1491 lines
44 KiB
C
1491 lines
44 KiB
C
/*
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* RTSP/SDP client
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* Copyright (c) 2002 Fabrice Bellard.
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avformat.h"
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#include <sys/time.h>
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#include <unistd.h> /* for select() prototype */
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#include "network.h"
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#include "avstring.h"
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#include "rtp_internal.h"
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//#define DEBUG
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//#define DEBUG_RTP_TCP
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enum RTSPClientState {
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RTSP_STATE_IDLE,
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RTSP_STATE_PLAYING,
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RTSP_STATE_PAUSED,
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};
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typedef struct RTSPState {
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URLContext *rtsp_hd; /* RTSP TCP connexion handle */
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int nb_rtsp_streams;
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struct RTSPStream **rtsp_streams;
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enum RTSPClientState state;
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int64_t seek_timestamp;
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/* XXX: currently we use unbuffered input */
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// ByteIOContext rtsp_gb;
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int seq; /* RTSP command sequence number */
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char session_id[512];
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enum RTSPProtocol protocol;
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char last_reply[2048]; /* XXX: allocate ? */
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RTPDemuxContext *cur_rtp;
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} RTSPState;
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typedef struct RTSPStream {
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URLContext *rtp_handle; /* RTP stream handle */
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RTPDemuxContext *rtp_ctx; /* RTP parse context */
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int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */
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int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */
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char control_url[1024]; /* url for this stream (from SDP) */
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int sdp_port; /* port (from SDP content - not used in RTSP) */
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struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */
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int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */
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int sdp_payload_type; /* payload type - only used in SDP */
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rtp_payload_data_t rtp_payload_data; /* rtp payload parsing infos from SDP */
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RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure)
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void *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol)
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} RTSPStream;
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static int rtsp_read_play(AVFormatContext *s);
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/* XXX: currently, the only way to change the protocols consists in
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changing this variable */
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int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_UDP);
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static int rtsp_probe(AVProbeData *p)
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{
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if (av_strstart(p->filename, "rtsp:", NULL))
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return AVPROBE_SCORE_MAX;
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return 0;
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}
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static int redir_isspace(int c)
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{
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return (c == ' ' || c == '\t' || c == '\n' || c == '\r');
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}
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static void skip_spaces(const char **pp)
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{
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const char *p;
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p = *pp;
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while (redir_isspace(*p))
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p++;
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*pp = p;
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}
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static void get_word_sep(char *buf, int buf_size, const char *sep,
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const char **pp)
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{
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const char *p;
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char *q;
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p = *pp;
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if (*p == '/')
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p++;
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skip_spaces(&p);
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q = buf;
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while (!strchr(sep, *p) && *p != '\0') {
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if ((q - buf) < buf_size - 1)
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*q++ = *p;
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p++;
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}
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if (buf_size > 0)
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*q = '\0';
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*pp = p;
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}
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static void get_word(char *buf, int buf_size, const char **pp)
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{
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const char *p;
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char *q;
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p = *pp;
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skip_spaces(&p);
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q = buf;
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while (!redir_isspace(*p) && *p != '\0') {
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if ((q - buf) < buf_size - 1)
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*q++ = *p;
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p++;
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}
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if (buf_size > 0)
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*q = '\0';
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*pp = p;
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}
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/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other
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params>] */
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static int sdp_parse_rtpmap(AVCodecContext *codec, RTSPStream *rtsp_st, int payload_type, const char *p)
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{
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char buf[256];
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int i;
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AVCodec *c;
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const char *c_name;
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/* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
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see if we can handle this kind of payload */
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get_word_sep(buf, sizeof(buf), "/", &p);
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if (payload_type >= RTP_PT_PRIVATE) {
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RTPDynamicProtocolHandler *handler= RTPFirstDynamicPayloadHandler;
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while(handler) {
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if (!strcmp(buf, handler->enc_name) && (codec->codec_type == handler->codec_type)) {
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codec->codec_id = handler->codec_id;
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rtsp_st->dynamic_handler= handler;
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if(handler->open) {
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rtsp_st->dynamic_protocol_context= handler->open();
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}
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break;
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}
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handler= handler->next;
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}
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} else {
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/* We are in a standard case ( from http://www.iana.org/assignments/rtp-parameters) */
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/* search into AVRtpPayloadTypes[] */
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for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
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if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec->codec_type == AVRtpPayloadTypes[i].codec_type)){
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codec->codec_id = AVRtpPayloadTypes[i].codec_id;
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break;
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}
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}
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c = avcodec_find_decoder(codec->codec_id);
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if (c && c->name)
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c_name = c->name;
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else
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c_name = (char *)NULL;
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if (c_name) {
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get_word_sep(buf, sizeof(buf), "/", &p);
