| /* |
| * RTSP/SDP client |
| * Copyright (c) 2002 Fabrice Bellard |
| * |
| * This file is part of FFmpeg. |
| * |
| * FFmpeg is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Lesser General Public |
| * License as published by the Free Software Foundation; either |
| * version 2.1 of the License, or (at your option) any later version. |
| * |
| * FFmpeg is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| * Lesser General Public License for more details. |
| * |
| * You should have received a copy of the GNU Lesser General Public |
| * License along with FFmpeg; if not, write to the Free Software |
| * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| */ |
| |
| #include "libavutil/avassert.h" |
| #include "libavutil/base64.h" |
| #include "libavutil/avstring.h" |
| #include "libavutil/intreadwrite.h" |
| #include "libavutil/mathematics.h" |
| #include "libavutil/parseutils.h" |
| #include "libavutil/random_seed.h" |
| #include "libavutil/dict.h" |
| #include "libavutil/opt.h" |
| #include "libavutil/time.h" |
| #include "avformat.h" |
| #include "avio_internal.h" |
| |
| #if HAVE_POLL_H |
| #include <poll.h> |
| #endif |
| #include "internal.h" |
| #include "network.h" |
| #include "os_support.h" |
| #include "http.h" |
| #include "rtsp.h" |
| |
| #include "rtpdec.h" |
| #include "rtpproto.h" |
| #include "rdt.h" |
| #include "rtpdec_formats.h" |
| #include "rtpenc_chain.h" |
| #include "url.h" |
| #include "rtpenc.h" |
| #include "mpegts.h" |
| |
| /* Timeout values for socket poll, in ms, |
| * and read_packet(), in seconds */ |
| #define POLL_TIMEOUT_MS 100 |
| #define READ_PACKET_TIMEOUT_S 10 |
| #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS |
| #define SDP_MAX_SIZE 16384 |
| #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH |
| #define DEFAULT_REORDERING_DELAY 100000 |
| |
| #define OFFSET(x) offsetof(RTSPState, x) |
| #define DEC AV_OPT_FLAG_DECODING_PARAM |
| #define ENC AV_OPT_FLAG_ENCODING_PARAM |
| |
| #define RTSP_FLAG_OPTS(name, longname) \ |
| { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \ |
| { "filter_src", "only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" } |
| |
| #define RTSP_MEDIATYPE_OPTS(name, longname) \ |
| { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \ |
| { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \ |
| { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \ |
| { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }, \ |
| { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, "allowed_media_types" } |
| |
| #define COMMON_OPTS() \ |
| { "reorder_queue_size", "set number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }, \ |
| { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC } \ |
| |
| |
| const AVOption ff_rtsp_options[] = { |
| { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DEC }, |
| FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags), |
| { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \ |
| { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \ |
| { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \ |
| { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" }, |
| { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" }, |
| RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"), |
| { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }, |
| { "prefer_tcp", "try RTP via TCP first, if available", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_PREFER_TCP}, 0, 0, DEC|ENC, "rtsp_flags" }, |
| RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"), |
| { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC }, |
| { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC }, |
| { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC }, |
| { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC }, |
| COMMON_OPTS(), |
| { "user-agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC }, |
| { NULL }, |
| }; |
| |
| static const AVOption sdp_options[] = { |
| RTSP_FLAG_OPTS("sdp_flags", "SDP flags"), |
| { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" }, |
| { "rtcp_to_source", "send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" }, |
| RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"), |
| COMMON_OPTS(), |
| { NULL }, |
| }; |
| |
| static const AVOption rtp_options[] = { |
| RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"), |
| COMMON_OPTS(), |
| { NULL }, |
| }; |
| |
| |
| static AVDictionary *map_to_opts(RTSPState *rt) |
| { |
| AVDictionary *opts = NULL; |
| char buf[256]; |
| |
| snprintf(buf, sizeof(buf), "%d", rt->buffer_size); |
| av_dict_set(&opts, "buffer_size", buf, 0); |
| |
| return opts; |
| } |
| |
| static void get_word_until_chars(char *buf, int buf_size, |
| const char *sep, const char **pp) |
| { |
| const char *p; |
| char *q; |
| |
| p = *pp; |
| p += strspn(p, SPACE_CHARS); |
| q = buf; |
| while (!strchr(sep, *p) && *p != '\0') { |
| if ((q - buf) < buf_size - 1) |
| *q++ = *p; |
| p++; |
| } |
| if (buf_size > 0) |
| *q = '\0'; |
| *pp = p; |
| } |
| |
| static void get_word_sep(char *buf, int buf_size, const char *sep, |
| const char **pp) |
| { |
| if (**pp == '/') (*pp)++; |
| get_word_until_chars(buf, buf_size, sep, pp); |
| } |
| |
| static void get_word(char *buf, int buf_size, const char **pp) |
| { |
| get_word_until_chars(buf, buf_size, SPACE_CHARS, pp); |
| } |
| |
| /** Parse a string p in the form of Range:npt=xx-xx, and determine the start |
| * and end time. |
| * Used for seeking in the rtp stream. |
| */ |
| static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end) |
| { |
| char buf[256]; |
| |
| p += strspn(p, SPACE_CHARS); |
| if (!av_stristart(p, "npt=", &p)) |
| return; |
| |
| *start = AV_NOPTS_VALUE; |
| *end = AV_NOPTS_VALUE; |
| |
| get_word_sep(buf, sizeof(buf), "-", &p); |
| if (av_parse_time(start, buf, 1) < 0) |
| return; |
| if (*p == '-') { |
| p++; |
| get_word_sep(buf, sizeof(buf), "-", &p); |
| if (av_parse_time(end, buf, 1) < 0) |
| av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf); |
| } |
| } |
| |
| static int get_sockaddr(AVFormatContext *s, |
| const char *buf, struct sockaddr_storage *sock) |
| { |
| struct addrinfo hints = { 0 }, *ai = NULL; |
| int ret; |
| |
| hints.ai_flags = AI_NUMERICHOST; |
| if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) { |
| av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n", |
| buf, |
| gai_strerror(ret)); |
| 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, AVStream *st) |
| { |
| AVCodecContext *codec = st ? st->codec : NULL; |
| if (!handler) |
| return; |
| if (codec) |
| codec->codec_id = handler->codec_id; |
| rtsp_st->dynamic_handler = handler; |
| if (st) |
| st->need_parsing = handler->need_parsing; |
| if (handler->priv_data_size) { |
| rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size); |
| if (!rtsp_st->dynamic_protocol_context) |
| rtsp_st->dynamic_handler = NULL; |
| } |
| } |
| |
| static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st, |
| AVStream *st) |
| { |
| if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) { |
| int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1, |
| rtsp_st->dynamic_protocol_context); |
| if (ret < 0) { |
| if (rtsp_st->dynamic_protocol_context) { |
| if (rtsp_st->dynamic_handler->close) |
| rtsp_st->dynamic_handler->close( |
| rtsp_st->dynamic_protocol_context); |
| av_free(rtsp_st->dynamic_protocol_context); |
| } |
| rtsp_st->dynamic_protocol_context = NULL; |
| rtsp_st->dynamic_handler = NULL; |
| } |
| } |
| } |
| |
| /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */ |
| static int sdp_parse_rtpmap(AVFormatContext *s, |
| AVStream *st, RTSPStream *rtsp_st, |
| int payload_type, const char *p) |
| { |
| AVCodecContext *codec = st->codec; |
| char buf[256]; |
| int i; |
| AVCodec *c; |
| const char *c_name; |
| |
| /* See if we can handle this kind of payload. |
| * The space should normally not be there but some Real streams or |
| * particular servers ("RealServer Version 6.1.3.970", see issue 1658) |
| * have a trailing space. */ |
| get_word_sep(buf, sizeof(buf), "/ ", &p); |
| if (payload_type < RTP_PT_PRIVATE) { |
| /* We are in a standard case |
| * (from http://www.iana.org/assignments/rtp-parameters). */ |
| codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type); |
| } |
| |
| if (codec->codec_id == AV_CODEC_ID_NONE) { |
| RTPDynamicProtocolHandler *handler = |
| ff_rtp_handler_find_by_name(buf, codec->codec_type); |
| init_rtp_handler(handler, rtsp_st, st); |
| /* If no dynamic handler was found, check with the list of standard |
| * allocated types, if such a stream for some reason happens to |
| * use a private payload type. This isn't handled in rtpdec.c, since |
| * the format name from the rtpmap line never is passed into rtpdec. */ |
| if (!rtsp_st->dynamic_handler) |
| codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type); |
| } |
| |
| c = avcodec_find_decoder(codec->codec_id); |
| if (c && c->name) |
| c_name = c->name; |
| else |
| c_name = "(null)"; |
| |
| get_word_sep(buf, sizeof(buf), "/", &p); |
| i = atoi(buf); |
| switch (codec->codec_type) { |
| case AVMEDIA_TYPE_AUDIO: |
| av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name); |
| codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE; |
| codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS; |
| if (i > 0) { |
| codec->sample_rate = i; |
