| /* |
| * 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/base64.h" |
| #include "libavutil/avstring.h" |
| #include "libavutil/intreadwrite.h" |
| #include "avformat.h" |
| |
| #include <sys/time.h> |
| #if HAVE_SYS_SELECT_H |
| #include <sys/select.h> |
| #endif |
| #include <strings.h> |
| #include "internal.h" |
| #include "network.h" |
| #include "os_support.h" |
| #include "rtsp.h" |
| |
| #include "rtpdec.h" |
| #include "rdt.h" |
| #include "rtpdec_asf.h" |
| |
| //#define DEBUG |
| //#define DEBUG_RTP_TCP |
| |
| #if LIBAVFORMAT_VERSION_INT < (53 << 16) |
| int rtsp_default_protocols = (1 << RTSP_LOWER_TRANSPORT_UDP); |
| #endif |
| |
| /* Timeout values for socket select, in ms, |
| * and read_packet(), in seconds */ |
| #define SELECT_TIMEOUT_MS 100 |
| #define READ_PACKET_TIMEOUT_S 10 |
| #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / SELECT_TIMEOUT_MS |
| |
| #define SPACE_CHARS " \t\r\n" |
| /* we use memchr() instead of strchr() here because strchr() will return |
| * the terminating '\0' of SPACE_CHARS instead of NULL if c is '\0'. */ |
| #define redir_isspace(c) memchr(SPACE_CHARS, c, 4) |
| static void skip_spaces(const char **pp) |
| { |
| const char *p; |
| p = *pp; |
| while (redir_isspace(*p)) |
| p++; |
| *pp = p; |
| } |
| |
| static void get_word_until_chars(char *buf, int buf_size, |
| const char *sep, const char **pp) |
| { |
| const char *p; |
| char *q; |
| |
| p = *pp; |
| skip_spaces(&p); |
| 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 the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */ |
| static int sdp_parse_rtpmap(AVFormatContext *s, |
| AVCodecContext *codec, RTSPStream *rtsp_st, |
| int payload_type, const char *p) |
| { |
| char buf[256]; |
| int i; |
| AVCodec *c; |
| const char *c_name; |
| |
| /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and |
| * 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) { |
| RTPDynamicProtocolHandler *handler; |
| for (handler = RTPFirstDynamicPayloadHandler; |
| handler; handler = handler->next) { |
| if (!strcasecmp(buf, handler->enc_name) && |
| codec->codec_type == handler->codec_type) { |
| codec->codec_id = handler->codec_id; |
| rtsp_st->dynamic_handler = handler; |
| if (handler->open) |
| rtsp_st->dynamic_protocol_context = handler->open(); |
| break; |
| } |
| } |
| } else { |
| /* We are in a standard case |
| * (from http://www.iana.org/assignments/rtp-parameters). */ |
| /* search into AVRtpPayloadTypes[] */ |
| codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type); |
| } |
| |
| c = avcodec_find_decoder(codec->codec_id); |
| 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; |
| get_word_sep(buf, sizeof(buf), "/", &p); |
| i = atoi(buf); |
| if (i > 0) |
| codec->channels = i; |
| // TODO: there is a bug here; if it is a mono stream, and |
| // less than 22000Hz, faad upconverts to stereo and twice |
| // the frequency. No problem, but the sample rate is being |
| // set here by the sdp line. Patch on its way. (rdm) |
| } |
| 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); |
| break; |
| default: |
| break; |
| } |
| return 0; |
| } |
| |
| /* return the length and optionally the data */ |
| static int hex_to_data(uint8_t *data, const char *p) |
| { |
| int c, len, v; |
| |
| len = 0; |
| v = 1; |
| for (;;) { |
| skip_spaces(&p); |
| if (*p == '\0') |
| break; |
| c = toupper((unsigned char) *p++); |
| if (c >= '0' && c <= '9') |
| c = c - '0'; |
| else if (c >= 'A' && c <= 'F') |
| c = c - 'A' + 10; |
| else |
| break; |
| v = (v << 4) | c; |
| if (v & 0x100) { |
| if (data) |
| data[len] = v; |
| len++; |
| v = 1; |
| } |
| } |
| return len; |
| } |
| |
| static void sdp_parse_fmtp_config(AVCodecContext * codec, void *ctx, |
| char *attr, char *value) |
| { |
| switch (codec->codec_id) { |
| case CODEC_ID_MPEG4: |
| case CODEC_ID_AAC: |
| if (!strcmp(attr, "config")) { |
| /* decode the hexa encoded parameter */ |
| int len = hex_to_data(NULL, value); |
| if (codec->extradata) |
| av_free(codec->extradata); |
| codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE); |
| if (!codec->extradata) |
| return; |
| codec->extradata_size = len; |
| hex_to_data(codec->extradata, value); |
| } |
| break; |
| default: |
| break; |
| } |
| return; |
| } |
| |
| typedef struct { |
| const char *str; |
| uint16_t type; |
| uint32_t offset; |
| } AttrNameMap; |
| |
| /* All known fmtp parameters and the corresponding RTPAttrTypeEnum */ |
| #define ATTR_NAME_TYPE_INT 0 |
| #define ATTR_NAME_TYPE_STR 1 |
| static const AttrNameMap attr_names[]= |
| { |
| { "SizeLength", ATTR_NAME_TYPE_INT, |
| offsetof(RTPPayloadData, sizelength) }, |
| { "IndexLength", ATTR_NAME_TYPE_INT, |
| offsetof(RTPPayloadData, indexlength) }, |
| { "IndexDeltaLength", ATTR_NAME_TYPE_INT, |
| offsetof(RTPPayloadData, indexdeltalength) }, |
| { "profile-level-id", ATTR_NAME_TYPE_INT, |
| offsetof(RTPPayloadData, profile_level_id) }, |
| { "StreamType", ATTR_NAME_TYPE_INT, |
| offsetof(RTPPayloadData, streamtype) }, |
| { "mode", ATTR_NAME_TYPE_STR, |
| offsetof(RTPPayloadData, mode) }, |
| { NULL, -1, -1 }, |
| }; |
| |
| /* 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) |
| { |
| skip_spaces(p); |
| 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; |
