72415b2adb
is deprecated and will be dropped at the next major bump. Originally committed as revision 22735 to svn://svn.ffmpeg.org/ffmpeg/trunk
2060 lines
68 KiB
C
2060 lines
68 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 "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 "rtsp.h"
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#include "rtpdec.h"
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#include "rdt.h"
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#include "rtpdec_asf.h"
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#include "rtpdec_vorbis.h"
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//#define DEBUG
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//#define DEBUG_RTP_TCP
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#if LIBAVFORMAT_VERSION_INT < (53 << 16)
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int rtsp_default_protocols = (1 << RTSP_LOWER_TRANSPORT_UDP);
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#endif
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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 SPACE_CHARS " \t\r\n"
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/* we use memchr() instead of strchr() here because strchr() will return
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* the terminating '\0' of SPACE_CHARS instead of NULL if c is '\0'. */
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#define redir_isspace(c) memchr(SPACE_CHARS, c, 4)
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static void skip_spaces(const char **pp)
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{
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const char *p;
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p = *pp;
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while (redir_isspace(*p))
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p++;
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*pp = p;
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}
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static void get_word_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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skip_spaces(&p);
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q = buf;
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while (!strchr(sep, *p) && *p != '\0') {
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if ((q - buf) < buf_size - 1)
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*q++ = *p;
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p++;
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}
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if (buf_size > 0)
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*q = '\0';
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*pp = p;
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}
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static void get_word_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 the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
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static int sdp_parse_rtpmap(AVFormatContext *s,
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AVCodecContext *codec, RTSPStream *rtsp_st,
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int payload_type, const char *p)
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{
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char buf[256];
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int i;
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AVCodec *c;
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const char *c_name;
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/* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
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* see if we can handle this kind of payload.
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* 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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for (handler = RTPFirstDynamicPayloadHandler;
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handler; handler = handler->next) {
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if (!strcasecmp(buf, handler->enc_name) &&
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codec->codec_type == handler->codec_type) {
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codec->codec_id = handler->codec_id;
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rtsp_st->dynamic_handler = handler;
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if (handler->open)
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rtsp_st->dynamic_protocol_context = handler->open();
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break;
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}
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}
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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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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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break;
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default:
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break;
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}
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return 0;
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}
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/* return the length and optionally the data */
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static int hex_to_data(uint8_t *data, const char *p)
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{
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int c, len, v;
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len = 0;
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v = 1;
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for (;;) {
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skip_spaces(&p);
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if (*p == '\0')
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break;
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c = toupper((unsigned char) *p++);
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if (c >= '0' && c <= '9')
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c = c - '0';
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else if (c >= 'A' && c <= 'F')
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c = c - 'A' + 10;
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else
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break;
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v = (v << 4) | c;
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if (v & 0x100) {
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if (data)
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data[len] = v;
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len++;
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v = 1;
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}
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}
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return len;
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}
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static void sdp_parse_fmtp_config(AVCodecContext * codec, void *ctx,
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char *attr, char *value)
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{
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switch (codec->codec_id) {
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case CODEC_ID_MPEG4:
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case CODEC_ID_AAC:
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if (!strcmp(attr, "config")) {
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/* decode the hexa encoded parameter */
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int len = hex_to_data(NULL, value);
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if (codec->extradata)
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av_free(codec->extradata);
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codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
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if (!codec->extradata)
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return;
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codec->extradata_size = len;
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hex_to_data(codec->extradata, value);
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}
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break;
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case CODEC_ID_VORBIS:
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ff_vorbis_parse_fmtp_config(codec, ctx, attr, value);
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break;
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default:
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break;
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}
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return;
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}
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typedef struct {
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const char *str;
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uint16_t type;
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uint32_t offset;
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} AttrNameMap;
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/* All known fmtp parameters and the corresponding RTPAttrTypeEnum */