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i = atoi(buf);
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switch (codec->codec_type) {
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case CODEC_TYPE_AUDIO:
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av_log(codec, AV_LOG_DEBUG, " audio codec set to : %s\n", c_name);
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codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
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codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
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if (i > 0) {
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codec->sample_rate = i;
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get_word_sep(buf, sizeof(buf), "/", &p);
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i = atoi(buf);
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if (i > 0)
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codec->channels = i;
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// TODO: there is a bug here; if it is a mono stream, and less than 22000Hz, faad upconverts to stereo and twice the
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// frequency. No problem, but the sample rate is being set here by the sdp line. Upcoming patch forthcoming. (rdm)
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}
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av_log(codec, AV_LOG_DEBUG, " audio samplerate set to : %i\n", codec->sample_rate);
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av_log(codec, AV_LOG_DEBUG, " audio channels set to : %i\n", codec->channels);
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break;
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case CODEC_TYPE_VIDEO:
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av_log(codec, AV_LOG_DEBUG, " video codec set to : %s\n", c_name);
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break;
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default:
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break;
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}
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return 0;
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}
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return -1;
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}
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/* return the length and optionnaly the data */
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static int hex_to_data(uint8_t *data, const char *p)
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{
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int c, len, v;
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len = 0;
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v = 1;
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for(;;) {
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skip_spaces(&p);
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if (p == '\0')
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break;
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c = toupper((unsigned char)*p++);
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if (c >= '0' && c <= '9')
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c = c - '0';
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else if (c >= 'A' && c <= 'F')
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c = c - 'A' + 10;
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else
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break;
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v = (v << 4) | c;
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if (v & 0x100) {
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if (data)
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data[len] = v;
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len++;
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v = 1;
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}
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}
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return len;
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}
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static void sdp_parse_fmtp_config(AVCodecContext *codec, char *attr, char *value)
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{
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switch (codec->codec_id) {
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case CODEC_ID_MPEG4:
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case CODEC_ID_AAC:
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if (!strcmp(attr, "config")) {
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/* decode the hexa encoded parameter */
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int len = hex_to_data(NULL, value);
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codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
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if (!codec->extradata)
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return;
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codec->extradata_size = len;
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hex_to_data(codec->extradata, value);
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}
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break;
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default:
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break;
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}
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return;
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}
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typedef struct attrname_map
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{
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const char *str;
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uint16_t type;
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uint32_t offset;
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} attrname_map_t;
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/* All known fmtp parmeters and the corresping RTPAttrTypeEnum */
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#define ATTR_NAME_TYPE_INT 0
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#define ATTR_NAME_TYPE_STR 1
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static attrname_map_t attr_names[]=
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{
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{"SizeLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, sizelength)},
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{"IndexLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexlength)},
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{"IndexDeltaLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexdeltalength)},
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{"profile-level-id", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, profile_level_id)},
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{"StreamType", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, streamtype)},
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{"mode", ATTR_NAME_TYPE_STR, offsetof(rtp_payload_data_t, mode)},
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{NULL, -1, -1},
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};
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/** parse the attribute line from the fmtp a line of an sdp resonse. This is broken out as a function
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* because it is used in rtp_h264.c, which is forthcoming.
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*/
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int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
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{
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skip_spaces(p);
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if(**p)
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{
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get_word_sep(attr, attr_size, "=", p);
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if (**p == '=')
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(*p)++;
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get_word_sep(value, value_size, ";", p);
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if (**p == ';')
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(*p)++;
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return 1;
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}
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return 0;
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}
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/* parse a SDP line and save stream attributes */
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static void sdp_parse_fmtp(AVStream *st, const char *p)
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{
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char attr[256];
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char value[4096];
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int i;
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RTSPStream *rtsp_st = st->priv_data;
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AVCodecContext *codec = st->codec;
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rtp_payload_data_t *rtp_payload_data = &rtsp_st->rtp_payload_data;
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/* loop on each attribute */
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while(rtsp_next_attr_and_value(&p, attr, sizeof(attr), value, sizeof(value)))
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{
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/* grab the codec extra_data from the config parameter of the fmtp line */
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sdp_parse_fmtp_config(codec, attr, value);
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/* Looking for a known attribute */
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for (i = 0; attr_names[i].str; ++i) {
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if (!strcasecmp(attr, attr_names[i].str)) {
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if (attr_names[i].type == ATTR_NAME_TYPE_INT)
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*(int *)((char *)rtp_payload_data + attr_names[i].offset) = atoi(value);
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else if (attr_names[i].type == ATTR_NAME_TYPE_STR)
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*(char **)((char *)rtp_payload_data + attr_names[i].offset) = av_strdup(value);
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}
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}
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}
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}
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/** Parse a string \p in the form of Range:npt=xx-xx, and determine the start
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* and end time.