| } |
| |
| /* parse a SDP line and save stream attributes */ |
| static void sdp_parse_fmtp(AVStream *st, const char *p) |
| { |
| char attr[256]; |
| /* Vorbis setup headers can be up to 12KB and are sent base64 |
| * encoded, giving a 12KB * (4/3) = 16KB FMTP line. */ |
| char value[16384]; |
| int i; |
| RTSPStream *rtsp_st = st->priv_data; |
| AVCodecContext *codec = st->codec; |
| RTPPayloadData *rtp_payload_data = &rtsp_st->rtp_payload_data; |
| |
| /* loop on each attribute */ |
| while (ff_rtsp_next_attr_and_value(&p, attr, sizeof(attr), |
| value, sizeof(value))) { |
| /* grab the codec extra_data from the config parameter of the fmtp |
| * line */ |
| sdp_parse_fmtp_config(codec, rtsp_st->dynamic_protocol_context, |
| attr, value); |
| /* Looking for a known attribute */ |
| for (i = 0; attr_names[i].str; ++i) { |
| if (!strcasecmp(attr, attr_names[i].str)) { |
| if (attr_names[i].type == ATTR_NAME_TYPE_INT) { |
| *(int *)((char *)rtp_payload_data + |
| attr_names[i].offset) = atoi(value); |
| } else if (attr_names[i].type == ATTR_NAME_TYPE_STR) |
| *(char **)((char *)rtp_payload_data + |
| attr_names[i].offset) = av_strdup(value); |
| } |
| } |
| } |
| } |
| |
| /** 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]; |
| |
| skip_spaces(&p); |
| if (!av_stristart(p, "npt=", &p)) |
| return; |
| |
| *start = AV_NOPTS_VALUE; |
| *end = AV_NOPTS_VALUE; |
| |
| get_word_sep(buf, sizeof(buf), "-", &p); |
| *start = parse_date(buf, 1); |
| if (*p == '-') { |
| p++; |
| get_word_sep(buf, sizeof(buf), "-", &p); |
| *end = parse_date(buf, 1); |
| } |
| // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start); |
| // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end); |
| } |
| |
| typedef struct SDPParseState { |
| /* SDP only */ |
| struct in_addr 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 in_addr sdp_ip; |
| int ttl; |
| |
| dprintf(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") != 0) |
| return; |
| get_word_sep(buf1, sizeof(buf1), "/", &p); |
| if (ff_inet_aton(buf1, &sdp_ip) == 0) |
| 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 { |
| st = s->streams[s->nb_streams - 1]; |
| rtsp_st = st->priv_data; |
| rtsp_st->sdp_ip = sdp_ip; |
| rtsp_st->sdp_ttl = ttl; |
| } |
| break; |
| case 's': |
| av_metadata_set2(&s->metadata, "title", p, 0); |
| break; |
| case 'i': |
| if (s->nb_streams == 0) { |
| av_metadata_set2(&s->metadata, "comment", p, 0); |
| break; |
| } |
| break; |
| case 'm': |
| /* new stream */ |
| s1->skip_media = 0; |
| 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 { |
| 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 (ignored) */ |
| |
| /* 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 */ |
| } else { |
| st = av_new_stream(s, 0); |
| if (!st) |
| return; |
| st->priv_data = rtsp_st; |
| rtsp_st->stream_index = st->index; |
| st->codec->codec_type = codec_type; |
| if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) { |
| /* if standard payload type, we can find the codec right now */ |
| ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type); |
| } |
| } |
| /* 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 */ |
| st = s->streams[s->nb_streams - 1]; |
| rtsp_st = st->priv_data; |
| |
| /* XXX: may need to add full url resolution */ |
| ff_url_split(proto, sizeof(proto), NULL, 0, NULL, 0, |
| NULL, NULL, 0, p); |
| if (proto[0] == '\0') { |
| /* relative control URL */ |
| if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/') |
| av_strlcat(rtsp_st->control_url, "/", |
| sizeof(rtsp_st->control_url)); |
| av_strlcat(rtsp_st->control_url, p, |
| sizeof(rtsp_st->control_url)); |
| } else |
| av_strlcpy(rtsp_st->control_url, p, |
| sizeof(rtsp_st->control_url)); |
| } |
| } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) { |
| /* NOTE: rtpmap is only supported AFTER the 'm=' tag */ |
| get_word(buf1, sizeof(buf1), &p); |
| payload_type = atoi(buf1); |
| st = s->streams[s->nb_streams - 1]; |
| rtsp_st = st->priv_data; |
| sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p); |
| } else if (av_strstart(p, "fmtp:", &p)) { |
| /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */ |
| get_word(buf1, sizeof(buf1), &p); |
| payload_type = atoi(buf1); |
| for (i = 0; i < s->nb_streams; i++) { |
| st = s->streams[i]; |
| rtsp_st = st->priv_data; |
| if (rtsp_st->sdp_payload_type == payload_type) { |
| if (!(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))) |
| sdp_parse_fmtp(st, p); |
| } |
| } |
| } else if (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); |
| for (i = 0; i < s->nb_streams; i++) { |
| st = s->streams[i]; |
| rtsp_st = st->priv_data; |
| 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 (rt->server_type == RTSP_SERVER_WMS) |
| ff_wms_parse_sdp_a_line(s, p); |
| if (s->nb_streams > 0) { |
| if (rt->server_type == RTSP_SERVER_REAL) |
| ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p); |
| |
| rtsp_st = s->streams[s->nb_streams - 1]->priv_data; |
| if (rtsp_st->dynamic_handler && |
| rtsp_st->dynamic_handler->parse_sdp_a_line) |
| rtsp_st->dynamic_handler->parse_sdp_a_line(s, |
| s->nb_streams - 1, |
| rtsp_st->dynamic_protocol_context, buf); |
| } |
| } |
| break; |
| } |
| } |
| |