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#define ATTR_NAME_TYPE_INT 0
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#define ATTR_NAME_TYPE_STR 1
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static const AttrNameMap attr_names[]=
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{
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{ "SizeLength", ATTR_NAME_TYPE_INT,
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offsetof(RTPPayloadData, sizelength) },
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{ "IndexLength", ATTR_NAME_TYPE_INT,
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offsetof(RTPPayloadData, indexlength) },
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{ "IndexDeltaLength", ATTR_NAME_TYPE_INT,
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offsetof(RTPPayloadData, indexdeltalength) },
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{ "profile-level-id", ATTR_NAME_TYPE_INT,
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offsetof(RTPPayloadData, profile_level_id) },
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{ "StreamType", ATTR_NAME_TYPE_INT,
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offsetof(RTPPayloadData, streamtype) },
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{ "mode", ATTR_NAME_TYPE_STR,
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offsetof(RTPPayloadData, mode) },
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{ NULL, -1, -1 },
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};
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/* parse the attribute line from the fmtp a line of an sdp 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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skip_spaces(p);
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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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/* parse a SDP line and save stream attributes */
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static void sdp_parse_fmtp(AVStream *st, const char *p)
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{
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char attr[256];
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/* Vorbis setup headers can be up to 12KB and are sent base64
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* encoded, giving a 12KB * (4/3) = 16KB FMTP line. */
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char value[16384];
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int i;
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RTSPStream *rtsp_st = st->priv_data;
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AVCodecContext *codec = st->codec;
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RTPPayloadData *rtp_payload_data = &rtsp_st->rtp_payload_data;
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/* loop on each attribute */
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while (ff_rtsp_next_attr_and_value(&p, attr, sizeof(attr),
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value, sizeof(value))) {
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/* grab the codec extra_data from the config parameter of the fmtp
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* line */
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sdp_parse_fmtp_config(codec, rtsp_st->dynamic_protocol_context,
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attr, value);
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/* Looking for a known attribute */
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for (i = 0; attr_names[i].str; ++i) {
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if (!strcasecmp(attr, attr_names[i].str)) {
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if (attr_names[i].type == ATTR_NAME_TYPE_INT) {
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*(int *)((char *)rtp_payload_data +
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attr_names[i].offset) = atoi(value);
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} else if (attr_names[i].type == ATTR_NAME_TYPE_STR)
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*(char **)((char *)rtp_payload_data +
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attr_names[i].offset) = av_strdup(value);
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}
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}
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}
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}
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/** Parse a string p in the form of Range:npt=xx-xx, and determine the start
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* and end time.
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* Used for seeking in the rtp stream.
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*/
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static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
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{
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char buf[256];
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skip_spaces(&p);
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if (!av_stristart(p, "npt=", &p))
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return;
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*start = AV_NOPTS_VALUE;
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*end = AV_NOPTS_VALUE;
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get_word_sep(buf, sizeof(buf), "-", &p);
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*start = parse_date(buf, 1);
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if (*p == '-') {
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p++;
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get_word_sep(buf, sizeof(buf), "-", &p);
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*end = parse_date(buf, 1);
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}
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// av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
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// av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
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}
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typedef struct SDPParseState {
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/* SDP only */
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struct in_addr default_ip;
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int default_ttl;
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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 in_addr 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") != 0)
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return;
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get_word_sep(buf1, sizeof(buf1), "/", &p);
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if (ff_inet_aton(buf1, &sdp_ip) == 0)
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return;
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ttl = 16;
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if (*p == '/') {
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p++;
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get_word_sep(buf1, sizeof(buf1), "/", &p);
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ttl = atoi(buf1);
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}
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if (s->nb_streams == 0) {
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s1->default_ip = sdp_ip;
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s1->default_ttl = ttl;
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} else {
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st = s->streams[s->nb_streams - 1];
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rtsp_st = st->priv_data;
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rtsp_st->sdp_ip = sdp_ip;
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rtsp_st->sdp_ttl = ttl;
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}
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break;
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case 's':
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av_metadata_set(&s->metadata, "title", p);
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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_set(&s->metadata, "comment", p);
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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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/* 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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}
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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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ff_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->codec, rtsp_st, payload_type, p);
|
|
} else if (av_strstart(p, "fmtp:", &p)) {
|
|
/* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
|
|
get_word(buf1, sizeof(buf1), &p);
|
|
payload_type = atoi(buf1);
|
|
for (i = 0; i < s->nb_streams; i++) {
|
|
st = s->streams[i];
|
|
rtsp_st = st->priv_data;
|
|
if (rtsp_st->sdp_payload_type == payload_type) {
|
|
if (!(rtsp_st->dynamic_handler &&
|
|
rtsp_st->dynamic_handler->parse_sdp_a_line &&
|
|
rtsp_st->dynamic_handler->parse_sdp_a_line(s,
|
|
i, rtsp_st->dynamic_protocol_context, buf)))
|
|
sdp_parse_fmtp(st, p);
|
|
}
|
|
}
|
|
} else if (av_strstart(p, "framesize:", &p)) {
|
|
// let dynamic protocol handlers have a stab at the line.
