X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtsp.c;h=a8c5a641cbcebbdb6740b813574d89f759402ffd;hb=6f34f031908b8f16482e951ee5232116fb42b46a;hp=975637cf54c0c75fbb315889cbf6eff07c6bed63;hpb=39278ff0de5b5e3397c22538978bffbb38ee099b;p=ffmpeg diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c index 975637cf54c..a8c5a641cbc 100644 --- a/libavformat/rtsp.c +++ b/libavformat/rtsp.c @@ -21,6 +21,7 @@ #include "libavutil/avassert.h" #include "libavutil/base64.h" +#include "libavutil/bprint.h" #include "libavutil/avstring.h" #include "libavutil/intreadwrite.h" #include "libavutil/mathematics.h" @@ -50,12 +51,8 @@ #include "rtpenc.h" #include "mpegts.h" -/* Timeout values for socket poll, in ms, - * and read_packet(), in seconds */ -#define POLL_TIMEOUT_MS 100 +/* Default timeout values for read packet in seconds */ #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 @@ -76,7 +73,8 @@ #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 } \ + { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC }, \ + { "pkt_size", "Underlying protocol send packet size", OFFSET(pkt_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, ENC } \ const AVOption ff_rtsp_options[] = { @@ -87,24 +85,18 @@ const AVOption ff_rtsp_options[] = { { "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" }, + { "https", "HTTPS tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTPS )}, 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" }, + { "satip_raw", "export raw MPEG-TS stream instead of demuxing", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_SATIP_RAW}, 0, 0, DEC, "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 }, { "listen_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 }, -#if FF_API_OLD_RTSP_OPTIONS - { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen) (deprecated, use listen_timeout)", 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 }, -#else { "timeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC }, -#endif COMMON_OPTS(), { "user_agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC }, -#if FF_API_OLD_RTSP_OPTIONS - { "user-agent", "override User-Agent header (deprecated, use user_agent)", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC }, -#endif { NULL }, }; @@ -112,6 +104,7 @@ 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" }, + { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = READ_PACKET_TIMEOUT_S}, INT_MIN, INT_MAX, DEC }, RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"), COMMON_OPTS(), { NULL }, @@ -119,6 +112,8 @@ static const AVOption sdp_options[] = { static const AVOption rtp_options[] = { RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"), + { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = READ_PACKET_TIMEOUT_S}, INT_MIN, INT_MAX, DEC }, + RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"), COMMON_OPTS(), { NULL }, }; @@ -131,6 +126,8 @@ static AVDictionary *map_to_opts(RTSPState *rt) snprintf(buf, sizeof(buf), "%d", rt->buffer_size); av_dict_set(&opts, "buffer_size", buf, 0); + snprintf(buf, sizeof(buf), "%d", rt->pkt_size); + av_dict_set(&opts, "pkt_size", buf, 0); return opts; } @@ -248,6 +245,35 @@ static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st, } } +static int init_satip_stream(AVFormatContext *s) +{ + RTSPState *rt = s->priv_data; + RTSPStream *rtsp_st = av_mallocz(sizeof(RTSPStream)); + if (!rtsp_st) + return AVERROR(ENOMEM); + dynarray_add(&rt->rtsp_streams, + &rt->nb_rtsp_streams, rtsp_st); + + rtsp_st->sdp_payload_type = 33; // MP2T + av_strlcpy(rtsp_st->control_url, + rt->control_uri, sizeof(rtsp_st->control_url)); + + if (rt->rtsp_flags & RTSP_FLAG_SATIP_RAW) { + AVStream *st = avformat_new_stream(s, NULL); + if (!st) + return AVERROR(ENOMEM); + st->id = rt->nb_rtsp_streams - 1; + rtsp_st->stream_index = st->index; + st->codecpar->codec_type = AVMEDIA_TYPE_DATA; + st->codecpar->codec_id = AV_CODEC_ID_MPEG2TS; + } else { + rtsp_st->stream_index = -1; + init_rtp_handler(&ff_mpegts_dynamic_handler, rtsp_st, NULL); + finalize_rtp_handler_init(s, rtsp_st, NULL); + } + return 0; +} + /* parse the rtpmap description: /[/] */ static int sdp_parse_rtpmap(AVFormatContext *s, AVStream *st, RTSPStream *rtsp_st, @@ -537,7 +563,7 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1, break; case 'a': if (av_strstart(p, "control:", &p)) { - if (s->nb_streams == 0) { + if (rt->nb_rtsp_streams == 0) { if (!strncmp(p, "rtsp://", 7)) av_strlcpy(rt->control_uri, p, sizeof(rt->control_uri)); @@ -683,15 +709,7 @@ int ff_sdp_parse(AVFormatContext *s, const char *content) { 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; + char buf[SDP_MAX_SIZE], *q; SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state; p = content; @@ -758,9 +776,7 @@ void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets) 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; + ffurl_closep(&rtsp_st->rtp_handle); } } @@ -1031,7 +1047,7 @@ static void handle_rtp_info(RTSPState *rt, const char *url, static void rtsp_parse_rtp_info(RTSPState *rt, const char *p) { int read = 0; - char key[20], value[1024], url[1024] = ""; + char key[20], value[MAX_URL_SIZE], url[MAX_URL_SIZE] = ""; uint32_t seq = 0, rtptime = 0; for (;;) { @@ -1122,6 +1138,9 @@ void ff_rtsp_parse_line(AVFormatContext *s, } else if (av_stristart(p, "Content-Type:", &p)) { p += strspn(p, SPACE_CHARS); av_strlcpy(reply->content_type, p, sizeof(reply->content_type)); + } else if (av_stristart(p, "com.ses.streamID:", &p)) { + p += strspn(p, SPACE_CHARS); + av_strlcpy(reply->stream_id, p, sizeof(reply->stream_id)); } } @@ -1130,7 +1149,7 @@ void ff_rtsp_skip_packet(AVFormatContext *s) { RTSPState *rt = s->priv_data; int ret, len, len1; - uint8_t buf[1024]; + uint8_t buf[MAX_URL_SIZE]; ret = ffurl_read_complete(rt->rtsp_hd, buf, 3); if (ret != 3) @@ -1156,7 +1175,7 @@ int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, int return_on_interleaved_data, const char *method) { RTSPState *rt = s->priv_data; - char buf[4096], buf1[1024], *q; + char buf[MAX_URL_SIZE], buf1[MAX_URL_SIZE], *q; unsigned char ch; const char *p; int ret, content_length, line_count = 0, request = 0; @@ -1226,7 +1245,8 @@ start: content = av_malloc(content_length + 1); if (!content) return AVERROR(ENOMEM); - ffurl_read_complete(rt->rtsp_hd, content, content_length); + if (ffurl_read_complete(rt->rtsp_hd, content, content_length) != content_length) + return AVERROR(EIO); content[content_length] = '\0'; } if (content_ptr) @@ -1235,7 +1255,7 @@ start: av_freep(&content); if (request) { - char buf[1024]; + char buf[MAX_URL_SIZE]; char base64buf[AV_BASE64_SIZE(sizeof(buf))]; const char* ptr = buf; @@ -1311,9 +1331,12 @@ static int rtsp_send_cmd_with_content_async(AVFormatContext *s, int send_content_length) { RTSPState *rt = s->priv_data; - char buf[4096], *out_buf; + char buf[MAX_URL_SIZE], *out_buf; char base64buf[AV_BASE64_SIZE(sizeof(buf))]; + if (!rt->rtsp_hd_out) + return AVERROR(ENOTCONN); + /* Add in RTSP headers */ out_buf = buf; rt->seq++; @@ -1418,7 +1441,7 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, int rtx = 0, j, i, err, interleave = 0, port_off; RTSPStream *rtsp_st; RTSPMessageHeader reply1, *reply = &reply1; - char cmd[2048]; + char cmd[MAX_URL_SIZE]; const char *trans_pref; if (rt->transport == RTSP_TRANSPORT_RDT) @@ -1439,7 +1462,7 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, port_off -= port_off & 0x01; for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) { - char transport[2048]; + char transport[MAX_URL_SIZE]; /* * WMS serves all UDP data over a single connection, the RTX, which @@ -1497,8 +1520,10 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, rtp_opened: port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle); have_port: - snprintf(transport, sizeof(transport) - 