* RTSP/SDP client
* Copyright (c) 2002 Fabrice Bellard
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/base64.h"
#include "libavutil/avstring.h"
#include "libavutil/intreadwrite.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/parseutils.h"
#include "libavutil/random_seed.h"
+#include "libavutil/dict.h"
#include "avformat.h"
+#include "avio_internal.h"
#include <sys/time.h>
-#if HAVE_SYS_SELECT_H
-#include <sys/select.h>
+#if HAVE_POLL_H
+#include <poll.h>
#endif
#include <strings.h>
#include "internal.h"
#include "rtpdec.h"
#include "rdt.h"
#include "rtpdec_formats.h"
+#include "rtpenc_chain.h"
+#include "url.h"
+#include "rtpenc.h"
//#define DEBUG
-//#define DEBUG_RTP_TCP
-#if LIBAVFORMAT_VERSION_INT < (53 << 16)
-int rtsp_default_protocols = (1 << RTSP_LOWER_TRANSPORT_UDP);
-#endif
-
-/* Timeout values for socket select, in ms,
+/* Timeout values for socket poll, in ms,
* and read_packet(), in seconds */
-#define SELECT_TIMEOUT_MS 100
+#define POLL_TIMEOUT_MS 100
#define READ_PACKET_TIMEOUT_S 10
-#define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / SELECT_TIMEOUT_MS
+#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 OFFSET(x) offsetof(RTSPState, x)
+#define DEC AV_OPT_FLAG_DECODING_PARAM
+#define ENC AV_OPT_FLAG_ENCODING_PARAM
+
+#define RTSP_FLAG_OPTS(name, longname) \
+ { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {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" }
+
+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 },
+ FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
+ { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
+ { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
+ { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
+ { "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"),
+ { NULL },
+};
+
+static const AVOption sdp_options[] = {
+ RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
+ { NULL },
+};
+
+static const AVOption rtp_options[] = {
+ RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
+ { NULL },
+};
+
static void get_word_until_chars(char *buf, int buf_size,
const char *sep, const char **pp)
{
get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
}
+/** Parse a string p in the form of Range:npt=xx-xx, and determine the start
+ * and end time.
+ * Used for seeking in the rtp stream.
+ */
+static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
+{
+ char buf[256];
+
+ p += strspn(p, SPACE_CHARS);
+ if (!av_stristart(p, "npt=", &p))
+ return;
+
+ *start = AV_NOPTS_VALUE;
+ *end = AV_NOPTS_VALUE;
+
+ get_word_sep(buf, sizeof(buf), "-", &p);
+ av_parse_time(start, buf, 1);
+ if (*p == '-') {
+ p++;
+ get_word_sep(buf, sizeof(buf), "-", &p);
+ av_parse_time(end, buf, 1);
+ }
+// av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
+// av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
+}
+
+static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
+{
+ struct addrinfo hints, *ai = NULL;
+ memset(&hints, 0, sizeof(hints));
+ hints.ai_flags = AI_NUMERICHOST;
+ if (getaddrinfo(buf, NULL, &hints, &ai))
+ return -1;
+ memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
+ freeaddrinfo(ai);
+ return 0;
+}
+
+#if CONFIG_RTPDEC
+static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
+ RTSPStream *rtsp_st, AVCodecContext *codec)
+{
+ if (!handler)
+ return;
+ codec->codec_id = handler->codec_id;
+ rtsp_st->dynamic_handler = handler;
+ if (handler->alloc)
+ rtsp_st->dynamic_protocol_context = handler->alloc();
+}
+
/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
static int sdp_parse_rtpmap(AVFormatContext *s,
- AVCodecContext *codec, RTSPStream *rtsp_st,
+ AVStream *st, RTSPStream *rtsp_st,
int payload_type, const char *p)
{
+ AVCodecContext *codec = st->codec;
char buf[256];
int i;
AVCodec *c;
* have a trailing space. */
get_word_sep(buf, sizeof(buf), "/ ", &p);
if (payload_type >= RTP_PT_PRIVATE) {
- RTPDynamicProtocolHandler *handler;
- for (handler = RTPFirstDynamicPayloadHandler;
- handler; handler = handler->next) {
- if (!strcasecmp(buf, handler->enc_name) &&
- codec->codec_type == handler->codec_type) {
- codec->codec_id = handler->codec_id;
- rtsp_st->dynamic_handler = handler;
- if (handler->open)
- rtsp_st->dynamic_protocol_context = handler->open();
- break;
- }
- }
+ RTPDynamicProtocolHandler *handler =
+ ff_rtp_handler_find_by_name(buf, codec->codec_type);
+ init_rtp_handler(handler, rtsp_st, codec);
/* If no dynamic handler was found, check with the list of standard
* allocated types, if such a stream for some reason happens to
* use a private payload type. This isn't handled in rtpdec.c, since
codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
if (i > 0) {
codec->sample_rate = i;
+ av_set_pts_info(st, 32, 1, codec->sample_rate);
get_word_sep(buf, sizeof(buf), "/", &p);
i = atoi(buf);
if (i > 0)
break;
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);
break;
default:
break;
return 0;
}
-/** Parse a string p in the form of Range:npt=xx-xx, and determine the start
- * and end time.
- * Used for seeking in the rtp stream.