| avpriv_set_pts_info(st, 32, 1, codec->sample_rate); |
| get_word_sep(buf, sizeof(buf), "/", &p); |
| i = atoi(buf); |
| if (i > 0) |
| codec->channels = i; |
| } |
| av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n", |
| codec->sample_rate); |
| av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n", |
| codec->channels); |
| break; |
| case AVMEDIA_TYPE_VIDEO: |
| av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name); |
| if (i > 0) |
| avpriv_set_pts_info(st, 32, 1, i); |
| break; |
| default: |
| break; |
| } |
| finalize_rtp_handler_init(s, rtsp_st, st); |
| return 0; |
| } |
| |
| /* parse the attribute line from the fmtp a line of an sdp response. This |
| * is broken out as a function because it is used in rtp_h264.c, which is |
| * forthcoming. */ |
| int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, |
| char *value, int value_size) |
| { |
| *p += strspn(*p, SPACE_CHARS); |
| if (**p) { |
| get_word_sep(attr, attr_size, "=", p); |
| if (**p == '=') |
| (*p)++; |
| get_word_sep(value, value_size, ";", p); |
| if (**p == ';') |
| (*p)++; |
| return 1; |
| } |
| return 0; |
| } |
| |
| typedef struct SDPParseState { |
| /* SDP only */ |
| struct sockaddr_storage default_ip; |
| int default_ttl; |
| int skip_media; ///< set if an unknown m= line occurs |
| int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */ |
| struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */ |
| int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */ |
| struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */ |
| int seen_rtpmap; |
| int seen_fmtp; |
| char delayed_fmtp[2048]; |
| } SDPParseState; |
| |
| static void copy_default_source_addrs(struct RTSPSource **addrs, int count, |
| struct RTSPSource ***dest, int *dest_count) |
| { |
| RTSPSource *rtsp_src, *rtsp_src2; |
| int i; |
| for (i = 0; i < count; i++) { |
| rtsp_src = addrs[i]; |
| rtsp_src2 = av_malloc(sizeof(*rtsp_src2)); |
| if (!rtsp_src2) |
| continue; |
| memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src)); |
| dynarray_add(dest, dest_count, rtsp_src2); |
| } |
| } |
| |
| static void parse_fmtp(AVFormatContext *s, RTSPState *rt, |
| int payload_type, const char *line) |
| { |
| int i; |
| |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| RTSPStream *rtsp_st = rt->rtsp_streams[i]; |
| if (rtsp_st->sdp_payload_type == payload_type && |
| rtsp_st->dynamic_handler && |
| rtsp_st->dynamic_handler->parse_sdp_a_line) { |
| rtsp_st->dynamic_handler->parse_sdp_a_line(s, i, |
| rtsp_st->dynamic_protocol_context, line); |
| } |
| } |
| } |
| |
| static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1, |
| int letter, const char *buf) |
| { |
| RTSPState *rt = s->priv_data; |
| char buf1[64], st_type[64]; |
| const char *p; |
| enum AVMediaType codec_type; |
| int payload_type; |
| AVStream *st; |
| RTSPStream *rtsp_st; |
| RTSPSource *rtsp_src; |
| struct sockaddr_storage sdp_ip; |
| int ttl; |
| |
| av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf); |
| |
| p = buf; |
| if (s1->skip_media && letter != 'm') |
| return; |
| switch (letter) { |
| case 'c': |
| get_word(buf1, sizeof(buf1), &p); |
| if (strcmp(buf1, "IN") != 0) |
| return; |
| get_word(buf1, sizeof(buf1), &p); |
| if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6")) |
| return; |
| get_word_sep(buf1, sizeof(buf1), "/", &p); |
| if (get_sockaddr(s, buf1, &sdp_ip)) |
| return; |
| ttl = 16; |
| if (*p == '/') { |
| p++; |
| get_word_sep(buf1, sizeof(buf1), "/", &p); |
| ttl = atoi(buf1); |
| } |
| if (s->nb_streams == 0) { |
| s1->default_ip = sdp_ip; |
| s1->default_ttl = ttl; |
| } else { |
| rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; |
| rtsp_st->sdp_ip = sdp_ip; |
| rtsp_st->sdp_ttl = ttl; |
| } |
| break; |
| case 's': |
| av_dict_set(&s->metadata, "title", p, 0); |
| break; |
| case 'i': |
| if (s->nb_streams == 0) { |
| av_dict_set(&s->metadata, "comment", p, 0); |
| break; |
| } |
| break; |
| case 'm': |
| /* new stream */ |
| s1->skip_media = 0; |
| s1->seen_fmtp = 0; |
| s1->seen_rtpmap = 0; |
| codec_type = AVMEDIA_TYPE_UNKNOWN; |
| get_word(st_type, sizeof(st_type), &p); |
| if (!strcmp(st_type, "audio")) { |
| codec_type = AVMEDIA_TYPE_AUDIO; |
| } else if (!strcmp(st_type, "video")) { |
| codec_type = AVMEDIA_TYPE_VIDEO; |
| } else if (!strcmp(st_type, "application")) { |
| codec_type = AVMEDIA_TYPE_DATA; |
| } else if (!strcmp(st_type, "text")) { |
| codec_type = AVMEDIA_TYPE_SUBTITLE; |
| } |
| if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) { |
| s1->skip_media = 1; |
| return; |
| } |
| rtsp_st = av_mallocz(sizeof(RTSPStream)); |
| if (!rtsp_st) |
| return; |
| rtsp_st->stream_index = -1; |
| dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st); |
| |
| rtsp_st->sdp_ip = s1->default_ip; |
| rtsp_st->sdp_ttl = s1->default_ttl; |
| |
| copy_default_source_addrs(s1->default_include_source_addrs, |
| s1->nb_default_include_source_addrs, |
| &rtsp_st->include_source_addrs, |
| &rtsp_st->nb_include_source_addrs); |
| copy_default_source_addrs(s1->default_exclude_source_addrs, |
| s1->nb_default_exclude_source_addrs, |
| &rtsp_st->exclude_source_addrs, |
| &rtsp_st->nb_exclude_source_addrs); |
| |
| get_word(buf1, sizeof(buf1), &p); /* port */ |
| rtsp_st->sdp_port = atoi(buf1); |
| |
| get_word(buf1, sizeof(buf1), &p); /* protocol */ |
| if (!strcmp(buf1, "udp")) |
| rt->transport = RTSP_TRANSPORT_RAW; |
| else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF")) |
| rtsp_st->feedback = 1; |
| |
| /* XXX: handle list of formats */ |
| get_word(buf1, sizeof(buf1), &p); /* format list */ |
| rtsp_st->sdp_payload_type = atoi(buf1); |
| |
| if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) { |
| /* no corresponding stream */ |
| if (rt->transport == RTSP_TRANSPORT_RAW) { |
| if (CONFIG_RTPDEC && !rt->ts) |
| rt->ts = avpriv_mpegts_parse_open(s); |
| } else { |
| RTPDynamicProtocolHandler *handler; |
| handler = ff_rtp_handler_find_by_id( |
| rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA); |
| init_rtp_handler(handler, rtsp_st, NULL); |
| finalize_rtp_handler_init(s, rtsp_st, NULL); |
| } |
| } else if (rt->server_type == RTSP_SERVER_WMS && |
| codec_type == AVMEDIA_TYPE_DATA) { |
| /* RTX stream, a stream that carries all the other actual |
| * audio/video streams. Don't expose this to the callers. */ |
| } else { |
| st = avformat_new_stream(s, NULL); |
| if (!st) |
| return; |
| st->id = rt->nb_rtsp_streams - 1; |
| rtsp_st->stream_index = st->index; |
| st->codec->codec_type = codec_type; |
| if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) { |
| RTPDynamicProtocolHandler *handler; |
| /* if standard payload type, we can find the codec right now */ |
| ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type); |
| if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO && |
| st->codec->sample_rate > 0) |
| avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate); |
| /* Even static payload types may need a custom depacketizer */ |
| handler = ff_rtp_handler_find_by_id( |
| rtsp_st->sdp_payload_type, st->codec->codec_type); |
| init_rtp_handler(handler, rtsp_st, st); |
| finalize_rtp_handler_init(s, rtsp_st, st); |
| } |
| if (rt->default_lang[0]) |
| av_dict_set(&st->metadata, "language", rt->default_lang, 0); |
| } |
| /* put a default control url */ |
| av_strlcpy(rtsp_st->control_url, rt->control_uri, |
| sizeof(rtsp_st->control_url)); |
| break; |
| case 'a': |
| if (av_strstart(p, "control:", &p)) { |
| if (s->nb_streams == 0) { |
| if (!strncmp(p, "rtsp://", 7)) |
| av_strlcpy(rt->control_uri, p, |
| sizeof(rt->control_uri)); |
| } else { |
| char proto[32]; |
| /* get the control url */ |
| rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; |
| |
| /* XXX: may need to add full url resolution */ |
| av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0, |
| NULL, NULL, 0, p); |
| if (proto[0] == '\0') { |
| /* relative control URL */ |
| if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/') |
| av_strlcat(rtsp_st->control_url, "/", |
| sizeof(rtsp_st->control_url)); |
| av_strlcat(rtsp_st->control_url, p, |
| sizeof(rtsp_st->control_url)); |
| } else |
| av_strlcpy(rtsp_st->control_url, p, |
| sizeof(rtsp_st->control_url)); |
| } |
| } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) { |
| /* NOTE: rtpmap is only supported AFTER the 'm=' tag */ |
| get_word(buf1, sizeof(buf1), &p); |
| payload_type = atoi(buf1); |
| rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; |
| if (rtsp_st->stream_index >= 0) { |
| st = s->streams[rtsp_st->stream_index]; |
| sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p); |
| } |
| s1->seen_rtpmap = 1; |
| if (s1->seen_fmtp) { |
| parse_fmtp(s, rt, payload_type, s1->delayed_fmtp); |
| } |
| } else if (av_strstart(p, "fmtp:", &p) || |
| av_strstart(p, "framesize:", &p)) { |
| // let dynamic protocol handlers have a stab at the line. |
| get_word(buf1, sizeof(buf1), &p); |
| payload_type = atoi(buf1); |
| if (s1->seen_rtpmap) { |
| parse_fmtp(s, rt, payload_type, buf); |
| } else { |
| s1->seen_fmtp = 1; |
| av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp)); |
| } |
| } else if (av_strstart(p, "range:", &p)) { |
| int64_t start, end; |
| |
| // this is so that seeking on a streamed file can work. |
| rtsp_parse_range_npt(p, &start, &end); |
| s->start_time = start; |
| /* AV_NOPTS_VALUE means live broadcast (and can't seek) */ |