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* Used for seeking in the rtp stream.
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*/
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static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
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{
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char buf[256];
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skip_spaces(&p);
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if (!av_stristart(p, "npt=", &p))
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return;
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*start = AV_NOPTS_VALUE;
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*end = AV_NOPTS_VALUE;
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get_word_sep(buf, sizeof(buf), "-", &p);
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*start = parse_date(buf, 1);
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if (*p == '-') {
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p++;
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get_word_sep(buf, sizeof(buf), "-", &p);
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*end = parse_date(buf, 1);
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}
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// av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
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// av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
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}
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typedef struct SDPParseState {
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/* SDP only */
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struct in_addr default_ip;
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int default_ttl;
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} SDPParseState;
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static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
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int letter, const char *buf)
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{
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RTSPState *rt = s->priv_data;
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char buf1[64], st_type[64];
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const char *p;
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int codec_type, payload_type, i;
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AVStream *st;
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RTSPStream *rtsp_st;
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struct in_addr sdp_ip;
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int ttl;
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#ifdef DEBUG
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printf("sdp: %c='%s'\n", letter, buf);
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#endif
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p = buf;
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switch(letter) {
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case 'c':
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get_word(buf1, sizeof(buf1), &p);
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if (strcmp(buf1, "IN") != 0)
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return;
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get_word(buf1, sizeof(buf1), &p);
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if (strcmp(buf1, "IP4") != 0)
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return;
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get_word_sep(buf1, sizeof(buf1), "/", &p);
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if (inet_aton(buf1, &sdp_ip) == 0)
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return;
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ttl = 16;
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if (*p == '/') {
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p++;
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get_word_sep(buf1, sizeof(buf1), "/", &p);
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ttl = atoi(buf1);
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}
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if (s->nb_streams == 0) {
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s1->default_ip = sdp_ip;
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s1->default_ttl = ttl;
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} else {
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st = s->streams[s->nb_streams - 1];
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rtsp_st = st->priv_data;
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rtsp_st->sdp_ip = sdp_ip;
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rtsp_st->sdp_ttl = ttl;
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}
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break;
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case 's':
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av_strlcpy(s->title, p, sizeof(s->title));
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break;
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case 'i':
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if (s->nb_streams == 0) {
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av_strlcpy(s->comment, p, sizeof(s->comment));
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break;
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}
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break;
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case 'm':
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/* new stream */
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get_word(st_type, sizeof(st_type), &p);
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if (!strcmp(st_type, "audio")) {
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codec_type = CODEC_TYPE_AUDIO;
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} else if (!strcmp(st_type, "video")) {
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codec_type = CODEC_TYPE_VIDEO;
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} else {
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return;
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}
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rtsp_st = av_mallocz(sizeof(RTSPStream));
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if (!rtsp_st)
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return;
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rtsp_st->stream_index = -1;
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dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
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rtsp_st->sdp_ip = s1->default_ip;
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rtsp_st->sdp_ttl = s1->default_ttl;
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get_word(buf1, sizeof(buf1), &p); /* port */
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rtsp_st->sdp_port = atoi(buf1);
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get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
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/* XXX: handle list of formats */
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get_word(buf1, sizeof(buf1), &p); /* format list */
