mirror of
https://gitee.com/openharmony/third_party_ffmpeg
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8c1ebdcea2
* qatar/master: shorten: Use separate pointers for the allocated memory for decoded samples. atrac3: Fix crash in tonal component decoding. ws_snd1: Fix wrong samples counts. movenc: Don't set a default sample duration when creating ismv rtp: Factorize the check for distinguishing RTCP packets from RTP golomb: avoid infinite loop on all-zero input (or end of buffer). bethsoftvid: synchronize video timestamps with audio sample rate bethsoftvid: add audio stream only after getting the first audio packet bethsoftvid: Set video packet duration instead of accumulating pts. bethsoftvid: set packet key frame flag for audio and I-frame video packets. bethsoftvid: fix read_packet() return codes. bethsoftvid: pass palette in side data instead of in a separate packet. sdp: Ignore RTCP packets when autodetecting RTP streams proresenc: initialise 'sign' variable mpegaudio: replace memcpy by SIMD code vc1: prevent using last_frame as a reference for I/P first frame. Conflicts: libavcodec/atrac3.c libavcodec/golomb.h libavcodec/shorten.c libavcodec/ws-snd1.c tests/ref/fate/bethsoft-vid Merged-by: Michael Niedermayer <michaelni@gmx.at>
1997 lines
71 KiB
C
1997 lines
71 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/mathematics.h"
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#include "libavutil/parseutils.h"
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#include "libavutil/random_seed.h"
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#include "libavutil/dict.h"
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#include "libavutil/opt.h"
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#include "avformat.h"
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#include "avio_internal.h"
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#include <sys/time.h>
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#if HAVE_POLL_H
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#include <poll.h>
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#endif
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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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#include "url.h"
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#include "rtpenc.h"
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//#define DEBUG
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/* Timeout values for socket poll, in ms,
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* and read_packet(), in seconds */
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#define POLL_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 / POLL_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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#define OFFSET(x) offsetof(RTSPState, x)
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#define DEC AV_OPT_FLAG_DECODING_PARAM
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#define ENC AV_OPT_FLAG_ENCODING_PARAM
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#define RTSP_FLAG_OPTS(name, longname) \
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{ name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
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{ "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
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#define RTSP_MEDIATYPE_OPTS(name, longname) \
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{ name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
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{ "video", "Video", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
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{ "audio", "Audio", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
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{ "data", "Data", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
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const AVOption ff_rtsp_options[] = {
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{ "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {0}, 0, 1, DEC },
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FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
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{ "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
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{ "udp", "UDP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
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{ "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
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{ "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
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{ "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {(1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
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RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
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RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
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{ "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
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{ "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
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{ NULL },
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};
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static const AVOption sdp_options[] = {
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RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
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RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
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{ NULL },
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};
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static const AVOption rtp_options[] = {
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RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
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{ NULL },
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};
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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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av_parse_time(start, 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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av_parse_time(end, 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->alloc) {
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rtsp_st->dynamic_protocol_context = handler->alloc();
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if (!rtsp_st->dynamic_protocol_context)
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rtsp_st->dynamic_handler = NULL;
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}
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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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avpriv_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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avpriv_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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if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
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rtsp_st->dynamic_handler->init(s, st->index,
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rtsp_st->dynamic_protocol_context);
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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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av_dlog(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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rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
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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_dict_set(&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_dict_set(&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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codec_type = AVMEDIA_TYPE_UNKNOWN;
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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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}
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if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
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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 = avformat_new_stream(s, NULL);
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if (!st)
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return;
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st->id = rt->nb_rtsp_streams - 1;
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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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avpriv_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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if (handler && handler->init)
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handler->init(s, st->index,
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rtsp_st->dynamic_protocol_context);
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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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rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
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|
|
|
/* XXX: may need to add full url resolution */
|
|
av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
|
|
NULL, NULL, 0, p);
|
|
if (proto[0] == '\0') {
|
|
/* relative control URL */
|
|
if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
|
|
av_strlcat(rtsp_st->control_url, "/",
|
|
sizeof(rtsp_st->control_url));
|
|
av_strlcat(rtsp_st->control_url, p,
|
|
sizeof(rtsp_st->control_url));
|
|
} else
|
|
av_strlcpy(rtsp_st->control_url, p,
|
|
sizeof(rtsp_st->control_url));
|
|
}
|
|
} else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
|
|
/* NOTE: rtpmap is only supported AFTER the 'm=' tag */
|
|
get_word(buf1, sizeof(buf1), &p);
|
|
payload_type = atoi(buf1);
|
|
st = s->streams[s->nb_streams - 1];
|
|
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
|
|
sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
|
|
} else if (av_strstart(p, "fmtp:", &p) ||
|
|
av_strstart(p, "framesize:", &p)) {
|
|
/* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
|
|
// let dynamic protocol handlers have a stab at the line.