|
|
get_word(buf1, sizeof(buf1), &p);
|
|
payload_type = atoi(buf1);
|
|
for (i = 0; i < s->nb_streams; i++) {
|
|
st = s->streams[i];
|
|
rtsp_st = st->priv_data;
|
|
if (rtsp_st->sdp_payload_type == payload_type &&
|
|
rtsp_st->dynamic_handler &&
|
|
rtsp_st->dynamic_handler->parse_sdp_a_line)
|
|
rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
|
|
rtsp_st->dynamic_protocol_context, buf);
|
|
}
|
|
} else if (av_strstart(p, "range:", &p)) {
|
|
int64_t start, end;
|
|
|
|
// this is so that seeking on a streamed file can work.
|
|
rtsp_parse_range_npt(p, &start, &end);
|
|
s->start_time = start;
|
|
/* AV_NOPTS_VALUE means live broadcast (and can't seek) */
|
|
s->duration = (end == AV_NOPTS_VALUE) ?
|
|
AV_NOPTS_VALUE : end - start;
|
|
} else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
|
|
if (atoi(p) == 1)
|
|
rt->transport = RTSP_TRANSPORT_RDT;
|
|
} else {
|
|
if (rt->server_type == RTSP_SERVER_WMS)
|
|
ff_wms_parse_sdp_a_line(s, p);
|
|
if (s->nb_streams > 0) {
|
|
if (rt->server_type == RTSP_SERVER_REAL)
|
|
ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
|
|
|
|
rtsp_st = s->streams[s->nb_streams - 1]->priv_data;
|
|
if (rtsp_st->dynamic_handler &&
|
|
rtsp_st->dynamic_handler->parse_sdp_a_line)
|
|
rtsp_st->dynamic_handler->parse_sdp_a_line(s,
|
|
s->nb_streams - 1,
|
|
rtsp_st->dynamic_protocol_context, buf);
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
static int sdp_parse(AVFormatContext *s, const char *content)
|
|
{
|
|
const char *p;
|
|
int letter;
|
|
/* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
|
|
* contain long SDP lines containing complete ASF Headers (several
|
|
* kB) or arrays of MDPR (RM stream descriptor) headers plus
|
|
* "rulebooks" describing their properties. Therefore, the SDP line
|
|
* buffer is large.
