#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"
-#include <sys/time.h>
#if HAVE_POLL_H
#include <poll.h>
#endif
-#include <strings.h>
#include "internal.h"
#include "network.h"
#include "os_support.h"
#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 RTSP_FLAG_OPTS(name, longname) \
{ name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
- { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
+ { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }, \
+ { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
+
+#define RTSP_MEDIATYPE_OPTS(name, longname) \
+ { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
+ { "video", "Video", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
+ { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
+ { "data", "Data", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
const AVOption ff_rtsp_options[] = {
{ "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {0}, 0, 1, DEC },
{ "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
{ "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {(1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
RTSP_FLAG_OPTS("rtsp_flags", "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, {RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
+ { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {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, {-1}, INT_MIN, INT_MAX, DEC },
+ { NULL },
+};
+
+static const AVOption sdp_options[] = {
+ RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
+ RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
+ { NULL },
+};
+
+static const AVOption rtp_options[] = {
+ RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
{ NULL },
};
static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
{
- struct addrinfo hints, *ai = NULL;
- memset(&hints, 0, sizeof(hints));
+ struct addrinfo hints = { 0 }, *ai = NULL;
hints.ai_flags = AI_NUMERICHOST;
if (getaddrinfo(buf, NULL, &hints, &ai))
return -1;
return;
codec->codec_id = handler->codec_id;
rtsp_st->dynamic_handler = handler;
- if (handler->alloc)
+ 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>] */
* 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) {
+ if (payload_type < RTP_PT_PRIVATE) {
+ /* We are in a standard case
+ * (from http://www.iana.org/assignments/rtp-parameters). */
+ /* search into AVRtpPayloadTypes[] */
+ codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
+ }
+
+ if (codec->codec_id == CODEC_ID_NONE) {
RTPDynamicProtocolHandler *handler =
ff_rtp_handler_find_by_name(buf, codec->codec_type);
init_rtp_handler(handler, rtsp_st, codec);
* 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);
- } else {
- /* We are in a standard case
- * (from http://www.iana.org/assignments/rtp-parameters). */
- /* search into AVRtpPayloadTypes[] */
- codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
}
c = avcodec_find_decoder(codec->codec_id);
codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
if (i > 0) {
codec->sample_rate = i;
- av_set_pts_info(st, 32, 1, codec->sample_rate);
+ avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
get_word_sep(buf, sizeof(buf), "/", &p);
i = atoi(buf);
if (i > 0)
case AVMEDIA_TYPE_VIDEO:
av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
if (i > 0)
- av_set_pts_info(st, 32, 1, i);
+ 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;
}
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;
codec_type = AVMEDIA_TYPE_VIDEO;
} else if (!strcmp(st_type, "application")) {
codec_type = AVMEDIA_TYPE_DATA;
- } else {
+ }
+ if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
s1->skip_media = 1;
return;
}
if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
/* no corresponding stream */
+ } 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 = av_new_stream(s, rt->nb_rtsp_streams - 1);
+ 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) {
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)
- av_set_pts_info(st, 32, 1, st->codec->sample_rate);
+ 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 */
/* NOTE: rtpmap is only supported AFTER the 'm=' tag */
get_word(buf1, sizeof(buf1), &p);
payload_type = atoi(buf1);
- st = s->streams[s->nb_streams - 1];
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
- sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
+ 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 */
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, s->nb_streams - 1, p);
+ ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
- rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
if (rtsp_st->dynamic_handler &&
rtsp_st->dynamic_handler->parse_sdp_a_line)
rtsp_st->dynamic_handler->parse_sdp_a_line(s,
- s->nb_streams - 1,
+ rtsp_st->stream_index,
rtsp_st->dynamic_protocol_context, buf);
}
}
* 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, *s1 = &sdp_parse_state;
+ SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
- memset(s1, 0, sizeof(SDPParseState));
p = content;
for (;;) {
p += strspn(p, SPACE_CHARS);
}
av_free(rt->rtsp_streams);
if (rt->asf_ctx) {
- av_close_input_stream (rt->asf_ctx);
- rt->asf_ctx = NULL;
+ avformat_close_input(&rt->asf_ctx);
}
av_free(rt->p);
av_free(rt->recvbuf);
}
-static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
