diff options
Diffstat (limited to 'ffmpeg/libavformat/rtsp.c')
| -rw-r--r-- | ffmpeg/libavformat/rtsp.c | 2243 |
1 files changed, 2243 insertions, 0 deletions
diff --git a/ffmpeg/libavformat/rtsp.c b/ffmpeg/libavformat/rtsp.c new file mode 100644 index 0000000..317893c --- /dev/null +++ b/ffmpeg/libavformat/rtsp.c @@ -0,0 +1,2243 @@ +/* + * 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 "rdt.h" +#include "rtpdec_formats.h" +#include "rtpenc_chain.h" +#include "url.h" +#include "rtpenc.h" +#include "mpegts.h" + +//#define DEBUG + +/* 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" }, \ + { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 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_DATA+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" } + +#define RTSP_REORDERING_OPTS() \ + { "reorder_queue_size", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC } + +const AVOption ff_rtsp_options[] = { + { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 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, {.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", "RTSP flags"), + RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"), + { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC }, + { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC }, + { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC }, + RTSP_REORDERING_OPTS(), + { NULL }, +}; + +static const AVOption sdp_options[] = { + RTSP_FLAG_OPTS("sdp_flags", "SDP flags"), + { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" }, + RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"), + RTSP_REORDERING_OPTS(), + { NULL }, +}; + +static const AVOption rtp_options[] = { + RTSP_FLAG_OPTS("rtp_flags", "RTP flags"), + RTSP_REORDERING_OPTS(), + { NULL }, +}; + +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); + av_parse_time(start, buf, 1); + if (*p == '-') { + p++; + get_word_sep(buf, sizeof(buf), "-", &p); + av_parse_time(end, buf, 1); + } +} + +static int get_sockaddr(const char *buf, struct sockaddr_storage *sock) +{ + struct addrinfo hints = { 0 }, *ai = NULL; + 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; + if (codec) + codec->codec_id = handler->codec_id; + rtsp_st->dynamic_handler = handler; + if (handler->alloc) { + rtsp_st->dynamic_protocol_context = handler->alloc(); + if (!rtsp_st->dynamic_protocol_context) + rtsp_st->dynamic_handler = NULL; + } +} + +/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */ +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, 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 + * 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; + } + if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) + rtsp_st->dynamic_handler->init(s, st->index, + rtsp_st->dynamic_protocol_context); + return 0; +} + +/* 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 +} SDPParseState; + +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, i; + AVStream *st; + RTSPStream *rtsp_st; + struct sockaddr_storage sdp_ip; + int ttl; + + av_dlog(s, "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(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; + 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; + } + 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; + + 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 (!rt->ts && CONFIG_RTPDEC) + rt->ts = ff_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); + if (handler && handler->init) + handler->init(s, -1, rtsp_st->dynamic_protocol_context); + } + } 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->codec); + if (handler && handler->init) + handler->init(s, st->index, + rtsp_st->dynamic_protocol_context); + } + } + /* 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); + } + } 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 < 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) + rtsp_st->dynamic_handler->parse_sdp_a_line(s, i, + rtsp_st->dynamic_protocol_context, buf); + } + } 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, "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 (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; + /* 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++; + } + 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 (rt->transport == RTSP_TRANSPORT_RTP && 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) +{ + RTSPState *rt = s->priv_data; + 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->dynamic_handler && rtsp_st->dynamic_protocol_context) + rtsp_st->dynamic_handler->free( + rtsp_st->dynamic_protocol_context); + av_free(rtsp_st); + } + } + av_free(rt->rtsp_streams); + if (rt->asf_ctx) { + avformat_close_input(&rt->asf_ctx); + } + if (rt->ts && CONFIG_RTPDEC) + ff_mpegts_parse_close(rt->ts); + av_free(rt->p); + av_free(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 (s->oformat && CONFIG_RTSP_MUXER) { + 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; + } else if (rt->transport == RTSP_TRANSPORT_RAW) { + return 0; // Don't need to open any parser here + } 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 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 (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) { + 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(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(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++; + } +} + +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, + 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(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_dlog(s, "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_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch); + if (ret != 1) + return AVERROR_EOF; + if (ch == '\n') + break; + if (ch == '$') { + /* XXX: only parse it if first char on line ? */ + 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_dlog(s, "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(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); + ffurl_read_complete(rt->rtsp_hd, content, content_length); + content[content_length] = '\0'; + } + if (content_ptr) + *content_ptr = content; + else + av_free(content); + + if (request) { + char buf[1024]; + char base64buf[AV_BASE64_SIZE(sizeof(buf))]; + const char* ptr = buf; + + if (!strcmp(reply->reason, "OPTIONS")) { + snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n"); + if (reply->seq) + av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq); + if (reply->session_id[0]) + av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", + reply->session_id); + } else { + snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n"); + } + av_strlcat(buf, "\r\n", sizeof(buf)); + + if (rt->control_transport == RTSP_MODE_TUNNEL) { + av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf)); + ptr = base64buf; + } + ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr)); + + rt->last_cmd_time = av_gettime(); + /* Even if the request from the server had data, it is not the data + * that the caller wants or expects. The memory could also be leaked + * if the actual following reply has content data. */ + if (content_ptr) + av_freep(content_ptr); + /* If method is set, this is called from ff_rtsp_send_cmd, + * where a reply to exactly this request is awaited. For + * callers from within packet receiving, we just want to + * return to the caller and go back to receiving packets. */ + if (method) + goto start; + return 0; + } + + if (rt->seq != reply->seq) { + av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n", + rt->seq, reply->seq); + } + + /* 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 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; + 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); + 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_dlog(s, "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(); + + return 0; +} + +int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, + const char *url, const char *headers) +{ + return ff_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 = ff_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) { + ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1, + "?localport=%d", j); + /* we will use two ports per rtp stream (rtp and rtcp) */ + j += 2; + if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE, + &s->interrupt_callback, NULL)) + goto rtp_opened; + } + av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n"); + 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 (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); + 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 = 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] = ""; + + 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, "%s", options); + } else { + ff_url_join(url, sizeof(url), "rtp", NULL, host, + 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; + } + /* Try to initialize the connection state in a + * 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 && + CONFIG_RTPDEC) + ff_rtp_send_punch_packets(rtsp_st->rtp_handle); + 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(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE, + &s->interrupt_callback, NULL) < 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); + 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 host[1024], path[1024], tcpname[1024], cmd[2048], auth[128]; + int port, err, tcp_fd; + RTSPMessageHeader reply1 = {0}, *reply = &reply1; + int lower_transport_mask = 0; + 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: + 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 (*auth) { + av_strlcpy(rt->auth, auth, sizeof(rt->auth)); + } + if (port < 0) + port = RTSP_DEFAULT_PORT; + + 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), "rtsp", 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 { + /* open the tcp connection */ + ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL); + if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE, + &s->interrupt_callback, NULL) < 0) { + err = AVERROR(EIO); + goto fail; + } + rt->rtsp_hd_out = rt->rtsp_hd; + } + rt->seq = 0; + + 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); + } + + /* 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 = 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 (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; + + do { + int lower_transport = ff_log2_tab[lower_transport_mask & + ~(lower_transport_mask - 1)]; + + 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)); + 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() < 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_free(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 (rt->ts && CONFIG_RTPDEC) { + ret = ff_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()) + 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) { + 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); + } + } + } + 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 (rt->ts && CONFIG_RTPDEC) { + ret = ff_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 (p + sizeof("c=IN IP") - 1 < p_end && + av_strstart(p, "c=IN IP", NULL)) + return AVPROBE_SCORE_MAX / 2; + + while (p < p_end - 1 && *p != '\n') p++; + if (++p >= p_end) + break; + if (*p == '\r') + p++; + } + return 0; +} + +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); + 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_free(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)) { + getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip), + namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST); + ff_url_join(url, sizeof(url), "rtp", NULL, + namebuf, rtsp_st->sdp_port, + "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port, + rtsp_st->sdp_ttl, + rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0); + if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE, + &s->interrupt_callback, NULL) < 0) { + err = AVERROR_INVALIDDATA; + goto fail; + } + } + if ((err = 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[1500]; + 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(&in, s->filename, AVIO_FLAG_READ, + &s->interrupt_callback, NULL); + 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_DATA+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 */ |