diff options
Diffstat (limited to 'contrib/ffmpeg/libavformat/rtsp.c')
| -rw-r--r-- | contrib/ffmpeg/libavformat/rtsp.c | 1486 |
1 files changed, 1486 insertions, 0 deletions
diff --git a/contrib/ffmpeg/libavformat/rtsp.c b/contrib/ffmpeg/libavformat/rtsp.c new file mode 100644 index 000000000..7d4c6bf78 --- /dev/null +++ b/contrib/ffmpeg/libavformat/rtsp.c @@ -0,0 +1,1486 @@ +/* + * 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 "avformat.h" + +#include <sys/time.h> +#include <unistd.h> /* for select() prototype */ +#include "network.h" + +#include "rtp_internal.h" + +//#define DEBUG +//#define DEBUG_RTP_TCP + +enum RTSPClientState { + RTSP_STATE_IDLE, + RTSP_STATE_PLAYING, + RTSP_STATE_PAUSED, +}; + +typedef struct RTSPState { + URLContext *rtsp_hd; /* RTSP TCP connexion handle */ + int nb_rtsp_streams; + struct RTSPStream **rtsp_streams; + + enum RTSPClientState state; + int64_t seek_timestamp; + + /* XXX: currently we use unbuffered input */ + // ByteIOContext rtsp_gb; + int seq; /* RTSP command sequence number */ + char session_id[512]; + enum RTSPProtocol protocol; + char last_reply[2048]; /* XXX: allocate ? */ + RTPDemuxContext *cur_rtp; +} RTSPState; + +typedef struct RTSPStream { + URLContext *rtp_handle; /* RTP stream handle */ + RTPDemuxContext *rtp_ctx; /* RTP parse context */ + + int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */ + int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */ + char control_url[1024]; /* url for this stream (from SDP) */ + + int sdp_port; /* port (from SDP content - not used in RTSP) */ + struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */ + int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */ + int sdp_payload_type; /* payload type - only used in SDP */ + rtp_payload_data_t rtp_payload_data; /* rtp payload parsing infos from SDP */ + + RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure) + void *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol) +} RTSPStream; + +static int rtsp_read_play(AVFormatContext *s); + +/* XXX: currently, the only way to change the protocols consists in + changing this variable */ + +int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_UDP); + +FFRTSPCallback *ff_rtsp_callback = NULL; + +static int rtsp_probe(AVProbeData *p) +{ + if (strstart(p->filename, "rtsp:", NULL)) + return AVPROBE_SCORE_MAX; + return 0; +} + +static int redir_isspace(int c) +{ + return (c == ' ' || c == '\t' || c == '\n' || c == '\r'); +} + +static void skip_spaces(const char **pp) +{ + const char *p; + p = *pp; + while (redir_isspace(*p)) + p++; + *pp = p; +} + +static void get_word_sep(char *buf, int buf_size, const char *sep, + const char **pp) +{ + const char *p; + char *q; + + p = *pp; + if (*p == '/') + p++; + 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(char *buf, int buf_size, const char **pp) +{ + const char *p; + char *q; + + p = *pp; + skip_spaces(&p); + q = buf; + while (!redir_isspace(*p) && *p != '\0') { + if ((q - buf) < buf_size - 1) + *q++ = *p; + p++; + } + if (buf_size > 0) + *q = '\0'; + *pp = p; +} + +/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other + params>] */ +static int sdp_parse_rtpmap(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 */ + get_word_sep(buf, sizeof(buf), "/", &p); + if (payload_type >= RTP_PT_PRIVATE) { + RTPDynamicProtocolHandler *handler= RTPFirstDynamicPayloadHandler; + while(handler) { + if (!strcmp(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; + } + handler= handler->next; + } + } else { + /* We are in a standard case ( from http://www.iana.org/assignments/rtp-parameters) */ + /* search into AVRtpPayloadTypes[] */ + for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i) + if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec->codec_type == AVRtpPayloadTypes[i].codec_type)){ + codec->codec_id = AVRtpPayloadTypes[i].codec_id; + break; + } + } + + c = avcodec_find_decoder(codec->codec_id); + if (c && c->name) + c_name = c->name; + else + c_name = (char *)NULL; + + if (c_name) { + get_word_sep(buf, sizeof(buf), "/", &p); + i = atoi(buf); + switch (codec->codec_type) { + case CODEC_TYPE_AUDIO: + av_log(codec, 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. Upcoming patch forthcoming. (rdm) + } + av_log(codec, AV_LOG_DEBUG, " audio samplerate set to : %i\n", codec->sample_rate); + av_log(codec, AV_LOG_DEBUG, " audio channels set to : %i\n", codec->channels); + break; + case CODEC_TYPE_VIDEO: + av_log(codec, AV_LOG_DEBUG, " video codec set to : %s\n", c_name); + break; + default: + break; + } + return 0; + } + + return -1; +} + +/* return the length and optionnaly 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, 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); + 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 attrname_map +{ + const char *str; + uint16_t type; + uint32_t offset; +} attrname_map_t; + +/* All known fmtp parmeters and the corresping RTPAttrTypeEnum */ +#define ATTR_NAME_TYPE_INT 0 +#define ATTR_NAME_TYPE_STR 1 +static attrname_map_t attr_names[]= +{ + {"SizeLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, sizelength)}, + {"IndexLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexlength)}, + {"IndexDeltaLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexdeltalength)}, + {"profile-level-id", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, profile_level_id)}, + {"StreamType", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, streamtype)}, + {"mode", ATTR_NAME_TYPE_STR, offsetof(rtp_payload_data_t, mode)}, + {NULL, -1, -1}, +}; + +/** parse the attribute line from the fmtp a line of an sdp resonse. This is broken out as a function +* because it is used in rtp_h264.c, which is forthcoming. +*/ +int 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]; + char value[4096]; + int i; + + RTSPStream *rtsp_st = st->priv_data; + AVCodecContext *codec = st->codec; + rtp_payload_data_t *rtp_payload_data = &rtsp_st->rtp_payload_data; + + /* loop on each attribute */ + while(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, 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 (!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; +} 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; + int codec_type, payload_type, i; + AVStream *st; + RTSPStream *rtsp_st; + struct in_addr sdp_ip; + int ttl; + +#ifdef DEBUG + printf("sdp: %c='%s'\n", letter, buf); +#endif + + p = buf; + 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 (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': + pstrcpy(s->title, sizeof(s->title), p); + break; + case 'i': + if (s->nb_streams == 0) { + pstrcpy(s->comment, sizeof(s->comment), p); + break; + } + break; + case 'm': + /* new stream */ + get_word(st_type, sizeof(st_type), &p); + if (!strcmp(st_type, "audio")) { + codec_type = CODEC_TYPE_AUDIO; + } else if (!strcmp(st_type, "video")) { + codec_type = CODEC_TYPE_VIDEO; + } else { + 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(AVRtpPayloadTypes[rtsp_st->sdp_payload_type].enc_name, "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 */ + rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type); + } + } + /* put a default control url */ + pstrcpy(rtsp_st->control_url, sizeof(rtsp_st->control_url), s->filename); + break; + case 'a': + if (strstart(p, "control:", &p) && s->nb_streams > 0) { + 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 */ + url_split(proto, sizeof(proto), NULL, 0, NULL, 0, NULL, NULL, 0, p); + if (proto[0] == '\0') { + /* relative control URL */ + pstrcat(rtsp_st->control_url, sizeof(rtsp_st->control_url), "/"); + pstrcat(rtsp_st->control_url, sizeof(rtsp_st->control_url), p); + } else { + pstrcpy(rtsp_st->control_url, sizeof(rtsp_st->control_url), p); + } + } else if (strstart(p, "rtpmap:", &p)) { + /* NOTE: rtpmap is only supported AFTER the 'm=' 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) { + sdp_parse_rtpmap(st->codec, rtsp_st, payload_type, p); + } + } + } else if (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) { + if(!rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf)) { + sdp_parse_fmtp(st, p); + } + } else { + sdp_parse_fmtp(st, p); + } + } + } + } else if(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) { + if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) { + rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf); + } + } + } + } else if(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; + s->duration= (end==AV_NOPTS_VALUE)?AV_NOPTS_VALUE:end-start; // AV_NOPTS_VALUE means live broadcast (and can't seek) + } + break; + } +} + +static int sdp_parse(AVFormatContext *s, const char *content) +{ + const char *p; + int letter; + char buf[1024], *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; +} + +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(RTSPHeader *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 (*p == '/') + p++; + get_word_sep(profile, sizeof(profile), "/;,", &p); + lower_transport[0] = '\0'; + if (*p == '/') { + p++; + get_word_sep(lower_transport, sizeof(lower_transport), + ";,", &p); + } + if (!strcasecmp(lower_transport, "TCP")) + th->protocol = RTSP_PROTOCOL_RTP_TCP; + else + th->protocol = RTSP_PROTOCOL_RTP_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->protocol == RTSP_PROTOCOL_RTP_UDP) + th->protocol = RTSP_PROTOCOL_RTP_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 (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 rtsp_parse_line(RTSPHeader *reply, const char *buf) +{ + const char *p; + + /* NOTE: we do case independent match for broken servers */ + p = buf; + if (stristart(p, "Session:", &p)) { + get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p); + } else if (stristart(p, "Content-Length:", &p)) { + reply->content_length = strtol(p, NULL, 10); + } else if (stristart(p, "Transport:", &p)) { + rtsp_parse_transport(reply, p); + } else if (stristart(p, "CSeq:", &p)) { + reply->seq = strtol(p, NULL, 10); + } else if (stristart(p, "Range:", &p)) { + rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end); + } +} + +static int url_readbuf(URLContext *h, unsigned char *buf, int size) +{ + int ret, len; + + len = 0; + while (len < size) { + ret = url_read(h, buf+len, size-len); + if (ret < 1) + return ret; + len += ret; + } + return len; +} + +/* skip a RTP/TCP interleaved packet */ +static void rtsp_skip_packet(AVFormatContext *s) +{ + RTSPState *rt = s->priv_data; + int ret, len, len1; + uint8_t buf[1024]; + + ret = url_readbuf(rt->rtsp_hd, buf, 3); + if (ret != 3) + return; + len = (buf[1] << 8) | buf[2]; +#ifdef DEBUG + printf("skipping RTP packet len=%d\n", len); +#endif + /* skip payload */ + while (len > 0) { + len1 = len; + if (len1 > sizeof(buf)) + len1 = sizeof(buf); + ret = url_readbuf(rt->rtsp_hd, buf, len1); + if (ret != len1) + return; + len -= len1; + } +} + +static void rtsp_send_cmd(AVFormatContext *s, + const char *cmd, RTSPHeader *reply, + unsigned char **content_ptr) +{ + RTSPState *rt = s->priv_data; + char buf[4096], buf1[1024], *q; + unsigned char ch; + const char *p; + int content_length, line_count; + unsigned char *content = NULL; + + memset(reply, 0, sizeof(RTSPHeader)); + + rt->seq++; + pstrcpy(buf, sizeof(buf), cmd); + snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq); + pstrcat(buf, sizeof(buf), buf1); + if (rt->session_id[0] != '\0' && !strstr(cmd, "\nIf-Match:")) { + snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id); + pstrcat(buf, sizeof(buf), buf1); + } + pstrcat(buf, sizeof(buf), "\r\n"); +#ifdef DEBUG + printf("Sending:\n%s--\n", buf); +#endif + url_write(rt->rtsp_hd, buf, strlen(buf)); + + /* parse reply (XXX: use buffers) */ + line_count = 0; + rt->last_reply[0] = '\0'; + for(;;) { + q = buf; + for(;;) { + if (url_readbuf(rt->rtsp_hd, &ch, 1) != 1) + break; + if (ch == '\n') + break; + if (ch == '$') { + /* XXX: only parse it if first char on line ? */ + rtsp_skip_packet(s); + } else if (ch != '\r') { + if ((q - buf) < sizeof(buf) - 1) + *q++ = ch; + } + } + *q = '\0'; +#ifdef DEBUG + printf("line='%s'\n", buf); +#endif + /* 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 { + rtsp_parse_line(reply, p); + pstrcat(rt->last_reply, sizeof(rt->last_reply), p); + pstrcat(rt->last_reply, sizeof(rt->last_reply), "\n"); + } + line_count++; + } + + if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0') + pstrcpy(rt->session_id, sizeof(rt->session_id), reply->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_readbuf(rt->rtsp_hd, content, content_length); + content[content_length] = '\0'; + } + if (content_ptr) + *content_ptr = content; +} + + +void rtsp_set_callback(FFRTSPCallback *rtsp_cb) +{ + ff_rtsp_callback = rtsp_cb; +} + + +/* close and free RTSP streams */ +static void rtsp_close_streams(RTSPState *rt) +{ + 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->rtp_ctx) + rtp_parse_close(rtsp_st->rtp_ctx); + 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(rtsp_st); + } + av_free(rt->rtsp_streams); +} + +static int rtsp_read_header(AVFormatContext *s, + AVFormatParameters *ap) +{ + RTSPState *rt = s->priv_data; + char host[1024], path[1024], tcpname[1024], cmd[2048]; + URLContext *rtsp_hd; + int port, i, j, ret, err; + RTSPHeader reply1, *reply = &reply1; + unsigned char *content = NULL; + RTSPStream *rtsp_st; + int protocol_mask; + AVStream *st; + + /* extract hostname and port */ + url_split(NULL, 0, NULL, 0, + host, sizeof(host), &port, path, sizeof(path), s->filename); + if (port < 0) + port = RTSP_DEFAULT_PORT; + + /* open the tcp connexion */ + snprintf(tcpname, sizeof(tcpname), "tcp://%s:%d", host, port); + if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0) + return AVERROR_IO; + rt->rtsp_hd = rtsp_hd; + rt->seq = 0; + + /* describe the stream */ + snprintf(cmd, sizeof(cmd), + "DESCRIBE %s RTSP/1.0\r\n" + "Accept: application/sdp\r\n", + s->filename); + rtsp_send_cmd(s, cmd, reply, &content); + if (!content) { + err = AVERROR_INVALIDDATA; + goto fail; + } + if (reply->status_code != RTSP_STATUS_OK) { + err = AVERROR_INVALIDDATA; + goto fail; + } + + /* now we got the SDP description, we parse it */ + ret = sdp_parse(s, (const char *)content); + av_freep(&content); + if (ret < 0) { + err = AVERROR_INVALIDDATA; + goto fail; + } + + protocol_mask = rtsp_default_protocols; + + /* 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]; + + rtsp_st = rt->rtsp_streams[i]; + + /* compute available transports */ + transport[0] = '\0'; + + /* RTP/UDP */ + if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP)) { + char buf[256]; + + /* first try in specified port range */ + if (RTSP_RTP_PORT_MIN != 0) { + while(j <= RTSP_RTP_PORT_MAX) { + snprintf(buf, sizeof(buf), "rtp://?localport=%d", j); + if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0) { + j += 2; /* we will use two port by rtp stream (rtp and rtcp) */ + goto rtp_opened; + } + } + } + +/* then try on any port +** if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) { +** err = AVERROR_INVALIDDATA; +** goto fail; +** } +*/ + + rtp_opened: + port = rtp_get_local_port(rtsp_st->rtp_handle); + if (transport[0] != '\0') + pstrcat(transport, sizeof(transport), ","); + snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1, + "RTP/AVP/UDP;unicast;client_port=%d-%d", + port, port + 1); + } + + /* RTP/TCP */ + else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_TCP)) { + if (transport[0] != '\0') + pstrcat(transport, sizeof(transport), ","); + snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1, + "RTP/AVP/TCP"); + } + + else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST)) { + if (transport[0] != '\0') + pstrcat(transport, sizeof(transport), ","); + snprintf(transport + strlen(transport), + sizeof(transport) - strlen(transport) - 1, + "RTP/AVP/UDP;multicast"); + } + snprintf(cmd, sizeof(cmd), + "SETUP %s RTSP/1.0\r\n" + "Transport: %s\r\n", + rtsp_st->control_url, transport); + rtsp_send_cmd(s, cmd, reply, NULL); + 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].protocol != rt->protocol) { + err = AVERROR_INVALIDDATA; + goto fail; + } + } else { + rt->protocol = reply->transports[0].protocol; + } + + /* close RTP connection if not choosen */ + if (reply->transports[0].protocol != RTSP_PROTOCOL_RTP_UDP && + (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP))) { + url_close(rtsp_st->rtp_handle); + rtsp_st->rtp_handle = NULL; + } + + switch(reply->transports[0].protocol) { + case RTSP_PROTOCOL_RTP_TCP: + rtsp_st->interleaved_min = reply->transports[0].interleaved_min; + rtsp_st->interleaved_max = reply->transports[0].interleaved_max; + break; + + case RTSP_PROTOCOL_RTP_UDP: + { + char url[1024]; + + /* XXX: also use address if specified */ + snprintf(url, sizeof(url), "rtp://%s:%d", + host, reply->transports[0].server_port_min); + if (rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) { + err = AVERROR_INVALIDDATA; + goto fail; + } + } + break; + case RTSP_PROTOCOL_RTP_UDP_MULTICAST: + { + char url[1024]; + int ttl; + + ttl = reply->transports[0].ttl; + if (!ttl) + ttl = 16; + snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d", + host, + reply->transports[0].server_port_min, + ttl); + if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) { + err = AVERROR_INVALIDDATA; + goto fail; + } + } + break; + } + /* open the RTP context */ + st = NULL; + if (rtsp_st->stream_index >= 0) + st = s->streams[rtsp_st->stream_index]; + if (!st) + s->ctx_flags |= AVFMTCTX_NOHEADER; + rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data); + + if (!rtsp_st->rtp_ctx) { + err = AVERROR_NOMEM; + goto fail; + } else { + if(rtsp_st->dynamic_handler) { + rtsp_st->rtp_ctx->dynamic_protocol_context= rtsp_st->dynamic_protocol_context; + rtsp_st->rtp_ctx->parse_packet= rtsp_st->dynamic_handler->parse_packet; + } + } + } + + /* use callback if available to extend setup */ + if (ff_rtsp_callback) { + if (ff_rtsp_callback(RTSP_ACTION_CLIENT_SETUP, rt->session_id, + NULL, 0, rt->last_reply) < 0) { + err = AVERROR_INVALIDDATA; + goto fail; + } + } + + + rt->state = RTSP_STATE_IDLE; + rt->seek_timestamp = 0; /* default is to start stream at position + zero */ + if (ap->initial_pause) { + /* do not start immediately */ + } else { + if (rtsp_read_play(s) < 0) { + err = AVERROR_INVALIDDATA; + goto