| s->duration = (end == AV_NOPTS_VALUE) ? |
| AV_NOPTS_VALUE : end - start; |
| } else if (av_strstart(p, "lang:", &p)) { |
| if (s->nb_streams > 0) { |
| get_word(buf1, sizeof(buf1), &p); |
| rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; |
| if (rtsp_st->stream_index >= 0) { |
| st = s->streams[rtsp_st->stream_index]; |
| av_dict_set(&st->metadata, "language", buf1, 0); |
| } |
| } else |
| get_word(rt->default_lang, sizeof(rt->default_lang), &p); |
| } else if (av_strstart(p, "IsRealDataType:integer;",&p)) { |
| if (atoi(p) == 1) |
| rt->transport = RTSP_TRANSPORT_RDT; |
| } else if (av_strstart(p, "SampleRate:integer;", &p) && |
| s->nb_streams > 0) { |
| st = s->streams[s->nb_streams - 1]; |
| st->codec->sample_rate = atoi(p); |
| } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) { |
| // RFC 4568 |
| rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; |
| get_word(buf1, sizeof(buf1), &p); // ignore tag |
| get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p); |
| p += strspn(p, SPACE_CHARS); |
| if (av_strstart(p, "inline:", &p)) |
| get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p); |
| } else if (av_strstart(p, "source-filter:", &p)) { |
| int exclude = 0; |
| get_word(buf1, sizeof(buf1), &p); |
| if (strcmp(buf1, "incl") && strcmp(buf1, "excl")) |
| return; |
| exclude = !strcmp(buf1, "excl"); |
| |
| get_word(buf1, sizeof(buf1), &p); |
| if (strcmp(buf1, "IN") != 0) |
| return; |
| get_word(buf1, sizeof(buf1), &p); |
| if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*")) |
| return; |
| // not checking that the destination address actually matches or is wildcard |
| get_word(buf1, sizeof(buf1), &p); |
| |
| while (*p != '\0') { |
| rtsp_src = av_mallocz(sizeof(*rtsp_src)); |
| if (!rtsp_src) |
| return; |
| get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p); |
| if (exclude) { |
| if (s->nb_streams == 0) { |
| dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src); |
| } else { |
| rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; |
| dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src); |
| } |
| } else { |
| if (s->nb_streams == 0) { |
| dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src); |
| } else { |
| rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; |
| dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src); |
| } |
| } |
| } |
| } else { |
| if (rt->server_type == RTSP_SERVER_WMS) |
| ff_wms_parse_sdp_a_line(s, p); |
| if (s->nb_streams > 0) { |
| rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; |
| |
| if (rt->server_type == RTSP_SERVER_REAL) |
| ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p); |
| |
| if (rtsp_st->dynamic_handler && |
| rtsp_st->dynamic_handler->parse_sdp_a_line) |
| rtsp_st->dynamic_handler->parse_sdp_a_line(s, |
| rtsp_st->stream_index, |
| rtsp_st->dynamic_protocol_context, buf); |
| } |
| } |
| break; |
| } |
| } |
| |
| int ff_sdp_parse(AVFormatContext *s, const char *content) |
| { |
| RTSPState *rt = s->priv_data; |
| const char *p; |
| int letter, i; |
| /* Some SDP lines, particularly for Realmedia or ASF RTSP streams, |
| * contain long SDP lines containing complete ASF Headers (several |
| * kB) or arrays of MDPR (RM stream descriptor) headers plus |
| * "rulebooks" describing their properties. Therefore, the SDP line |
| * buffer is large. |
| * |
| * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line |
| * in rtpdec_xiph.c. */ |
| char buf[16384], *q; |
| SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state; |
| |
| p = content; |
| for (;;) { |
| p += strspn(p, SPACE_CHARS); |
| letter = *p; |
| if (letter == '\0') |
| break; |
| p++; |
| if (*p != '=') |
| goto next_line; |
| p++; |
| /* get the content */ |
| q = buf; |
| while (*p != '\n' && *p != '\r' && *p != '\0') { |
| if ((q - buf) < sizeof(buf) - 1) |
| *q++ = *p; |
| p++; |
| } |
| *q = '\0'; |
| sdp_parse_line(s, s1, letter, buf); |
| next_line: |
| while (*p != '\n' && *p != '\0') |
| p++; |
| if (*p == '\n') |
| p++; |
| } |
| |
| for (i = 0; i < s1->nb_default_include_source_addrs; i++) |
| av_freep(&s1->default_include_source_addrs[i]); |
| av_freep(&s1->default_include_source_addrs); |
| for (i = 0; i < s1->nb_default_exclude_source_addrs; i++) |
| av_freep(&s1->default_exclude_source_addrs[i]); |
| av_freep(&s1->default_exclude_source_addrs); |
| |
| rt->p = av_malloc_array(rt->nb_rtsp_streams + 1, sizeof(struct pollfd) * 2); |
| if (!rt->p) return AVERROR(ENOMEM); |
| return 0; |
| } |
| #endif /* CONFIG_RTPDEC */ |
| |
| void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets) |
| { |
| 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) { |
| if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets) |
| ff_rtsp_tcp_write_packet(s, rtsp_st); |
| ffio_free_dyn_buf(&rtpctx->pb); |
| } else { |
| avio_closep(&rtpctx->pb); |
| } |
| avformat_free_context(rtpctx); |
| } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT) |
| ff_rdt_parse_close(rtsp_st->transport_priv); |
| else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) |
| ff_rtp_parse_close(rtsp_st->transport_priv); |
| } |
| rtsp_st->transport_priv = NULL; |
| if (rtsp_st->rtp_handle) |
| ffurl_close(rtsp_st->rtp_handle); |
| rtsp_st->rtp_handle = NULL; |
| } |
| } |
| |
| /* close and free RTSP streams */ |
| void ff_rtsp_close_streams(AVFormatContext *s) |
| { |
| RTSPState *rt = s->priv_data; |
| int i, j; |
| RTSPStream *rtsp_st; |
| |
| ff_rtsp_undo_setup(s, 0); |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| rtsp_st = rt->rtsp_streams[i]; |
| if (rtsp_st) { |
| if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) { |
| if (rtsp_st->dynamic_handler->close) |
| rtsp_st->dynamic_handler->close( |
| rtsp_st->dynamic_protocol_context); |
| av_free(rtsp_st->dynamic_protocol_context); |
| } |
| for (j = 0; j < rtsp_st->nb_include_source_addrs; j++) |
| av_freep(&rtsp_st->include_source_addrs[j]); |
| av_freep(&rtsp_st->include_source_addrs); |
| for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++) |
| av_freep(&rtsp_st->exclude_source_addrs[j]); |
| av_freep(&rtsp_st->exclude_source_addrs); |
| |
| av_freep(&rtsp_st); |
| } |
| } |
| av_freep(&rt->rtsp_streams); |
| if (rt->asf_ctx) { |
| avformat_close_input(&rt->asf_ctx); |
| } |
| if (CONFIG_RTPDEC && rt->ts) |
| avpriv_mpegts_parse_close(rt->ts); |
| av_freep(&rt->p); |
| av_freep(&rt->recvbuf); |
| } |
| |
| int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st) |
| { |
| RTSPState *rt = s->priv_data; |
| AVStream *st = NULL; |
| int reordering_queue_size = rt->reordering_queue_size; |
| if (reordering_queue_size < 0) { |
| if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay) |
| reordering_queue_size = 0; |
| else |
| reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE; |
| } |
| |
| /* open the RTP context */ |
| if (rtsp_st->stream_index >= 0) |
| st = s->streams[rtsp_st->stream_index]; |
| if (!st) |
| s->ctx_flags |= AVFMTCTX_NOHEADER; |
| |
| if (CONFIG_RTSP_MUXER && s->oformat && st) { |
| int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, |
| s, st, rtsp_st->rtp_handle, |
| RTSP_TCP_MAX_PACKET_SIZE, |
| rtsp_st->stream_index); |
| /* Ownership of rtp_handle is passed to the rtp mux context */ |
| rtsp_st->rtp_handle = NULL; |
| if (ret < 0) |
| return ret; |
| st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base; |
| } else if (rt->transport == RTSP_TRANSPORT_RAW) { |
| return 0; // Don't need to open any parser here |
| } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st) |
| rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index, |
| rtsp_st->dynamic_protocol_context, |
| rtsp_st->dynamic_handler); |
| else if (CONFIG_RTPDEC) |
| rtsp_st->transport_priv = ff_rtp_parse_open(s, st, |
| rtsp_st->sdp_payload_type, |
| reordering_queue_size); |
| |
| if (!rtsp_st->transport_priv) { |
| return AVERROR(ENOMEM); |
| } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) { |
| if (rtsp_st->dynamic_handler) { |
| ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv, |
| rtsp_st->dynamic_protocol_context, |
| rtsp_st->dynamic_handler); |
| } |
| if (rtsp_st->crypto_suite[0]) |
| ff_rtp_parse_set_crypto(rtsp_st->transport_priv, |
| rtsp_st->crypto_suite, |
| rtsp_st->crypto_params); |
| } |
| |
| return 0; |
| } |
| |
| #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER |
| static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp) |
| { |
| const char *q; |
| char *p; |
| int v; |
| |
| q = *pp; |
| q += strspn(q, SPACE_CHARS); |
| v = strtol(q, &p, 10); |
| if (*p == '-') { |
| p++; |
| *min_ptr = v; |
| v = strtol(p, &p, 10); |
| *max_ptr = v; |
| } else { |
| *min_ptr = v; |
| *max_ptr = v; |
| } |
| *pp = p; |
| } |
| |
| /* XXX: only one transport specification is parsed */ |
| static void rtsp_parse_transport(AVFormatContext *s, |
| RTSPMessageHeader *reply, const char *p) |
| { |
| char transport_protocol[16]; |
| char profile[16]; |
| char lower_transport[16]; |
| char parameter[16]; |
| RTSPTransportField *th; |
| char buf[256]; |
| |
| reply->nb_transports = 0; |
| |
| for (;;) { |
| p += strspn(p, SPACE_CHARS); |
| if (*p == '\0') |
| break; |
| |
| th = &reply->transports[reply->nb_transports]; |
| |
| get_word_sep(transport_protocol, sizeof(transport_protocol), |
| "/", &p); |
| if (!av_strcasecmp (transport_protocol, "rtp")) { |
| get_word_sep(profile, sizeof(profile), "/;,", &p); |