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rtsp_st->sdp_payload_type = atoi(buf1);
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if (!strcmp(AVRtpPayloadTypes[rtsp_st->sdp_payload_type].enc_name, "MP2T")) {
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/* no corresponding stream */
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} else {
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st = av_new_stream(s, 0);
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if (!st)
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return;
|
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st->priv_data = rtsp_st;
|
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rtsp_st->stream_index = st->index;
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st->codec->codec_type = codec_type;
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if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
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/* if standard payload type, we can find the codec right now */
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rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
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}
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}
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/* put a default control url */
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av_strlcpy(rtsp_st->control_url, s->filename, sizeof(rtsp_st->control_url));
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break;
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case 'a':
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if (av_strstart(p, "control:", &p) && s->nb_streams > 0) {
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char proto[32];
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/* get the control url */
|
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st = s->streams[s->nb_streams - 1];
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rtsp_st = st->priv_data;
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|
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/* XXX: may need to add full url resolution */
|
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url_split(proto, sizeof(proto), NULL, 0, NULL, 0, NULL, NULL, 0, p);
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if (proto[0] == '\0') {
|
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/* relative control URL */
|
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av_strlcat(rtsp_st->control_url, "/", sizeof(rtsp_st->control_url));
|
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av_strlcat(rtsp_st->control_url, p, sizeof(rtsp_st->control_url));
|
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} else {
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av_strlcpy(rtsp_st->control_url, p, sizeof(rtsp_st->control_url));
|
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}
|
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} else if (av_strstart(p, "rtpmap:", &p)) {
|
|
/* NOTE: rtpmap is only supported AFTER the 'm=' tag */
|
|
get_word(buf1, sizeof(buf1), &p);
|
|
payload_type = atoi(buf1);
|
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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 (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) {
|
|
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(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) {
|
|
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(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;
|
|
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 (av_stristart(p, "Session:", &p)) {
|
|
get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
|
|
} 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);
|
|
}
|
|
}
|
|
|
|
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 = AV_RB16(buf + 1);
|
|
#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++;
|
|
av_strlcpy(buf, cmd, sizeof(buf));
|
|
snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq);
|
|
av_strlcat(buf, buf1, sizeof(buf));
|
|
if (rt->session_id[0] != '\0' && !strstr(cmd, "\nIf-Match:")) {
|
|
snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id);
|
|
av_strlcat(buf, buf1, sizeof(buf));
|
|
}
|
|
av_strlcat(buf, "\r\n", sizeof(buf));
|
|
#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);
|
|
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_readbuf(rt->rtsp_hd, content, content_length);
|
|
content[content_length] = '\0';
|
|
}
|
|
if (content_ptr)
|
|
*content_ptr = content;
|
|
else
|
|
av_free(content);
|
|
}
|
|
|
|
|
|
/* 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], *option_list, *option;
|
|
URLContext *rtsp_hd;
|
|
int port, i, j, ret, err;
|
|
RTSPHeader reply1, *reply = &reply1;
|
|
unsigned char *content = NULL;
|
|
RTSPStream *rtsp_st;
|
|
int protocol_mask = 0;
|
|
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;
|
|
|
|
/* search for options */
|
|
option_list = strchr(path, '?');
|
|
if (option_list) {
|
|
/* remove the options from the path */
|
|
*option_list++ = 0;
|
|
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") == 0)
|
|
protocol_mask = (1<< RTSP_PROTOCOL_RTP_UDP);
|
|
else if (strcmp(option, "multicast") == 0)
|
|
protocol_mask = (1<< RTSP_PROTOCOL_RTP_UDP_MULTICAST);
|
|
else if (strcmp(option, "tcp") == 0)
|
|
protocol_mask = (1<< RTSP_PROTOCOL_RTP_TCP);
|
|
}
|
|
}
|
|
|
|
if (!protocol_mask)
|
|
protocol_mask = rtsp_default_protocols;
|
|
|
|
/* open the tcp connexion */
|
|
snprintf(tcpname, sizeof(tcpname), "tcp://%s:%d", host, port);
|
|
if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0)
|
|
return AVERROR(EIO);
|
|
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;
|
|
}
|
|
|
|
/* 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')
|
|
av_strlcat(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')
|
|
av_strlcat(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')
|
|
av_strlcat(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];
|
|
struct in_addr in;
|
|
|
|
in.s_addr = htonl(reply->transports[0].destination);
|
|
snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d",
|
|
inet_ntoa(in),
|
|
reply->transports[0].port_min,
|
|
reply->transports[0].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(ENOMEM);
|
|
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;
|
|
}
|
|
}
|
|
}
|
|
|
|
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 = AV_RB16(buf + 1);
|
|
#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 (len >=0 && rtsp_st->rtp_ctx)
|
|
rtp_check_and_send_back_rr(rtsp_st->rtp_ctx, len);
|
|
break;
|
|
}
|
|
if (len < 0)
|
|
return AVERROR(EIO);
|
|
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);
|
|
|
|
rtsp_close_streams(rt);
|
|
url_close(rt->rtsp_hd);
|
|
return 0;
|
|
}
|
|
|
|
#ifdef CONFIG_RTSP_DEMUXER
|
|
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,
|
|
};
|
|
#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];
|
|
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(ENOMEM);
|
|
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
|
|
|
|
#ifdef CONFIG_REDIR_DEMUXER
|
|
/* 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 (av_strstart(p, "http://", NULL) ||
|
|
av_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(EIO);
|
|
else
|
|
return 0;
|
|
}
|
|
|
|
AVInputFormat redir_demuxer = {
|
|
"redir",
|
|
"Redirector format",
|
|
0,
|
|
redir_probe,
|
|
NULL,
|
|
NULL,
|
|
NULL,
|
|
};
|
|
#endif
|