mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2024-12-15 19:55:08 +00:00
bbd8f5477d
This also reverts SVN rev 26016, which incorrectly overwrote the time base with 90 kHz for all streams, regardless of what was set by the SDP parsing. The stream that triggered the fix in 26016 still works after this commit. Originally committed as revision 26022 to svn://svn.ffmpeg.org/ffmpeg/trunk
1837 lines
64 KiB
C
1837 lines
64 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 "libavutil/base64.h"
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#include "libavutil/avstring.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/random_seed.h"
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#include "avformat.h"
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#include <sys/time.h>
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#if HAVE_SYS_SELECT_H
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#include <sys/select.h>
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#endif
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#include <strings.h>
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#include "internal.h"
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#include "network.h"
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#include "os_support.h"
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#include "http.h"
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#include "rtsp.h"
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#include "rtpdec.h"
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#include "rdt.h"
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#include "rtpdec_formats.h"
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#include "rtpenc_chain.h"
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//#define DEBUG
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//#define DEBUG_RTP_TCP
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/* Timeout values for socket select, in ms,
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* and read_packet(), in seconds */
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#define SELECT_TIMEOUT_MS 100
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#define READ_PACKET_TIMEOUT_S 10
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#define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / SELECT_TIMEOUT_MS
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#define SDP_MAX_SIZE 16384
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#define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
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static void get_word_until_chars(char *buf, int buf_size,
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const char *sep, 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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p += strspn(p, SPACE_CHARS);
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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_sep(char *buf, int buf_size, const char *sep,
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const char **pp)
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{
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if (**pp == '/') (*pp)++;
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get_word_until_chars(buf, buf_size, sep, pp);
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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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get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
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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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p += strspn(p, SPACE_CHARS);
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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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static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
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{
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struct addrinfo hints, *ai = NULL;
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memset(&hints, 0, sizeof(hints));
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hints.ai_flags = AI_NUMERICHOST;
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if (getaddrinfo(buf, NULL, &hints, &ai))
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return -1;
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memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
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freeaddrinfo(ai);
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return 0;
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}
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#if CONFIG_RTPDEC
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static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
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RTSPStream *rtsp_st, AVCodecContext *codec)
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{
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if (!handler)
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return;
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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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/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
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static int sdp_parse_rtpmap(AVFormatContext *s,
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AVStream *st, RTSPStream *rtsp_st,
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int payload_type, const char *p)
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{
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AVCodecContext *codec = st->codec;
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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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* The space should normally not be there but some Real streams or
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* particular servers ("RealServer Version 6.1.3.970", see issue 1658)
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* have a trailing space. */
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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 =
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ff_rtp_handler_find_by_name(buf, codec->codec_type);
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init_rtp_handler(handler, rtsp_st, codec);
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/* If no dynamic handler was found, check with the list of standard
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* allocated types, if such a stream for some reason happens to
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* use a private payload type. This isn't handled in rtpdec.c, since
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* the format name from the rtpmap line never is passed into rtpdec. */
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if (!rtsp_st->dynamic_handler)
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codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
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} else {
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/* We are in a standard case
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* (from http://www.iana.org/assignments/rtp-parameters). */
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/* search into AVRtpPayloadTypes[] */
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codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
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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 = "(null)";
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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 AVMEDIA_TYPE_AUDIO:
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av_log(s, 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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av_set_pts_info(st, 32, 1, codec->sample_rate);
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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
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// less than 22000Hz, faad upconverts to stereo and twice
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// the frequency. No problem, but the sample rate is being
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// set here by the sdp line. Patch on its way. (rdm)
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}
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av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
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codec->sample_rate);
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av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
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codec->channels);
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break;
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case AVMEDIA_TYPE_VIDEO:
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av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
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if (i > 0)
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av_set_pts_info(st, 32, 1, i);
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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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/* parse the attribute line from the fmtp a line of an sdp response. This
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* is broken out as a function because it is used in rtp_h264.c, which is
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* forthcoming. */
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int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
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char *value, int value_size)
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{
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*p += strspn(*p, SPACE_CHARS);
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if (**p) {
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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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typedef struct SDPParseState {
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/* SDP only */
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struct sockaddr_storage default_ip;
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int default_ttl;
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int skip_media; ///< set if an unknown m= line occurs
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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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enum AVMediaType codec_type;
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int payload_type, i;
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AVStream *st;
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RTSPStream *rtsp_st;
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struct sockaddr_storage sdp_ip;
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int ttl;
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dprintf(s, "sdp: %c='%s'\n", letter, buf);
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p = buf;
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if (s1->skip_media && letter != 'm')
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return;
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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") && strcmp(buf1, "IP6"))
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return;
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get_word_sep(buf1, sizeof(buf1), "/", &p);
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if (get_sockaddr(buf1, &sdp_ip))
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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_metadata_set2(&s->metadata, "title", p, 0);
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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_metadata_set2(&s->metadata, "comment", p, 0);
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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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s1->skip_media = 0;
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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 = AVMEDIA_TYPE_AUDIO;
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} else if (!strcmp(st_type, "video")) {
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codec_type = AVMEDIA_TYPE_VIDEO;
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} else if (!strcmp(st_type, "application")) {
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codec_type = AVMEDIA_TYPE_DATA;
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} else {
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s1->skip_media = 1;
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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(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "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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RTPDynamicProtocolHandler *handler;
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/* if standard payload type, we can find the codec right now */
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ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
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if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
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st->codec->sample_rate > 0)
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av_set_pts_info(st, 32, 1, st->codec->sample_rate);
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/* Even static payload types may need a custom depacketizer */
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handler = ff_rtp_handler_find_by_id(
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rtsp_st->sdp_payload_type, st->codec->codec_type);
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init_rtp_handler(handler, rtsp_st, st->codec);
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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, rt->control_uri,
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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)) {
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if (s->nb_streams == 0) {
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if (!strncmp(p, "rtsp://", 7))
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av_strlcpy(rt->control_uri, p,
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sizeof(rt->control_uri));
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} else {
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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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/* XXX: may need to add full url resolution */
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av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
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NULL, NULL, 0, p);
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if (proto[0] == '\0') {
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/* relative control URL */
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if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
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av_strlcat(rtsp_st->control_url, "/",
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sizeof(rtsp_st->control_url));
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av_strlcat(rtsp_st->control_url, p,
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sizeof(rtsp_st->control_url));
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} else
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av_strlcpy(rtsp_st->control_url, p,
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sizeof(rtsp_st->control_url));
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}
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} else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
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/* NOTE: rtpmap is only supported AFTER the 'm=' tag */
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get_word(buf1, sizeof(buf1), &p);
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payload_type = atoi(buf1);
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st = s->streams[s->nb_streams - 1];
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rtsp_st = st->priv_data;
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sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
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} else if (av_strstart(p, "fmtp:", &p) ||
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av_strstart(p, "framesize:", &p)) {
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/* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
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// let dynamic protocol handlers have a stab at the line.