| static int sdp_parse(AVFormatContext *s, const char *content) |
| { |
| 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 sdp_parse_fmtp. */ |
| char buf[16384], *q; |
| SDPParseState sdp_parse_state, *s1 = &sdp_parse_state; |
| |
| memset(s1, 0, sizeof(SDPParseState)); |
| p = content; |
| for (;;) { |
| skip_spaces(&p); |
| 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++; |
| } |
| return 0; |
| } |
| |
| /* close and free RTSP streams */ |
| void ff_rtsp_close_streams(AVFormatContext *s) |
| { |
| RTSPState *rt = s->priv_data; |
| int i; |
| RTSPStream *rtsp_st; |
| |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| rtsp_st = rt->rtsp_streams[i]; |
| if (rtsp_st) { |
| if (rtsp_st->transport_priv) { |
| if (s->oformat) { |
| AVFormatContext *rtpctx = rtsp_st->transport_priv; |
| av_write_trailer(rtpctx); |
| if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) { |
| uint8_t *ptr; |
| url_close_dyn_buf(rtpctx->pb, &ptr); |
| av_free(ptr); |
| } else { |
| url_fclose(rtpctx->pb); |
| } |
| av_metadata_free(&rtpctx->streams[0]->metadata); |
| av_metadata_free(&rtpctx->metadata); |
| av_free(rtpctx->streams[0]); |
| av_free(rtpctx); |
| } else if (rt->transport == RTSP_TRANSPORT_RDT) |
| ff_rdt_parse_close(rtsp_st->transport_priv); |
| else |
| rtp_parse_close(rtsp_st->transport_priv); |
| } |
| if (rtsp_st->rtp_handle) |
| url_close(rtsp_st->rtp_handle); |
| if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) |
| rtsp_st->dynamic_handler->close( |
| rtsp_st->dynamic_protocol_context); |
| } |
| } |
| av_free(rt->rtsp_streams); |
| if (rt->asf_ctx) { |
| av_close_input_stream (rt->asf_ctx); |
| rt->asf_ctx = NULL; |
| } |
| } |
| |
| static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st, |
| URLContext *handle) |
| { |
| RTSPState *rt = s->priv_data; |
| AVFormatContext *rtpctx; |
| int ret; |
| AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL); |
| |
| if (!rtp_format) |
| return NULL; |
| |
| /* Allocate an AVFormatContext for each output stream */ |
| rtpctx = avformat_alloc_context(); |
| if (!rtpctx) |
| return NULL; |
| |
| rtpctx->oformat = rtp_format; |
| if (!av_new_stream(rtpctx, 0)) { |
| av_free(rtpctx); |
| return NULL; |
| } |
| /* Copy the max delay setting; the rtp muxer reads this. */ |
| rtpctx->max_delay = s->max_delay; |
| /* Copy other stream parameters. */ |
| rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio; |
| |
| /* Set the synchronized start time. */ |
| rtpctx->start_time_realtime = rt->start_time; |
| |
| /* Remove the local codec, link to the original codec |
| * context instead, to give the rtp muxer access to |
| * codec parameters. */ |
| av_free(rtpctx->streams[0]->codec); |
| rtpctx->streams[0]->codec = st->codec; |
| |
| if (handle) { |
| url_fdopen(&rtpctx->pb, handle); |
| } else |
| url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE); |
| ret = av_write_header(rtpctx); |
| |
| if (ret) { |
| if (handle) { |
| url_fclose(rtpctx->pb); |
| } else { |
| uint8_t *ptr; |
| url_close_dyn_buf(rtpctx->pb, &ptr); |
| av_free(ptr); |
| } |
| av_free(rtpctx->streams[0]); |
| av_free(rtpctx); |
| return NULL; |
| } |
| |
| /* Copy the RTP AVStream timebase back to the original AVStream */ |
| st->time_base = rtpctx->streams[0]->time_base; |
| return rtpctx; |
| } |
| |
| static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st) |
| { |
| RTSPState *rt = s->priv_data; |
| AVStream *st = NULL; |
| |
| /* 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) { |
| rtsp_st->transport_priv = rtsp_rtp_mux_open(s, st, rtsp_st->rtp_handle); |
| /* Ownership of rtp_handle is passed to the rtp mux context */ |
| rtsp_st->rtp_handle = NULL; |
| } else if (rt->transport == RTSP_TRANSPORT_RDT) |
| rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index, |
| rtsp_st->dynamic_protocol_context, |
| rtsp_st->dynamic_handler); |
| else |
| rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle, |
| rtsp_st->sdp_payload_type, |
| &rtsp_st->rtp_payload_data); |
| |
| if (!rtsp_st->transport_priv) { |
| return AVERROR(ENOMEM); |
| } else if (rt->transport != RTSP_TRANSPORT_RDT) { |
| if (rtsp_st->dynamic_handler) { |
| rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv, |
| rtsp_st->dynamic_protocol_context, |
| rtsp_st->dynamic_handler); |
| } |
| } |
| |
| return 0; |
| } |
| |
| #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER |
| static int rtsp_probe(AVProbeData *p) |
| { |
| if (av_strstart(p->filename, "rtsp:", NULL)) |
| return AVPROBE_SCORE_MAX; |
| return 0; |
| } |
| |
| static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp) |
| { |
| const char *p; |
| int v; |
| |
| p = *pp; |
| skip_spaces(&p); |
| v = strtol(p, (char **)&p, 10); |
| if (*p == '-') { |
| p++; |
| *min_ptr = v; |
| v = strtol(p, (char **)&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 (;;) { |
| skip_spaces(&p); |
| if (*p == '\0') |
| break; |
| |
| th = &reply->transports[reply->nb_transports]; |
| |
| get_word_sep(transport_protocol, sizeof(transport_protocol), |
| "/", &p); |
| if (!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 (!strcasecmp (transport_protocol, "x-pn-tng") || |
| !strcasecmp (transport_protocol, "x-real-rdt")) { |
| /* x-pn-tng/<protocol> */ |