1, - "%s/UDP;", trans_pref); + av_strlcpy(transport, trans_pref, sizeof(transport)); + av_strlcat(transport, + rt->server_type == RTSP_SERVER_SATIP ? ";" : "/UDP;", + sizeof(transport)); if (rt->server_type != RTSP_SERVER_REAL) av_strlcat(transport, "unicast;", sizeof(transport)); av_strlcatf(transport, sizeof(transport), @@ -1561,6 +1586,15 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, goto fail; } + if (rt->server_type == RTSP_SERVER_SATIP && reply->stream_id[0]) { + char proto[128], host[128], path[512], auth[128]; + int port; + av_url_split(proto, sizeof(proto), auth, sizeof(auth), host, sizeof(host), + &port, path, sizeof(path), rt->control_uri); + ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL, host, + port, "/stream=%s", reply->stream_id); + } + /* XXX: same protocol for all streams is required */ if (i > 0) { if (reply->transports[0].lower_transport != rt->lower_transport || @@ -1588,7 +1622,7 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, break; case RTSP_LOWER_TRANSPORT_UDP: { - char url[1024], options[30] = ""; + char url[MAX_URL_SIZE], options[30] = ""; const char *peer = host; if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC) @@ -1606,9 +1640,10 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, break; } case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: { - char url[1024], namebuf[50], optbuf[20] = ""; + char url[MAX_URL_SIZE], namebuf[50], optbuf[20] = ""; struct sockaddr_storage addr; int port, ttl; + AVDictionary *opts = map_to_opts(rt); if (reply->transports[0].destination.ss_family) { addr = reply->transports[0].destination; @@ -1625,8 +1660,11 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, 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, s->protocol_blacklist, NULL) < 0) { + err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE, + &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL); + av_dict_free(&opts); + + if (err < 0) { err = AVERROR_INVALIDDATA; goto fail; } @@ -1654,21 +1692,23 @@ fail: 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; + if (rt->rtsp_hd_out != rt->rtsp_hd) + ffurl_closep(&rt->rtsp_hd_out); + rt->rtsp_hd_out = NULL; + ffurl_closep(&rt->rtsp_hd); } 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]; + char tcpname[1024], cmd[MAX_URL_SIZE], auth[128]; const char *lower_rtsp_proto = "tcp"; int port, err, tcp_fd; RTSPMessageHeader reply1, *reply = &reply1; int lower_transport_mask = 0; int default_port = RTSP_DEFAULT_PORT; + int https_tunnel = 0; char real_challenge[64] = ""; struct sockaddr_storage peer; socklen_t peer_len = sizeof(peer); @@ -1687,7 +1727,9 @@ int ff_rtsp_connect(AVFormatContext *s) 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)) { + if (rt->lower_transport_mask & ((1 << RTSP_LOWER_TRANSPORT_HTTP) | + (1 << RTSP_LOWER_TRANSPORT_HTTPS))) { + https_tunnel = !!(rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTPS)); rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP; rt->control_transport = RTSP_MODE_TUNNEL; } @@ -1704,6 +1746,9 @@ redirect: lower_rtsp_proto = "tls"; default_port = RTSPS_DEFAULT_PORT; rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP; + } else if (!strcmp(proto, "satip")) { + av_strlcpy(proto, "rtsp", sizeof(proto)); + rt->server_type = RTSP_SERVER_SATIP; } if (*auth) { @@ -1740,8 +1785,11 @@ redirect: char httpname[1024]; char sessioncookie[17]; char headers[1024]; + AVDictionary *options = NULL; + + av_dict_set_int(&options, "timeout", rt->stimeout, 0); - ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path); + ff_url_join(httpname, sizeof(httpname), https_tunnel ? "https" : "http", auth, host, port, "%s", path); snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x", av_get_random_seed(), av_get_random_seed()); @@ -1770,7 +1818,8 @@ redirect: } /* complete the connection */ - if (ffurl_connect(rt->rtsp_hd, NULL)) { + if (ffurl_connect(rt->rtsp_hd, &options)) { + av_dict_free(&options); err = AVERROR(EIO); goto fail; } @@ -1793,6 +1842,7 @@ redirect: 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); + av_opt_set(rt->rtsp_hd_out->priv_data, "send_expect_100", "0", 0); /* Initialize the authentication state for the POST session. The HTTP * protocol implementation doesn't properly handle multi-pass @@ -1813,10 +1863,12 @@ redirect: ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd); /* complete the connection */ - if (ffurl_connect(rt->rtsp_hd_out, NULL)) { + if (ffurl_connect(rt->rtsp_hd_out, &options)) { + av_dict_free(&options); err = AVERROR(EIO); goto fail; } + av_dict_free(&options); } else { int ret; /* open the tcp connection */ @@ -1844,7 +1896,9 @@ redirect: /* request options supported by the server; this also detects server * type */ - for (rt->server_type = RTSP_SERVER_RTP;;) { + if (rt->server_type != RTSP_SERVER_SATIP) + rt->server_type = RTSP_SERVER_RTP; + for (;;) { cmd[0] = 0; if (rt->server_type == RTSP_SERVER_REAL) av_strlcat(cmd, @@ -1879,9 +1933,15 @@ redirect: break; } - if (CONFIG_RTSP_DEMUXER && s->iformat) - err = ff_rtsp_setup_input_streams(s, reply); - else if (CONFIG_RTSP_MUXER) +#if CONFIG_RTSP_DEMUXER + if (s->iformat) { + if (rt->server_type == RTSP_SERVER_SATIP) + err = init_satip_stream(s); + else + err = ff_rtsp_setup_input_streams(s, reply); + } else +#endif + if (CONFIG_RTSP_MUXER) err = ff_rtsp_setup_output_streams(s, host); else av_assert0(0); @@ -1943,13 +2003,9 @@ static int parse_rtsp_message(AVFormatContext *s) 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"); + return ff_rtsp_parse_streaming_commands(s); } else - return 0; + return AVERROR_EOF; } else { RTSPMessageHeader reply; ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL); @@ -1968,12 +2024,13 @@ static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, { RTSPState *rt = s->priv_data; RTSPStream *rtsp_st; - int n, i, ret, timeout_cnt = 0; + int n, i, ret; struct pollfd *p = rt->p; int *fds = NULL, fdsnum, fdsidx; + int runs = rt->initial_timeout * 1000LL / POLLING_TIME; if (!p) { - p = rt->p = av_malloc_array(2 * (rt->nb_rtsp_streams + 1), sizeof(struct pollfd)); + p = rt->p = av_malloc_array(2 * rt->nb_rtsp_streams + 1, sizeof(*p)); if (!p) return AVERROR(ENOMEM); @@ -2008,10 +2065,9 @@ static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, return AVERROR_EXIT; if (wait_end && wait_end - av_gettime_relative() < 0) return AVERROR(EAGAIN); - n = poll(p, rt->max_p, POLL_TIMEOUT_MS); + n = poll(p, rt->max_p, POLLING_TIME); if (n > 0) { int j = rt->rtsp_hd ? 1 : 0; - timeout_cnt = 0; for (i = 0; i < rt->nb_rtsp_streams; i++) { rtsp_st = rt->rtsp_streams[i]; if (rtsp_st->rtp_handle) { @@ -2032,7 +2088,7 @@ static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, } } #endif - } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) { + } else if (n == 0 && rt->initial_timeout > 0 && --runs <= 0) { return AVERROR(ETIMEDOUT); } else if (n < 0 && errno != EINTR) return AVERROR(errno); @@ -2238,9 +2294,7 @@ redo: 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); + av_rescale_q (rtpctx->rtcp_ts_offset, rtpctx->st->time_base, AV_TIME_BASE_Q); } } } @@ -2281,7 +2335,7 @@ end: #endif /* CONFIG_RTPDEC */ #if CONFIG_SDP_DEMUXER -static int sdp_probe(AVProbeData *p1) +static int sdp_probe(const AVProbeData *p1) { const char *p = p1->buf, *p_end = p1->buf + p1->buf_size; @@ -2317,7 +2371,7 @@ static int sdp_read_header(AVFormatContext *s) RTSPStream *rtsp_st; int size, i, err; char *content; - char url[1024]; + char url[MAX_URL_SIZE]; if (!ff_network_init()) return AVERROR(EIO); @@ -2330,11 +2384,14 @@ static int sdp_read_header(AVFormatContext *s) /* read the whole sdp file */ /* XXX: better loading */ content = av_malloc(SDP_MAX_SIZE); - if (!content) + if (!content) { + ff_network_close(); return AVERROR(ENOMEM); + } size = avio_read(s->pb, content, SDP_MAX_SIZE - 1); if (size <= 0) { av_free(content); + ff_network_close(); return AVERROR_INVALIDDATA; } content[size] ='\0'; @@ -2420,7 +2477,7 @@ AVInputFormat ff_sdp_demuxer = { #endif /* CONFIG_SDP_DEMUXER */ #if CONFIG_RTP_DEMUXER -static int rtp_probe(AVProbeData *p) +static int rtp_probe(const AVProbeData *p) { if (av_strstart(p->filename, "rtp:", NULL)) return AVPROBE_SCORE_MAX; @@ -2430,7 +2487,7 @@ static int rtp_probe(AVProbeData *p) static int rtp_read_header(AVFormatContext *s) { uint8_t recvbuf[RTP_MAX_PACKET_LENGTH]; - char host[500], sdp[500]; + char host[500], filters_buf[1000]; int ret, port; URLContext* in = NULL; int payload_type; @@ -2439,12 +2496,17 @@ static int rtp_read_header(AVFormatContext *s) AVIOContext pb; socklen_t addrlen = sizeof(addr); RTSPState *rt = s->priv_data; + const char *p; + AVBPrint sdp; + AVDictionary *opts = NULL; if (!ff_network_init()) return AVERROR(EIO); + opts = map_to_opts(rt); ret = ffurl_open_whitelist(&in, s->url, AVIO_FLAG_READ, - &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL); + &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL); + av_dict_free(&opts); if (ret) goto fail; @@ -2472,8 +2534,7 @@ static int rtp_read_header(AVFormatContext *s) break; } getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen); - ffurl_close(in); - in = NULL; + ffurl_closep(&in); par = avcodec_parameters_alloc(); if (!par) { @@ -2485,6 +2546,7 @@ static int rtp_read_header(AVFormatContext *s) av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d " "without an SDP file describing it\n", payload_type); + ret = AVERROR_INVALIDDATA; goto fail; } if (par->codec_type != AVMEDIA_TYPE_DATA) { @@ -2496,31 +2558,60 @@ static int rtp_read_header(AVFormatContext *s) av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port, NULL, 0, s->url); - 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, - par->codec_type == AVMEDIA_TYPE_DATA ? "application" : - par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio", - port, payload_type); - av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp); + av_bprint_init(&sdp, 0, AV_BPRINT_SIZE_UNLIMITED); + av_bprintf(&sdp, "v=0\r\nc=IN IP%d %s\r\n", + addr.ss_family == AF_INET ? 4 : 6, host); + + p = strchr(s->url, '?'); + if (p) { + static const char filters[][2][8] = { { "sources", "incl" }, + { "block", "excl" } }; + int i; + char *q; + for (i = 0; i < FF_ARRAY_ELEMS(filters); i++) { + if (av_find_info_tag(filters_buf, sizeof(filters_buf), filters[i][0], p)) { + q = filters_buf; + while ((q = strchr(q, ',')) != NULL) + *q = ' '; + av_bprintf(&sdp, "a=source-filter:%s IN IP%d %s %s\r\n", + filters[i][1], + addr.ss_family == AF_INET ? 4 : 6, host, + filters_buf); + } + } + } + + av_bprintf(&sdp, "m=%s %d RTP/AVP %d\r\n", + par->codec_type == AVMEDIA_TYPE_DATA ? "application" : + par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio", + port, payload_type); + av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp.str); + if (!av_bprint_is_complete(&sdp)) + goto fail_nobuf; avcodec_parameters_free(&par); - ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL); + ffio_init_context(&pb, sdp.str, sdp.len, 0, NULL, NULL, NULL, NULL); s->pb = &pb; - /* sdp_read_header initializes this again */ + /* if sdp_read_header() fails then following ff_network_close() cancels out */ + /* ff_network_init() at the start of this function. Otherwise it cancels out */ + /* ff_network_init() inside sdp_read_header() */ ff_network_close(); rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1; ret = sdp_read_header(s); s->pb = NULL; + av_bprint_finalize(&sdp, NULL); return ret; +fail_nobuf: + ret = AVERROR(ENOMEM); + av_log(s, AV_LOG_ERROR, "rtp_read_header(): not enough buffer space for sdp-headers\n"); + av_bprint_finalize(&sdp, NULL); fail: avcodec_parameters_free(&par); - if (in) - ffurl_close(in); + ffurl_closep(&in); ff_network_close(); return ret; }