- */
-static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
-{
- char buf[256];
-
- p += strspn(p, SPACE_CHARS);
- if (!av_stristart(p, "npt=", &p))
- return;
-
- *start = AV_NOPTS_VALUE;
- *end = AV_NOPTS_VALUE;
-
- get_word_sep(buf, sizeof(buf), "-", &p);
- *start = parse_date(buf, 1);
- if (*p == '-') {
- p++;
- get_word_sep(buf, sizeof(buf), "-", &p);
- *end = parse_date(buf, 1);
- }
-// av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
-// av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
-}
-
-static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
-{
- struct addrinfo hints, *ai = NULL;
- memset(&hints, 0, sizeof(hints));
- hints.ai_flags = AI_NUMERICHOST;
- if (getaddrinfo(buf, NULL, &hints, &ai))
- return -1;
- memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
- freeaddrinfo(ai);
- return 0;
-}
-
typedef struct SDPParseState {
/* SDP only */
struct sockaddr_storage default_ip;
struct sockaddr_storage sdp_ip;
int ttl;
- dprintf(s, "sdp: %c='%s'\n", letter, buf);
+ av_dlog(s, "sdp: %c='%s'\n", letter, buf);
p = buf;
if (s1->skip_media && letter != 'm')
s1->default_ip = sdp_ip;
s1->default_ttl = ttl;
} else {
- st = s->streams[s->nb_streams - 1];
- rtsp_st = st->priv_data;
+ rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
rtsp_st->sdp_ip = sdp_ip;
rtsp_st->sdp_ttl = ttl;
}
break;
case 's':
- av_metadata_set2(&s->metadata, "title", p, 0);
+ av_dict_set(&s->metadata, "title", p, 0);
break;
case 'i':
if (s->nb_streams == 0) {
- av_metadata_set2(&s->metadata, "comment", p, 0);
+ av_dict_set(&s->metadata, "comment", p, 0);
break;
}
break;
if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
/* no corresponding stream */
} else {
- st = av_new_stream(s, 0);
+ st = avformat_new_stream(s, NULL);
if (!st)
return;
- st->priv_data = rtsp_st;
+ st->id = rt->nb_rtsp_streams - 1;
rtsp_st->stream_index = st->index;
st->codec->codec_type = codec_type;
if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
+ RTPDynamicProtocolHandler *handler;
/* if standard payload type, we can find the codec right now */
ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
+ if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
+ st->codec->sample_rate > 0)
+ av_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);
}
}
/* put a default control url */
av_strlcpy(rt->control_uri, p,
sizeof(rt->control_uri));
} else {
- char proto[32];
- /* get the control url */
- st = s->streams[s->nb_streams - 1];
- rtsp_st = st->priv_data;
-
- /* XXX: may need to add full url resolution */
- av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
- NULL, NULL, 0, p);
- if (proto[0] == '\0') {
- /* relative control URL */
- if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
- av_strlcat(rtsp_st->control_url, "/",
- sizeof(rtsp_st->control_url));
- av_strlcat(rtsp_st->control_url, p,
- sizeof(rtsp_st->control_url));
- } else
- av_strlcpy(rtsp_st->control_url, p,
- sizeof(rtsp_st->control_url));
+ char proto[32];
+ /* get the control url */
+ rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
+
+ /* XXX: may need to add full url resolution */
+ av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
+ NULL, NULL, 0, p);
+ if (proto[0] == '\0') {
+ /* relative control URL */
+ if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
+ av_strlcat(rtsp_st->control_url, "/",
+ sizeof(rtsp_st->control_url));
+ av_strlcat(rtsp_st->control_url, p,
+ sizeof(rtsp_st->control_url));
+ } else
+ av_strlcpy(rtsp_st->control_url, p,
+ sizeof(rtsp_st->control_url));
}
} else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
/* NOTE: rtpmap is only supported AFTER the 'm=' tag */
get_word(buf1, sizeof(buf1), &p);
payload_type = atoi(buf1);
st = s->streams[s->nb_streams - 1];
- rtsp_st = st->priv_data;
- sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p);
+ rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
+ sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
} else if (av_strstart(p, "fmtp:", &p) ||
av_strstart(p, "framesize:", &p)) {
/* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
// let dynamic protocol handlers have a stab at the line.
get_word(buf1, sizeof(buf1), &p);
payload_type = atoi(buf1);
- for (i = 0; i < s->nb_streams; i++) {
- st = s->streams[i];
- rtsp_st = st->priv_data;
+ for (i = 0; i < rt->nb_rtsp_streams; i++) {
+ rtsp_st = rt->rtsp_streams[i];
if (rtsp_st->sdp_payload_type == payload_type &&
rtsp_st->dynamic_handler &&
rtsp_st->dynamic_handler->parse_sdp_a_line)
} else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
if (atoi(p) == 1)
rt->transport = RTSP_TRANSPORT_RDT;
+ } else if (av_strstart(p, "SampleRate:integer;", &p) &&
+ s->nb_streams > 0) {
+ st = s->streams[s->nb_streams - 1];
+ st->codec->sample_rate = atoi(p);
} else {
if (rt->server_type == RTSP_SERVER_WMS)
ff_wms_parse_sdp_a_line(s, p);
if (rt->server_type == RTSP_SERVER_REAL)
ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
- rtsp_st = s->streams[s->nb_streams - 1]->priv_data;
+ 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,
}
}
-static int sdp_parse(AVFormatContext *s, const char *content)
+int ff_sdp_parse(AVFormatContext *s, const char *content)
{
+ RTSPState *rt = s->priv_data;
const char *p;
int letter;
/* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
if (*p == '\n')
p++;
}
+ rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
+ if (!rt->p) return AVERROR(ENOMEM);
return 0;
}
+#endif /* CONFIG_RTPDEC */
+
+void ff_rtsp_undo_setup(AVFormatContext *s)
+{
+ RTSPState *rt = s->priv_data;
+ int i;
+
+ for (i = 0; i < rt->nb_rtsp_streams; i++) {
+ RTSPStream *rtsp_st = rt->rtsp_streams[i];
+ if (!rtsp_st)
+ continue;
+ if (rtsp_st->transport_priv) {
+ if (s->oformat) {
+ AVFormatContext *rtpctx = rtsp_st->transport_priv;
+ av_write_trailer(rtpctx);
+ if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
+ uint8_t *ptr;
+ avio_close_dyn_buf(rtpctx->pb, &ptr);
+ av_free(ptr);
+ } else {
+ avio_close(rtpctx->pb);
+ }
+ avformat_free_context(rtpctx);
+ } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
+ ff_rdt_parse_close(rtsp_st->transport_priv);
+ else if (CONFIG_RTPDEC)
+ ff_rtp_parse_close(rtsp_st->transport_priv);
+ }
+ rtsp_st->transport_priv = NULL;
+ if (rtsp_st->rtp_handle)
+ ffurl_close(rtsp_st->rtp_handle);
+ rtsp_st->rtp_handle = NULL;
+ }
+}
/* close and free RTSP streams */
void ff_rtsp_close_streams(AVFormatContext *s)
int i;
RTSPStream *rtsp_st;
+ ff_rtsp_undo_setup(s);
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
if (rtsp_st) {
- if (rtsp_st->transport_priv) {
- if (s->oformat) {
- AVFormatContext *rtpctx = rtsp_st->transport_priv;
- av_write_trailer(rtpctx);
- if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
- uint8_t *ptr;
- url_close_dyn_buf(rtpctx->pb, &ptr);
- av_free(ptr);
- } else {
- url_fclose(rtpctx->pb);
- }
- av_metadata_free(&rtpctx->streams[0]->metadata);
- av_metadata_free(&rtpctx->metadata);
- av_free(rtpctx->streams[0]);
- av_free(rtpctx);
- } else if (rt->transport == RTSP_TRANSPORT_RDT)
- ff_rdt_parse_close(rtsp_st->transport_priv);
- else
- rtp_parse_close(rtsp_st->transport_priv);
- }
- if (rtsp_st->rtp_handle)
- url_close(rtsp_st->rtp_handle);
if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
- rtsp_st->dynamic_handler->close(
+ rtsp_st->dynamic_handler->free(
rtsp_st->dynamic_protocol_context);
+ av_free(rtsp_st);
}
}
av_free(rt->rtsp_streams);
av_close_input_stream (rt->asf_ctx);
rt->asf_ctx = NULL;
}
+ av_free(rt->p);
av_free(rt->recvbuf);
}
-static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st,
- URLContext *handle)
-{
- RTSPState *rt = s->priv_data;