|
|
*
|
|
* The Vorbis FMTP line can be up to 16KB - see sdp_parse_fmtp. */
|
|
char buf[16384], *q;
|
|
SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
|
|
|
|
memset(s1, 0, sizeof(SDPParseState));
|
|
p = content;
|
|
for (;;) {
|
|
skip_spaces(&p);
|
|
letter = *p;
|
|
if (letter == '\0')
|
|
break;
|
|
p++;
|
|
if (*p != '=')
|
|
goto next_line;
|
|
p++;
|
|
/* get the content */
|
|
q = buf;
|
|
while (*p != '\n' && *p != '\r' && *p != '\0') {
|
|
if ((q - buf) < sizeof(buf) - 1)
|
|
*q++ = *p;
|
|
p++;
|
|
}
|
|
*q = '\0';
|
|
sdp_parse_line(s, s1, letter, buf);
|
|
next_line:
|
|
while (*p != '\n' && *p != '\0')
|
|
p++;
|
|
if (*p == '\n')
|
|
p++;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/* close and free RTSP streams */
|
|
void ff_rtsp_close_streams(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int i;
|
|
RTSPStream *rtsp_st;
|
|
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
if (rtsp_st) {
|
|
if (rtsp_st->transport_priv) {
|
|
if (s->oformat) {
|
|
AVFormatContext *rtpctx = rtsp_st->transport_priv;
|
|
av_write_trailer(rtpctx);
|
|
if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
|
|
uint8_t *ptr;
|
|
url_close_dyn_buf(rtpctx->pb, &ptr);
|
|
av_free(ptr);
|
|
} else {
|
|
url_fclose(rtpctx->pb);
|
|
}
|
|
av_metadata_free(&rtpctx->streams[0]->metadata);
|
|
av_metadata_free(&rtpctx->metadata);
|
|
av_free(rtpctx->streams[0]);
|
|
av_free(rtpctx);
|
|
} else if (rt->transport == RTSP_TRANSPORT_RDT)
|
|
ff_rdt_parse_close(rtsp_st->transport_priv);
|
|
else
|
|
rtp_parse_close(rtsp_st->transport_priv);
|
|
}
|
|
if (rtsp_st->rtp_handle)
|
|
url_close(rtsp_st->rtp_handle);
|
|
if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
|
|
rtsp_st->dynamic_handler->close(
|
|
rtsp_st->dynamic_protocol_context);
|
|
}
|
|
}
|
|
av_free(rt->rtsp_streams);
|
|
if (rt->asf_ctx) {
|
|
av_close_input_stream (rt->asf_ctx);
|
|
rt->asf_ctx = NULL;
|
|
}
|
|
}
|
|
|
|
static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st,
|
|
URLContext *handle)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
AVFormatContext *rtpctx;
|
|
int ret;
|
|
AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
|
|
|
|
if (!rtp_format)
|
|
return NULL;
|
|
|
|
/* Allocate an AVFormatContext for each output stream */
|
|
rtpctx = avformat_alloc_context();
|
|
if (!rtpctx)
|
|
return NULL;
|
|
|
|
rtpctx->oformat = rtp_format;
|
|
if (!av_new_stream(rtpctx, 0)) {
|
|
av_free(rtpctx);
|
|
return NULL;
|
|
}
|
|
/* Copy the max delay setting; the rtp muxer reads this. */
|
|
rtpctx->max_delay = s->max_delay;
|
|
/* Copy other stream parameters. */
|
|
rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio;
|
|
|
|
/* Set the synchronized start time. */
|
|
rtpctx->start_time_realtime = rt->start_time;
|
|
|
|
/* Remove the local codec, link to the original codec
|
|
* context instead, to give the rtp muxer access to
|
|
* codec parameters. */
|
|
av_free(rtpctx->streams[0]->codec);
|
|
rtpctx->streams[0]->codec = st->codec;
|
|
|
|
if (handle) {
|
|
url_fdopen(&rtpctx->pb, handle);
|
|
} else
|
|
url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
|
|
ret = av_write_header(rtpctx);
|
|
|
|
if (ret) {
|
|
if (handle) {
|
|
url_fclose(rtpctx->pb);
|
|
} else {
|
|
uint8_t *ptr;
|
|
url_close_dyn_buf(rtpctx->pb, &ptr);
|
|
av_free(ptr);
|
|
}
|
|
av_free(rtpctx->streams[0]);
|
|
av_free(rtpctx);
|
|
return NULL;
|
|
}
|
|
|
|
/* Copy the RTP AVStream timebase back to the original AVStream */
|
|