| get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p); |
| profile[0] = '\0'; |
| th->transport = RTSP_TRANSPORT_RDT; |
| } |
| if (!strcasecmp(lower_transport, "TCP")) |
| 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 == '=') { |
| p++; |
| th->ttl = strtol(p, (char **)&p, 10); |
| } |
| } else if (!strcmp(parameter, "destination")) { |
| struct in_addr ipaddr; |
| |
| if (*p == '=') { |
| p++; |
| get_word_sep(buf, sizeof(buf), ";,", &p); |
| if (ff_inet_aton(buf, &ipaddr)) |
| th->destination = ntohl(ipaddr.s_addr); |
| } |
| } |
| while (*p != ';' && *p != '\0' && *p != ',') |
| p++; |
| if (*p == ';') |
| p++; |
| } |
| if (*p == ',') |
| p++; |
| |
| reply->nb_transports++; |
| } |
| } |
| |
| void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf, |
| HTTPAuthState *auth_state) |
| { |
| 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)) { |
| skip_spaces(&p); |
| av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge)); |
| } else if (av_stristart(p, "Server:", &p)) { |
| skip_spaces(&p); |
| 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)) { |
| skip_spaces(&p); |
| av_strlcpy(reply->location, p , sizeof(reply->location)); |
| } else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) { |
| skip_spaces(&p); |
| ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p); |
| } else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) { |
| skip_spaces(&p); |
| ff_http_auth_handle_header(auth_state, "Authentication-Info", p); |
| } |
| } |
| |
| /* 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 = url_read_complete(rt->rtsp_hd, buf, 3); |
| if (ret != 3) |
| return; |
| len = AV_RB16(buf + 1); |
| |
| dprintf(s, "skipping RTP packet len=%d\n", len); |
| |
| /* skip payload */ |
| while (len > 0) { |
| len1 = len; |
| if (len1 > sizeof(buf)) |
| len1 = sizeof(buf); |
| ret = url_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) |
| { |
| RTSPState *rt = s->priv_data; |
| char buf[4096], buf1[1024], *q; |
| unsigned char ch; |
| const char *p; |
| int ret, content_length, line_count = 0; |
| unsigned char *content = NULL; |
| |
| memset(reply, 0, sizeof(*reply)); |
| |
| /* parse reply (XXX: use buffers) */ |
| rt->last_reply[0] = '\0'; |
| for (;;) { |
| q = buf; |
| for (;;) { |
| ret = url_read_complete(rt->rtsp_hd, &ch, 1); |
| #ifdef DEBUG_RTP_TCP |
| dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch); |
| #endif |
| if (ret != 1) |
| return -1; |
| 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'; |
| |
| dprintf(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); |
| get_word(buf1, sizeof(buf1), &p); |
| reply->status_code = atoi(buf1); |
| } else { |
| ff_rtsp_parse_line(reply, p, &rt->auth_state); |
| 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') |
| 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); |
| (void)url_read_complete(rt->rtsp_hd, content, content_length); |
| content[content_length] = '\0'; |
| } |
| if (content_ptr) |
| *content_ptr = content; |
| else |
| av_free(content); |
| |
| 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; |
| } |
| |
| void 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]; |
| |
| 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)); |
| |
| dprintf(s, "Sending:\n%s--\n", buf); |
| |
| url_write(rt->rtsp_hd, buf, strlen(buf)); |
| if (send_content_length > 0 && send_content) |
| url_write(rt->rtsp_hd, send_content, send_content_length); |
| rt->last_cmd_time = av_gettime(); |
| } |
| |
| void ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, |
| const char *url, const char *headers) |
| { |
| ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0); |
| } |
| |
| void ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url, |
| const char *headers, RTSPMessageHeader *reply, |
| unsigned char **content_ptr) |
| { |
| ff_rtsp_send_cmd_with_content(s, method, url, headers, reply, |
| content_ptr, NULL, 0); |
| } |
| |
| void 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; |
| |
| retry: |
| cur_auth_type = rt->auth_state.auth_type; |
| ff_rtsp_send_cmd_with_content_async(s, method, url, header, |
| send_content, send_content_length); |
| |
| ff_rtsp_read_reply(s, reply, content_ptr, 0); |
| |
| if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE && |
| rt->auth_state.auth_type != HTTP_AUTH_NONE) |
| goto retry; |
| } |
| |
| /** |
| * @return 0 on success, <0 on error, 1 if protocol is unavailable. |
| */ |
| static int make_setup_request(AVFormatContext *s, const char *host, int port, |
| int lower_transport, const char *real_challenge) |
| { |
| RTSPState *rt = s->priv_data; |
| int rtx, j, i, err, interleave = 0; |
| 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 |
| trans_pref = "RTP/AVP"; |
| |
| /* default timeout: 1 minute */ |
| rt->timeout = 60; |
| |
| /* for each stream, make the setup request */ |
| /* XXX: we assume the same server is used for the control of each |
| * RTSP stream */ |
| |
| for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) { |
| 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 */ |
| if (RTSP_RTP_PORT_MIN != 0) { |
| while (j <= RTSP_RTP_PORT_MAX) { |
| ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1, |
| "?localport=%d", j); |