| lower_transport[0] = '\0'; |
| /* rtp/avp/<protocol> */ |
| if (*p == '/') { |
| get_word_sep(lower_transport, sizeof(lower_transport), |
| ";,", &p); |
| } |
| th->transport = RTSP_TRANSPORT_RTP; |
| } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") || |
| !av_strcasecmp (transport_protocol, "x-real-rdt")) { |
| /* x-pn-tng/<protocol> */ |
| get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p); |
| profile[0] = '\0'; |
| th->transport = RTSP_TRANSPORT_RDT; |
| } else if (!av_strcasecmp(transport_protocol, "raw")) { |
| get_word_sep(profile, sizeof(profile), "/;,", &p); |
| lower_transport[0] = '\0'; |
| /* raw/raw/<protocol> */ |
| if (*p == '/') { |
| get_word_sep(lower_transport, sizeof(lower_transport), |
| ";,", &p); |
| } |
| th->transport = RTSP_TRANSPORT_RAW; |
| } |
| if (!av_strcasecmp(lower_transport, "TCP")) |
| th->lower_transport = RTSP_LOWER_TRANSPORT_TCP; |
| else |
| th->lower_transport = RTSP_LOWER_TRANSPORT_UDP; |
| |
| if (*p == ';') |
| p++; |
| /* get each parameter */ |
| while (*p != '\0' && *p != ',') { |
| get_word_sep(parameter, sizeof(parameter), "=;,", &p); |
| if (!strcmp(parameter, "port")) { |
| if (*p == '=') { |
| p++; |
| rtsp_parse_range(&th->port_min, &th->port_max, &p); |
| } |
| } else if (!strcmp(parameter, "client_port")) { |
| if (*p == '=') { |
| p++; |
| rtsp_parse_range(&th->client_port_min, |
| &th->client_port_max, &p); |
| } |
| } else if (!strcmp(parameter, "server_port")) { |
| if (*p == '=') { |
| p++; |
| rtsp_parse_range(&th->server_port_min, |
| &th->server_port_max, &p); |
| } |
| } else if (!strcmp(parameter, "interleaved")) { |
| if (*p == '=') { |
| p++; |
| rtsp_parse_range(&th->interleaved_min, |
| &th->interleaved_max, &p); |
| } |
| } else if (!strcmp(parameter, "multicast")) { |
| if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP) |
| th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST; |
| } else if (!strcmp(parameter, "ttl")) { |
| if (*p == '=') { |
| char *end; |
| p++; |
| th->ttl = strtol(p, &end, 10); |
| p = end; |
| } |
| } else if (!strcmp(parameter, "destination")) { |
| if (*p == '=') { |
| p++; |
| get_word_sep(buf, sizeof(buf), ";,", &p); |
| get_sockaddr(s, buf, &th->destination); |
| } |
| } else if (!strcmp(parameter, "source")) { |
| if (*p == '=') { |
| p++; |
| get_word_sep(buf, sizeof(buf), ";,", &p); |
| av_strlcpy(th->source, buf, sizeof(th->source)); |
| } |
| } else if (!strcmp(parameter, "mode")) { |
| if (*p == '=') { |
| p++; |
| get_word_sep(buf, sizeof(buf), ";, ", &p); |
| if (!strcmp(buf, "record") || |
| !strcmp(buf, "receive")) |
| th->mode_record = 1; |
| } |
| } |
| |
| while (*p != ';' && *p != '\0' && *p != ',') |
| p++; |
| if (*p == ';') |
| p++; |
| } |
| if (*p == ',') |
| p++; |
| |
| reply->nb_transports++; |
| if (reply->nb_transports >= RTSP_MAX_TRANSPORTS) |
| break; |
| } |
| } |
| |
| 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(AVFormatContext *s, |
| RTSPMessageHeader *reply, const char *buf, |
| RTSPState *rt, const char *method) |
| { |
| const char *p; |
| |
| /* NOTE: we do case independent match for broken servers */ |
| p = buf; |
| if (av_stristart(p, "Session:", &p)) { |
| int t; |
| get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p); |
| if (av_stristart(p, ";timeout=", &p) && |
| (t = strtol(p, NULL, 10)) > 0) { |
| reply->timeout = t; |
| } |
| } else if (av_stristart(p, "Content-Length:", &p)) { |
| reply->content_length = strtol(p, NULL, 10); |
| } else if (av_stristart(p, "Transport:", &p)) { |
| rtsp_parse_transport(s, reply, p); |
| } else if (av_stristart(p, "CSeq:", &p)) { |
| reply->seq = strtol(p, NULL, 10); |
| } else if (av_stristart(p, "Range:", &p)) { |
| rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end); |
| } else if (av_stristart(p, "RealChallenge1:", &p)) { |
| p += strspn(p, SPACE_CHARS); |
| av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge)); |
| } else if (av_stristart(p, "Server:", &p)) { |
| p += strspn(p, SPACE_CHARS); |
| av_strlcpy(reply->server, p, sizeof(reply->server)); |
| } else if (av_stristart(p, "Notice:", &p) || |
| av_stristart(p, "X-Notice:", &p)) { |
| reply->notice = strtol(p, NULL, 10); |
| } else if (av_stristart(p, "Location:", &p)) { |
| p += strspn(p, SPACE_CHARS); |
| av_strlcpy(reply->location, p , sizeof(reply->location)); |
| } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) { |
| p += strspn(p, SPACE_CHARS); |
| ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p); |
| } else if (av_stristart(p, "Authentication-Info:", &p) && rt) { |
| p += strspn(p, SPACE_CHARS); |
| ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p); |
| } else if (av_stristart(p, "Content-Base:", &p) && rt) { |
| p += strspn(p, SPACE_CHARS); |
| if (method && !strcmp(method, "DESCRIBE")) |
| av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri)); |
| } else if (av_stristart(p, "RTP-Info:", &p) && rt) { |
| p += strspn(p, SPACE_CHARS); |
| if (method && !strcmp(method, "PLAY")) |
| rtsp_parse_rtp_info(rt, p); |
| } else if (av_stristart(p, "Public:", &p) && rt) { |
| if (strstr(p, "GET_PARAMETER") && |
| method && !strcmp(method, "OPTIONS")) |
| rt->get_parameter_supported = 1; |
| } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) { |
| p += strspn(p, SPACE_CHARS); |
| rt->accept_dynamic_rate = atoi(p); |
| } else if (av_stristart(p, "Content-Type:", &p)) { |
| p += strspn(p, SPACE_CHARS); |
| av_strlcpy(reply->content_type, p, sizeof(reply->content_type)); |
| } |
| } |
| |
| /* skip a RTP/TCP interleaved packet */ |
| void ff_rtsp_skip_packet(AVFormatContext *s) |
| { |
| RTSPState *rt = s->priv_data; |
| int ret, len, len1; |
| uint8_t buf[1024]; |
| |
| ret = ffurl_read_complete(rt->rtsp_hd, buf, 3); |
| if (ret != 3) |
| return; |
| len = AV_RB16(buf + 1); |
| |
| av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len); |
| |
| /* skip payload */ |
| while (len > 0) { |
| len1 = len; |
| if (len1 > sizeof(buf)) |
| len1 = sizeof(buf); |
| ret = ffurl_read_complete(rt->rtsp_hd, buf, len1); |
| if (ret != len1) |
| return; |
| len -= len1; |
| } |
| } |
| |
| int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, |
| unsigned char **content_ptr, |
| int return_on_interleaved_data, const char *method) |
| { |
| RTSPState *rt = s->priv_data; |
| char buf[4096], buf1[1024], *q; |
| unsigned char ch; |
| const char *p; |
| int ret, content_length, line_count = 0, request = 0; |
| unsigned char *content = NULL; |
| |
| start: |
| line_count = 0; |
| request = 0; |
| content = NULL; |
| memset(reply, 0, sizeof(*reply)); |
| |
| /* parse reply (XXX: use buffers) */ |
| rt->last_reply[0] = '\0'; |
| for (;;) { |
| q = buf; |
| for (;;) { |
| ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1); |
| av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch); |
| if (ret != 1) |
| return AVERROR_EOF; |
| if (ch == '\n') |
| break; |
| if (ch == '$' && q == buf) { |
| if (return_on_interleaved_data) { |
| return 1; |
| } else |
| ff_rtsp_skip_packet(s); |
| } else if (ch != '\r') { |
| if ((q - buf) < sizeof(buf) - 1) |
| *q++ = ch; |
| } |
| } |
| *q = '\0'; |
| |
| av_log(s, AV_LOG_TRACE, "line='%s'\n", buf); |
| |
| /* test if last line */ |
| if (buf[0] == '\0') |
| break; |
| p = buf; |
| if (line_count == 0) { |
| /* get reply code */ |
| get_word(buf1, sizeof(buf1), &p); |
| if (!strncmp(buf1, "RTSP/", 5)) { |
| get_word(buf1, sizeof(buf1), &p); |
| reply->status_code = atoi(buf1); |
| av_strlcpy(reply->reason, p, sizeof(reply->reason)); |
| } else { |
| av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method |
| get_word(buf1, sizeof(buf1), &p); // object |
| request = 1; |
| } |
| } else { |
| ff_rtsp_parse_line(s, reply, p, rt, method); |
| av_strlcat(rt->last_reply, p, sizeof(rt->last_reply)); |
| av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply)); |
| } |
| line_count++; |
| } |
| |
| if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request) |
| av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id)); |
| |
| content_length = reply->content_length; |
| if (content_length > 0) { |
| /* leave some room for a trailing '\0' (useful for simple parsing) */ |
| content = av_malloc(content_length + 1); |
| if (!content) |
| return AVERROR(ENOMEM); |
| ffurl_read_complete(rt->rtsp_hd, content, content_length); |
| content[content_length] = '\0'; |
| } |
| if (content_ptr) |
| *content_ptr = content; |
| else |
| av_freep(&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_relative(); |
| /* Even if the request from the server had data, it is not the data |
| * that the caller wants or expects. The memory could also be leaked |
| * if the actual following reply has content data. */ |
| if (content_ptr) |
| av_freep(content_ptr); |
| /* If method is set, this is called from ff_rtsp_send_cmd, |
| * where a reply to exactly this request is awaited. For |
| * callers from within packet receiving, we just want to |
| * return to the caller and go back to receiving packets. */ |
| if (method) |
| goto start; |
| return 0; |
| } |
| |
| if (rt->seq != reply->seq) { |
| av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n", |
| rt->seq, reply->seq); |
| } |
| |
| /* EOS */ |
| if (reply->notice == 2101 /* End-of-Stream Reached */ || |
| reply->notice == 2104 /* Start-of-Stream Reached */ || |
| reply->notice == 2306 /* Continuous Feed Terminated */) { |
| rt->state = RTSP_STATE_IDLE; |