st->time_base = rtpctx->streams[0]->time_base;
|
|
return rtpctx;
|
|
}
|
|
|
|
static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
AVStream *st = NULL;
|
|
|
|
/* open the RTP context */
|
|
if (rtsp_st->stream_index >= 0)
|
|
st = s->streams[rtsp_st->stream_index];
|
|
if (!st)
|
|
s->ctx_flags |= AVFMTCTX_NOHEADER;
|
|
|
|
if (s->oformat) {
|
|
rtsp_st->transport_priv = rtsp_rtp_mux_open(s, st, rtsp_st->rtp_handle);
|
|
/* Ownership of rtp_handle is passed to the rtp mux context */
|
|
rtsp_st->rtp_handle = NULL;
|
|
} else if (rt->transport == RTSP_TRANSPORT_RDT)
|
|
rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
|
|
rtsp_st->dynamic_protocol_context,
|
|
rtsp_st->dynamic_handler);
|
|
else
|
|
rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
|
|
rtsp_st->sdp_payload_type,
|
|
&rtsp_st->rtp_payload_data);
|
|
|
|
if (!rtsp_st->transport_priv) {
|
|
return AVERROR(ENOMEM);
|
|
} else if (rt->transport != RTSP_TRANSPORT_RDT) {
|
|
if (rtsp_st->dynamic_handler) {
|
|
rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
|
|
rtsp_st->dynamic_protocol_context,
|
|
rtsp_st->dynamic_handler);
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
#if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
|
|
static int rtsp_probe(AVProbeData *p)
|
|
{
|
|
if (av_strstart(p->filename, "rtsp:", NULL))
|
|
return AVPROBE_SCORE_MAX;
|
|
return 0;
|
|
}
|
|
|
|
static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
|
|
{
|
|
const char *p;
|
|
int v;
|
|
|
|
p = *pp;
|
|
skip_spaces(&p);
|
|
v = strtol(p, (char **)&p, 10);
|
|
if (*p == '-') {
|
|
p++;
|
|
*min_ptr = v;
|
|
v = strtol(p, (char **)&p, 10);
|
|
*max_ptr = v;
|
|
} else {
|
|
*min_ptr = v;
|
|
*max_ptr = v;
|
|
}
|
|
*pp = p;
|
|
}
|
|
|
|
/* XXX: only one transport specification is parsed */
|
|
static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
|
|
{
|
|
char transport_protocol[16];
|
|
char profile[16];
|
|
char lower_transport[16];
|
|
char parameter[16];
|
|
RTSPTransportField *th;
|
|
char buf[256];
|
|
|
|
reply->nb_transports = 0;
|
|
|
|
for (;;) {
|
|
skip_spaces(&p);
|
|
if (*p == '\0')
|
|
break;
|
|
|
|
th = &reply->transports[reply->nb_transports];
|
|
|
|
get_word_sep(transport_protocol, sizeof(transport_protocol),
|
|
"/", &p);
|
|
if (!strcasecmp (transport_protocol, "rtp")) {
|
|
get_word_sep(profile, sizeof(profile), "/;,", &p);
|
|
lower_transport[0] = '\0';
|
|
/* rtp/avp/<protocol> */
|
|
if (*p == '/') {
|
|
get_word_sep(lower_transport, sizeof(lower_transport),
|
|
";,", &p);
|
|
}
|
|
th->transport = RTSP_TRANSPORT_RTP;
|
|
} else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
|
|
!strcasecmp (transport_protocol, "x-real-rdt")) {
|
|
/* x-pn-tng/<protocol> */
|
|
get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
|
|
profile[0] = '\0';
|
|
th->transport = RTSP_TRANSPORT_RDT;
|
|
}
|
|
if (!strcasecmp(lower_transport, "TCP"))
|
|
th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
|
|
else
|
|
th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
|
|
|
|
if (*p == ';')
|
|
p++;
|
|
/* get each parameter */
|
|
while (*p != '\0' && *p != ',') {
|
|
get_word_sep(parameter, sizeof(parameter), "=;,", &p);
|
|
if (!strcmp(parameter, "port")) {
|
|
if (*p == '=') {
|
|
p++;
|
|
rtsp_parse_range(&th->port_min, &th->port_max, &p);
|
|
}
|
|
} else if (!strcmp(parameter, "client_port")) {
|
|
if (*p == '=') {
|
|
p++;
|
|
rtsp_parse_range(&th->client_port_min,
|
|
&th->client_port_max, &p);
|
|
}