fail; + } + } + return 0; + fail: + rtsp_close_streams(rt); + av_freep(&content); + url_close(rt->rtsp_hd); + return err; +} + +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 + printf("tcp_read_packet:\n"); +#endif + redo: + for(;;) { + ret = url_readbuf(rt->rtsp_hd, buf, 1); +#ifdef DEBUG_RTP_TCP + printf("ret=%d c=%02x [%c]\n", ret, buf[0], buf[0]); +#endif + if (ret != 1) + return -1; + if (buf[0] == '$') + break; + } + ret = url_readbuf(rt->rtsp_hd, buf, 3); + if (ret != 3) + return -1; + id = buf[0]; + len = (buf[1] << 8) | buf[2]; +#ifdef DEBUG_RTP_TCP + printf("id=%d len=%d\n", id, len); +#endif + if (len > buf_size || len < 12) + goto redo; + /* get the data */ + ret = url_readbuf(rt->rtsp_hd, buf, len); + if (ret != len) + 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 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 fd1, fd2, fd_max, n, i, ret; + struct timeval tv; + + for(;;) { + if (url_interrupt_cb()) + return -1; + FD_ZERO(&rfds); + fd_max = -1; + for(i = 0; i < rt->nb_rtsp_streams; i++) { + rtsp_st = rt->rtsp_streams[i]; + /* currently, we cannot probe RTCP handle because of blocking restrictions */ + rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2); + if (fd1 > fd_max) + fd_max = fd1; + FD_SET(fd1, &rfds); + } + tv.tv_sec = 0; + tv.tv_usec = 100 * 1000; + n = select(fd_max + 1, &rfds, NULL, NULL, &tv); + if (n > 0) { + for(i = 0; i < rt->nb_rtsp_streams; i++) { + rtsp_st = rt->rtsp_streams[i]; + rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2); + if (FD_ISSET(fd1, &rfds)) { + ret = url_read(rtsp_st->rtp_handle, buf, buf_size); + if (ret > 0) { + *prtsp_st = rtsp_st; + return ret; + } + } + } + } + } +} + +static int rtsp_read_packet(AVFormatContext *s, + AVPacket *pkt) +{ + RTSPState *rt = s->priv_data; + RTSPStream *rtsp_st; + int ret, len; + uint8_t buf[RTP_MAX_PACKET_LENGTH]; + + /* get next frames from the same RTP packet */ + if (rt->cur_rtp) { + ret = rtp_parse_packet(rt->cur_rtp, pkt, NULL, 0); + if (ret == 0) { + rt->cur_rtp = NULL; + return 0; + } else if (ret == 1) { + return 0; + } else { + rt->cur_rtp = NULL; + } + } + + /* read next RTP packet */ + redo: + switch(rt->protocol) { + default: + case RTSP_PROTOCOL_RTP_TCP: + len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf)); + break; + case RTSP_PROTOCOL_RTP_UDP: + case RTSP_PROTOCOL_RTP_UDP_MULTICAST: + len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf)); + if (rtsp_st->rtp_ctx) + rtp_check_and_send_back_rr(rtsp_st->rtp_ctx, len); + break; + } + if (len < 0) + return AVERROR_IO; + ret = rtp_parse_packet(rtsp_st->rtp_ctx, pkt, buf, len); + if (ret < 0) + goto redo; + if (ret == 1) { + /* more packets may follow, so we save the RTP context */ + rt->cur_rtp = rtsp_st->rtp_ctx; + } + return 0; +} + +static int rtsp_read_play(AVFormatContext *s) +{ + RTSPState *rt = s->priv_data; + RTSPHeader reply1, *reply = &reply1; + char cmd[1024]; + + av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state); + + if (rt->state == RTSP_STATE_PAUSED) { + snprintf(cmd, sizeof(cmd), + "PLAY %s RTSP/1.0\r\n", + s->filename); + } else { + snprintf(cmd, sizeof(cmd), + "PLAY %s RTSP/1.0\r\n" + "Range: npt=%0.3f-\r\n", + s->filename, + (double)rt->seek_timestamp / AV_TIME_BASE); + } + rtsp_send_cmd(s, cmd, reply, NULL); + if (reply->status_code != RTSP_STATUS_OK) { + return -1; + } else { + rt->state = RTSP_STATE_PLAYING; + return 0; + } +} + +/* pause the stream */ +static int rtsp_read_pause(AVFormatContext *s) +{ + RTSPState *rt = s->priv_data; + RTSPHeader reply1, *reply = &reply1; + char cmd[1024]; + + rt = s->priv_data; + + if (rt->state != RTSP_STATE_PLAYING) + return 