| /* we will use two ports per rtp stream (rtp and rtcp) */ |
| j += 2; |
| if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0) |
| goto rtp_opened; |
| } |
| } |
| |
| #if 0 |
| /* then try on any port */ |
| if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) { |
| err = AVERROR_INVALIDDATA; |
| goto fail; |
| } |
| #endif |
| |
| rtp_opened: |
| port = rtp_get_local_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 && |
| s->streams[rtsp_st->stream_index]->codec->codec_type == |
| AVMEDIA_TYPE_DATA) |
| continue; |
| snprintf(transport, sizeof(transport) - 1, |
| "%s/TCP;", trans_pref); |
| if (rt->server_type == RTSP_SERVER_WMS) |
| 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=receive", 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 (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 = 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; |
| } |
| |
| /* close RTP connection if not choosen */ |
| if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP && |
| (lower_transport == RTSP_LOWER_TRANSPORT_UDP)) { |
| url_close(rtsp_st->rtp_handle); |
| rtsp_st->rtp_handle = NULL; |
| } |
| |
| 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]; |
| |
| /* XXX: also use address if specified */ |
| ff_url_join(url, sizeof(url), "rtp", NULL, host, |
| reply->transports[0].server_port_min, NULL); |
| if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && |
| 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) |
| rtp_send_punch_packets(rtsp_st->rtp_handle); |
| break; |
| } |
| case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: { |
| char url[1024]; |
| struct in_addr in; |
| int port, ttl; |
| |
| if (reply->transports[0].destination) { |
| in.s_addr = htonl(reply->transports[0].destination); |
| port = reply->transports[0].port_min; |
| ttl = reply->transports[0].ttl; |
| } else { |
| in = rtsp_st->sdp_ip; |
| port = rtsp_st->sdp_port; |
| ttl = rtsp_st->sdp_ttl; |
| } |
| ff_url_join(url, sizeof(url), "rtp", NULL, inet_ntoa(in), |
| port, "?ttl=%d", ttl); |
| if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) { |
| err = AVERROR_INVALIDDATA; |
| goto fail; |
| } |
| break; |
| } |
| } |
| |
| if ((err = rtsp_open_transport_ctx(s, rtsp_st))) |
| goto fail; |
| } |
| |
| if (reply->timeout > 0) |
| rt->timeout = reply->timeout; |
| |
| if (rt->server_type == RTSP_SERVER_REAL) |
| rt->need_subscription = 1; |
| |
| return 0; |
| |
| fail: |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| if (rt->rtsp_streams[i]->rtp_handle) { |
| url_close(rt->rtsp_streams[i]->rtp_handle); |
| rt->rtsp_streams[i]->rtp_handle = NULL; |
| } |
| } |
| return err; |
| } |
| |
| static int rtsp_read_play(AVFormatContext *s) |
| { |
| RTSPState *rt = s->priv_data; |
| RTSPMessageHeader reply1, *reply = &reply1; |
| int i; |
| char cmd[1024]; |
| |
| av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state); |
| |
| if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) { |
| if (rt->state == RTSP_STATE_PAUSED) { |
| cmd[0] = 0; |
| } else { |
| snprintf(cmd, sizeof(cmd), |
| "Range: npt=%0.3f-\r\n", |
| (double)rt->seek_timestamp / AV_TIME_BASE); |
| } |
| ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL); |
| if (reply->status_code != RTSP_STATUS_OK) { |
| return -1; |
| } |
| if (reply->range_start != AV_NOPTS_VALUE && |
| rt->transport == RTSP_TRANSPORT_RTP) { |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| RTSPStream *rtsp_st = rt->rtsp_streams[i]; |
| RTPDemuxContext *rtpctx = rtsp_st->transport_priv; |
| AVStream *st = NULL; |
| if (!rtpctx) |
| continue; |
| if (rtsp_st->stream_index >= 0) |
| st = s->streams[rtsp_st->stream_index]; |
| rtpctx->last_rtcp_ntp_time = AV_NOPTS_VALUE; |
| rtpctx->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
| if (st) |
| rtpctx->range_start_offset = av_rescale_q(reply->range_start, |
| AV_TIME_BASE_Q, |
| st->time_base); |
| } |
| } |
| } |
| rt->state = RTSP_STATE_STREAMING; |
| return 0; |
| } |
| |
| static int rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply) |
| { |
| RTSPState *rt = s->priv_data; |
| char cmd[1024]; |
| unsigned char *content = NULL; |
| int ret; |
| |
| /* describe the stream */ |
| snprintf(cmd, sizeof(cmd), |
| "Accept: application/sdp\r\n"); |
| if (rt->server_type == RTSP_SERVER_REAL) { |
| /** |
| * The Require: attribute is needed for proper streaming from |
| * Realmedia servers. |
| */ |
| av_strlcat(cmd, |
| "Require: com.real.retain-entity-for-setup\r\n", |
| sizeof(cmd)); |
| } |
| ff_rtsp_send_cmd(s, "DESCRIBE", rt->control_uri, cmd, reply, &content); |
| if (!content) |
| return AVERROR_INVALIDDATA; |
| if (reply->status_code != RTSP_STATUS_OK) { |
| av_freep(&content); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| /* now we got the SDP description, we parse it */ |
| ret = sdp_parse(s, (const char *)content); |
| av_freep(&content); |
| if (ret < 0) |
| return AVERROR_INVALIDDATA; |
| |
| return 0; |
| } |
| |
| static int rtsp_setup_output_streams(AVFormatContext *s, const char *addr) |
| { |
| RTSPState *rt = s->priv_data; |
| RTSPMessageHeader reply1, *reply = &reply1; |
| int i; |
| char *sdp; |
| AVFormatContext sdp_ctx, *ctx_array[1]; |
| |
| rt->start_time = av_gettime(); |
| |
| /* Announce the stream */ |