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get_word(buf1, sizeof(buf1), &p);
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payload_type = atoi(buf1);
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for (i = 0; i < s->nb_streams; i++) {
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st = s->streams[i];
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rtsp_st = st->priv_data;
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if (rtsp_st->sdp_payload_type == payload_type &&
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rtsp_st->dynamic_handler &&
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rtsp_st->dynamic_handler->parse_sdp_a_line)
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rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
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rtsp_st->dynamic_protocol_context, buf);
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}
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} else if (av_strstart(p, "range:", &p)) {
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int64_t start, end;
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// this is so that seeking on a streamed file can work.
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rtsp_parse_range_npt(p, &start, &end);
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s->start_time = start;
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/* AV_NOPTS_VALUE means live broadcast (and can't seek) */
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s->duration = (end == AV_NOPTS_VALUE) ?
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AV_NOPTS_VALUE : end - start;
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} else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
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if (atoi(p) == 1)
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rt->transport = RTSP_TRANSPORT_RDT;
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} else if (av_strstart(p, "SampleRate:integer;", &p) &&
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s->nb_streams > 0) {
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st = s->streams[s->nb_streams - 1];
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st->codec->sample_rate = atoi(p);
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} else {
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if (rt->server_type == RTSP_SERVER_WMS)
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ff_wms_parse_sdp_a_line(s, p);
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if (s->nb_streams > 0) {
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if (rt->server_type == RTSP_SERVER_REAL)
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ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
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rtsp_st = s->streams[s->nb_streams - 1]->priv_data;
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if (rtsp_st->dynamic_handler &&
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rtsp_st->dynamic_handler->parse_sdp_a_line)
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rtsp_st->dynamic_handler->parse_sdp_a_line(s,
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s->nb_streams - 1,
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rtsp_st->dynamic_protocol_context, buf);
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}
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}
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break;
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}
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}
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|
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int ff_sdp_parse(AVFormatContext *s, const char *content)
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{
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const char *p;
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int letter;
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/* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
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* contain long SDP lines containing complete ASF Headers (several
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* kB) or arrays of MDPR (RM stream descriptor) headers plus
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* "rulebooks" describing their properties. Therefore, the SDP line
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* buffer is large.
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*
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* The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
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* in rtpdec_xiph.c. */
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char buf[16384], *q;
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SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
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memset(s1, 0, sizeof(SDPParseState));
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p = content;
|
|
for (;;) {
|
|
p += strspn(p, SPACE_CHARS);
|
|
letter = *p;
|
|
if (letter == '\0')
|
|
break;
|
|
p++;
|
|
if (*p != '=')
|
|
goto next_line;
|
|
p++;
|
|
/* get the content */
|
|
q = buf;
|
|
while (*p != '\n' && *p != '\r' && *p != '\0') {
|
|
if ((q - buf) < sizeof(buf) - 1)
|
|
*q++ = *p;
|
|
p++;
|
|
}
|
|
*q = '\0';
|
|
sdp_parse_line(s, s1, letter, buf);
|
|
next_line:
|
|
while (*p != '\n' && *p != '\0')
|
|
p++;
|
|
if (*p == '\n')
|
|
p++;
|
|
}
|
|
return 0;
|
|
}
|
|
#endif /* CONFIG_RTPDEC */
|
|
|
|
/* close and free RTSP streams */
|
|
void ff_rtsp_close_streams(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int i;
|
|
RTSPStream *rtsp_st;
|
|
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
if (rtsp_st) {
|
|
if (rtsp_st->transport_priv) {
|
|
if (s->oformat) {
|
|
AVFormatContext *rtpctx = rtsp_st->transport_priv;
|
|
av_write_trailer(rtpctx);
|
|
if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
|
|
uint8_t *ptr;
|
|
url_close_dyn_buf(rtpctx->pb, &ptr);
|
|
av_free(ptr);
|
|
} else {
|
|
url_fclose(rtpctx->pb);
|
|
}
|
|
av_metadata_free(&rtpctx->streams[0]->metadata);
|
|
av_metadata_free(&rtpctx->metadata);
|
|
av_free(rtpctx->streams[0]);
|
|
av_free(rtpctx);
|
|
} else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
|
|
ff_rdt_parse_close(rtsp_st->transport_priv);
|
|
else if (CONFIG_RTPDEC)
|
|
rtp_parse_close(rtsp_st->transport_priv);
|
|
}
|
|
if (rtsp_st->rtp_handle)
|
|
url_close(rtsp_st->rtp_handle);
|
|
if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
|
|
rtsp_st->dynamic_handler->close(
|
|
rtsp_st->dynamic_protocol_context);
|
|
}
|
|
}
|
|
av_free(rt->rtsp_streams);
|
|
if (rt->asf_ctx) {
|
|
av_close_input_stream (rt->asf_ctx);
|
|
rt->asf_ctx = NULL;
|
|
}
|
|
av_free(rt->recvbuf);
|
|
}
|
|
|
|