0; + + snprintf(cmd, sizeof(cmd), + "PAUSE %s RTSP/1.0\r\n", + s->filename); + rtsp_send_cmd(s, cmd, reply, NULL); + if (reply->status_code != RTSP_STATUS_OK) { + return -1; + } else { + 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 = timestamp; + switch(rt->state) { + default: + case RTSP_STATE_IDLE: + break; + case RTSP_STATE_PLAYING: + 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; + RTSPHeader reply1, *reply = &reply1; + char cmd[1024]; + +#if 0 + /* NOTE: it is valid to flush the buffer here */ + if (rt->protocol == RTSP_PROTOCOL_RTP_TCP) { + url_fclose(&rt->rtsp_gb); + } +#endif + snprintf(cmd, sizeof(cmd), + "TEARDOWN %s RTSP/1.0\r\n", + s->filename); + rtsp_send_cmd(s, cmd, reply, NULL); + + if (ff_rtsp_callback) { + ff_rtsp_callback(RTSP_ACTION_CLIENT_TEARDOWN, rt->session_id, + NULL, 0, NULL); + } + + rtsp_close_streams(rt); + url_close(rt->rtsp_hd); + return 0; +} + +AVInputFormat rtsp_demuxer = { + "rtsp", + "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, +}; + +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 && 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]; + AVStream *st; + + /* 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]; + + snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d", + inet_ntoa(rtsp_st->sdp_ip), + rtsp_st->sdp_port, + rtsp_st->sdp_ttl); + if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) { + err = AVERROR_INVALIDDATA; + goto fail; + } + /* open the RTP context */ + st = NULL; + if (rtsp_st->stream_index >= 0) + st = s->streams[rtsp_st->stream_index]; + if (!st) + s->ctx_flags |= AVFMTCTX_NOHEADER; + rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data); + if (!rtsp_st->rtp_ctx) { + err = AVERROR_NOMEM; + goto fail; + } else { + if(rtsp_st->dynamic_handler) { + rtsp_st->rtp_ctx->dynamic_protocol_context= rtsp_st->dynamic_protocol_context; + rtsp_st->rtp_ctx->parse_packet= rtsp_st->dynamic_handler->parse_packet; + } + } + } + return 0; + fail: + rtsp_close_streams(rt); + return err; +} + +static int sdp_read_packet(AVFormatContext *s, + AVPacket *pkt) +{ + return rtsp_read_packet(s, pkt); +} + +static int sdp_read_close(AVFormatContext *s) +{ + RTSPState *rt = s->priv_data; + rtsp_close_streams(rt); + return 0; +} + +#ifdef CONFIG_SDP_DEMUXER +AVInputFormat sdp_demuxer = { + "sdp", + "SDP", + sizeof(RTSPState), + sdp_probe, + sdp_read_header, + sdp_read_packet, + sdp_read_close, +}; +#endif + +/* dummy redirector format (used directly in av_open_input_file now) */ +static int redir_probe(AVProbeData *pd) +{ + const char *p; + p = pd->buf; + while (redir_isspace(*p)) + p++; + if (strstart(p, "http://", NULL) || + strstart(p, "rtsp://", NULL)) + return AVPROBE_SCORE_MAX; + return 0; +} + +/* called from utils.c */ +int redir_open(AVFormatContext **ic_ptr, ByteIOContext *f) +{ + char buf[4096], *q; + int c; + AVFormatContext *ic = NULL; + + /* parse each URL and try to open it */ + c = url_fgetc(f); + while (c != URL_EOF) { + /* skip spaces */ + for(;;) { + if (!redir_isspace(c)) + break; + c = url_fgetc(f); + } + if (c == URL_EOF) + break; + /* record url */ + q = buf; + for(;;) { + if (c == URL_EOF || redir_isspace(c)) + break; + if ((q - buf) < sizeof(buf) - 1) + *q++ = c; + c = url_fgetc(f); + } + *q = '\0'; + //printf("URL='%s'\n", buf); + /* try to open the media file */ + if (av_open_input_file(&ic, buf, NULL, 0, NULL) == 0) + break; + } + *ic_ptr = ic; + if (!ic) + return AVERROR_IO; + else + return 0; +} + +AVInputFormat redir_demuxer = { + "redir", + "Redirector format", + 0, + redir_probe, + NULL, + NULL, + NULL, +}; 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