|
|
get_word(buf1, sizeof(buf1), &p);
|
|
payload_type = atoi(buf1);
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
if (rtsp_st->sdp_payload_type == payload_type &&
|
|
rtsp_st->dynamic_handler &&
|
|
rtsp_st->dynamic_handler->parse_sdp_a_line)
|
|
rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
|
|
rtsp_st->dynamic_protocol_context, buf);
|
|
}
|
|
} else if (av_strstart(p, "range:", &p)) {
|
|
int64_t start, end;
|
|
|
|
// this is so that seeking on a streamed file can work.
|
|
rtsp_parse_range_npt(p, &start, &end);
|
|
s->start_time = start;
|
|
/* AV_NOPTS_VALUE means live broadcast (and can't seek) */
|
|
s->duration = (end == AV_NOPTS_VALUE) ?
|
|
AV_NOPTS_VALUE : end - start;
|
|
} else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
|
|
if (atoi(p) == 1)
|
|
rt->transport = RTSP_TRANSPORT_RDT;
|
|
} else if (av_strstart(p, "SampleRate:integer;", &p) &&
|
|
s->nb_streams > 0) {
|
|
st = s->streams[s->nb_streams - 1];
|
|
st->codec->sample_rate = atoi(p);
|
|
} else {
|
|
if (rt->server_type == RTSP_SERVER_WMS)
|
|
ff_wms_parse_sdp_a_line(s, p);
|
|
if (s->nb_streams > 0) {
|
|
if (rt->server_type == RTSP_SERVER_REAL)
|
|
ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
|
|
|
|
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
|
|
if (rtsp_st->dynamic_handler &&
|
|
rtsp_st->dynamic_handler->parse_sdp_a_line)
|
|
rtsp_st->dynamic_handler->parse_sdp_a_line(s,
|
|
s->nb_streams - 1,
|
|
rtsp_st->dynamic_protocol_context, buf);
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
int ff_sdp_parse(AVFormatContext *s, const char *content)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
const char *p;
|
|
int letter;
|
|
/* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
|
|
* contain long SDP lines containing complete ASF Headers (several
|
|
* kB) or arrays of MDPR (RM stream descriptor) headers plus
|
|
* "rulebooks" describing their properties. Therefore, the SDP line
|
|
* buffer is large.