static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
AVStream *st = NULL;
|
|
|
|
/* open the RTP context */
|
|
if (rtsp_st->stream_index >= 0)
|
|
st = s->streams[rtsp_st->stream_index];
|
|
if (!st)
|
|
s->ctx_flags |= AVFMTCTX_NOHEADER;
|
|
|
|
if (s->oformat && CONFIG_RTSP_MUXER) {
|
|
rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
|
|
rtsp_st->rtp_handle,
|
|
RTSP_TCP_MAX_PACKET_SIZE);
|
|
/* Ownership of rtp_handle is passed to the rtp mux context */
|
|
rtsp_st->rtp_handle = NULL;
|
|
} else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
|
|
rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
|
|
rtsp_st->dynamic_protocol_context,
|
|
rtsp_st->dynamic_handler);
|
|
else if (CONFIG_RTPDEC)
|
|
rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
|
|
rtsp_st->sdp_payload_type,
|
|
(rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
|
|
? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
|
|
|
|
if (!rtsp_st->transport_priv) {
|
|
return AVERROR(ENOMEM);
|
|
} else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
|
|
if (rtsp_st->dynamic_handler) {
|
|
rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
|
|
rtsp_st->dynamic_protocol_context,
|
|
rtsp_st->dynamic_handler);
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
#if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
|
|
static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
|
|
{
|
|
const char *p;
|
|
int v;
|
|
|
|
p = *pp;
|
|
p += strspn(p, SPACE_CHARS);
|
|
v = strtol(p, (char **)&p, 10);
|
|
if (*p == '-') {
|
|
p++;
|
|
*min_ptr = v;
|
|
v = strtol(p, (char **)&p, 10);
|
|
*max_ptr = v;
|
|
} else {
|
|
*min_ptr = v;
|
|
*max_ptr = v;
|
|
}
|
|
*pp = p;
|
|
}
|
|
|
|
/* XXX: only one transport specification is parsed */
|
|
static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
|
|
{
|
|
char transport_protocol[16];
|
|
char profile[16];
|
|
char lower_transport[16];
|
|
char parameter[16];
|
|
RTSPTransportField *th;
|
|
char buf[256];
|
|
|
|
reply->nb_transports = 0;
|
|
|
|
for (;;) {
|
|
p += strspn(p, SPACE_CHARS);
|
|
if (*p == '\0')
|
|
break;
|
|
|
|
th = &reply->transports[reply->nb_transports];
|
|
|
|
get_word_sep(transport_protocol, sizeof(transport_protocol),
|
|
"/", &p);
|
|
if (!strcasecmp (transport_protocol, "rtp")) {
|
|
get_word_sep(profile, sizeof(profile), "/;,", &p);
|
|
lower_transport[0] = '\0';
|
|
/* rtp/avp/<protocol> */
|
|
if (*p == '/') {
|
|
get_word_sep(lower_transport, sizeof(lower_transport),
|
|
";,", &p);
|
|
}
|
|
th->transport = RTSP_TRANSPORT_RTP;
|
|
} else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
|
|
!strcasecmp (transport_protocol, "x-real-rdt")) {
|
|
/* x-pn-tng/<protocol> */
|
|
get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
|
|
profile[0] = '\0';
|
|
th->transport = RTSP_TRANSPORT_RDT;
|
|
}
|
|
if (!strcasecmp(lower_transport, "TCP"))
|
|
th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
|
|
else
|
|
th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
|
|
|
|
if (*p == ';')
|
|
p++;
|
|
/* get each parameter */
|
|
while (*p != '\0' && *p != ',') {
|
|
get_word_sep(parameter, sizeof(parameter), "=;,", &p);
|
|
if (!strcmp(parameter, "port")) {
|
|
if (*p == '=') {
|
|
p++;
|
|
rtsp_parse_range(&th->port_min, &th->port_max, &p);
|
|
}
|
|
} else if (!strcmp(parameter, "client_port")) {
|
|
if (*p == '=') {
|
|
p++;
|
|
rtsp_parse_range(&th->client_port_min,
|
|
&th->client_port_max, &p);
|
|
}
|
|
} else if (!strcmp(parameter, "server_port")) {
|
|
if (*p == '=') {
|
|
p++;
|
|
rtsp_parse_range(&th->server_port_min,
|
|
&th->server_port_max, &p);
|
|
}
|
|
} else if (!strcmp(parameter, "interleaved")) {
|
|
if (*p == '=') {
|
|
p++;
|
|
rtsp_parse_range(&th->interleaved_min,
|
|
&th->interleaved_max, &p);
|
|
}
|
|
} else if (!strcmp(parameter, "multicast")) {
|
|
if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
|
|
th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
|
|
} else if (!strcmp(parameter, "ttl")) {
|
|
if (*p == '=') {
|
|
p++;
|
|
th->ttl = strtol(p, (char **)&p, 10);
|
|
}
|
|
} else if (!strcmp(parameter, "destination")) {
|
|
if (*p == '=') {
|
|
p++;
|
|
get_word_sep(buf, sizeof(buf), ";,", &p);
|
|
get_sockaddr(buf, &th->destination);
|
|
}
|
|
} else if (!strcmp(parameter, "source")) {
|
|
if (*p == '=') {
|
|
p++;
|
|
get_word_sep(buf, sizeof(buf), ";,", &p);
|
|
av_strlcpy(th->source, buf, sizeof(th->source));
|
|
}
|
|
}
|
|
|
|
while (*p != ';' && *p != '\0' && *p != ',')
|
|
p++;
|
|
if (*p == ';')
|
|
p++;
|
|
}
|
|
if (*p == ',')
|
|
p++;
|
|
|
|
reply->nb_transports++;
|
|
}
|
|
}
|
|
|
|
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
|
|
HTTPAuthState *auth_state)
|
|
{
|
|
const char *p;
|
|
|
|
/* NOTE: we do case independent match for broken servers */
|
|
p = buf;
|
|
if (av_stristart(p, "Session:", &p)) {
|
|
int t;
|
|
get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
|
|
if (av_stristart(p, ";timeout=", &p) &&
|
|
(t = strtol(p, NULL, 10)) > 0) {
|
|
reply->timeout = t;
|
|
}
|
|
} else if (av_stristart(p, "Content-Length:", &p)) {
|
|
reply->content_length = strtol(p, NULL, 10);
|
|
} else if (av_stristart(p, "Transport:", &p)) {
|
|
rtsp_parse_transport(reply, p);
|
|
} else if (av_stristart(p, "CSeq:", &p)) {
|
|
reply->seq = strtol(p, NULL, 10);
|
|
} else if (av_stristart(p, "Range:", &p)) {
|
|
rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
|
|
} else if (av_stristart(p, "RealChallenge1:", &p)) {
|
|
p += strspn(p, SPACE_CHARS);
|
|
av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
|
|
} else if (av_stristart(p, "Server:", &p)) {
|
|
p += strspn(p, SPACE_CHARS);
|
|
av_strlcpy(reply->server, p, sizeof(reply->server));
|
|
} else if (av_stristart(p, "Notice:", &p) ||
|
|
av_stristart(p, "X-Notice:", &p)) {
|
|
reply->notice = strtol(p, NULL, 10);
|
|
} else if (av_stristart(p, "Location:", &p)) {
|
|
p += strspn(p, SPACE_CHARS);
|
|
av_strlcpy(reply->location, p , sizeof(reply->location));
|
|
} else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) {
|
|
p += strspn(p, SPACE_CHARS);
|
|
ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p);
|
|
} else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) {
|
|
p += strspn(p, SPACE_CHARS);
|
|
ff_http_auth_handle_header(auth_state, "Authentication-Info", p);
|
|
} else if (av_stristart(p, "Content-Base:", &p)) {
|
|
p += strspn(p, SPACE_CHARS);
|
|
av_strlcpy(reply->content_base, p , sizeof(reply->content_base));
|
|
}
|
|
}
|
|
|
|
/* skip a RTP/TCP interleaved packet */
|
|
void ff_rtsp_skip_packet(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int ret, len, len1;
|
|
uint8_t buf[1024];
|
|
|
|
ret = url_read_complete(rt->rtsp_hd, buf, 3);
|
|
if (ret != 3)
|
|
return;
|
|
len = AV_RB16(buf + 1);
|
|
|
|
dprintf(s, "skipping RTP packet len=%d\n", len);
|
|
|
|
/* skip payload */
|
|
while (len > 0) {
|
|
len1 = len;
|
|
if (len1 > sizeof(buf))
|
|
len1 = sizeof(buf);
|
|
ret = url_read_complete(rt->rtsp_hd, buf, len1);