| sdp = av_mallocz(8192); |
| if (sdp == NULL) |
| return AVERROR(ENOMEM); |
| /* We create the SDP based on the RTSP AVFormatContext where we |
| * aren't allowed to change the filename field. (We create the SDP |
| * based on the RTSP context since the contexts for the RTP streams |
| * don't exist yet.) In order to specify a custom URL with the actual |
| * peer IP instead of the originally specified hostname, we create |
| * a temporary copy of the AVFormatContext, where the custom URL is set. |
| * |
| * FIXME: Create the SDP without copying the AVFormatContext. |
| * This either requires setting up the RTP stream AVFormatContexts |
| * already here (complicating things immensely) or getting a more |
| * flexible SDP creation interface. |
| */ |
| sdp_ctx = *s; |
| ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename), |
| "rtsp", NULL, addr, -1, NULL); |
| ctx_array[0] = &sdp_ctx; |
| if (avf_sdp_create(ctx_array, 1, sdp, 8192)) { |
| av_free(sdp); |
| return AVERROR_INVALIDDATA; |
| } |
| av_log(s, AV_LOG_INFO, "SDP:\n%s\n", sdp); |
| ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri, |
| "Content-Type: application/sdp\r\n", |
| reply, NULL, sdp, strlen(sdp)); |
| av_free(sdp); |
| if (reply->status_code != RTSP_STATUS_OK) |
| return AVERROR_INVALIDDATA; |
| |
| /* Set up the RTSPStreams for each AVStream */ |
| for (i = 0; i < s->nb_streams; i++) { |
| RTSPStream *rtsp_st; |
| AVStream *st = s->streams[i]; |
| |
| rtsp_st = av_mallocz(sizeof(RTSPStream)); |
| if (!rtsp_st) |
| return AVERROR(ENOMEM); |
| dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st); |
| |
| st->priv_data = rtsp_st; |
| rtsp_st->stream_index = i; |
| |
| av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url)); |
| /* Note, this must match the relative uri set in the sdp content */ |
| av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url), |
| "/streamid=%d", i); |
| } |
| |
| return 0; |
| } |
| |
| int ff_rtsp_connect(AVFormatContext *s) |
| { |
| RTSPState *rt = s->priv_data; |
| char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128]; |
| char *option_list, *option, *filename; |
| URLContext *rtsp_hd; |
| int port, err, tcp_fd; |
| RTSPMessageHeader reply1 = {}, *reply = &reply1; |
| int lower_transport_mask = 0; |
| char real_challenge[64]; |
| struct sockaddr_storage peer; |
| socklen_t peer_len = sizeof(peer); |
| |
| if (!ff_network_init()) |
| return AVERROR(EIO); |
| redirect: |
| /* extract hostname and port */ |
| ff_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; |
| |
| /* search for options */ |
| option_list = strrchr(path, '?'); |
| if (option_list) { |
| /* Strip out the RTSP specific options, write out the rest of |
| * the options back into the same string. */ |
| filename = option_list; |
| while (option_list) { |
| /* move the option pointer */ |
| option = ++option_list; |
| option_list = strchr(option_list, '&'); |
| if (option_list) |
| *option_list = 0; |
| |
| /* handle the options */ |
| if (!strcmp(option, "udp")) { |
| lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP); |
| } else if (!strcmp(option, "multicast")) { |
| lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST); |
| } else if (!strcmp(option, "tcp")) { |
| lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP); |
| } else { |
| /* Write options back into the buffer, using memmove instead |
| * of strcpy since the strings may overlap. */ |
| int len = strlen(option); |
| memmove(++filename, option, len); |
| filename += len; |
| if (option_list) *filename = '&'; |
| } |
| } |
| *filename = 0; |
| } |
| |
| 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) { |
| 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); |
| |
| /* open the tcp connexion */ |
| ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL); |
| if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0) { |
| err = AVERROR(EIO); |
| goto fail; |
| } |
| rt->rtsp_hd = rtsp_hd; |
| rt->seq = 0; |
| |
| tcp_fd = url_get_file_handle(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. |
| * @param CompanyID is a 16-byte ID in base64 |
| * @param ClientChallenge is a 16-byte ID in hex |
| */ |
| "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 (!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) |
| err = rtsp_setup_input_streams(s, reply); |
| else |
| err = 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 = make_setup_request(s, host, port, lower_transport, |
| rt->server_type == RTSP_SERVER_REAL ? |
| real_challenge : NULL); |
| if (err < 0) |
| goto fail; |
| lower_transport_mask &= ~(1 << lower_transport); |
| if (lower_transport_mask == 0 && err == 1) { |
| err = FF_NETERROR(EPROTONOSUPPORT); |
| goto fail; |
| } |
| } while (err); |
| |
| 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); |
| url_close(rt->rtsp_hd); |
| 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 |
| |
| #if CONFIG_RTSP_DEMUXER |
| static int rtsp_read_header(AVFormatContext *s, |
| AVFormatParameters *ap) |
| { |
| RTSPState *rt = s->priv_data; |
| int ret; |
| |
| ret = ff_rtsp_connect(s); |
| if (ret) |
| return ret; |
| |
| if (ap->initial_pause) { |
| /* do not start immediately */ |
| } else { |
| if (rtsp_read_play(s) < 0) { |
| ff_rtsp_close_streams(s); |