|
|
*
|
|
* The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
|
|
* in rtpdec_xiph.c. */
|
|
char buf[16384], *q;
|
|
SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
|
|
|
|
memset(s1, 0, sizeof(SDPParseState));
|
|
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++;
|
|
}
|
|
rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
|
|
if (!rt->p) return AVERROR(ENOMEM);
|
|
return 0;
|
|
}
|
|
#endif /* CONFIG_RTPDEC */
|
|
|
|
void ff_rtsp_undo_setup(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int i;
|
|
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
RTSPStream *rtsp_st = rt->rtsp_streams[i];
|
|
if (!rtsp_st)
|
|
continue;
|
|
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;
|
|
avio_close_dyn_buf(rtpctx->pb, &ptr);
|
|
av_free(ptr);
|
|
} else {
|
|
avio_close(rtpctx->pb);
|
|
}
|
|
avformat_free_context(rtpctx);
|
|
} else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
|
|
ff_rdt_parse_close(rtsp_st->transport_priv);
|
|
else if (CONFIG_RTPDEC)
|
|
ff_rtp_parse_close(rtsp_st->transport_priv);
|
|
}
|
|
rtsp_st->transport_priv = NULL;
|
|
if (rtsp_st->rtp_handle)
|
|
ffurl_close(rtsp_st->rtp_handle);
|
|
rtsp_st->rtp_handle = NULL;
|
|
}
|
|
}
|
|
|
|
/* close and free RTSP streams */
|
|
void ff_rtsp_close_streams(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int i;
|
|
RTSPStream *rtsp_st;
|
|
|
|
ff_rtsp_undo_setup(s);
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
if (rtsp_st) {
|
|
if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
|
|
rtsp_st->dynamic_handler->free(
|
|
rtsp_st->dynamic_protocol_context);
|
|
av_free(rtsp_st);
|
|
}
|
|
}
|
|
av_free(rt->rtsp_streams);
|
|
if (rt->asf_ctx) {
|
|
avformat_close_input(&rt->asf_ctx);
|
|
}
|
|
av_free(rt->p);
|
|
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 = ff_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) {
|
|
ff_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 (!av_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 (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
|
|
!av_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 (!av_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++;
|
|
}
|
|
}
|
|
|
|
static void handle_rtp_info(RTSPState *rt, const char *url,
|
|
uint32_t seq, uint32_t rtptime)
|
|
{
|
|
int i;
|
|
if (!rtptime || !url[0])
|
|
return;
|
|
if (rt->transport != RTSP_TRANSPORT_RTP)
|
|
return;
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
RTSPStream *rtsp_st = rt->rtsp_streams[i];
|
|
RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
|
|
if (!rtpctx)
|
|
continue;
|
|
if (!strcmp(rtsp_st->control_url, url)) {
|
|
rtpctx->base_timestamp = rtptime;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
|
|
{
|
|
int read = 0;
|
|
char key[20], value[1024], url[1024] = "";
|
|
uint32_t seq = 0, rtptime = 0;
|
|
|
|
for (;;) {
|
|
p += strspn(p, SPACE_CHARS);
|
|
if (!*p)
|
|
break;
|
|
get_word_sep(key, sizeof(key), "=", &p);
|
|
if (*p != '=')
|
|
break;
|
|
p++;
|
|
get_word_sep(value, sizeof(value), ";, ", &p);
|
|
read++;
|
|
if (!strcmp(key, "url"))
|
|
av_strlcpy(url, value, sizeof(url));
|
|
else if (!strcmp(key, "seq"))
|
|
seq = strtoul(value, NULL, 10);
|
|
else if (!strcmp(key, "rtptime"))
|
|
rtptime = strtoul(value, NULL, 10);
|
|
if (*p == ',') {
|
|
handle_rtp_info(rt, url, seq, rtptime);
|
|
url[0] = '\0';
|
|
seq = rtptime = 0;
|
|
read = 0;
|
|
}
|
|
if (*p)
|
|
p++;
|
|
}
|
|
if (read > 0)
|
|
handle_rtp_info(rt, url, seq, rtptime);
|
|
}
|
|
|
|
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
|
|
RTSPState *rt, const char *method)
|
|
{
|
|
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) && rt) {
|
|
p += strspn(p, SPACE_CHARS);
|
|
ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
|
|
} else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
|
|
p += strspn(p, SPACE_CHARS);
|
|
ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
|
|
} else if (av_stristart(p, "Content-Base:", &p) && rt) {
|
|
p += strspn(p, SPACE_CHARS);
|
|
if (method && !strcmp(method, "DESCRIBE"))
|
|
av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
|
|
} else if (av_stristart(p, "RTP-Info:", &p) && rt) {
|
|
p += strspn(p, SPACE_CHARS);
|
|
if (method && !strcmp(method, "PLAY"))
|
|
rtsp_parse_rtp_info(rt, p);
|
|
} else if (av_stristart(p, "Public:", &p) && rt) {
|
|
if (strstr(p, "GET_PARAMETER") &&
|
|
method && !strcmp(method, "OPTIONS"))
|
|
rt->get_parameter_supported = 1;
|
|
} else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
|
|
p += strspn(p, SPACE_CHARS);
|
|
rt->accept_dynamic_rate = atoi(p);
|
|
}
|
|
}
|
|
|
|
/* skip a RTP/TCP interleaved packet */
|
|
void ff_rtsp_skip_packet(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int ret, len, len1;
|
|
uint8_t buf[1024];
|
|
|
|
ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
|
|
if (ret != 3)
|
|
return;
|
|
len = AV_RB16(buf + 1);
|
|
|
|
av_dlog(s, "skipping RTP packet len=%d\n", len);
|
|
|
|
/* skip payload */
|
|
while (len > 0) {
|
|
len1 = len;
|
|
if (len1 > sizeof(buf))
|
|
len1 = sizeof(buf);
|
|
ret = ffurl_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, const char *method)
|
|
{
|
|
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 = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
|
|
av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
|
|
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';
|
|
|
|
av_dlog(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, method);
|
|
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);
|
|
ffurl_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;
|
|
}
|
|
|
|
/**
|
|
* Send a command to the RTSP server without waiting for the reply.