|
|
if (ret != len1)
|
|
return;
|
|
len -= len1;
|
|
}
|
|
}
|
|
|
|
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
|
|
unsigned char **content_ptr,
|
|
int return_on_interleaved_data)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
char buf[4096], buf1[1024], *q;
|
|
unsigned char ch;
|
|
const char *p;
|
|
int ret, content_length, line_count = 0;
|
|
unsigned char *content = NULL;
|
|
|
|
memset(reply, 0, sizeof(*reply));
|
|
|
|
/* parse reply (XXX: use buffers) */
|
|
rt->last_reply[0] = '\0';
|
|
for (;;) {
|
|
q = buf;
|
|
for (;;) {
|
|
ret = url_read_complete(rt->rtsp_hd, &ch, 1);
|
|
#ifdef DEBUG_RTP_TCP
|
|
dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
|
|
#endif
|
|
if (ret != 1)
|
|
return AVERROR_EOF;
|
|
if (ch == '\n')
|
|
break;
|
|
if (ch == '$') {
|
|
/* XXX: only parse it if first char on line ? */
|
|
if (return_on_interleaved_data) {
|
|
return 1;
|
|
} else
|
|
ff_rtsp_skip_packet(s);
|
|
} else if (ch != '\r') {
|
|
if ((q - buf) < sizeof(buf) - 1)
|
|
*q++ = ch;
|
|
}
|
|
}
|
|
*q = '\0';
|
|
|
|
dprintf(s, "line='%s'\n", buf);
|
|
|
|
/* test if last line */
|
|
if (buf[0] == '\0')
|
|
break;
|
|
p = buf;
|
|
if (line_count == 0) {
|
|
/* get reply code */
|
|
get_word(buf1, sizeof(buf1), &p);
|
|
get_word(buf1, sizeof(buf1), &p);
|
|
reply->status_code = atoi(buf1);
|
|
av_strlcpy(reply->reason, p, sizeof(reply->reason));
|
|
} else {
|
|
ff_rtsp_parse_line(reply, p, &rt->auth_state);
|
|
av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
|
|
av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
|
|
}
|
|
line_count++;
|
|
}
|
|
|
|
if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
|
|
av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
|
|
|
|
content_length = reply->content_length;
|
|
if (content_length > 0) {
|
|
/* leave some room for a trailing '\0' (useful for simple parsing) */
|
|
content = av_malloc(content_length + 1);
|
|
(void)url_read_complete(rt->rtsp_hd, content, content_length);
|
|
content[content_length] = '\0';
|
|
}
|
|
if (content_ptr)
|
|
*content_ptr = content;
|
|
else
|
|
av_free(content);
|
|
|
|
if (rt->seq != reply->seq) {
|
|
av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
|
|
rt->seq, reply->seq);
|
|
}
|
|
|
|
/* EOS */
|
|
if (reply->notice == 2101 /* End-of-Stream Reached */ ||
|
|
reply->notice == 2104 /* Start-of-Stream Reached */ ||
|
|
reply->notice == 2306 /* Continuous Feed Terminated */) {
|
|
rt->state = RTSP_STATE_IDLE;
|
|
} else if (reply->notice >= 4400 && reply->notice < 5500) {
|
|
return AVERROR(EIO); /* data or server error */
|
|
} else if (reply->notice == 2401 /* Ticket Expired */ ||
|
|
(reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
|
|
return AVERROR(EPERM);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
|
|
const char *method, const char *url,
|
|
const char *headers,
|
|
const unsigned char *send_content,
|
|
int send_content_length)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
char buf[4096], *out_buf;
|
|
char base64buf[AV_BASE64_SIZE(sizeof(buf))];
|
|
|
|
/* Add in RTSP headers */
|
|
out_buf = buf;
|
|
rt->seq++;
|
|
snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
|
|
if (headers)
|
|
av_strlcat(buf, headers, sizeof(buf));
|
|
av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
|
|
if (rt->session_id[0] != '\0' && (!headers ||
|
|
!strstr(headers, "\nIf-Match:"))) {
|
|
av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
|
|
}
|
|
if (rt->auth[0]) {
|
|
char *str = ff_http_auth_create_response(&rt->auth_state,
|
|
rt->auth, url, method);
|
|
if (str)
|
|
av_strlcat(buf, str, sizeof(buf));
|
|
av_free(str);
|
|
}
|
|
if (send_content_length > 0 && send_content)
|
|
av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
|
|
av_strlcat(buf, "\r\n", sizeof(buf));
|
|
|
|
/* base64 encode rtsp if tunneling */
|
|
if (rt->control_transport == RTSP_MODE_TUNNEL) {
|
|
av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
|
|
out_buf = base64buf;
|
|
}
|
|
|
|
dprintf(s, "Sending:\n%s--\n", buf);
|
|
|
|
url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
|
|
if (send_content_length > 0 && send_content) {
|
|
if (rt->control_transport == RTSP_MODE_TUNNEL) {
|
|
av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
|
|
"with content data not supported\n");
|
|
return AVERROR_PATCHWELCOME;
|
|
}
|
|
url_write(rt->rtsp_hd_out, send_content, send_content_length);
|
|
}
|
|
rt->last_cmd_time = av_gettime();
|
|
|
|
return 0;
|
|
}
|
|
|
|
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
|
|
const char *url, const char *headers)
|
|
{
|
|
return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
|
|
}
|
|
|
|
int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
|
|
const char *headers, RTSPMessageHeader *reply,
|
|
unsigned char **content_ptr)
|
|
{
|
|
return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
|
|
content_ptr, NULL, 0);
|
|
}
|
|
|
|
int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
|
|
const char *method, const char *url,
|
|
const char *header,
|
|
RTSPMessageHeader *reply,
|
|
unsigned char **content_ptr,
|
|
const unsigned char *send_content,
|
|
int send_content_length)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
HTTPAuthType cur_auth_type;
|
|
int ret;
|
|
|
|
retry:
|
|
cur_auth_type = rt->auth_state.auth_type;
|
|
if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
|
|
send_content,
|
|
send_content_length)))
|
|
return ret;
|
|
|
|
if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0) ) < 0)
|
|
return ret;
|
|
|
|
if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
|
|
rt->auth_state.auth_type != HTTP_AUTH_NONE)
|
|
goto retry;
|
|
|
|
if (reply->status_code > 400){
|
|
av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
|
|
method,
|
|
reply->status_code,
|
|
reply->reason);
|
|
av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* @return 0 on success, <0 on error, 1 if protocol is unavailable.
|
|
*/
|
|
static int make_setup_request(AVFormatContext *s, const char *host, int port,
|
|
int lower_transport, const char *real_challenge)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int rtx, j, i, err, interleave = 0;
|
|
RTSPStream *rtsp_st;
|
|
RTSPMessageHeader reply1, *reply = &reply1;
|
|
char cmd[2048];
|
|
const char *trans_pref;
|
|
|
|
if (rt->transport == RTSP_TRANSPORT_RDT)
|
|
trans_pref = "x-pn-tng";
|
|
else
|
|
trans_pref = "RTP/AVP";
|
|
|
|
/* default timeout: 1 minute */
|
|
rt->timeout = 60;
|
|
|
|
/* for each stream, make the setup request */
|
|
/* XXX: we assume the same server is used for the control of each
|
|
* RTSP stream */
|
|
|
|
for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
|
|
char transport[2048];
|
|
|
|
/**
|
|
* WMS serves all UDP data over a single connection, the RTX, which
|
|
* isn't necessarily the first in the SDP but has to be the first
|
|
* to be set up, else the second/third SETUP will fail with a 461.