- AVFormatContext *rtpctx;
- int ret;
- AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
-
- if (!rtp_format)
- return NULL;
-
- /* Allocate an AVFormatContext for each output stream */
- rtpctx = avformat_alloc_context();
- if (!rtpctx)
- return NULL;
-
- rtpctx->oformat = rtp_format;
- if (!av_new_stream(rtpctx, 0)) {
- av_free(rtpctx);
- return NULL;
- }
- /* Copy the max delay setting; the rtp muxer reads this. */
- rtpctx->max_delay = s->max_delay;
- /* Copy other stream parameters. */
- rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio;
-
- /* Set the synchronized start time. */
- rtpctx->start_time_realtime = rt->start_time;
-
- /* Remove the local codec, link to the original codec
- * context instead, to give the rtp muxer access to
- * codec parameters. */
- av_free(rtpctx->streams[0]->codec);
- rtpctx->streams[0]->codec = st->codec;
-
- if (handle) {
- url_fdopen(&rtpctx->pb, handle);
- } else
- url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
- ret = av_write_header(rtpctx);
-
- if (ret) {
- if (handle) {
- url_fclose(rtpctx->pb);
- } else {
- uint8_t *ptr;
- url_close_dyn_buf(rtpctx->pb, &ptr);
- av_free(ptr);
- }
- av_free(rtpctx->streams[0]);
- av_free(rtpctx);
- return NULL;
- }
-
- /* Copy the RTP AVStream timebase back to the original AVStream */
- st->time_base = rtpctx->streams[0]->time_base;
- return rtpctx;
-}
-
static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
{
RTSPState *rt = s->priv_data;
if (!st)
s->ctx_flags |= AVFMTCTX_NOHEADER;
- if (s->oformat) {
- rtsp_st->transport_priv = rtsp_rtp_mux_open(s, st, rtsp_st->rtp_handle);
+ 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);
/* Ownership of rtp_handle is passed to the rtp mux context */
rtsp_st->rtp_handle = NULL;
- } else if (rt->transport == RTSP_TRANSPORT_RDT)
+ } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
rtsp_st->dynamic_protocol_context,
rtsp_st->dynamic_handler);
- else
- rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
+ else if (CONFIG_RTPDEC)
+ rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
rtsp_st->sdp_payload_type,
(rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
if (!rtsp_st->transport_priv) {
return AVERROR(ENOMEM);
- } else if (rt->transport != RTSP_TRANSPORT_RDT) {
+ } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
if (rtsp_st->dynamic_handler) {
- rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
- rtsp_st->dynamic_protocol_context,
- rtsp_st->dynamic_handler);
+ ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
+ rtsp_st->dynamic_protocol_context,
+ rtsp_st->dynamic_handler);
}
}
}
#if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
-static int rtsp_probe(AVProbeData *p)
-{
- if (av_strstart(p->filename, "rtsp:", NULL))
- return AVPROBE_SCORE_MAX;
- return 0;
-}
-
static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
{
const char *p;
}
}
+static void handle_rtp_info(RTSPState *rt, const char *url,
+ uint32_t seq, uint32_t rtptime)
+{
+ int i;
+ if (!rtptime || !url[0])
+ return;
+ if (rt->transport != RTSP_TRANSPORT_RTP)
+ return;
+ for (i = 0; i < rt->nb_rtsp_streams; i++) {
+ RTSPStream *rtsp_st = rt->rtsp_streams[i];
+ RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
+ if (!rtpctx)
+ continue;
+ if (!strcmp(rtsp_st->control_url, url)) {
+ rtpctx->base_timestamp = rtptime;
+ break;
+ }
+ }
+}
+
+static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
+{
+ int read = 0;
+ char key[20], value[1024], url[1024] = "";
+ uint32_t seq = 0, rtptime = 0;
+
+ for (;;) {
+ p += strspn(p, SPACE_CHARS);
+ if (!*p)
+ break;
+ get_word_sep(key, sizeof(key), "=", &p);
+ if (*p != '=')
+ break;
+ p++;
+ get_word_sep(value, sizeof(value), ";, ", &p);
+ read++;
+ if (!strcmp(key, "url"))
+ av_strlcpy(url, value, sizeof(url));
+ else if (!strcmp(key, "seq"))
+ seq = strtoul(value, NULL, 10);
+ else if (!strcmp(key, "rtptime"))
+ rtptime = strtoul(value, NULL, 10);
+ if (*p == ',') {
+ handle_rtp_info(rt, url, seq, rtptime);
+ url[0] = '\0';
+ seq = rtptime = 0;
+ read = 0;
+ }
+ if (*p)
+ p++;
+ }
+ if (read > 0)
+ handle_rtp_info(rt, url, seq, rtptime);
+}
+
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
- HTTPAuthState *auth_state)
+ RTSPState *rt, const char *method)
{
const char *p;
} else if (av_stristart(p, "Location:", &p)) {
p += strspn(p, SPACE_CHARS);
av_strlcpy(reply->location, p , sizeof(reply->location));
- } else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) {
+ } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
+ p += strspn(p, SPACE_CHARS);
+ ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
+ } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
p += strspn(p, SPACE_CHARS);
- ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p);
- } else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) {
+ ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
+ } else if (av_stristart(p, "Content-Base:", &p) && rt) {
p += strspn(p, SPACE_CHARS);
- ff_http_auth_handle_header(auth_state, "Authentication-Info", p);
+ if (method && !strcmp(method, "DESCRIBE"))
+ av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
+ } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
+ p += strspn(p, SPACE_CHARS);
+ if (method && !strcmp(method, "PLAY"))
+ rtsp_parse_rtp_info(rt, p);
+ } else if (av_stristart(p, "Public:", &p) && rt) {
+ if (strstr(p, "GET_PARAMETER") &&
+ method && !strcmp(method, "OPTIONS"))
+ rt->get_parameter_supported = 1;
+ } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
+ p += strspn(p, SPACE_CHARS);
+ rt->accept_dynamic_rate = atoi(p);
}
}
int ret, len, len1;
uint8_t buf[1024];
- ret = url_read_complete(rt->rtsp_hd, buf, 3);
+ ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
if (ret != 3)
return;
len = AV_RB16(buf + 1);
- dprintf(s, "skipping RTP packet len=%d\n", len);
+ av_dlog(s, "skipping RTP packet len=%d\n", len);
/* skip payload */
while (len > 0) {
len1 = len;
if (len1 > sizeof(buf))
len1 = sizeof(buf);
- ret = url_read_complete(rt->rtsp_hd, buf, len1);
+ ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
if (ret != len1)
return;
len -= len1;
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
unsigned char **content_ptr,
- int return_on_interleaved_data)
+ int return_on_interleaved_data, const char *method)
{
RTSPState *rt = s->priv_data;
char buf[4096], buf1[1024], *q;
for (;;) {
q = buf;
for (;;) {
- ret = url_read_complete(rt->rtsp_hd, &ch, 1);
-#ifdef DEBUG_RTP_TCP
- dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
-#endif
+ ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
+ av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
if (ret != 1)
return AVERROR_EOF;
if (ch == '\n')
}
*q = '\0';
- dprintf(s, "line='%s'\n", buf);
+ av_dlog(s, "line='%s'\n", buf);
/* test if last line */
if (buf[0] == '\0')
reply->status_code = atoi(buf1);
av_strlcpy(reply->reason, p, sizeof(reply->reason));
} else {
- ff_rtsp_parse_line(reply, p, &rt->auth_state);
+ ff_rtsp_parse_line(reply, p, rt, method);
av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
}
if (content_length > 0) {
/* leave some room for a trailing '\0' (useful for simple parsing) */
content = av_malloc(content_length + 1);
- (void)url_read_complete(rt->rtsp_hd, content, content_length);
+ ffurl_read_complete(rt->rtsp_hd, content, content_length);
content[content_length] = '\0';
}
if (content_ptr)
return 0;
}
-int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
- const char *method, const char *url,
- const char *headers,
- const unsigned char *send_content,
- int send_content_length)
+/**
+ * Send a command to the RTSP server without waiting for the reply.