|
|
*
|
|
* @param s RTSP (de)muxer context
|
|
* @param method the method for the request
|
|
* @param url the target url for the request
|
|
* @param headers extra header lines to include in the request
|
|
* @param send_content if non-null, the data to send as request body content
|
|
* @param send_content_length the length of the send_content data, or 0 if
|
|
* send_content is null
|
|
*
|
|
* @return zero if success, nonzero otherwise
|
|
*/
|
|
static 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;
|
|
}
|
|
|
|
av_dlog(s, "Sending:\n%s--\n", buf);
|
|
|
|
ffurl_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;
|
|
}
|
|
ffurl_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, method) ) < 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;
|
|
}
|
|
|
|
int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
|
|
int lower_transport, const char *real_challenge)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int rtx = 0, j, i, err, interleave = 0, port_off;
|
|
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;
|
|
|
|
/* Choose a random starting offset within the first half of the
|
|
* port range, to allow for a number of ports to try even if the offset
|
|
* happens to be at the end of the random range. */
|
|
port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
|
|
/* even random offset */
|
|
port_off -= port_off & 0x01;
|
|
|
|
for (j = rt->rtp_port_min + port_off, 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 */
|
|
while (j <= rt->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 (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
|
|
&s->interrupt_callback, NULL))
|
|
goto rtp_opened;
|
|
}
|
|
av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
|
|
err = AVERROR(EIO);
|
|
goto fail;
|
|
|
|
rtp_opened:
|
|
port = ff_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->transport != RTSP_TRANSPORT_RDT)
|
|
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 (rt->accept_dynamic_rate)
|
|
av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
|
|
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;
|
|
}
|
|
|
|
/* Fail if the server responded with another lower transport mode
|
|
* than what we requested. */
|
|
if (reply->transports[0].lower_transport != lower_transport) {
|
|
av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
|
|
err = AVERROR_INVALIDDATA;
|
|
goto fail;
|
|
}
|
|
|
|
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], options[30] = "";
|
|
|
|
if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
|
|
av_strlcpy(options, "?connect=1", sizeof(options));
|
|
/* 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, "%s", options);
|
|
} else {
|
|
ff_url_join(url, sizeof(url), "rtp", NULL, host,
|
|
reply->transports[0].server_port_min, "%s", options);
|
|
}
|
|
if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
|
|
ff_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)
|
|
ff_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 (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
|
|
&s->interrupt_callback, NULL) < 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:
|
|
ff_rtsp_undo_setup(s);
|
|
return err;
|
|
}
|
|
|
|
void ff_rtsp_close_connections(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
|
|
ffurl_close(rt->rtsp_hd);
|
|
rt->rtsp_hd = rt->rtsp_hd_out = NULL;
|
|
}
|
|
|
|
int ff_rtsp_connect(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
|
|
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 (rt->rtp_port_max < rt->rtp_port_min) {
|
|
av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
|
|
"than min port %d\n", rt->rtp_port_max,
|
|
rt->rtp_port_min);
|
|
return AVERROR(EINVAL);
|
|
}
|
|
|
|
if (!ff_network_init())
|
|
return AVERROR(EIO);
|
|
|
|
rt->control_transport = RTSP_MODE_PLAIN;