|
|
*/
|
|
if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
|
|
rt->server_type == RTSP_SERVER_WMS) {
|
|
if (i == 0) {
|
|
/* rtx first */
|
|
for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
|
|
int len = strlen(rt->rtsp_streams[rtx]->control_url);
|
|
if (len >= 4 &&
|
|
!strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
|
|
"/rtx"))
|
|
break;
|
|
}
|
|
if (rtx == rt->nb_rtsp_streams)
|
|
return -1; /* no RTX found */
|
|
rtsp_st = rt->rtsp_streams[rtx];
|
|
} else
|
|
rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
|
|
} else
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
|
|
/* RTP/UDP */
|
|
if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
|
|
char buf[256];
|
|
|
|
if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
|
|
port = reply->transports[0].client_port_min;
|
|
goto have_port;
|
|
}
|
|
|
|
/* first try in specified port range */
|
|
if (RTSP_RTP_PORT_MIN != 0) {
|
|
while (j <= RTSP_RTP_PORT_MAX) {
|
|
ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
|
|
"?localport=%d", j);
|
|
/* we will use two ports per rtp stream (rtp and rtcp) */
|
|
j += 2;
|
|
if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
|
|
goto rtp_opened;
|
|
}
|
|
}
|
|
|
|
#if 0
|
|
/* then try on any port */
|
|
if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
|
|
err = AVERROR_INVALIDDATA;
|
|
goto fail;
|
|
}
|
|
#endif
|
|
|
|
rtp_opened:
|
|
port = rtp_get_local_rtp_port(rtsp_st->rtp_handle);
|
|
have_port:
|
|
snprintf(transport, sizeof(transport) - 1,
|
|
"%s/UDP;", trans_pref);
|
|
if (rt->server_type != RTSP_SERVER_REAL)
|
|
av_strlcat(transport, "unicast;", sizeof(transport));
|
|
av_strlcatf(transport, sizeof(transport),
|
|
"client_port=%d", port);
|
|
if (rt->transport == RTSP_TRANSPORT_RTP &&
|
|
!(rt->server_type == RTSP_SERVER_WMS && i > 0))
|
|
av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
|
|
}
|
|
|
|
/* RTP/TCP */
|
|
else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
|
|
/** For WMS streams, the application streams are only used for
|
|
* UDP. When trying to set it up for TCP streams, the server
|
|
* will return an error. Therefore, we skip those streams. */
|
|
if (rt->server_type == RTSP_SERVER_WMS &&
|
|
s->streams[rtsp_st->stream_index]->codec->codec_type ==
|
|
AVMEDIA_TYPE_DATA)
|
|
continue;
|
|
snprintf(transport, sizeof(transport) - 1,
|
|
"%s/TCP;", trans_pref);
|
|
if (rt->server_type == RTSP_SERVER_WMS)
|
|
av_strlcat(transport, "unicast;", sizeof(transport));
|
|
av_strlcatf(transport, sizeof(transport),
|
|
"interleaved=%d-%d",
|
|
interleave, interleave + 1);
|
|
interleave += 2;
|
|
}
|
|
|
|
else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
|
|
snprintf(transport, sizeof(transport) - 1,
|
|
"%s/UDP;multicast", trans_pref);
|
|
}
|
|
if (s->oformat) {
|
|
av_strlcat(transport, ";mode=receive", sizeof(transport));
|
|
} else if (rt->server_type == RTSP_SERVER_REAL ||
|
|
rt->server_type == RTSP_SERVER_WMS)
|
|
av_strlcat(transport, ";mode=play", sizeof(transport));
|
|
snprintf(cmd, sizeof(cmd),
|
|
"Transport: %s\r\n",
|
|
transport);
|
|
if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
|
|
char real_res[41], real_csum[9];
|
|
ff_rdt_calc_response_and_checksum(real_res, real_csum,
|
|
real_challenge);
|
|
av_strlcatf(cmd, sizeof(cmd),
|
|
"If-Match: %s\r\n"
|
|
"RealChallenge2: %s, sd=%s\r\n",
|
|
rt->session_id, real_res, real_csum);
|
|
}
|
|
ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
|
|
if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
|
|
err = 1;
|
|
goto fail;
|
|
} else if (reply->status_code != RTSP_STATUS_OK ||
|
|
reply->nb_transports != 1) {
|
|
err = AVERROR_INVALIDDATA;
|
|
goto fail;
|
|
}
|
|
|
|
/* XXX: same protocol for all streams is required */
|
|
if (i > 0) {
|
|
if (reply->transports[0].lower_transport != rt->lower_transport ||
|
|
reply->transports[0].transport != rt->transport) {
|
|
err = AVERROR_INVALIDDATA;
|
|
goto fail;
|
|
}
|
|
} else {
|
|
rt->lower_transport = reply->transports[0].lower_transport;
|
|
rt->transport = reply->transports[0].transport;
|
|
}
|
|
|
|
/* close RTP connection if not chosen */
|
|
if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&
|
|
(lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {
|
|
url_close(rtsp_st->rtp_handle);
|
|
rtsp_st->rtp_handle = NULL;
|
|
}
|
|
|
|
switch(reply->transports[0].lower_transport) {
|
|
case RTSP_LOWER_TRANSPORT_TCP:
|
|
rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
|
|
rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
|
|
break;
|
|
|
|
case RTSP_LOWER_TRANSPORT_UDP: {
|
|
char url[1024];
|
|
|
|
/* Use source address if specified */
|
|
if (reply->transports[0].source[0]) {
|
|
ff_url_join(url, sizeof(url), "rtp", NULL,
|
|
reply->transports[0].source,
|
|
reply->transports[0].server_port_min, NULL);
|
|
} else {
|
|
ff_url_join(url, sizeof(url), "rtp", NULL, host,
|
|
reply->transports[0].server_port_min, NULL);
|
|
}
|
|
if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
|
|
rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
|
|
err = AVERROR_INVALIDDATA;
|
|
goto fail;
|
|
}
|
|
/* Try to initialize the connection state in a
|
|
* potential NAT router by sending dummy packets.
|
|
* RTP/RTCP dummy packets are used for RDT, too.