+ *
+ * @param s RTSP (de)muxer context
+ * @param method the method for the request
+ * @param url the target url for the request
+ * @param headers extra header lines to include in the request
+ * @param send_content if non-null, the data to send as request body content
+ * @param send_content_length the length of the send_content data, or 0 if
+ * send_content is null
+ *
+ * @return zero if success, nonzero otherwise
+ */
+static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
+ const char *method, const char *url,
+ const char *headers,
+ const unsigned char *send_content,
+ int send_content_length)
{
RTSPState *rt = s->priv_data;
char buf[4096], *out_buf;
out_buf = base64buf;
}
- dprintf(s, "Sending:\n%s--\n", buf);
+ av_dlog(s, "Sending:\n%s--\n", buf);
- url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
+ ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
if (send_content_length > 0 && send_content) {
if (rt->control_transport == RTSP_MODE_TUNNEL) {
av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
"with content data not supported\n");
return AVERROR_PATCHWELCOME;
}
- url_write(rt->rtsp_hd_out, send_content, send_content_length);
+ ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
}
rt->last_cmd_time = av_gettime();
send_content_length)))
return ret;
- if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0) ) < 0)
+ if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
return ret;
if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
return 0;
}
-/**
- * @return 0 on success, <0 on error, 1 if protocol is unavailable.
- */
-static int make_setup_request(AVFormatContext *s, const char *host, int port,
+int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
int lower_transport, const char *real_challenge)
{
RTSPState *rt = s->priv_data;
for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
char transport[2048];
- /**
+ /*
* WMS serves all UDP data over a single connection, the RTX, which
* isn't necessarily the first in the SDP but has to be the first
* to be set up, else the second/third SETUP will fail with a 461.
"?localport=%d", j);
/* we will use two ports per rtp stream (rtp and rtcp) */
j += 2;
- if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
+ if (ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE) == 0)
goto rtp_opened;
}
}
-#if 0
- /* then try on any port */
- if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
- err = AVERROR_INVALIDDATA;
- goto fail;
- }
-#endif
+ av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
+ err = AVERROR(EIO);
+ goto fail;
rtp_opened:
- port = rtp_get_local_port(rtsp_st->rtp_handle);
+ port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
have_port:
snprintf(transport, sizeof(transport) - 1,
"%s/UDP;", trans_pref);
/* RTP/TCP */
else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
- /** For WMS streams, the application streams are only used for
+ /* For WMS streams, the application streams are only used for
* UDP. When trying to set it up for TCP streams, the server
* will return an error. Therefore, we skip those streams. */
if (rt->server_type == RTSP_SERVER_WMS &&
continue;
snprintf(transport, sizeof(transport) - 1,
"%s/TCP;", trans_pref);
- if (rt->server_type == RTSP_SERVER_WMS)
+ if (rt->transport != RTSP_TRANSPORT_RDT)
av_strlcat(transport, "unicast;", sizeof(transport));
av_strlcatf(transport, sizeof(transport),
"interleaved=%d-%d",
snprintf(cmd, sizeof(cmd),
"Transport: %s\r\n",
transport);
- if (i == 0 && rt->server_type == RTSP_SERVER_REAL) {
+ if (rt->accept_dynamic_rate)
+ av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
+ if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
char real_res[41], real_csum[9];
ff_rdt_calc_response_and_checksum(real_res, real_csum,
real_challenge);
rt->transport = reply->transports[0].transport;
}
- /* close RTP connection if not chosen */
- if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&
- (lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {
- url_close(rtsp_st->rtp_handle);
- rtsp_st->rtp_handle = NULL;
+ /* Fail if the server responded with another lower transport mode
+ * than what we requested. */
+ if (reply->transports[0].lower_transport != lower_transport) {
+ av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
+ err = AVERROR_INVALIDDATA;
+ goto fail;
}
switch(reply->transports[0].lower_transport) {
break;
case RTSP_LOWER_TRANSPORT_UDP: {
- char url[1024];
+ char url[1024], options[30] = "";
+ if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
+ av_strlcpy(options, "?connect=1", sizeof(options));
/* Use source address if specified */
if (reply->transports[0].source[0]) {
ff_url_join(url, sizeof(url), "rtp", NULL,
reply->transports[0].source,
- reply->transports[0].server_port_min, NULL);
+ reply->transports[0].server_port_min, "%s", options);
} else {
ff_url_join(url, sizeof(url), "rtp", NULL, host,
- reply->transports[0].server_port_min, NULL);
+ reply->transports[0].server_port_min, "%s", options);
}
if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
- rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
+ ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
* potential NAT router by sending dummy packets.
* RTP/RTCP dummy packets are used for RDT, too.