|
|
if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
|
|
rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
|
|
rt->control_transport = RTSP_MODE_TUNNEL;
|
|
}
|
|
/* Only pass through valid flags from here */
|
|
rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
|
|
|
|
redirect:
|
|
lower_transport_mask = rt->lower_transport_mask;
|
|
/* 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;
|
|
|
|
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 (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
|
|
&s->interrupt_callback) < 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);
|
|
av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
|
|
|
|
/* complete the connection */
|
|
if (ffurl_connect(rt->rtsp_hd, NULL)) {
|
|
err = AVERROR(EIO);
|
|
goto fail;
|
|
}
|
|
|
|
/* POST requests */
|
|
if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
|
|
&s->interrupt_callback) < 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);
|
|
av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
|
|
av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
|
|
|
|
/* 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 (ffurl_connect(rt->rtsp_hd_out, NULL)) {
|
|
err = AVERROR(EIO);
|
|
goto fail;
|
|
}
|
|
} else {
|
|
/* open the tcp connection */
|
|
ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
|
|
if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
|
|
&s->interrupt_callback, NULL) < 0) {
|
|
err = AVERROR(EIO);
|
|
goto fail;
|
|
}
|
|
rt->rtsp_hd_out = rt->rtsp_hd;
|
|
}
|
|
rt->seq = 0;
|
|
|
|
tcp_fd = ffurl_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.
|
|
* ClientChallenge is a 16-byte ID in hex
|
|
* CompanyID is a 16-byte ID in base64
|
|
*/
|
|
"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 (!av_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 = ff_rtsp_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 = AVERROR(EPROTONOSUPPORT);
|
|
goto fail;
|
|
}
|
|
} while (err);
|
|
|
|
rt->lower_transport_mask = lower_transport_mask;
|
|
av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
|
|
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;
|
|
int n, i, ret, tcp_fd, timeout_cnt = 0;
|
|
int max_p = 0;
|
|
struct pollfd *p = rt->p;
|
|
|
|
for (;;) {
|
|
if (ff_check_interrupt(&s->interrupt_callback))
|
|
return AVERROR_EXIT;
|
|
if (wait_end && wait_end - av_gettime() < 0)
|
|
return AVERROR(EAGAIN);
|
|
max_p = 0;
|
|
if (rt->rtsp_hd) {
|
|
tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
|
|
p[max_p].fd = tcp_fd;
|
|
p[max_p++].events = POLLIN;
|
|
} else {
|
|
tcp_fd = -1;
|
|
}
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
if (rtsp_st->rtp_handle) {
|
|
p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
|
|
p[max_p++].events = POLLIN;
|
|
p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
|
|
p[max_p++].events = POLLIN;
|
|
}
|
|
}
|
|
n = poll(p, max_p, POLL_TIMEOUT_MS);
|
|
if (n > 0) {
|
|
int j = 1 - (tcp_fd == -1);
|
|
timeout_cnt = 0;
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
if (rtsp_st->rtp_handle) {
|
|
if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
|
|
ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
|
|
if (ret > 0) {
|
|
*prtsp_st = rtsp_st;
|
|
return ret;
|
|
}
|
|
}
|
|
j+=2;
|
|
}
|
|
}
|
|
#if CONFIG_RTSP_DEMUXER
|
|
if (tcp_fd != -1 && p[0].revents & POLLIN) {
|
|
RTSPMessageHeader reply;
|
|
|
|
ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
|
|
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 AVERROR(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 = ff_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;
|
|
if (!rtpctx)
|
|
continue;
|
|
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)
|
|