| } else if (reply->notice >= 4400 && reply->notice < 5500) { |
| return AVERROR(EIO); /* data or server error */ |
| } else if (reply->notice == 2401 /* Ticket Expired */ || |
| (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ ) |
| return AVERROR(EPERM); |
| |
| return 0; |
| } |
| |
| /** |
| * Send a command to the RTSP server without waiting for the reply. |
| * |
| * @param s RTSP (de)muxer context |
| * @param method the method for the request |
| * @param url the target url for the request |
| * @param headers extra header lines to include in the request |
| * @param send_content if non-null, the data to send as request body content |
| * @param send_content_length the length of the send_content data, or 0 if |
| * send_content is null |
| * |
| * @return zero if success, nonzero otherwise |
| */ |
| static int rtsp_send_cmd_with_content_async(AVFormatContext *s, |
| const char *method, const char *url, |
| const char *headers, |
| const unsigned char *send_content, |
| int send_content_length) |
| { |
| RTSPState *rt = s->priv_data; |
| char buf[4096], *out_buf; |
| char base64buf[AV_BASE64_SIZE(sizeof(buf))]; |
| |
| /* Add in RTSP headers */ |
| out_buf = buf; |
| rt->seq++; |
| snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url); |
| if (headers) |
| av_strlcat(buf, headers, sizeof(buf)); |
| av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq); |
| av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent); |
| if (rt->session_id[0] != '\0' && (!headers || |
| !strstr(headers, "\nIf-Match:"))) { |
| av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id); |
| } |
| if (rt->auth[0]) { |
| char *str = ff_http_auth_create_response(&rt->auth_state, |
| rt->auth, url, method); |
| if (str) |
| av_strlcat(buf, str, sizeof(buf)); |
| av_free(str); |
| } |
| if (send_content_length > 0 && send_content) |
| av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length); |
| av_strlcat(buf, "\r\n", sizeof(buf)); |
| |
| /* base64 encode rtsp if tunneling */ |
| if (rt->control_transport == RTSP_MODE_TUNNEL) { |
| av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf)); |
| out_buf = base64buf; |
| } |
| |
| av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf); |
| |
| ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf)); |
| if (send_content_length > 0 && send_content) { |
| if (rt->control_transport == RTSP_MODE_TUNNEL) { |
| av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests " |
| "with content data not supported\n"); |
| return AVERROR_PATCHWELCOME; |
| } |
| ffurl_write(rt->rtsp_hd_out, send_content, send_content_length); |
| } |
| rt->last_cmd_time = av_gettime_relative(); |
| |
| return 0; |
| } |
| |
| int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, |
| const char *url, const char *headers) |
| { |
| return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0); |
| } |
| |
| int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url, |
| const char *headers, RTSPMessageHeader *reply, |
| unsigned char **content_ptr) |
| { |
| return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply, |
| content_ptr, NULL, 0); |
| } |
| |
| int ff_rtsp_send_cmd_with_content(AVFormatContext *s, |
| const char *method, const char *url, |
| const char *header, |
| RTSPMessageHeader *reply, |
| unsigned char **content_ptr, |
| const unsigned char *send_content, |
| int send_content_length) |
| { |
| RTSPState *rt = s->priv_data; |
| HTTPAuthType cur_auth_type; |
| int ret, attempts = 0; |
| |
| retry: |
| cur_auth_type = rt->auth_state.auth_type; |
| if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header, |
| send_content, |
| send_content_length))) |
| return ret; |
| |
| if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0) |
| return ret; |
| attempts++; |
| |
| if (reply->status_code == 401 && |
| (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) && |
| rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2) |
| goto retry; |
| |
| if (reply->status_code > 400){ |
| av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n", |
| method, |
| reply->status_code, |
| reply->reason); |
| av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply); |
| } |
| |
| return 0; |
| } |
| |
| int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, |
| int lower_transport, const char *real_challenge) |
| { |
| RTSPState *rt = s->priv_data; |
| int rtx = 0, j, i, err, interleave = 0, port_off; |
| RTSPStream *rtsp_st; |
| RTSPMessageHeader reply1, *reply = &reply1; |
| char cmd[2048]; |
| const char *trans_pref; |
| |
| if (rt->transport == RTSP_TRANSPORT_RDT) |
| trans_pref = "x-pn-tng"; |
| else if (rt->transport == RTSP_TRANSPORT_RAW) |
| trans_pref = "RAW/RAW"; |
| else |
| trans_pref = "RTP/AVP"; |
| |
| /* default timeout: 1 minute */ |
| rt->timeout = 60; |
| |
| /* Choose a random starting offset within the first half of the |
| * port range, to allow for a number of ports to try even if the offset |
| * happens to be at the end of the random range. */ |
| port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2); |
| /* even random offset */ |
| port_off -= port_off & 0x01; |
| |
| for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) { |
| char transport[2048]; |
| |
| /* |
| * WMS serves all UDP data over a single connection, the RTX, which |
| * isn't necessarily the first in the SDP but has to be the first |
| * to be set up, else the second/third SETUP will fail with a 461. |
| */ |
| if (lower_transport == RTSP_LOWER_TRANSPORT_UDP && |
| rt->server_type == RTSP_SERVER_WMS) { |
| if (i == 0) { |
| /* rtx first */ |
| for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) { |
| int len = strlen(rt->rtsp_streams[rtx]->control_url); |
| if (len >= 4 && |
| !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4, |
| "/rtx")) |
| break; |
| } |
| if (rtx == rt->nb_rtsp_streams) |
| return -1; /* no RTX found */ |
| rtsp_st = rt->rtsp_streams[rtx]; |
| } else |
| rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1]; |
| } else |
| rtsp_st = rt->rtsp_streams[i]; |
| |
| /* RTP/UDP */ |
| if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) { |
| char buf[256]; |
| |
| if (rt->server_type == RTSP_SERVER_WMS && i > 1) { |
| port = reply->transports[0].client_port_min; |
| goto have_port; |
| } |
| |
| /* first try in specified port range */ |
| while (j <= rt->rtp_port_max) { |
| AVDictionary *opts = map_to_opts(rt); |
| |
| 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; |
| err = ffurl_open_whitelist(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE, |
| &s->interrupt_callback, &opts, s->protocol_whitelist); |
| |
| av_dict_free(&opts); |
| |
| if (!err) |
| goto rtp_opened; |
| } |
| av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n"); |
| err = AVERROR(EIO); |
| goto fail; |
| |
| rtp_opened: |
| port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle); |
| have_port: |
| snprintf(transport, sizeof(transport) - 1, |
| "%s/UDP;", trans_pref); |
| if (rt->server_type != RTSP_SERVER_REAL) |
| av_strlcat(transport, "unicast;", sizeof(transport)); |
| av_strlcatf(transport, sizeof(transport), |
| "client_port=%d", port); |
| if (rt->transport == RTSP_TRANSPORT_RTP && |
| !(rt->server_type == RTSP_SERVER_WMS && i > 0)) |
| av_strlcatf(transport, sizeof(transport), "-%d", port + 1); |
| } |
| |
| /* RTP/TCP */ |
| else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) { |
| /* For WMS streams, the application streams are only used for |
| * UDP. When trying to set it up for TCP streams, the server |
| * will return an error. Therefore, we skip those streams. */ |
| if (rt->server_type == RTSP_SERVER_WMS && |
| (rtsp_st->stream_index < 0 || |
| s->streams[rtsp_st->stream_index]->codec->codec_type == |
| AVMEDIA_TYPE_DATA)) |
| continue; |
| snprintf(transport, sizeof(transport) - 1, |
| "%s/TCP;", trans_pref); |
| if (rt->transport != RTSP_TRANSPORT_RDT) |
| av_strlcat(transport, "unicast;", sizeof(transport)); |
| av_strlcatf(transport, sizeof(transport), |
| "interleaved=%d-%d", |
| interleave, interleave + 1); |
| interleave += 2; |
| } |
| |
| else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) { |
| snprintf(transport, sizeof(transport) - 1, |
| "%s/UDP;multicast", trans_pref); |
| } |
| if (s->oformat) { |
| av_strlcat(transport, ";mode=record", sizeof(transport)); |
| } else if (rt->server_type == RTSP_SERVER_REAL || |
| rt->server_type == RTSP_SERVER_WMS) |
| av_strlcat(transport, ";mode=play", sizeof(transport)); |
| snprintf(cmd, sizeof(cmd), |
| "Transport: %s\r\n", |
| transport); |
| if (rt->accept_dynamic_rate) |
| av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd)); |
| if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) { |
| char real_res[41], real_csum[9]; |
| ff_rdt_calc_response_and_checksum(real_res, real_csum, |
| real_challenge); |
| av_strlcatf(cmd, sizeof(cmd), |
| "If-Match: %s\r\n" |
| "RealChallenge2: %s, sd=%s\r\n", |
| rt->session_id, real_res, real_csum); |
| } |
| ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL); |
| if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) { |
| err = 1; |
| goto fail; |
| } else if (reply->status_code != RTSP_STATUS_OK || |
| reply->nb_transports != 1) { |
| err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA); |
| goto fail; |
| } |
| |
| /* XXX: same protocol for all streams is required */ |
| if (i > 0) { |
