to the resource indicated by url, but do not initiate the connectio...
char * av_base64_encode(char *out, int out_size, const uint8_t *in, int in_size)
Encode data to base64 and null-terminate.
void avcodec_parameters_free(AVCodecParameters **par)
Free an AVCodecParameters instance and everything associated with it and write NULL to the supplied p...
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
const AVCodecDescriptor * avcodec_descriptor_get(enum AVCodecID id)
int stream_index
corresponding stream index, if any.
MpegTSContext * ff_mpegts_parse_open(AVFormatContext *s)
int buf_size
Size of buf except extra allocated bytes.
int seq
RTSP command sequence number.
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
Number of elements in AVFormatContext.streams.
#define RTSP_FLAG_CUSTOM_IO
Do all IO via the AVIOContext.
AVFormatContext * asf_ctx
The following are used for RTP/ASF streams.
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...
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
enum AVStreamParseType need_parsing
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)
const URLProtocol ** protocols
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
static void get_word(char *buf, int buf_size, const char **pp)
Usually treated as AVMEDIA_TYPE_DATA.
int ffurl_get_file_handle(URLContext *h)
Return the file descriptor associated with this URL.
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
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.
void(* close)(PayloadContext *protocol_data)
Free any data needed by the rtp parsing for this dynamic data.
int sdp_payload_type
payload type
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, RTPDynamicProtocolHandler *handler)
int nb_exclude_source_addrs
Number of source-specific multicast exclude source IP addresses (from SDP content) ...
void ffio_free_dyn_buf(AVIOContext **s)
Free a dynamic buffer.
static int read_header(FFV1Context *f)
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio)
#define AVERROR_PATCHWELCOME
Not yet implemented in Libav, patches welcome.
int ff_url_join(char *str, int size, const char *proto, const char *authorization, const char *hostname, int port, const char *fmt,...)
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
char addr[128]
Source-specific multicast include source IP address (from SDP content)
#define AV_LOG_INFO
Standard information.
struct sockaddr_storage sdp_ip
IP address (from SDP content)
void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
Undo the effect of ff_rtsp_make_setup_request, close the transport_priv and rtp_handle fields...
int ff_check_interrupt(AVIOInterruptCB *cb)
Check if the user has requested to interrupt a blocking function associated with cb.
AVIOContext * pb
I/O context.
int media_type_mask
Mask of all requested media types.
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
#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.
uint32_t ssrc
SSRC for this stream, to allow identifying RTCP packets before the first RTP packet.
#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(* parse_sdp_a_line)(AVFormatContext *s, int st_index, PayloadContext *priv_data, const char *line)
Parse the a= line from the sdp field.
#define RTSPS_DEFAULT_PORT
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)
#define AVIO_FLAG_READ_WRITE
read-write pseudo flag
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.
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
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.
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...
const char * name
Name of the codec described by this descriptor.
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)
This struct describes the properties of a single codec described by an AVCodecID. ...
int64_t av_gettime_relative(void)
Get the current time in microseconds since some unspecified starting point.
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.
int64_t start_time
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.
int sample_rate
Audio only.
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)
static AVDictionary * map_to_opts(RTSPState *rt)
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 need_subscription
The following are used for Real stream selection.
RTPDynamicProtocolHandler * dynamic_handler
The following are used for dynamic protocols (rtpdec_*.c/rdt.c)
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...
struct AVInputFormat * iformat
The input container format.
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)
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...
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
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.
char * protocol_blacklist
A comma-separated list of protocol names that will not be used internally by libavformat.
int64_t duration
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.
AVCodecParameters * codecpar
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
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
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...
#define RTP_REORDER_QUEUE_DEFAULT_SIZE
const URLProtocol ** ffurl_get_protocols(const char *whitelist, const char *blacklist)
Construct a list of protocols matching a given whitelist and/or blacklist.
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)
#define AV_NOPTS_VALUE
Undefined timestamp value.
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...