|
|
*/
|
|
if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
|
|
CONFIG_RTPDEC)
|
|
rtp_send_punch_packets(rtsp_st->rtp_handle);
|
|
break;
|
|
}
|
|
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
|
|
char url[1024], namebuf[50];
|
|
struct sockaddr_storage addr;
|
|
int port, ttl;
|
|
|
|
if (reply->transports[0].destination.ss_family) {
|
|
addr = reply->transports[0].destination;
|
|
port = reply->transports[0].port_min;
|
|
ttl = reply->transports[0].ttl;
|
|
} else {
|
|
addr = rtsp_st->sdp_ip;
|
|
port = rtsp_st->sdp_port;
|
|
ttl = rtsp_st->sdp_ttl;
|
|
}
|
|
getnameinfo((struct sockaddr*) &addr, sizeof(addr),
|
|
namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
|
|
ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
|
|
port, "?ttl=%d", ttl);
|
|
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
|
|
err = AVERROR_INVALIDDATA;
|
|
goto fail;
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
|
|
goto fail;
|
|
}
|
|
|
|
if (reply->timeout > 0)
|
|
rt->timeout = reply->timeout;
|
|
|
|
if (rt->server_type == RTSP_SERVER_REAL)
|
|
rt->need_subscription = 1;
|
|
|
|
return 0;
|
|
|
|
fail:
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
if (rt->rtsp_streams[i]->rtp_handle) {
|
|
url_close(rt->rtsp_streams[i]->rtp_handle);
|
|
rt->rtsp_streams[i]->rtp_handle = NULL;
|
|
}
|
|
}
|
|
return err;
|
|
}
|
|
|
|
void ff_rtsp_close_connections(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
|
|
url_close(rt->rtsp_hd);
|
|
rt->rtsp_hd = rt->rtsp_hd_out = NULL;
|
|
}
|
|
|
|
int ff_rtsp_connect(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
|
|
char *option_list, *option, *filename;
|
|
int port, err, tcp_fd;
|
|
RTSPMessageHeader reply1 = {0}, *reply = &reply1;
|
|
int lower_transport_mask = 0;
|
|
char real_challenge[64];
|
|
struct sockaddr_storage peer;
|
|
socklen_t peer_len = sizeof(peer);
|
|
|
|
if (!ff_network_init())
|
|
return AVERROR(EIO);
|
|
redirect:
|
|
rt->control_transport = RTSP_MODE_PLAIN;
|
|
/* extract hostname and port */
|
|
av_url_split(NULL, 0, auth, sizeof(auth),
|
|
host, sizeof(host), &port, path, sizeof(path), s->filename);
|
|
if (*auth) {
|
|
av_strlcpy(rt->auth, auth, sizeof(rt->auth));
|
|
}
|
|
if (port < 0)
|
|
port = RTSP_DEFAULT_PORT;
|
|
|
|
/* search for options */
|
|
option_list = strrchr(path, '?');
|
|
if (option_list) {
|
|
/* Strip out the RTSP specific options, write out the rest of
|
|
* the options back into the same string. */
|
|
filename = option_list;
|
|
while (option_list) {
|
|
/* move the option pointer */
|
|
option = ++option_list;
|
|
option_list = strchr(option_list, '&');
|
|
if (option_list)
|
|
*option_list = 0;
|
|
|
|
/* handle the options */
|
|
if (!strcmp(option, "udp")) {
|
|
lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
|
|
} else if (!strcmp(option, "multicast")) {
|
|
lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
|
|
} else if (!strcmp(option, "tcp")) {
|
|
lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
|
|
} else if(!strcmp(option, "http")) {
|
|
lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
|
|
rt->control_transport = RTSP_MODE_TUNNEL;
|
|
} else {
|
|
/* Write options back into the buffer, using memmove instead
|
|
* of strcpy since the strings may overlap. */
|
|
int len = strlen(option);
|
|
memmove(++filename, option, len);
|
|
filename += len;
|
|
if (option_list) *filename = '&';
|
|
}
|
|
}
|
|
*filename = 0;
|
|
}
|
|
|
|
if (!lower_transport_mask)
|
|
lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
|
|
|
|
if (s->oformat) {
|
|
/* Only UDP or TCP - UDP multicast isn't supported. */
|
|
lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
|
|
(1 << RTSP_LOWER_TRANSPORT_TCP);
|
|
if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
|
|
av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
|
|
"only UDP and TCP are supported for output.\n");
|
|
err = AVERROR(EINVAL);
|
|
goto fail;
|
|
}
|
|
}
|
|
|
|
/* Construct the URI used in request; this is similar to s->filename,
|
|
* but with authentication credentials removed and RTSP specific options
|
|
* stripped out. */
|
|
ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
|
|
host, port, "%s", path);
|
|
|
|
if (rt->control_transport == RTSP_MODE_TUNNEL) {
|
|
/* set up initial handshake for tunneling */
|
|
char httpname[1024];
|
|
char sessioncookie[17];
|
|
char headers[1024];
|
|
|
|
ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
|
|
snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
|
|
av_get_random_seed(), av_get_random_seed());
|
|
|
|
/* GET requests */
|
|
if (url_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) {
|
|
err = AVERROR(EIO);
|
|
goto fail;
|
|
}
|
|
|
|
/* generate GET headers */
|
|
snprintf(headers, sizeof(headers),
|
|
"x-sessioncookie: %s\r\n"
|
|
"Accept: application/x-rtsp-tunnelled\r\n"
|
|
"Pragma: no-cache\r\n"
|
|
"Cache-Control: no-cache\r\n",
|
|
sessioncookie);
|
|
ff_http_set_headers(rt->rtsp_hd, headers);
|
|
|
|
/* complete the connection */
|
|
if (url_connect(rt->rtsp_hd)) {
|
|
err = AVERROR(EIO);
|
|
goto fail;
|
|
}
|
|
|
|
/* POST requests */
|
|
if (url_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
|
|
err = AVERROR(EIO);
|
|
goto fail;
|
|
}
|
|
|
|
/* generate POST headers */
|
|
snprintf(headers, sizeof(headers),
|
|
"x-sessioncookie: %s\r\n"
|
|
"Content-Type: application/x-rtsp-tunnelled\r\n"
|
|
"Pragma: no-cache\r\n"
|
|
"Cache-Control: no-cache\r\n"
|
|
"Content-Length: 32767\r\n"
|
|
"Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
|
|
sessioncookie);
|
|
ff_http_set_headers(rt->rtsp_hd_out, headers);
|
|
ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
|
|
|
|
/* Initialize the authentication state for the POST session. The HTTP
|
|
* protocol implementation doesn't properly handle multi-pass
|
|
* authentication for POST requests, since it would require one of
|
|
* the following:
|
|
* - implementing Expect: 100-continue, which many HTTP servers
|
|
* don't support anyway, even less the RTSP servers that do HTTP
|
|
* tunneling
|
|
* - sending the whole POST data until getting a 401 reply specifying
|
|
* what authentication method to use, then resending all that data
|
|
* - waiting for potential 401 replies directly after sending the
|
|
* POST header (waiting for some unspecified time)
|
|
* Therefore, we copy the full auth state, which works for both basic
|
|
* and digest. (For digest, we would have to synchronize the nonce
|
|
* count variable between the two sessions, if we'd do more requests
|
|
* with the original session, though.)
|
|
*/
|
|
ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
|
|
|
|
/* complete the connection */
|
|
if (url_connect(rt->rtsp_hd_out)) {
|
|
err = AVERROR(EIO);
|
|
goto fail;
|
|
}
|
|
} else {
|
|
/* open the tcp connection */
|
|
ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
|
|
if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
|
|
err = AVERROR(EIO);
|
|
goto fail;
|
|
}
|
|
rt->rtsp_hd_out = rt->rtsp_hd;
|
|
}
|
|
rt->seq = 0;
|
|
|
|
tcp_fd = url_get_file_handle(rt->rtsp_hd);
|
|
if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
|
|
getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
|
|
NULL, 0, NI_NUMERICHOST);
|
|
}
|
|
|
|
/* request options supported by the server; this also detects server
|
|
* type */
|
|
for (rt->server_type = RTSP_SERVER_RTP;;) {
|
|
cmd[0] = 0;
|
|
if (rt->server_type == RTSP_SERVER_REAL)
|
|
av_strlcat(cmd,
|
|
/**
|
|
* The following entries are required for proper
|
|
* streaming from a Realmedia server. They are
|
|
* interdependent in some way although we currently
|
|
* don't quite understand how. Values were copied
|
|
* from mplayer SVN r23589.