*/
- if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat)
- rtp_send_punch_packets(rtsp_st->rtp_handle);
+ if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
+ CONFIG_RTPDEC)
+ ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
break;
}
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
port, "?ttl=%d", ttl);
- if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
+ if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
return 0;
fail:
- for (i = 0; i < rt->nb_rtsp_streams; i++) {
- if (rt->rtsp_streams[i]->rtp_handle) {
- url_close(rt->rtsp_streams[i]->rtp_handle);
- rt->rtsp_streams[i]->rtp_handle = NULL;
- }
- }
+ ff_rtsp_undo_setup(s);
return err;
}
-static int rtsp_read_play(AVFormatContext *s)
-{
- RTSPState *rt = s->priv_data;
- RTSPMessageHeader reply1, *reply = &reply1;
- int i;
- char cmd[1024];
-
- av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
- rt->nb_byes = 0;
-
- if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
- if (rt->state == RTSP_STATE_PAUSED) {
- cmd[0] = 0;
- } else {
- snprintf(cmd, sizeof(cmd),
- "Range: npt=%0.3f-\r\n",
- (double)rt->seek_timestamp / AV_TIME_BASE);
- }
- ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL);
- if (reply->status_code != RTSP_STATUS_OK) {
- return -1;
- }
- if (rt->transport == RTSP_TRANSPORT_RTP) {
- for (i = 0; i < rt->nb_rtsp_streams; i++) {
- RTSPStream *rtsp_st = rt->rtsp_streams[i];
- RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
- AVStream *st = NULL;
- if (!rtpctx)
- continue;
- if (rtsp_st->stream_index >= 0)
- st = s->streams[rtsp_st->stream_index];
- ff_rtp_reset_packet_queue(rtpctx);
- if (reply->range_start != AV_NOPTS_VALUE) {
- rtpctx->last_rtcp_ntp_time = AV_NOPTS_VALUE;
- rtpctx->first_rtcp_ntp_time = AV_NOPTS_VALUE;
- if (st)
- rtpctx->range_start_offset =
- av_rescale_q(reply->range_start, AV_TIME_BASE_Q,
- st->time_base);
- }
- }
- }
- }
- rt->state = RTSP_STATE_STREAMING;
- return 0;
-}
-
-static int rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
-{
- RTSPState *rt = s->priv_data;
- char cmd[1024];
- unsigned char *content = NULL;
- int ret;
-
- /* describe the stream */
- snprintf(cmd, sizeof(cmd),
- "Accept: application/sdp\r\n");
- if (rt->server_type == RTSP_SERVER_REAL) {
- /**
- * The Require: attribute is needed for proper streaming from
- * Realmedia servers.
- */
- av_strlcat(cmd,
- "Require: com.real.retain-entity-for-setup\r\n",
- sizeof(cmd));
- }
- ff_rtsp_send_cmd(s, "DESCRIBE", rt->control_uri, cmd, reply, &content);
- if (!content)
- return AVERROR_INVALIDDATA;
- if (reply->status_code != RTSP_STATUS_OK) {
- av_freep(&content);
- return AVERROR_INVALIDDATA;
- }
-
- /* now we got the SDP description, we parse it */
- ret = sdp_parse(s, (const char *)content);
- av_freep(&content);
- if (ret < 0)
- return AVERROR_INVALIDDATA;
-
- return 0;
-}
-
-static int rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
-{
- RTSPState *rt = s->priv_data;
- RTSPMessageHeader reply1, *reply = &reply1;
- int i;
- char *sdp;
- AVFormatContext sdp_ctx, *ctx_array[1];
-
- rt->start_time = av_gettime();
-
- /* Announce the stream */
- sdp = av_mallocz(SDP_MAX_SIZE);
- if (sdp == NULL)
- return AVERROR(ENOMEM);
- /* We create the SDP based on the RTSP AVFormatContext where we
- * aren't allowed to change the filename field. (We create the SDP
- * based on the RTSP context since the contexts for the RTP streams
- * don't exist yet.) In order to specify a custom URL with the actual
- * peer IP instead of the originally specified hostname, we create
- * a temporary copy of the AVFormatContext, where the custom URL is set.
- *
- * FIXME: Create the SDP without copying the AVFormatContext.
- * This either requires setting up the RTP stream AVFormatContexts
- * already here (complicating things immensely) or getting a more
- * flexible SDP creation interface.
- */
- sdp_ctx = *s;
- ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
- "rtsp", NULL, addr, -1, NULL);
- ctx_array[0] = &sdp_ctx;
- if (avf_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
- av_free(sdp);
- return AVERROR_INVALIDDATA;
- }
- av_log(s, AV_LOG_INFO, "SDP:\n%s\n", sdp);
- ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
- "Content-Type: application/sdp\r\n",
- reply, NULL, sdp, strlen(sdp));
- av_free(sdp);
- if (reply->status_code != RTSP_STATUS_OK)
- return AVERROR_INVALIDDATA;
-
- /* Set up the RTSPStreams for each AVStream */
- for (i = 0; i < s->nb_streams; i++) {
- RTSPStream *rtsp_st;
- AVStream *st = s->streams[i];
-
- rtsp_st = av_mallocz(sizeof(RTSPStream));
- if (!rtsp_st)
- return AVERROR(ENOMEM);
- dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
-
- st->priv_data = rtsp_st;
- rtsp_st->stream_index = i;
-
- av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
- /* Note, this must match the relative uri set in the sdp content */
- av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
- "/streamid=%d", i);
- }
-
- return 0;
-}
-
void ff_rtsp_close_connections(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
- if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
- url_close(rt->rtsp_hd);
+ if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
+ ffurl_close(rt->rtsp_hd);
rt->rtsp_hd = rt->rtsp_hd_out = NULL;
}
int port, err, tcp_fd;
RTSPMessageHeader reply1 = {0}, *reply = &reply1;
int lower_transport_mask = 0;
- char real_challenge[64];
+ char real_challenge[64] = "";
struct sockaddr_storage peer;
socklen_t peer_len = sizeof(peer);
if (!ff_network_init())
return AVERROR(EIO);
-redirect:
+
rt->control_transport = RTSP_MODE_PLAIN;
+ if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
+ rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
+ rt->control_transport = RTSP_MODE_TUNNEL;
+ }
+ /* Only pass through valid flags from here */
+ rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
+
+redirect:
+ lower_transport_mask = rt->lower_transport_mask;
/* extract hostname and port */
av_url_split(NULL, 0, auth, sizeof(auth),
host, sizeof(host), &port, path, sizeof(path), s->filename);
if (port < 0)
port = RTSP_DEFAULT_PORT;
+#if FF_API_RTSP_URL_OPTIONS
/* search for options */
option_list = strrchr(path, '?');
if (option_list) {
* 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, '&');
} 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. */
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 (url_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) {
+ if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ) < 0) {
err = AVERROR(EIO);
goto fail;
}
ff_http_set_headers(rt->rtsp_hd, headers);
/* complete the connection */
- if (url_connect(rt->rtsp_hd)) {
+ if (ffurl_connect(rt->rtsp_hd)) {
err = AVERROR(EIO);
goto fail;
}
/* POST requests */
- if (url_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
+ if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE) < 0 ) {
err = AVERROR(EIO);
goto fail;
}
ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
/* complete the connection */
- if (url_connect(rt->rtsp_hd_out)) {
+ if (ffurl_connect(rt->rtsp_hd_out)) {
err = AVERROR(EIO);
goto fail;
}
} else {
/* open the tcp connection */
ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
- if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
+ if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE) < 0) {
err = AVERROR(EIO);
goto fail;
}
}
rt->seq = 0;
- tcp_fd = url_get_file_handle(rt->rtsp_hd);
+ tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
NULL, 0, NI_NUMERICHOST);
cmd[0] = 0;
if (rt->server_type == RTSP_SERVER_REAL)
av_strlcat(cmd,
- /**
+ /*
* The following entries are required for proper
* streaming from a Realmedia server. They are
* interdependent in some way although we currently
* don't quite understand how. Values were copied
* from mplayer SVN r23589.