| url_close(rt->rtsp_hd); |
| return AVERROR_INVALIDDATA; |
| } |
| } |
| |
| return 0; |
| } |
| |
| static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, |
| uint8_t *buf, int buf_size) |
| { |
| RTSPState *rt = s->priv_data; |
| RTSPStream *rtsp_st; |
| fd_set rfds; |
| int fd, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0; |
| struct timeval tv; |
| |
| for (;;) { |
| if (url_interrupt_cb()) |
| return AVERROR(EINTR); |
| FD_ZERO(&rfds); |
| if (rt->rtsp_hd) { |
| tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd); |
| FD_SET(tcp_fd, &rfds); |
| } else { |
| fd_max = 0; |
| tcp_fd = -1; |
| } |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| rtsp_st = rt->rtsp_streams[i]; |
| if (rtsp_st->rtp_handle) { |
| /* currently, we cannot probe RTCP handle because of |
| * blocking restrictions */ |
| fd = url_get_file_handle(rtsp_st->rtp_handle); |
| if (fd > fd_max) |
| fd_max = fd; |
| FD_SET(fd, &rfds); |
| } |
| } |
| tv.tv_sec = 0; |
| tv.tv_usec = SELECT_TIMEOUT_MS * 1000; |
| n = select(fd_max + 1, &rfds, NULL, NULL, &tv); |
| if (n > 0) { |
| timeout_cnt = 0; |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| rtsp_st = rt->rtsp_streams[i]; |
| if (rtsp_st->rtp_handle) { |
| fd = url_get_file_handle(rtsp_st->rtp_handle); |
| if (FD_ISSET(fd, &rfds)) { |
| ret = url_read(rtsp_st->rtp_handle, buf, buf_size); |
| if (ret > 0) { |
| *prtsp_st = rtsp_st; |
| return ret; |
| } |
| } |
| } |
| } |
| #if CONFIG_RTSP_DEMUXER |
| if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) { |
| RTSPMessageHeader reply; |
| |
| ret = ff_rtsp_read_reply(s, &reply, NULL, 0); |
| 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 FF_NETERROR(ETIMEDOUT); |
| } else if (n < 0 && errno != EINTR) |
| return AVERROR(errno); |
| } |
| } |
| |
| static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, |
| uint8_t *buf, int buf_size) |
| { |
| RTSPState *rt = s->priv_data; |
| int id, len, i, ret; |
| RTSPStream *rtsp_st; |
| |
| #ifdef DEBUG_RTP_TCP |
| dprintf(s, "tcp_read_packet:\n"); |
| #endif |
| redo: |
| for (;;) { |
| RTSPMessageHeader reply; |
| |
| ret = ff_rtsp_read_reply(s, &reply, NULL, 1); |
| if (ret == -1) |
| return -1; |
| if (ret == 1) /* received '$' */ |
| break; |
| /* XXX: parse message */ |
| if (rt->state != RTSP_STATE_STREAMING) |
| return 0; |
| } |
| ret = url_read_complete(rt->rtsp_hd, buf, 3); |
| if (ret != 3) |
| return -1; |
| id = buf[0]; |
| len = AV_RB16(buf + 1); |
| #ifdef DEBUG_RTP_TCP |
| dprintf(s, "id=%d len=%d\n", id, len); |
| #endif |
| if (len > buf_size || len < 12) |
| goto redo; |
| /* get the data */ |
| ret = url_read_complete(rt->rtsp_hd, buf, len); |
| if (ret != len) |
| return -1; |
| if (rt->transport == RTSP_TRANSPORT_RDT && |
| ff_rdt_parse_header(buf, len, &id, NULL, NULL, NULL, NULL) < 0) |
| return -1; |
| |
| /* find the matching stream */ |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| rtsp_st = rt->rtsp_streams[i]; |
| if (id >= rtsp_st->interleaved_min && |
| id <= rtsp_st->interleaved_max) |
| goto found; |
| } |
| goto redo; |
| found: |
| *prtsp_st = rtsp_st; |
| return len; |
| } |
| |
| static int rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt) |
| { |
| RTSPState *rt = s->priv_data; |
| int ret, len; |
| uint8_t buf[10 * RTP_MAX_PACKET_LENGTH]; |
| RTSPStream *rtsp_st; |
| |
| /* 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 |
| ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0); |
| if (ret == 0) { |
| rt->cur_transport_priv = NULL; |
| return 0; |
| } else if (ret == 1) { |
| return 0; |
| } else |
| rt->cur_transport_priv = NULL; |
| } |
| |
| /* read next RTP packet */ |
| redo: |
| switch(rt->lower_transport) { |
| default: |
| #if CONFIG_RTSP_DEMUXER |
| case RTSP_LOWER_TRANSPORT_TCP: |
| len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf)); |
| break; |
| #endif |
| case RTSP_LOWER_TRANSPORT_UDP: |
| case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: |
| len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf)); |
| if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP) |
| rtp_check_and_send_back_rr(rtsp_st->transport_priv, len); |
| break; |
| } |
| 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, buf, len); |
| } else { |
| ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, buf, len); |
| 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; |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| RTPDemuxContext *rtpctx2 = rtsp_st->transport_priv; |
| if (rtpctx2 && |
| rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) |
| rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time; |
| } |
| } |
| } |
| } |
| 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; |
| } |
| |
| static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt) |
| { |
| RTSPState *rt = s->priv_data; |
| int ret; |
| RTSPMessageHeader reply1, *reply = &reply1; |
| char cmd[1024]; |
| |
| if (rt->server_type == RTSP_SERVER_REAL) { |
| int i; |
| enum AVDiscard cache[MAX_STREAMS]; |
| |
| for (i = 0; i < s->nb_streams; i++) |
| cache[i] = s->streams[i]->discard; |
| |
| if (!rt->need_subscription) { |
| if (memcmp (cache, rt->real_setup_cache, |
| sizeof(enum AVDiscard) * s->nb_streams)) { |
| snprintf(cmd, sizeof(cmd), |