| if (reply->transports[0].lower_transport != rt->lower_transport || |
| reply->transports[0].transport != rt->transport) { |
| err = AVERROR_INVALIDDATA; |
| goto fail; |
| } |
| } else { |
| rt->lower_transport = reply->transports[0].lower_transport; |
| rt->transport = reply->transports[0].transport; |
| } |
| |
| /* Fail if the server responded with another lower transport mode |
| * than what we requested. */ |
| if (reply->transports[0].lower_transport != lower_transport) { |
| av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n"); |
| err = AVERROR_INVALIDDATA; |
| goto fail; |
| } |
| |
| switch(reply->transports[0].lower_transport) { |
| case RTSP_LOWER_TRANSPORT_TCP: |
| rtsp_st->interleaved_min = reply->transports[0].interleaved_min; |
| rtsp_st->interleaved_max = reply->transports[0].interleaved_max; |
| break; |
| |
| case RTSP_LOWER_TRANSPORT_UDP: { |
| char url[1024], options[30] = ""; |
| const char *peer = host; |
| |
| if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC) |
| av_strlcpy(options, "?connect=1", sizeof(options)); |
| /* Use source address if specified */ |
| if (reply->transports[0].source[0]) |
| peer = reply->transports[0].source; |
| ff_url_join(url, sizeof(url), "rtp", NULL, peer, |
| reply->transports[0].server_port_min, "%s", options); |
| if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && |
| ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) { |
| err = AVERROR_INVALIDDATA; |
| goto fail; |
| } |
| break; |
| } |
| case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: { |
| char url[1024], namebuf[50], optbuf[20] = ""; |
| struct sockaddr_storage addr; |
| int port, ttl; |
| |
| if (reply->transports[0].destination.ss_family) { |
| addr = reply->transports[0].destination; |
| port = reply->transports[0].port_min; |
| ttl = reply->transports[0].ttl; |
| } else { |
| addr = rtsp_st->sdp_ip; |
| port = rtsp_st->sdp_port; |
| ttl = rtsp_st->sdp_ttl; |
| } |
| if (ttl > 0) |
| snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl); |
| getnameinfo((struct sockaddr*) &addr, sizeof(addr), |
| namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST); |
| ff_url_join(url, sizeof(url), "rtp", NULL, namebuf, |
| port, "%s", optbuf); |
| if (ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE, |
| &s->interrupt_callback, NULL, s->protocol_whitelist) < 0) { |
| err = AVERROR_INVALIDDATA; |
| goto fail; |
| } |
| break; |
| } |
| } |
| |
| if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st))) |
| goto fail; |
| } |
| |
| if (rt->nb_rtsp_streams && reply->timeout > 0) |
| rt->timeout = reply->timeout; |
| |
| if (rt->server_type == RTSP_SERVER_REAL) |
| rt->need_subscription = 1; |
| |
| return 0; |
| |
| fail: |
| ff_rtsp_undo_setup(s, 0); |
| return err; |
| } |
| |
| void ff_rtsp_close_connections(AVFormatContext *s) |
| { |
| RTSPState *rt = s->priv_data; |
| if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out); |
| ffurl_close(rt->rtsp_hd); |
| rt->rtsp_hd = rt->rtsp_hd_out = NULL; |
| } |
| |
| int ff_rtsp_connect(AVFormatContext *s) |
| { |
| RTSPState *rt = s->priv_data; |
| char proto[128], host[1024], path[1024]; |
| char tcpname[1024], cmd[2048], auth[128]; |
| const char *lower_rtsp_proto = "tcp"; |
| int port, err, tcp_fd; |
| RTSPMessageHeader reply1 = {0}, *reply = &reply1; |
| int lower_transport_mask = 0; |
| int default_port = RTSP_DEFAULT_PORT; |
| char real_challenge[64] = ""; |
| struct sockaddr_storage peer; |
| socklen_t peer_len = sizeof(peer); |
| |
| if (rt->rtp_port_max < rt->rtp_port_min) { |
| av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less " |
| "than min port %d\n", rt->rtp_port_max, |
| rt->rtp_port_min); |
| return AVERROR(EINVAL); |
| } |
| |
| if (!ff_network_init()) |
| return AVERROR(EIO); |
| |
| if (s->max_delay < 0) /* Not set by the caller */ |
| s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0; |
| |
| rt->control_transport = RTSP_MODE_PLAIN; |
| if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) { |
| rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP; |
| rt->control_transport = RTSP_MODE_TUNNEL; |
| } |
| /* Only pass through valid flags from here */ |
| rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1; |
| |
| redirect: |
| /* extract hostname and port */ |
| av_url_split(proto, sizeof(proto), auth, sizeof(auth), |
| host, sizeof(host), &port, path, sizeof(path), s->filename); |
| |
| if (!strcmp(proto, "rtsps")) { |
| lower_rtsp_proto = "tls"; |
| default_port = RTSPS_DEFAULT_PORT; |
| rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP; |
| } |
| |
| if (*auth) { |
| av_strlcpy(rt->auth, auth, sizeof(rt->auth)); |
| } |
| if (port < 0) |
| port = default_port; |
| |
| lower_transport_mask = rt->lower_transport_mask; |
| |
| if (!lower_transport_mask) |
| lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1; |
| |
| if (s->oformat) { |
| /* Only UDP or TCP - UDP multicast isn't supported. */ |
| lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) | |
| (1 << RTSP_LOWER_TRANSPORT_TCP); |
| if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) { |
| av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, " |
| "only UDP and TCP are supported for output.\n"); |
| err = AVERROR(EINVAL); |
| goto fail; |
| } |
| } |
| |
| /* Construct the URI used in request; this is similar to s->filename, |
| * but with authentication credentials removed and RTSP specific options |
| * stripped out. */ |
| ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL, |
| host, port, "%s", path); |
| |
| if (rt->control_transport == RTSP_MODE_TUNNEL) { |
| /* set up initial handshake for tunneling */ |
| char httpname[1024]; |
| char sessioncookie[17]; |
| char headers[1024]; |
| |
| ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path); |
| snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x", |
| av_get_random_seed(), av_get_random_seed()); |
| |
| /* GET requests */ |
| if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ, |
| &s->interrupt_callback) < 0) { |
| err = AVERROR(EIO); |
| goto fail; |
| } |
| |
| /* generate GET headers */ |
| snprintf(headers, sizeof(headers), |
| "x-sessioncookie: %s\r\n" |
| "Accept: application/x-rtsp-tunnelled\r\n" |
| "Pragma: no-cache\r\n" |
| "Cache-Control: no-cache\r\n", |
| sessioncookie); |
| av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0); |
| |
| /* complete the connection */ |
| if (ffurl_connect(rt->rtsp_hd, NULL)) { |
| err = AVERROR(EIO); |
| goto fail; |
| } |
| |
| /* POST requests */ |
| if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE, |
| &s->interrupt_callback) < 0 ) { |
| err = AVERROR(EIO); |
| goto fail; |
| } |
| |
| /* generate POST headers */ |
| snprintf(headers, sizeof(headers), |
| "x-sessioncookie: %s\r\n" |
| "Content-Type: application/x-rtsp-tunnelled\r\n" |
| "Pragma: no-cache\r\n" |
| "Cache-Control: no-cache\r\n" |
| "Content-Length: 32767\r\n" |
| "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n", |
| sessioncookie); |
| av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0); |
| av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0); |
| |
| /* Initialize the authentication state for the POST session. The HTTP |
| * protocol implementation doesn't properly handle multi-pass |
| * authentication for POST requests, since it would require one of |
| * the following: |
| * - implementing Expect: 100-continue, which many HTTP servers |
| * don't support anyway, even less the RTSP servers that do HTTP |
| * tunneling |
| * - sending the whole POST data until getting a 401 reply specifying |
| * what authentication method to use, then resending all that data |
| * - waiting for potential 401 replies directly after sending the |
| * POST header (waiting for some unspecified time) |
| * Therefore, we copy the full auth state, which works for both basic |
| * and digest. (For digest, we would have to synchronize the nonce |
| * count variable between the two sessions, if we'd do more requests |
| * with the original session, though.) |
| */ |
| ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd); |
| |
| /* complete the connection */ |
| if (ffurl_connect(rt->rtsp_hd_out, NULL)) { |
| err = AVERROR(EIO); |
| goto fail; |
| } |
| } else { |
| int ret; |
| /* open the tcp connection */ |
| ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL, |
| host, port, |
| "?timeout=%d", rt->stimeout); |
| if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE, |
| &s->interrupt_callback, NULL, s->protocol_whitelist)) < 0) { |
| err = ret; |
| goto fail; |
| } |
| rt->rtsp_hd_out = rt->rtsp_hd; |
| } |
| rt->seq = 0; |
| |
| tcp_fd = ffurl_get_file_handle(rt->rtsp_hd); |
| if (tcp_fd < 0) { |
| err = tcp_fd; |
| goto fail; |
| } |
| if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) { |
| getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host), |
| NULL, 0, NI_NUMERICHOST); |
| } |
| |
| /* request options supported by the server; this also detects server |
| * type */ |
| for (rt->server_type = RTSP_SERVER_RTP;;) { |
| cmd[0] = 0; |
| if (rt->server_type == RTSP_SERVER_REAL) |
| av_strlcat(cmd, |
| /* |
| * The following entries are required for proper |
| * streaming from a Realmedia server. They are |
| * interdependent in some way although we currently |
| * don't quite understand how. Values were copied |
| * from mplayer SVN r23589. |
| * ClientChallenge is a 16-byte ID in hex |