ff_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 = ff_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 = ff_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;
|
|
AVStream *st = NULL;
|
|
if (rtsp_st->stream_index >= 0)
|
|
st = s->streams[rtsp_st->stream_index];
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
|
|
AVStream *st2 = NULL;
|
|
if (rt->rtsp_streams[i]->stream_index >= 0)
|
|
st2 = s->streams[rt->rtsp_streams[i]->stream_index];
|
|
if (rtpctx2 && st && st2 &&
|
|
rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
|
|
rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
|
|
rtpctx2->rtcp_ts_offset = av_rescale_q(
|
|
rtpctx->rtcp_ts_offset, st->time_base,
|
|
st2->time_base);
|
|
}
|
|
}
|
|
}
|
|
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)
|
|
{
|
|
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 = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
|
|
if (size <= 0) {
|
|
av_free(content);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
content[size] ='\0';
|
|
|
|
err = ff_sdp_parse(s, content);
|
|
av_free(content);
|
|
if (err) goto fail;
|
|
|
|
/* 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&connect=%d", rtsp_st->sdp_port,
|
|
rtsp_st->sdp_ttl,
|
|
rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
|
|
if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
|
|
&s->interrupt_callback, NULL) < 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;
|
|
}
|
|
|
|
static const AVClass sdp_demuxer_class = {
|
|
.class_name = "SDP demuxer",
|
|
.item_name = av_default_item_name,
|
|
.option = sdp_options,
|
|
.version = LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
AVInputFormat ff_sdp_demuxer = {
|
|
.name = "sdp",
|
|
.long_name = NULL_IF_CONFIG_SMALL("SDP"),
|
|
.priv_data_size = sizeof(RTSPState),
|
|
.read_probe = sdp_probe,
|
|
.read_header = sdp_read_header,
|
|
.read_packet = ff_rtsp_fetch_packet,
|
|
.read_close = sdp_read_close,
|
|
.priv_class = &sdp_demuxer_class
|
|
};
|
|
#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)
|
|
{
|
|
uint8_t recvbuf[1500];
|
|
char host[500], sdp[500];
|
|
int ret, port;
|
|
URLContext* in = NULL;
|
|
int payload_type;
|
|
AVCodecContext codec;
|
|
struct sockaddr_storage addr;
|
|
AVIOContext pb;
|
|
socklen_t addrlen = sizeof(addr);
|
|
RTSPState *rt = s->priv_data;
|
|
|
|
if (!ff_network_init())
|
|
return AVERROR(EIO);
|
|
|
|
ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
|
|
&s->interrupt_callback, NULL);
|
|
if (ret)
|
|
goto fail;
|
|
|
|
while (1) {
|
|
ret = ffurl_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;
|
|
}
|
|
|
|
if (RTP_PT_IS_RTCP(recvbuf[1]))
|
|
continue;
|
|
|
|
payload_type = recvbuf[1] & 0x7f;
|
|
break;
|
|
}
|
|
getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
|
|
ffurl_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);
|
|
|
|
ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
|
|
s->pb = &pb;
|
|
|
|
/* sdp_read_header initializes this again */
|
|
ff_network_close();
|
|
|
|
rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
|
|
|
|
ret = sdp_read_header(s);
|
|
s->pb = NULL;
|
|
return ret;
|
|
|
|
fail:
|
|
if (in)
|
|
ffurl_close(in);
|
|
ff_network_close();
|
|
return ret;
|
|
}
|
|
|
|
static const AVClass rtp_demuxer_class = {
|
|
.class_name = "RTP demuxer",
|
|
.item_name = av_default_item_name,
|
|
.option = rtp_options,
|
|
.version = LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
AVInputFormat ff_rtp_demuxer = {
|
|
.name = "rtp",
|
|
.long_name = NULL_IF_CONFIG_SMALL("RTP input format"),
|
|
.priv_data_size = sizeof(RTSPState),
|
|
.read_probe = rtp_probe,
|
|
.read_header = rtp_read_header,
|
|
.read_packet = ff_rtsp_fetch_packet,
|
|
.read_close = sdp_read_close,
|
|
.flags = AVFMT_NOFILE,
|
|
.priv_class = &rtp_demuxer_class
|
|
};
|
|
#endif /* CONFIG_RTP_DEMUXER */
|