- * @param CompanyID is a 16-byte ID in base64
- * @param ClientChallenge is a 16-byte ID in hex
+ * ClientChallenge is a 16-byte ID in hex
+ * CompanyID is a 16-byte ID in base64
*/
"ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
"PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
break;
}
- if (s->iformat)
- err = rtsp_setup_input_streams(s, reply);
- else
- err = rtsp_setup_output_streams(s, host);
+ if (s->iformat && CONFIG_RTSP_DEMUXER)
+ err = ff_rtsp_setup_input_streams(s, reply);
+ else if (CONFIG_RTSP_MUXER)
+ err = ff_rtsp_setup_output_streams(s, host);
if (err)
goto fail;
int lower_transport = ff_log2_tab[lower_transport_mask &
~(lower_transport_mask - 1)];
- err = make_setup_request(s, host, port, lower_transport,
+ err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
rt->server_type == RTSP_SERVER_REAL ?
real_challenge : NULL);
if (err < 0)
goto fail;
lower_transport_mask &= ~(1 << lower_transport);
if (lower_transport_mask == 0 && err == 1) {
- err = FF_NETERROR(EPROTONOSUPPORT);
+ err = AVERROR(EPROTONOSUPPORT);
goto fail;
}
} while (err);
+ rt->lower_transport_mask = lower_transport_mask;
+ av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
rt->state = RTSP_STATE_IDLE;
rt->seek_timestamp = 0; /* default is to start stream at position zero */
return 0;
ff_network_close();
return err;
}
-#endif
-
-#if CONFIG_RTSP_DEMUXER
-static int rtsp_read_header(AVFormatContext *s,
- AVFormatParameters *ap)
-{
- RTSPState *rt = s->priv_data;
- int ret;
-
- ret = ff_rtsp_connect(s);
- if (ret)
- return ret;
-
- rt->real_setup_cache = av_mallocz(2 * s->nb_streams * sizeof(*rt->real_setup_cache));
- if (!rt->real_setup_cache)
- return AVERROR(ENOMEM);
- rt->real_setup = rt->real_setup_cache + s->nb_streams * sizeof(*rt->real_setup);
-
- if (ap->initial_pause) {
- /* do not start immediately */
- } else {
- if (rtsp_read_play(s) < 0) {
- ff_rtsp_close_streams(s);
- ff_rtsp_close_connections(s);
- return AVERROR_INVALIDDATA;
- }
- }
-
- return 0;
-}
+#endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
+#if CONFIG_RTPDEC
static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
- uint8_t *buf, int buf_size)
+ uint8_t *buf, int buf_size, int64_t wait_end)
{
RTSPState *rt = s->priv_data;
RTSPStream *rtsp_st;
- fd_set rfds;
- int fd, fd_rtcp, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
- struct timeval tv;
+ int n, i, ret, tcp_fd, timeout_cnt = 0;
+ int max_p = 0;
+ struct pollfd *p = rt->p;
for (;;) {
if (url_interrupt_cb())
- return AVERROR(EINTR);
- FD_ZERO(&rfds);
+ return AVERROR_EXIT;
+ if (wait_end && wait_end - av_gettime() < 0)
+ return AVERROR(EAGAIN);
+ max_p = 0;
if (rt->rtsp_hd) {
- tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
- FD_SET(tcp_fd, &rfds);
+ tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
+ p[max_p].fd = tcp_fd;
+ p[max_p++].events = POLLIN;
} else {
- fd_max = 0;
tcp_fd = -1;
}
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
if (rtsp_st->rtp_handle) {
- fd = url_get_file_handle(rtsp_st->rtp_handle);
- fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
- if (FFMAX(fd, fd_rtcp) > fd_max)
- fd_max = FFMAX(fd, fd_rtcp);
- FD_SET(fd, &rfds);
- FD_SET(fd_rtcp, &rfds);
+ p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
+ p[max_p++].events = POLLIN;
+ p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
+ p[max_p++].events = POLLIN;
}
}
- tv.tv_sec = 0;
- tv.tv_usec = SELECT_TIMEOUT_MS * 1000;
- n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
+ n = poll(p, max_p, POLL_TIMEOUT_MS);
if (n > 0) {
+ int j = 1 - (tcp_fd == -1);
timeout_cnt = 0;
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
if (rtsp_st->rtp_handle) {
- fd = url_get_file_handle(rtsp_st->rtp_handle);
- fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
- if (FD_ISSET(fd_rtcp, &rfds) || FD_ISSET(fd, &rfds)) {
- ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
+ if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
+ ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
if (ret > 0) {
*prtsp_st = rtsp_st;
return ret;
}
}
+ j+=2;
}
}
#if CONFIG_RTSP_DEMUXER
- if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
+ if (tcp_fd != -1 && p[0].revents & POLLIN) {
RTSPMessageHeader reply;
- ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
+ ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
if (ret < 0)
return ret;
/* XXX: parse message */
}
#endif
} else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
- return FF_NETERROR(ETIMEDOUT);
+ return AVERROR(ETIMEDOUT);
} else if (n < 0 && errno != EINTR)
return AVERROR(errno);
}
}
-static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
- uint8_t *buf, int buf_size)
-{
- RTSPState *rt = s->priv_data;
- int id, len, i, ret;
- RTSPStream *rtsp_st;
-
-#ifdef DEBUG_RTP_TCP
- dprintf(s, "tcp_read_packet:\n");
-#endif
-redo:
- for (;;) {
- RTSPMessageHeader reply;
-
- ret = ff_rtsp_read_reply(s, &reply, NULL, 1);
- if (ret < 0)
- return ret;
- if (ret == 1) /* received '$' */
- break;
- /* XXX: parse message */
- if (rt->state != RTSP_STATE_STREAMING)
- return 0;
- }
- ret = url_read_complete(rt->rtsp_hd, buf, 3);
- if (ret != 3)
- return -1;
- id = buf[0];
- len = AV_RB16(buf + 1);
-#ifdef DEBUG_RTP_TCP
- dprintf(s, "id=%d len=%d\n", id, len);
-#endif
- if (len > buf_size || len < 12)
- goto redo;
- /* get the data */
- ret = url_read_complete(rt->rtsp_hd, buf, len);
- if (ret != len)
- return -1;
- if (rt->transport == RTSP_TRANSPORT_RDT &&
- ff_rdt_parse_header(buf, len, &id, NULL, NULL, NULL, NULL) < 0)
- return -1;
-
- /* find the matching stream */
- for (i = 0; i < rt->nb_rtsp_streams; i++) {
- rtsp_st = rt->rtsp_streams[i];