|
|
* @param CompanyID is a 16-byte ID in base64
|
|
* @param ClientChallenge is a 16-byte ID in hex
|
|
*/
|
|
"ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
|
|
"PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
|
|
"CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
|
|
"GUID: 00000000-0000-0000-0000-000000000000\r\n",
|
|
sizeof(cmd));
|
|
ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
|
|
if (reply->status_code != RTSP_STATUS_OK) {
|
|
err = AVERROR_INVALIDDATA;
|
|
goto fail;
|
|
}
|
|
|
|
/* detect server type if not standard-compliant RTP */
|
|
if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
|
|
rt->server_type = RTSP_SERVER_REAL;
|
|
continue;
|
|
} else if (!strncasecmp(reply->server, "WMServer/", 9)) {
|
|
rt->server_type = RTSP_SERVER_WMS;
|
|
} else if (rt->server_type == RTSP_SERVER_REAL)
|
|
strcpy(real_challenge, reply->real_challenge);
|
|
break;
|
|
}
|
|
|
|
if (s->iformat && CONFIG_RTSP_DEMUXER)
|
|
err = ff_rtsp_setup_input_streams(s, reply);
|
|
else if (CONFIG_RTSP_MUXER)
|
|
err = ff_rtsp_setup_output_streams(s, host);
|
|
if (err)
|
|
goto fail;
|
|
|
|
do {
|
|
int lower_transport = ff_log2_tab[lower_transport_mask &
|
|
~(lower_transport_mask - 1)];
|
|
|
|
err = make_setup_request(s, host, port, lower_transport,
|
|
rt->server_type == RTSP_SERVER_REAL ?
|
|
real_challenge : NULL);
|
|
if (err < 0)
|
|
goto fail;
|
|
lower_transport_mask &= ~(1 << lower_transport);
|
|
if (lower_transport_mask == 0 && err == 1) {
|
|
err = FF_NETERROR(EPROTONOSUPPORT);
|
|
goto fail;
|
|
}
|
|
} while (err);
|
|
|
|
rt->state = RTSP_STATE_IDLE;
|
|
rt->seek_timestamp = 0; /* default is to start stream at position zero */
|
|
return 0;
|
|
fail:
|
|
ff_rtsp_close_streams(s);
|
|
ff_rtsp_close_connections(s);
|
|
if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
|
|
av_strlcpy(s->filename, reply->location, sizeof(s->filename));
|
|
av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
|
|
reply->status_code,
|
|
s->filename);
|
|
goto redirect;
|
|
}
|
|
ff_network_close();
|
|
return err;
|
|
}
|
|
#endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
|
|
|
|
#if CONFIG_RTPDEC
|
|
static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
|
|
uint8_t *buf, int buf_size, int64_t wait_end)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
RTSPStream *rtsp_st;
|
|
fd_set rfds;
|
|
int fd, fd_rtcp, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
|
|
struct timeval tv;
|
|
|
|
for (;;) {
|
|
if (url_interrupt_cb())
|
|
return AVERROR(EINTR);
|
|
if (wait_end && wait_end - av_gettime() < 0)
|
|
return AVERROR(EAGAIN);
|
|
FD_ZERO(&rfds);
|
|
if (rt->rtsp_hd) {
|
|
tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
|
|
FD_SET(tcp_fd, &rfds);
|
|
} else {
|
|
fd_max = 0;
|
|
tcp_fd = -1;
|
|
}
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
if (rtsp_st->rtp_handle) {
|
|
fd = url_get_file_handle(rtsp_st->rtp_handle);
|
|
fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
|
|
if (FFMAX(fd, fd_rtcp) > fd_max)
|
|
fd_max = FFMAX(fd, fd_rtcp);
|
|
FD_SET(fd, &rfds);
|
|
FD_SET(fd_rtcp, &rfds);
|
|
}
|
|
}
|
|
tv.tv_sec = 0;
|
|
tv.tv_usec = SELECT_TIMEOUT_MS * 1000;
|
|
n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
|
|
if (n > 0) {
|
|
timeout_cnt = 0;
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
if (rtsp_st->rtp_handle) {
|
|
fd = url_get_file_handle(rtsp_st->rtp_handle);
|
|
fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
|
|
if (FD_ISSET(fd_rtcp, &rfds) || FD_ISSET(fd, &rfds)) {
|
|
ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
|
|
if (ret > 0) {
|
|
*prtsp_st = rtsp_st;
|
|
return ret;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
#if CONFIG_RTSP_DEMUXER
|
|
if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
|
|
RTSPMessageHeader reply;
|
|
|
|
ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
|
|
if (ret < 0)
|
|
return ret;
|
|
/* XXX: parse message */
|
|
if (rt->state != RTSP_STATE_STREAMING)
|
|
return 0;
|
|
}
|
|
#endif
|
|
} else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
|
|
return FF_NETERROR(ETIMEDOUT);
|
|
} else if (n < 0 && errno != EINTR)
|
|
return AVERROR(errno);
|
|
}
|
|
}
|
|
|
|
int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int ret, len;
|
|
RTSPStream *rtsp_st, *first_queue_st = NULL;
|
|
int64_t wait_end = 0;
|
|
|
|
if (rt->nb_byes == rt->nb_rtsp_streams)
|
|
return AVERROR_EOF;
|
|
|
|
/* get next frames from the same RTP packet */
|
|
if (rt->cur_transport_priv) {
|
|
if (rt->transport == RTSP_TRANSPORT_RDT) {
|
|
ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
|
|
} else
|
|
ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
|
|
if (ret == 0) {
|
|
rt->cur_transport_priv = NULL;
|
|
return 0;
|
|
} else if (ret == 1) {
|
|
return 0;
|
|
} else
|
|
rt->cur_transport_priv = NULL;
|
|
}
|
|
|
|
if (rt->transport == RTSP_TRANSPORT_RTP) {
|
|
int i;
|
|
int64_t first_queue_time = 0;
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
|
|
int64_t queue_time = ff_rtp_queued_packet_time(rtpctx);
|
|
if (queue_time && (queue_time - first_queue_time < 0 ||
|
|
!first_queue_time)) {
|
|
first_queue_time = queue_time;
|
|
first_queue_st = rt->rtsp_streams[i];
|
|
}
|
|
}
|
|
if (first_queue_time)
|
|
wait_end = first_queue_time + s->max_delay;
|
|
}
|
|
|
|
/* read next RTP packet */
|
|
redo:
|
|
if (!rt->recvbuf) {
|
|
rt->recvbuf = av_malloc(RECVBUF_SIZE);
|
|
if (!rt->recvbuf)
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
switch(rt->lower_transport) {
|
|
default:
|
|
#if CONFIG_RTSP_DEMUXER
|
|
case RTSP_LOWER_TRANSPORT_TCP:
|
|
len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
|
|
break;
|
|
#endif
|
|
case RTSP_LOWER_TRANSPORT_UDP:
|
|
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
|
|
len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
|
|
if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
|
|
rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
|
|
break;
|
|
}
|
|
if (len == AVERROR(EAGAIN) && first_queue_st &&
|
|
rt->transport == RTSP_TRANSPORT_RTP) {
|
|
rtsp_st = first_queue_st;
|
|
ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
|
|
goto end;
|