| * CompanyID is a 16-byte ID in base64 |
| */ |
| "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n" |
| "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n" |
| "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n" |
| "GUID: 00000000-0000-0000-0000-000000000000\r\n", |
| sizeof(cmd)); |
| ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL); |
| if (reply->status_code != RTSP_STATUS_OK) { |
| err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA); |
| goto fail; |
| } |
| |
| /* detect server type if not standard-compliant RTP */ |
| if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) { |
| rt->server_type = RTSP_SERVER_REAL; |
| continue; |
| } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) { |
| rt->server_type = RTSP_SERVER_WMS; |
| } else if (rt->server_type == RTSP_SERVER_REAL) |
| strcpy(real_challenge, reply->real_challenge); |
| break; |
| } |
| |
| if (CONFIG_RTSP_DEMUXER && s->iformat) |
| err = ff_rtsp_setup_input_streams(s, reply); |
| else if (CONFIG_RTSP_MUXER) |
| err = ff_rtsp_setup_output_streams(s, host); |
| else |
| av_assert0(0); |
| if (err) |
| goto fail; |
| |
| do { |
| int lower_transport = ff_log2_tab[lower_transport_mask & |
| ~(lower_transport_mask - 1)]; |
| |
| if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP)) |
| && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP)) |
| lower_transport = RTSP_LOWER_TRANSPORT_TCP; |
| |
| err = ff_rtsp_make_setup_request(s, host, port, lower_transport, |
| rt->server_type == RTSP_SERVER_REAL ? |
| real_challenge : NULL); |
| if (err < 0) |
| goto fail; |
| lower_transport_mask &= ~(1 << lower_transport); |
| if (lower_transport_mask == 0 && err == 1) { |
| err = AVERROR(EPROTONOSUPPORT); |
| goto fail; |
| } |
| } while (err); |
| |
| rt->lower_transport_mask = lower_transport_mask; |
| av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge)); |
| rt->state = RTSP_STATE_IDLE; |
| rt->seek_timestamp = 0; /* default is to start stream at position zero */ |
| return 0; |
| fail: |
| ff_rtsp_close_streams(s); |
| ff_rtsp_close_connections(s); |
| if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) { |
| av_strlcpy(s->filename, reply->location, sizeof(s->filename)); |
| rt->session_id[0] = '\0'; |
| av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n", |
| reply->status_code, |
| s->filename); |
| goto redirect; |
| } |
| ff_network_close(); |
| return err; |
| } |
| #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */ |
| |
| #if CONFIG_RTPDEC |
| static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, |
| uint8_t *buf, int buf_size, int64_t wait_end) |
| { |
| RTSPState *rt = s->priv_data; |
| RTSPStream *rtsp_st; |
| int n, i, ret, tcp_fd, timeout_cnt = 0; |
| int max_p = 0; |
| struct pollfd *p = rt->p; |
| int *fds = NULL, fdsnum, fdsidx; |
| |
| for (;;) { |
| if (ff_check_interrupt(&s->interrupt_callback)) |
| return AVERROR_EXIT; |
| if (wait_end && wait_end - av_gettime_relative() < 0) |
| return AVERROR(EAGAIN); |
| max_p = 0; |
| if (rt->rtsp_hd) { |
| tcp_fd = ffurl_get_file_handle(rt->rtsp_hd); |
| p[max_p].fd = tcp_fd; |
| p[max_p++].events = POLLIN; |
| } else { |
| tcp_fd = -1; |
| } |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| rtsp_st = rt->rtsp_streams[i]; |
| if (rtsp_st->rtp_handle) { |
| if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle, |
| &fds, &fdsnum)) { |
| av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n"); |
| return ret; |
| } |
| if (fdsnum != 2) { |
| av_log(s, AV_LOG_ERROR, |
| "Number of fds %d not supported\n", fdsnum); |
| return AVERROR_INVALIDDATA; |
| } |
| for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) { |
| p[max_p].fd = fds[fdsidx]; |
| p[max_p++].events = POLLIN; |
| } |
| av_freep(&fds); |
| } |
| } |
| n = poll(p, max_p, POLL_TIMEOUT_MS); |
| if (n > 0) { |
| int j = 1 - (tcp_fd == -1); |
| timeout_cnt = 0; |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| rtsp_st = rt->rtsp_streams[i]; |
| if (rtsp_st->rtp_handle) { |
| if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) { |
| ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size); |
| if (ret > 0) { |
| *prtsp_st = rtsp_st; |
| return ret; |
| } |
| } |
| j+=2; |
| } |
| } |
| #if CONFIG_RTSP_DEMUXER |
| if (tcp_fd != -1 && p[0].revents & POLLIN) { |
| if (rt->rtsp_flags & RTSP_FLAG_LISTEN) { |
| if (rt->state == RTSP_STATE_STREAMING) { |
| if (!ff_rtsp_parse_streaming_commands(s)) |
| return AVERROR_EOF; |
| else |
| av_log(s, AV_LOG_WARNING, |
| "Unable to answer to TEARDOWN\n"); |
| } else |
| return 0; |
| } else { |
| RTSPMessageHeader reply; |
| ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL); |
| if (ret < 0) |
| return ret; |
| /* XXX: parse message */ |
| if (rt->state != RTSP_STATE_STREAMING) |
| return 0; |
| } |
| } |
| #endif |
| } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) { |
| return AVERROR(ETIMEDOUT); |
| } else if (n < 0 && errno != EINTR) |
| return AVERROR(errno); |
| } |
| } |
| |
| static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st, |
| const uint8_t *buf, int len) |
| { |
| RTSPState *rt = s->priv_data; |
| int i; |
| if (len < 0) |
| return len; |
| if (rt->nb_rtsp_streams == 1) { |
| *rtsp_st = rt->rtsp_streams[0]; |
| return len; |
| } |
| if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) { |
| if (RTP_PT_IS_RTCP(rt->recvbuf[1])) { |
| int no_ssrc = 0; |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv; |
| if (!rtpctx) |
| continue; |
| if (rtpctx->ssrc == AV_RB32(&buf[4])) { |
| *rtsp_st = rt->rtsp_streams[i]; |
| return len; |
| } |
| if (!rtpctx->ssrc) |
| no_ssrc = 1; |
| } |
| if (no_ssrc) { |
| av_log(s, AV_LOG_WARNING, |
| "Unable to pick stream for packet - SSRC not known for " |
| "all streams\n"); |
| return AVERROR(EAGAIN); |
| } |
| } else { |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) { |
| *rtsp_st = rt->rtsp_streams[i]; |
| return len; |
| } |
| } |
| } |
| } |
| av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n"); |
| return AVERROR(EAGAIN); |
| } |
| |
| int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt) |
| { |
| RTSPState *rt = s->priv_data; |
| int ret, len; |
| RTSPStream *rtsp_st, *first_queue_st = NULL; |
| int64_t wait_end = 0; |
| |
| if (rt->nb_byes == rt->nb_rtsp_streams) |
| return AVERROR_EOF; |
| |
| /* get next frames from the same RTP packet */ |
| if (rt->cur_transport_priv) { |
| if (rt->transport == RTSP_TRANSPORT_RDT) { |
| ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0); |
| } else if (rt->transport == RTSP_TRANSPORT_RTP) { |
| ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0); |
| } else if (CONFIG_RTPDEC && rt->ts) { |
| ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos); |
| if (ret >= 0) { |
| rt->recvbuf_pos += ret; |
| ret = rt->recvbuf_pos < rt->recvbuf_len; |
| } |
| } else |
| ret = -1; |
| if (ret == 0) { |
| rt->cur_transport_priv = NULL; |
| return 0; |
| } else if (ret == 1) { |
| return 0; |
| } else |
| rt->cur_transport_priv = NULL; |
| } |
| |
| redo: |
| if (rt->transport == RTSP_TRANSPORT_RTP) { |
| int i; |
| int64_t first_queue_time = 0; |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv; |
| int64_t queue_time; |
| if (!rtpctx) |
| continue; |
| queue_time = ff_rtp_queued_packet_time(rtpctx); |
| if (queue_time && (queue_time - first_queue_time < 0 || |
| !first_queue_time)) { |
| first_queue_time = queue_time; |
| first_queue_st = rt->rtsp_streams[i]; |
| } |
| } |
| if (first_queue_time) { |
| wait_end = first_queue_time + s->max_delay; |
| } else { |
| wait_end = 0; |
| first_queue_st = NULL; |
| } |
| } |
| |
| /* read next RTP packet */ |
| if (!rt->recvbuf) { |
| rt->recvbuf = av_malloc(RECVBUF_SIZE); |
| if (!rt->recvbuf) |
| return AVERROR(ENOMEM); |
| } |
| |
| switch(rt->lower_transport) { |
| default: |
| #if CONFIG_RTSP_DEMUXER |
| case RTSP_LOWER_TRANSPORT_TCP: |
| len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE); |
| break; |
| #endif |
| case RTSP_LOWER_TRANSPORT_UDP: |
| case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: |
| len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end); |
| if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP) |
| ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len); |
| break; |
| case RTSP_LOWER_TRANSPORT_CUSTOM: |
| if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP && |
| wait_end && wait_end < av_gettime_relative()) |
| len = AVERROR(EAGAIN); |
| else |
| len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE); |
| len = pick_stream(s, &rtsp_st, rt->recvbuf, len); |
| if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP) |
| ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len); |
| break; |
| } |
| if (len == AVERROR(EAGAIN) && first_queue_st && |
| rt->transport == RTSP_TRANSPORT_RTP) { |
| av_log(s, AV_LOG_WARNING, |
| "max delay reached. need to consume packet\n"); |
| rtsp_st = first_queue_st; |
| ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0); |
| goto end; |
| } |
| if (len < 0) |
| return len; |
| if (len == 0) |
| return AVERROR_EOF; |
| if (rt->transport == RTSP_TRANSPORT_RDT) { |
| ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len); |
| } else if (rt->transport == RTSP_TRANSPORT_RTP) { |
| ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len); |
| if (rtsp_st->feedback) { |
| AVIOContext *pb = NULL; |
| if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM) |
| pb = s->pb; |
| ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb); |
| } |
| if (ret < 0) { |
| /* Either bad packet, or a RTCP packet. Check if the |
| * first_rtcp_ntp_time field was initialized. */ |
| RTPDemuxContext *rtpctx = rtsp_st->transport_priv; |
| if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) { |
| /* first_rtcp_ntp_time has been initialized for this stream, |
| * copy the same value to all other uninitialized streams, |
| * in order to map their timestamp origin to the same ntp time |
| * as this one. */ |
| int i; |
| AVStream *st = NULL; |
| if (rtsp_st->stream_index >= 0) |
| st = s->streams[rtsp_st->stream_index]; |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv; |
| AVStream *st2 = NULL; |
| if (rt->rtsp_streams[i]->stream_index >= 0) |
| st2 = s->streams[rt->rtsp_streams[i]->stream_index]; |
| if (rtpctx2 && st && st2 && |
| rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) { |
| rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time; |
| rtpctx2->rtcp_ts_offset = av_rescale_q( |
| rtpctx->rtcp_ts_offset, st->time_base, |
| st2->time_base); |
| } |
| } |
| // Make real NTP start time available in AVFormatContext |
| if (s->start_time_realtime == AV_NOPTS_VALUE) { |
| s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32); |
| if (rtpctx->st) { |
| s->start_time_realtime -= |
| av_rescale (rtpctx->rtcp_ts_offset, |
| (uint64_t) rtpctx->st->time_base.num * 1000000, |
| rtpctx->st->time_base.den); |
| } |
| } |
| } |
| if (ret == -RTCP_BYE) { |
| rt->nb_byes++; |
| |
| av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n", |
| rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams); |
| |
| if (rt->nb_byes == rt->nb_rtsp_streams) |
| return AVERROR_EOF; |
| } |
| } |
| } else if (CONFIG_RTPDEC && rt->ts) { |
| ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len); |
| if (ret >= 0) { |
| if (ret < len) { |
| rt->recvbuf_len = len; |
| rt->recvbuf_pos = ret; |
| rt->cur_transport_priv = rt->ts; |
| return 1; |
| } else { |
| ret = 0; |
| } |
| } |
| } else { |
| return AVERROR_INVALIDDATA; |
| } |
| end: |
| if (ret < 0) |
| goto redo; |
| if (ret == 1) |
| /* more packets may follow, so we save the RTP context */ |
| rt->cur_transport_priv = rtsp_st->transport_priv; |
| |
| return ret; |
| } |
| #endif /* CONFIG_RTPDEC */ |
| |
| #if CONFIG_SDP_DEMUXER |
| static int sdp_probe(AVProbeData *p1) |
| { |
| const char *p = p1->buf, *p_end = p1->buf + p1->buf_size; |
| |
| /* we look for a line beginning "c=IN IP" */ |
| while (p < p_end && *p != '\0') { |
| if (sizeof("c=IN IP") - 1 < p_end - p && |
| av_strstart(p, "c=IN IP", NULL)) |
| return AVPROBE_SCORE_EXTENSION; |
| |
| while (p < p_end - 1 && *p != '\n') p++; |
| if (++p >= p_end) |
| break; |
| if (*p == '\r') |
| p++; |
| } |
| return 0; |
| } |
| |
| static void append_source_addrs(char *buf, int size, const char *name, |
| int count, struct RTSPSource **addrs) |
| { |
| int i; |
| if (!count) |
| return; |
| av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr); |
| for (i = 1; i < count; i++) |
| av_strlcatf(buf, size, ",%s", addrs[i]->addr); |
| } |
| |
| static int sdp_read_header(AVFormatContext *s) |
| { |
| RTSPState *rt = s->priv_data; |
| RTSPStream *rtsp_st; |
| int size, i, err; |
| char *content; |
| char url[1024]; |
| |
| if (!ff_network_init()) |
| return AVERROR(EIO); |
| |
| if (s->max_delay < 0) /* Not set by the caller */ |
| s->max_delay = DEFAULT_REORDERING_DELAY; |
| if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO) |
| rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM; |
| |
| /* read the whole sdp file */ |
| /* XXX: better loading */ |
| content = av_malloc(SDP_MAX_SIZE); |
| if (!content) |
| return AVERROR(ENOMEM); |
| size = avio_read(s->pb, content, SDP_MAX_SIZE - 1); |
| if (size <= 0) { |
| av_free(content); |
| return AVERROR_INVALIDDATA; |
| } |
| content[size] ='\0'; |
| |
| err = ff_sdp_parse(s, content); |
| av_freep(&content); |
| if (err) goto fail; |
| |
| /* open each RTP stream */ |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| char namebuf[50]; |
| rtsp_st = rt->rtsp_streams[i]; |
| |
| if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) { |
| AVDictionary *opts = map_to_opts(rt); |
| |
| err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, |
| sizeof(rtsp_st->sdp_ip), |
| namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST); |
| if (err) { |
| av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err)); |
| err = AVERROR(EIO); |
| av_dict_free(&opts); |
| goto fail; |
| } |
| ff_url_join(url, sizeof(url), "rtp", NULL, |
| namebuf, rtsp_st->sdp_port, |
| "?localport=%d&ttl=%d&connect=%d&write_to_source=%d", |
| rtsp_st->sdp_port, rtsp_st->sdp_ttl, |
| rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0, |
| rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0); |
| |
| append_source_addrs(url, sizeof(url), "sources", |
| rtsp_st->nb_include_source_addrs, |
| rtsp_st->include_source_addrs); |
| append_source_addrs(url, sizeof(url), "block", |
| rtsp_st->nb_exclude_source_addrs, |
| rtsp_st->exclude_source_addrs); |
| err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE, |
| &s->interrupt_callback, &opts, s->protocol_whitelist); |
| |
| av_dict_free(&opts); |
| |
| if (err < 0) { |
| err = AVERROR_INVALIDDATA; |
| goto fail; |
| } |
| } |
| if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st))) |
| goto fail; |
| } |
| return 0; |
| fail: |
| ff_rtsp_close_streams(s); |
| ff_network_close(); |
| return err; |
| } |
| |
| static int sdp_read_close(AVFormatContext *s) |
| { |
| ff_rtsp_close_streams(s); |
| ff_network_close(); |
| return 0; |
| } |
| |
| static const AVClass sdp_demuxer_class = { |
| .class_name = "SDP demuxer", |
| .item_name = av_default_item_name, |
| .option = sdp_options, |
| .version = LIBAVUTIL_VERSION_INT, |
| }; |
| |
| AVInputFormat ff_sdp_demuxer = { |
| .name = "sdp", |
| .long_name = NULL_IF_CONFIG_SMALL("SDP"), |
| .priv_data_size = sizeof(RTSPState), |
| .read_probe = sdp_probe, |
| .read_header = sdp_read_header, |
| .read_packet = ff_rtsp_fetch_packet, |
| .read_close = sdp_read_close, |
| .priv_class = &sdp_demuxer_class, |
| }; |
| #endif /* CONFIG_SDP_DEMUXER */ |
| |
| #if CONFIG_RTP_DEMUXER |
| static int rtp_probe(AVProbeData *p) |
| { |
| if (av_strstart(p->filename, "rtp:", NULL)) |
| return AVPROBE_SCORE_MAX; |
| return 0; |
| } |
| |
| static int rtp_read_header(AVFormatContext *s) |
| { |
| uint8_t recvbuf[RTP_MAX_PACKET_LENGTH]; |
| char host[500], sdp[500]; |
| int ret, port; |
| URLContext* in = NULL; |
| int payload_type; |
| AVCodecContext codec = { 0 }; |
| struct sockaddr_storage addr; |
| AVIOContext pb; |
| socklen_t addrlen = sizeof(addr); |
| RTSPState *rt = s->priv_data; |
| |
| if (!ff_network_init()) |
| return AVERROR(EIO); |
| |
| ret = ffurl_open_whitelist(&in, s->filename, AVIO_FLAG_READ, |
| &s->interrupt_callback, NULL, s->protocol_whitelist); |
| if (ret) |
| goto fail; |
| |
| while (1) { |
| ret = ffurl_read(in, recvbuf, sizeof(recvbuf)); |
| if (ret == AVERROR(EAGAIN)) |
| continue; |
| if (ret < 0) |
| goto fail; |
| if (ret < 12) { |
| av_log(s, AV_LOG_WARNING, "Received too short packet\n"); |
| continue; |
| } |
| |
| if ((recvbuf[0] & 0xc0) != 0x80) { |
| av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet " |
| "received\n"); |
| continue; |
| } |
| |
| if (RTP_PT_IS_RTCP(recvbuf[1])) |
| continue; |
| |
| payload_type = recvbuf[1] & 0x7f; |
| break; |
| } |
| getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen); |
| ffurl_close(in); |
| in = NULL; |
| |
| if (ff_rtp_get_codec_info(&codec, payload_type)) { |
| av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d " |
| "without an SDP file describing it\n", |
| payload_type); |
| goto fail; |
| } |
| if (codec.codec_type != AVMEDIA_TYPE_DATA) { |
| av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received " |
| "properly you need an SDP file " |
| "describing it\n"); |
| } |
| |
| av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port, |
| NULL, 0, s->filename); |
| |
| snprintf(sdp, sizeof(sdp), |
| "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n", |
| addr.ss_family == AF_INET ? 4 : 6, host, |
| codec.codec_type == AVMEDIA_TYPE_DATA ? "application" : |
| codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio", |
| port, payload_type); |
| av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp); |
| |
| ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL); |
| s->pb = &pb; |
| |
| /* sdp_read_header initializes this again */ |
| ff_network_close(); |
| |
| rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1; |
| |
| ret = sdp_read_header(s); |
| s->pb = NULL; |
| return ret; |
| |
| fail: |
| if (in) |
| ffurl_close(in); |
| ff_network_close(); |
| return ret; |
| } |
| |
| static const AVClass rtp_demuxer_class = { |
| .class_name = "RTP demuxer", |
| .item_name = av_default_item_name, |
| .option = rtp_options, |
| .version = LIBAVUTIL_VERSION_INT, |
| }; |
| |
| AVInputFormat ff_rtp_demuxer = { |
| .name = "rtp", |
| .long_name = NULL_IF_CONFIG_SMALL("RTP input"), |
| .priv_data_size = sizeof(RTSPState), |
| .read_probe = rtp_probe, |
| .read_header = rtp_read_header, |
| .read_packet = ff_rtsp_fetch_packet, |
| .read_close = sdp_read_close, |
| .flags = AVFMT_NOFILE, |
| .priv_class = &rtp_demuxer_class, |
| }; |
| #endif /* CONFIG_RTP_DEMUXER */ |