- if (id >= rtsp_st->interleaved_min &&
- id <= rtsp_st->interleaved_max)
- goto found;
- }
- goto redo;
-found:
- *prtsp_st = rtsp_st;
- return len;
-}
-
-static int rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
+int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
{
RTSPState *rt = s->priv_data;
int ret, len;
- RTSPStream *rtsp_st;
+ RTSPStream *rtsp_st, *first_queue_st = NULL;
+ int64_t wait_end = 0;
if (rt->nb_byes == rt->nb_rtsp_streams)
return AVERROR_EOF;
if (rt->transport == RTSP_TRANSPORT_RDT) {
ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
} else
- ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
+ ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
if (ret == 0) {
rt->cur_transport_priv = NULL;
return 0;
rt->cur_transport_priv = NULL;
}
+ if (rt->transport == RTSP_TRANSPORT_RTP) {
+ int i;
+ int64_t first_queue_time = 0;
+ for (i = 0; i < rt->nb_rtsp_streams; i++) {
+ RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
+ int64_t queue_time;
+ if (!rtpctx)
+ continue;
+ queue_time = ff_rtp_queued_packet_time(rtpctx);
+ if (queue_time && (queue_time - first_queue_time < 0 ||
+ !first_queue_time)) {
+ first_queue_time = queue_time;
+ first_queue_st = rt->rtsp_streams[i];
+ }
+ }
+ if (first_queue_time)
+ wait_end = first_queue_time + s->max_delay;
+ }
+
/* read next RTP packet */
redo:
if (!rt->recvbuf) {
default:
#if CONFIG_RTSP_DEMUXER
case RTSP_LOWER_TRANSPORT_TCP:
- len = tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
+ len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
break;
#endif
case RTSP_LOWER_TRANSPORT_UDP:
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
- len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
- if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
- rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
+ len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
+ if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
+ ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
break;
}
+ if (len == AVERROR(EAGAIN) && first_queue_st &&
+ rt->transport == RTSP_TRANSPORT_RTP) {
+ rtsp_st = first_queue_st;
+ ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
+ goto end;
+ }
if (len < 0)
return len;
if (len == 0)
if (rt->transport == RTSP_TRANSPORT_RDT) {
ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
} else {
- ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
+ ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
if (ret < 0) {
/* Either bad packet, or a RTCP packet. Check if the
* first_rtcp_ntp_time field was initialized. */
* in order to map their timestamp origin to the same ntp time
* as this one. */
int i;
+ AVStream *st = NULL;
+ if (rtsp_st->stream_index >= 0)
+ st = s->streams[rtsp_st->stream_index];
for (i = 0; i < rt->nb_rtsp_streams; i++) {
RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
- if (rtpctx2 &&
- rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE)
+ AVStream *st2 = NULL;
+ if (rt->rtsp_streams[i]->stream_index >= 0)
+ st2 = s->streams[rt->rtsp_streams[i]->stream_index];
+ if (rtpctx2 && st && st2 &&
+ rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
+ rtpctx2->rtcp_ts_offset = av_rescale_q(
+ rtpctx->rtcp_ts_offset, st->time_base,
+ st2->time_base);
+ }
}
}
if (ret == -RTCP_BYE) {
}
}
}
+end:
if (ret < 0)
goto redo;
if (ret == 1)
return ret;
}
+#endif /* CONFIG_RTPDEC */
-static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt)
-{
- RTSPState *rt = s->priv_data;
- int ret;
- RTSPMessageHeader reply1, *reply = &reply1;
- char cmd[1024];
-
- if (rt->server_type == RTSP_SERVER_REAL) {
- int i;
-
- for (i = 0; i < s->nb_streams; i++)
- rt->real_setup[i] = s->streams[i]->discard;
-
- if (!rt->need_subscription) {
- if (memcmp (rt->real_setup, rt->real_setup_cache,
- sizeof(enum AVDiscard) * s->nb_streams)) {
- snprintf(cmd, sizeof(cmd),
- "Unsubscribe: %s\r\n",
- rt->last_subscription);
- ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
- cmd, reply, NULL);
- if (reply->status_code != RTSP_STATUS_OK)
- return AVERROR_INVALIDDATA;
- rt->need_subscription = 1;
- }
- }
-
- if (rt->need_subscription) {
- int r, rule_nr, first = 1;
-
- memcpy(rt->real_setup_cache, rt->real_setup,
- sizeof(enum AVDiscard) * s->nb_streams);
- rt->last_subscription[0] = 0;
-
- snprintf(cmd, sizeof(cmd),
- "Subscribe: ");
- for (i = 0; i < rt->nb_rtsp_streams; i++) {
- rule_nr = 0;
- for (r = 0; r < s->nb_streams; r++) {
- if (s->streams[r]->priv_data == rt->rtsp_streams[i]) {
- if (s->streams[r]->discard != AVDISCARD_ALL) {
- if (!first)
- av_strlcat(rt->last_subscription, ",",
- sizeof(rt->last_subscription));
- ff_rdt_subscribe_rule(
- rt->last_subscription,
- sizeof(rt->last_subscription), i, rule_nr);
- first = 0;
- }
- rule_nr++;
- }
- }
- }
- av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription);
- ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
- cmd, reply, NULL);
- if (reply->status_code != RTSP_STATUS_OK)
- return AVERROR_INVALIDDATA;
- rt->need_subscription = 0;
-
- if (rt->state == RTSP_STATE_STREAMING)
- rtsp_read_play (s);
- }
- }
-
- ret = rtsp_fetch_packet(s, pkt);
- if (ret < 0)
- return ret;
-
- /* send dummy request to keep TCP connection alive */
- if ((av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
- if (rt->server_type == RTSP_SERVER_WMS) {
- ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
- } else {
- ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL);
- }
- }
-
- return 0;
-}
-
-/* pause the stream */
-static int rtsp_read_pause(AVFormatContext *s)
-{
- RTSPState *rt = s->priv_data;
- RTSPMessageHeader reply1, *reply = &reply1;
-
- if (rt->state != RTSP_STATE_STREAMING)
- return 0;