|
}
|
|
if (len < 0)
|
|
return len;
|
|
if (len == 0)
|
|
return AVERROR_EOF;
|
|
if (rt->transport == RTSP_TRANSPORT_RDT) {
|
|
ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
|
|
} else {
|
|
ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
|
|
if (ret < 0) {
|
|
/* Either bad packet, or a RTCP packet. Check if the
|
|
* first_rtcp_ntp_time field was initialized. */
|
|
RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
|
|
if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
|
|
/* first_rtcp_ntp_time has been initialized for this stream,
|
|
* copy the same value to all other uninitialized streams,
|
|
* in order to map their timestamp origin to the same ntp time
|
|
* as this one. */
|
|
int i;
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
|
|
if (rtpctx2 &&
|
|
rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE)
|
|
rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
|
|
}
|
|
}
|
|
if (ret == -RTCP_BYE) {
|
|
rt->nb_byes++;
|
|
|
|
av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
|
|
rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
|
|
|
|
if (rt->nb_byes == rt->nb_rtsp_streams)
|
|
return AVERROR_EOF;
|
|
}
|
|
}
|
|
}
|
|
end:
|
|
if (ret < 0)
|
|
goto redo;
|
|
if (ret == 1)
|
|
/* more packets may follow, so we save the RTP context */
|
|
rt->cur_transport_priv = rtsp_st->transport_priv;
|
|
|
|
return ret;
|
|
}
|
|
#endif /* CONFIG_RTPDEC */
|
|
|
|
#if CONFIG_SDP_DEMUXER
|
|
static int sdp_probe(AVProbeData *p1)
|
|
{
|
|
const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
|
|
|
|
/* we look for a line beginning "c=IN IP" */
|
|
while (p < p_end && *p != '\0') {
|
|
if (p + sizeof("c=IN IP") - 1 < p_end &&
|
|
av_strstart(p, "c=IN IP", NULL))
|
|
return AVPROBE_SCORE_MAX / 2;
|
|
|
|
while (p < p_end - 1 && *p != '\n') p++;
|
|
if (++p >= p_end)
|
|
break;
|
|
if (*p == '\r')
|
|
p++;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
RTSPStream *rtsp_st;
|
|
int size, i, err;
|
|
char *content;
|
|
char url[1024];
|
|
|
|
if (!ff_network_init())
|
|
return AVERROR(EIO);
|
|
|
|
/* read the whole sdp file */
|
|
/* XXX: better loading */
|
|
content = av_malloc(SDP_MAX_SIZE);
|
|
size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
|
|
if (size <= 0) {
|
|
av_free(content);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
content[size] ='\0';
|
|
|
|
ff_sdp_parse(s, content);
|
|
av_free(content);
|
|
|
|
/* open each RTP stream */
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
char namebuf[50];
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
|
|
getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
|
|
namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
|
|
ff_url_join(url, sizeof(url), "rtp", NULL,
|
|
namebuf, rtsp_st->sdp_port,
|
|
"?localport=%d&ttl=%d", rtsp_st->sdp_port,
|
|
rtsp_st->sdp_ttl);
|
|
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
|
|
err = AVERROR_INVALIDDATA;
|
|
goto fail;
|
|
}
|
|
if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
|
|
goto fail;
|
|
}
|
|
return 0;
|
|
fail:
|
|
ff_rtsp_close_streams(s);
|
|
ff_network_close();
|
|
return err;
|
|
}
|
|
|
|
static int sdp_read_close(AVFormatContext *s)
|
|
{
|
|
ff_rtsp_close_streams(s);
|
|
ff_network_close();
|
|
return 0;
|
|
}
|
|
|
|
AVInputFormat sdp_demuxer = {
|
|
"sdp",
|
|
NULL_IF_CONFIG_SMALL("SDP"),
|
|
sizeof(RTSPState),
|
|
sdp_probe,
|
|
sdp_read_header,
|
|
ff_rtsp_fetch_packet,
|
|
sdp_read_close,
|
|
};
|
|
#endif /* CONFIG_SDP_DEMUXER */
|
|
|
|
#if CONFIG_RTP_DEMUXER
|
|
static int rtp_probe(AVProbeData *p)
|
|
{
|
|
if (av_strstart(p->filename, "rtp:", NULL))
|
|
return AVPROBE_SCORE_MAX;
|
|
return 0;
|
|
}
|
|
|
|
static int rtp_read_header(AVFormatContext *s,
|
|
AVFormatParameters *ap)
|
|
{
|
|
uint8_t recvbuf[1500];
|
|
char host[500], sdp[500];
|
|
int ret, port;
|
|
URLContext* in = NULL;
|
|
int payload_type;
|
|
AVCodecContext codec;
|
|
struct sockaddr_storage addr;
|
|
ByteIOContext pb;
|
|
socklen_t addrlen = sizeof(addr);
|
|
|
|
if (!ff_network_init())
|
|
return AVERROR(EIO);
|
|
|
|
ret = url_open(&in, s->filename, URL_RDONLY);
|
|
if (ret)
|
|
goto fail;
|
|
|
|
while (1) {
|
|
ret = url_read(in, recvbuf, sizeof(recvbuf));
|
|
if (ret == AVERROR(EAGAIN))
|
|
continue;
|
|
if (ret < 0)
|
|
goto fail;
|
|
if (ret < 12) {
|
|
av_log(s, AV_LOG_WARNING, "Received too short packet\n");
|
|
continue;
|
|
}
|
|
|
|
if ((recvbuf[0] & 0xc0) != 0x80) {
|
|
av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
|
|
"received\n");
|
|
continue;
|
|
}
|
|
|
|
payload_type = recvbuf[1] & 0x7f;
|
|
break;
|
|
}
|
|
getsockname(url_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
|
|
url_close(in);
|
|
in = NULL;
|
|
|
|
memset(&codec, 0, sizeof(codec));
|
|
if (ff_rtp_get_codec_info(&codec, payload_type)) {
|
|
av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
|
|
"without an SDP file describing it\n",
|
|
payload_type);
|
|
goto fail;
|
|
}
|
|
if (codec.codec_type != AVMEDIA_TYPE_DATA) {
|
|
av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
|
|
"properly you need an SDP file "
|
|
"describing it\n");
|
|
}
|
|
|
|
av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
|
|
NULL, 0, s->filename);
|
|
|
|
snprintf(sdp, sizeof(sdp),
|
|
"v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
|
|
addr.ss_family == AF_INET ? 4 : 6, host,
|
|
codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
|
|
codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
|
|
port, payload_type);
|
|
av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
|
|
|
|
init_put_byte(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
|
|
s->pb = &pb;
|
|
|
|
/* sdp_read_header initializes this again */
|
|
ff_network_close();
|
|
|
|
ret = sdp_read_header(s, ap);
|
|
s->pb = NULL;
|
|
return ret;
|
|
|
|
fail:
|
|
if (in)
|
|
url_close(in);
|
|
ff_network_close();
|
|
return ret;
|
|
}
|
|
|
|
AVInputFormat rtp_demuxer = {
|
|
"rtp",
|
|
NULL_IF_CONFIG_SMALL("RTP input format"),
|
|
sizeof(RTSPState),
|
|
rtp_probe,
|
|
rtp_read_header,
|
|
ff_rtsp_fetch_packet,
|
|
sdp_read_close,
|
|
.flags = AVFMT_NOFILE,
|
|
};
|
|
#endif /* CONFIG_RTP_DEMUXER */
|
|
|