| "Unsubscribe: %s\r\n", |
| rt->last_subscription); |
| ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri, |
| cmd, reply, NULL); |
| if (reply->status_code != RTSP_STATUS_OK) |
| return AVERROR_INVALIDDATA; |
| rt->need_subscription = 1; |
| } |
| } |
| |
| if (rt->need_subscription) { |
| int r, rule_nr, first = 1; |
| |
| memcpy(rt->real_setup_cache, cache, |
| sizeof(enum AVDiscard) * s->nb_streams); |
| rt->last_subscription[0] = 0; |
| |
| snprintf(cmd, sizeof(cmd), |
| "Subscribe: "); |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| rule_nr = 0; |
| for (r = 0; r < s->nb_streams; r++) { |
| if (s->streams[r]->priv_data == rt->rtsp_streams[i]) { |
| if (s->streams[r]->discard != AVDISCARD_ALL) { |
| if (!first) |
| av_strlcat(rt->last_subscription, ",", |
| sizeof(rt->last_subscription)); |
| ff_rdt_subscribe_rule( |
| rt->last_subscription, |
| sizeof(rt->last_subscription), i, rule_nr); |
| first = 0; |
| } |
| rule_nr++; |
| } |
| } |
| } |
| av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription); |
| ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri, |
| cmd, reply, NULL); |
| if (reply->status_code != RTSP_STATUS_OK) |
| return AVERROR_INVALIDDATA; |
| rt->need_subscription = 0; |
| |
| if (rt->state == RTSP_STATE_STREAMING) |
| rtsp_read_play (s); |
| } |
| } |
| |
| ret = rtsp_fetch_packet(s, pkt); |
| if (ret < 0) |
| return ret; |
| |
| /* send dummy request to keep TCP connection alive */ |
| if ((rt->server_type == RTSP_SERVER_WMS || |
| rt->server_type == RTSP_SERVER_REAL) && |
| (av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) { |
| if (rt->server_type == RTSP_SERVER_WMS) { |
| ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL); |
| } else { |
| ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL); |
| } |
| } |
| |
| return 0; |
| } |
| |
| /* pause the stream */ |
| static int rtsp_read_pause(AVFormatContext *s) |
| { |
| RTSPState *rt = s->priv_data; |
| RTSPMessageHeader reply1, *reply = &reply1; |
| |
| if (rt->state != RTSP_STATE_STREAMING) |
| return 0; |
| else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) { |
| ff_rtsp_send_cmd(s, "PAUSE", rt->control_uri, NULL, reply, NULL); |
| if (reply->status_code != RTSP_STATUS_OK) { |
| return -1; |
| } |
| } |
| rt->state = RTSP_STATE_PAUSED; |
| return 0; |
| } |
| |
| static int rtsp_read_seek(AVFormatContext *s, int stream_index, |
| int64_t timestamp, int flags) |
| { |
| RTSPState *rt = s->priv_data; |
| |
| rt->seek_timestamp = av_rescale_q(timestamp, |
| s->streams[stream_index]->time_base, |
| AV_TIME_BASE_Q); |
| switch(rt->state) { |
| default: |
| case RTSP_STATE_IDLE: |
| break; |
| case RTSP_STATE_STREAMING: |
| if (rtsp_read_pause(s) != 0) |
| return -1; |
| rt->state = RTSP_STATE_SEEKING; |
| if (rtsp_read_play(s) != 0) |
| return -1; |
| break; |
| case RTSP_STATE_PAUSED: |
| rt->state = RTSP_STATE_IDLE; |
| break; |
| } |
| return 0; |
| } |
| |
| static int rtsp_read_close(AVFormatContext *s) |
| { |
| RTSPState *rt = s->priv_data; |
| |
| #if 0 |
| /* NOTE: it is valid to flush the buffer here */ |
| if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) { |
| url_fclose(&rt->rtsp_gb); |
| } |
| #endif |
| ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL); |
| |
| ff_rtsp_close_streams(s); |
| url_close(rt->rtsp_hd); |
| ff_network_close(); |
| return 0; |
| } |
| |
| AVInputFormat rtsp_demuxer = { |
| "rtsp", |
| NULL_IF_CONFIG_SMALL("RTSP input format"), |
| sizeof(RTSPState), |
| rtsp_probe, |
| rtsp_read_header, |
| rtsp_read_packet, |
| rtsp_read_close, |
| rtsp_read_seek, |
| .flags = AVFMT_NOFILE, |
| .read_play = rtsp_read_play, |
| .read_pause = rtsp_read_pause, |
| }; |
| #endif |
| |
| 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 IP4" */ |
| while (p < p_end && *p != '\0') { |
| if (p + sizeof("c=IN IP4") - 1 < p_end && |
| av_strstart(p, "c=IN IP4", 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; |
| } |
| |
| #define SDP_MAX_SIZE 8192 |
| |
| static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap) |
| { |
| RTSPState *rt = s->priv_data; |
| RTSPStream *rtsp_st; |
| int size, i, err; |
| char *content; |
| char url[1024]; |
| |
| if (!ff_network_init()) |
| return AVERROR(EIO); |
| |
| /* read the whole sdp file */ |
| /* XXX: better loading */ |
| content = av_malloc(SDP_MAX_SIZE); |
| size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1); |
| if (size <= 0) { |
| av_free(content); |
| return AVERROR_INVALIDDATA; |
| } |
| content[size] ='\0'; |
| |
| sdp_parse(s, content); |
| av_free(content); |
| |
| /* open each RTP stream */ |
| for (i = 0; i < rt->nb_rtsp_streams; i++) { |
| rtsp_st = rt->rtsp_streams[i]; |
| |
| ff_url_join(url, sizeof(url), "rtp", NULL, |
| inet_ntoa(rtsp_st->sdp_ip), rtsp_st->sdp_port, |
| "?localport=%d&ttl=%d", rtsp_st->sdp_port, |
| rtsp_st->sdp_ttl); |
| if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) { |
| err = AVERROR_INVALIDDATA; |
| goto fail; |
| } |
| if ((err = 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; |
| } |
| |
| AVInputFormat sdp_demuxer = { |
| "sdp", |
| NULL_IF_CONFIG_SMALL("SDP"), |
| sizeof(RTSPState), |
| sdp_probe, |
| sdp_read_header, |
| rtsp_fetch_packet, |
| sdp_read_close, |
| }; |