- else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
- ff_rtsp_send_cmd(s, "PAUSE", rt->control_uri, NULL, reply, NULL);
- if (reply->status_code != RTSP_STATUS_OK) {
- return -1;
- }
- }
- rt->state = RTSP_STATE_PAUSED;
- return 0;
-}
-
-static int rtsp_read_seek(AVFormatContext *s, int stream_index,
- int64_t timestamp, int flags)
-{
- RTSPState *rt = s->priv_data;
-
- rt->seek_timestamp = av_rescale_q(timestamp,
- s->streams[stream_index]->time_base,
- AV_TIME_BASE_Q);
- switch(rt->state) {
- default:
- case RTSP_STATE_IDLE:
- break;
- case RTSP_STATE_STREAMING:
- if (rtsp_read_pause(s) != 0)
- return -1;
- rt->state = RTSP_STATE_SEEKING;
- if (rtsp_read_play(s) != 0)
- return -1;
- break;
- case RTSP_STATE_PAUSED:
- rt->state = RTSP_STATE_IDLE;
- break;
- }
- return 0;
-}
-
-static int rtsp_read_close(AVFormatContext *s)
-{
- RTSPState *rt = s->priv_data;
-
-#if 0
- /* NOTE: it is valid to flush the buffer here */
- if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
- url_fclose(&rt->rtsp_gb);
- }
-#endif
- ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
-
- ff_rtsp_close_streams(s);
- ff_rtsp_close_connections(s);
- ff_network_close();
- rt->real_setup = NULL;
- av_freep(&rt->real_setup_cache);
- return 0;
-}
-
-AVInputFormat rtsp_demuxer = {
- "rtsp",
- NULL_IF_CONFIG_SMALL("RTSP input format"),
- sizeof(RTSPState),
- rtsp_probe,
- rtsp_read_header,
- rtsp_read_packet,
- rtsp_read_close,
- rtsp_read_seek,
- .flags = AVFMT_NOFILE,
- .read_play = rtsp_read_play,
- .read_pause = rtsp_read_pause,
-};
-#endif
-
+#if CONFIG_SDP_DEMUXER
static int sdp_probe(AVProbeData *p1)
{
const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
/* read the whole sdp file */
/* XXX: better loading */
content = av_malloc(SDP_MAX_SIZE);
- size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
+ size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
if (size <= 0) {
av_free(content);
return AVERROR_INVALIDDATA;
}
content[size] ='\0';
- sdp_parse(s, content);
+ err = ff_sdp_parse(s, content);
av_free(content);
+ if (err) goto fail;
/* open each RTP stream */
for (i = 0; i < rt->nb_rtsp_streams; i++) {
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 (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 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) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
return 0;
}
-AVInputFormat sdp_demuxer = {
- "sdp",
- NULL_IF_CONFIG_SMALL("SDP"),
- sizeof(RTSPState),
- sdp_probe,
- sdp_read_header,
- rtsp_fetch_packet,
- sdp_read_close,
+static const AVClass sdp_demuxer_class = {
+ .class_name = "SDP demuxer",
+ .item_name = av_default_item_name,
+ .option = sdp_options,
+ .version = LIBAVUTIL_VERSION_INT,
};
+
+AVInputFormat ff_sdp_demuxer = {
+ .name = "sdp",
+ .long_name = NULL_IF_CONFIG_SMALL("SDP"),
+ .priv_data_size = sizeof(RTSPState),
+ .read_probe = sdp_probe,
+ .read_header = sdp_read_header,
+ .read_packet = ff_rtsp_fetch_packet,
+ .read_close = sdp_read_close,
+ .priv_class = &sdp_demuxer_class
+};
+#endif /* CONFIG_SDP_DEMUXER */
+
+#if CONFIG_RTP_DEMUXER
+static int rtp_probe(AVProbeData *p)
+{
+ if (av_strstart(p->filename, "rtp:", NULL))
+ return AVPROBE_SCORE_MAX;
+ return 0;
+}
+
+static int rtp_read_header(AVFormatContext *s,
+ AVFormatParameters *ap)
+{
+ uint8_t recvbuf[1500];
+ char host[500], sdp[500];
+ int ret, port;
+ URLContext* in = NULL;
+ int payload_type;
+ AVCodecContext codec;
+ struct sockaddr_storage addr;
+ AVIOContext pb;
+ socklen_t addrlen = sizeof(addr);
+
+ if (!ff_network_init())
+ return AVERROR(EIO);
+
+ ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ);
+ if (ret)
+ goto fail;
+
+ while (1) {
+ ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
+ if (ret == AVERROR(EAGAIN))
+ continue;
+ if (ret < 0)
+ goto fail;
+ if (ret < 12) {
+ av_log(s, AV_LOG_WARNING, "Received too short packet\n");
+ continue;
+ }
+
+ if ((recvbuf[0] & 0xc0) != 0x80) {
+ av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
+ "received\n");
+ continue;
+ }
+
+ payload_type = recvbuf[1] & 0x7f;
+ break;
+ }
+ getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
+ 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",
+ payload_type);
+ goto fail;
+ }
+ if (codec.codec_type != AVMEDIA_TYPE_DATA) {
+ av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
+ "properly you need an SDP file "
+ "describing it\n");
+ }
+
+ av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
+ NULL, 0, s->filename);
+
+ snprintf(sdp, sizeof(sdp),
+ "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
+ addr.ss_family == AF_INET ? 4 : 6, host,
+ codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
+ codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
+ port, payload_type);
+ av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
+
+ ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
+ s->pb = &pb;
+
+ /* sdp_read_header initializes this again */
+ ff_network_close();
+
+ ret = sdp_read_header(s, ap);
+ s->pb = NULL;
+ return ret;
+
+fail:
+ if (in)
+ ffurl_close(in);
+ ff_network_close();
+ return ret;
+}
+
+static const AVClass rtp_demuxer_class = {
+ .class_name = "RTP demuxer",
+ .item_name = av_default_item_name,
+ .option = rtp_options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+AVInputFormat ff_rtp_demuxer = {
+ .name = "rtp",
+ .long_name = NULL_IF_CONFIG_SMALL("RTP input format"),
+ .priv_data_size = sizeof(RTSPState),
+ .read_probe = rtp_probe,
+ .read_header = rtp_read_header,
+ .read_packet = ff_rtsp_fetch_packet,
+ .read_close = sdp_read_close,
+ .flags = AVFMT_NOFILE,
+ .priv_class = &rtp_demuxer_class
+};
+#endif /* CONFIG_RTP_DEMUXER */
+