+int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
{
RTSPState *rt = s->priv_data;
AVStream *st = NULL;
s->ctx_flags |= AVFMTCTX_NOHEADER;
if (s->oformat && CONFIG_RTSP_MUXER) {
- rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
- rtsp_st->rtp_handle,
- RTSP_TCP_MAX_PACKET_SIZE);
+ int ret = ff_rtp_chain_mux_open(&rtsp_st->transport_priv, s, st,
+ rtsp_st->rtp_handle,
+ RTSP_TCP_MAX_PACKET_SIZE);
/* 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_RDT && CONFIG_RTPDEC)
rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
rtsp_st->dynamic_protocol_context,
#if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
{
- const char *p;
+ const char *q;
+ char *p;
int v;
- p = *pp;
- p += strspn(p, SPACE_CHARS);
- v = strtol(p, (char **)&p, 10);
+ q = *pp;
+ q += strspn(q, SPACE_CHARS);
+ v = strtol(q, &p, 10);
if (*p == '-') {
p++;
*min_ptr = v;
- v = strtol(p, (char **)&p, 10);
+ v = strtol(p, &p, 10);
*max_ptr = v;
} else {
*min_ptr = v;
get_word_sep(transport_protocol, sizeof(transport_protocol),
"/", &p);
- if (!strcasecmp (transport_protocol, "rtp")) {
+ if (!av_strcasecmp (transport_protocol, "rtp")) {
get_word_sep(profile, sizeof(profile), "/;,", &p);
lower_transport[0] = '\0';
/* rtp/avp/<protocol> */
";,", &p);
}
th->transport = RTSP_TRANSPORT_RTP;
- } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
- !strcasecmp (transport_protocol, "x-real-rdt")) {
+ } 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;
}
- if (!strcasecmp(lower_transport, "TCP"))
+ if (!av_strcasecmp(lower_transport, "TCP"))
th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
else
th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
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 != ',')
} 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));
}
}
char buf[4096], buf1[1024], *q;
unsigned char ch;
const char *p;
- int ret, content_length, line_count = 0;
+ 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) */
if (line_count == 0) {
/* get reply code */
get_word(buf1, sizeof(buf1), &p);
- get_word(buf1, sizeof(buf1), &p);
- reply->status_code = atoi(buf1);
- av_strlcpy(reply->reason, p, sizeof(reply->reason));
+ 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));
line_count++;
}
- if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
+ 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;
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);
{
RTSPState *rt = s->priv_data;
HTTPAuthType cur_auth_type;
- int ret;
+ int ret, attempts = 0;
retry:
cur_auth_type = rt->auth_state.auth_type;
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.auth_type != HTTP_AUTH_NONE)
+ 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){
int lower_transport, const char *real_challenge)
{
RTSPState *rt = s->priv_data;
- int rtx, j, i, err, interleave = 0;
+ int rtx = 0, j, i, err, interleave = 0, port_off;
RTSPStream *rtsp_st;
RTSPMessageHeader reply1, *reply = &reply1;
char cmd[2048];
/* XXX: we assume the same server is used for the control of each
* RTSP stream */
- for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
+ /* 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];
/*
}
/* first try in specified port range */
- if (RTSP_RTP_PORT_MIN != 0) {
- while (j <= RTSP_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) == 0)
- goto rtp_opened;
- }
+ 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");
* 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 &&
- s->streams[rtsp_st->stream_index]->codec->codec_type ==
- AVMEDIA_TYPE_DATA)
+ (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);
"%s/UDP;multicast", trans_pref);
}
if (s->oformat) {
- av_strlcat(transport, ";mode=receive", sizeof(transport));
+ 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));
break;
}
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
- char url[1024], namebuf[50];
+ char url[1024], namebuf[50], optbuf[20] = "";
struct sockaddr_storage addr;
int port, ttl;
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, "?ttl=%d", ttl);
- if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE) < 0) {
+ 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;
}
}
}
- if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
+ if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
goto fail;
}
- if (reply->timeout > 0)
+ if (rt->nb_rtsp_streams && reply->timeout > 0)
rt->timeout = reply->timeout;
if (rt->server_type == RTSP_SERVER_REAL)
{
RTSPState *rt = s->priv_data;
char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
- char *option_list, *option, *filename;
int port, err, tcp_fd;
RTSPMessageHeader reply1 = {0}, *reply = &reply1;
int lower_transport_mask = 0;
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;
if (port < 0)
port = RTSP_DEFAULT_PORT;
-#if FF_API_RTSP_URL_OPTIONS
- /* search for options */
- option_list = strrchr(path, '?');
- if (option_list) {
- /* Strip out the RTSP specific options, write out the rest of
- * the options back into the same string. */
- filename = option_list;
- while (option_list) {
- int handled = 1;
- /* move the option pointer */
- option = ++option_list;
- option_list = strchr(option_list, '&');
- if (option_list)
- *option_list = 0;
-
- /* handle the options */
- if (!strcmp(option, "udp")) {
- lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
- } else if (!strcmp(option, "multicast")) {
- lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
- } else if (!strcmp(option, "tcp")) {
- lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
- } else if(!strcmp(option, "http")) {
- lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
- rt->control_transport = RTSP_MODE_TUNNEL;
- } else if (!strcmp(option, "filter_src")) {
- rt->rtsp_flags |= RTSP_FLAG_FILTER_SRC;
- } else {
- /* Write options back into the buffer, using memmove instead
- * of strcpy since the strings may overlap. */
- int len = strlen(option);
- memmove(++filename, option, len);
- filename += len;
- if (option_list) *filename = '&';
- handled = 0;
- }
- if (handled)
- av_log(s, AV_LOG_WARNING, "Options passed via URL are "
- "deprecated, use -rtsp_transport "
- "and -rtsp_flags instead.\n");
- }
- *filename = 0;
- }
-#endif
-
if (!lower_transport_mask)
lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
av_get_random_seed(), av_get_random_seed());
/* GET requests */
- if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ) < 0) {
+ if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
+ &s->interrupt_callback) < 0) {
err = AVERROR(EIO);
goto fail;
}
"Pragma: no-cache\r\n"
"Cache-Control: no-cache\r\n",
sessioncookie);
- ff_http_set_headers(rt->rtsp_hd, headers);
+ av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
/* complete the connection */
- if (ffurl_connect(rt->rtsp_hd)) {
+ 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) < 0 ) {
+ if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
+ &s->interrupt_callback) < 0 ) {
err = AVERROR(EIO);
goto fail;
}
"Content-Length: 32767\r\n"
"Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
sessioncookie);
- ff_http_set_headers(rt->rtsp_hd_out, headers);
- ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
+ 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
ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
/* complete the connection */
- if (ffurl_connect(rt->rtsp_hd_out)) {
+ 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, NULL);
- if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE) < 0) {
+ if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
+ &s->interrupt_callback, NULL) < 0) {
err = AVERROR(EIO);
goto fail;
}
if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
rt->server_type = RTSP_SERVER_REAL;
continue;
- } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
+ } 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);
struct pollfd *p = rt->p;
for (;;) {
- if (url_interrupt_cb())
+ if (ff_check_interrupt(&s->interrupt_callback))
return AVERROR_EXIT;
if (wait_end && wait_end - av_gettime() < 0)
return AVERROR(EAGAIN);
}
#if CONFIG_RTSP_DEMUXER
if (tcp_fd != -1 && p[0].revents & POLLIN) {
- 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;
+ 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 0;
}
-static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
+static int sdp_read_header(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
RTSPStream *rtsp_st;
if (!ff_network_init())
return AVERROR(EIO);
+ if (s->max_delay < 0) /* Not set by the caller */
+ s->max_delay = DEFAULT_REORDERING_DELAY;
+
/* read the whole sdp file */
/* XXX: better loading */
content = av_malloc(SDP_MAX_SIZE);
namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
ff_url_join(url, sizeof(url), "rtp", NULL,
namebuf, rtsp_st->sdp_port,
- "?localport=%d&ttl=%d", rtsp_st->sdp_port,
- rtsp_st->sdp_ttl);
- if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE) < 0) {
+ "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
+ rtsp_st->sdp_ttl,
+ rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
+ if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
+ &s->interrupt_callback, NULL) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
- if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
+ if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
goto fail;
}
return 0;
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"),
.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 */
return 0;
}
-static int rtp_read_header(AVFormatContext *s,
- AVFormatParameters *ap)
+static int rtp_read_header(AVFormatContext *s)
{
uint8_t recvbuf[1500];
char host[500], sdp[500];
int ret, port;
URLContext* in = NULL;
int payload_type;
- AVCodecContext codec;
+ 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);
+ ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
+ &s->interrupt_callback, NULL);
if (ret)
goto fail;
continue;
}
+ if (RTP_PT_IS_RTCP(recvbuf[1]))
+ continue;
+
payload_type = recvbuf[1] & 0x7f;
break;
}
ffurl_close(in);
in = NULL;
- memset(&codec, 0, sizeof(codec));
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",
/* sdp_read_header initializes this again */
ff_network_close();
- ret = sdp_read_header(s, ap);
+ rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
+
+ ret = sdp_read_header(s);
s->pb = NULL;
return ret;
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 format"),
+ .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,
+ .flags = AVFMT_NOFILE,
+ .priv_class = &rtp_demuxer_class,
};
#endif /* CONFIG_RTP_DEMUXER */
-
projects / ffmpeg / blobdiff