|
|
} else if (!strcmp(parameter, "server_port")) {
|
|
if (*p == '=') {
|
|
p++;
|
|
rtsp_parse_range(&th->server_port_min,
|
|
&th->server_port_max, &p);
|
|
}
|
|
} else if (!strcmp(parameter, "interleaved")) {
|
|
if (*p == '=') {
|
|
p++;
|
|
rtsp_parse_range(&th->interleaved_min,
|
|
&th->interleaved_max, &p);
|
|
}
|
|
} else if (!strcmp(parameter, "multicast")) {
|
|
if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
|
|
th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
|
|
} else if (!strcmp(parameter, "ttl")) {
|
|
if (*p == '=') {
|
|
p++;
|
|
th->ttl = strtol(p, (char **)&p, 10);
|
|
}
|
|
} else if (!strcmp(parameter, "destination")) {
|
|
struct in_addr ipaddr;
|
|
|
|
if (*p == '=') {
|
|
p++;
|
|
get_word_sep(buf, sizeof(buf), ";,", &p);
|
|
if (ff_inet_aton(buf, &ipaddr))
|
|
th->destination = ntohl(ipaddr.s_addr);
|
|
}
|
|
}
|
|
while (*p != ';' && *p != '\0' && *p != ',')
|
|
p++;
|
|
if (*p == ';')
|
|
p++;
|
|
}
|
|
if (*p == ',')
|
|
p++;
|
|
|
|
reply->nb_transports++;
|
|
}
|
|
}
|
|
|
|
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
|
|
HTTPAuthState *auth_state)
|
|
{
|
|
const char *p;
|
|
|
|
/* NOTE: we do case independent match for broken servers */
|
|
p = buf;
|
|
if (av_stristart(p, "Session:", &p)) {
|
|
int t;
|
|
get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
|
|
if (av_stristart(p, ";timeout=", &p) &&
|
|
(t = strtol(p, NULL, 10)) > 0) {
|
|
reply->timeout = t;
|
|
}
|
|
} else if (av_stristart(p, "Content-Length:", &p)) {
|
|
reply->content_length = strtol(p, NULL, 10);
|
|
} else if (av_stristart(p, "Transport:", &p)) {
|
|
rtsp_parse_transport(reply, p);
|
|
} else if (av_stristart(p, "CSeq:", &p)) {
|
|
reply->seq = strtol(p, NULL, 10);
|
|
} else if (av_stristart(p, "Range:", &p)) {
|
|
rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
|
|
} else if (av_stristart(p, "RealChallenge1:", &p)) {
|
|
skip_spaces(&p);
|
|
av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
|
|
} else if (av_stristart(p, "Server:", &p)) {
|
|
skip_spaces(&p);
|
|
av_strlcpy(reply->server, p, sizeof(reply->server));
|
|
} else if (av_stristart(p, "Notice:", &p) ||
|
|
av_stristart(p, "X-Notice:", &p)) {
|
|
reply->notice = strtol(p, NULL, 10);
|
|
} else if (av_stristart(p, "Location:", &p)) {
|
|
skip_spaces(&p);
|
|
av_strlcpy(reply->location, p , sizeof(reply->location));
|
|
} else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) {
|
|
skip_spaces(&p);
|
|
ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p);
|
|
} else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) {
|
|
skip_spaces(&p);
|
|
ff_http_auth_handle_header(auth_state, "Authentication-Info", p);
|
|
}
|
|
}
|
|
|
|
/* skip a RTP/TCP interleaved packet */
|
|
void ff_rtsp_skip_packet(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int ret, len, len1;
|
|
uint8_t buf[1024];
|
|
|
|
ret = url_read_complete(rt->rtsp_hd, buf, 3);
|
|
if (ret != 3)
|
|
return;
|
|
len = AV_RB16(buf + 1);
|
|
|
|
dprintf(s, "skipping RTP packet len=%d\n", len);
|
|
|
|
/* skip payload */
|
|
while (len > 0) {
|
|
len1 = len;
|
|
if (len1 > sizeof(buf))
|
|
len1 = sizeof(buf);
|
|
ret = url_read_complete(rt->rtsp_hd, buf, len1);
|
|
if (ret != len1)
|
|
return;
|
|
len -= len1;
|
|
}
|
|
}
|
|
|
|
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
|
|
unsigned char **content_ptr,
|
|
int return_on_interleaved_data)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
char buf[4096], buf1[1024], *q;
|
|
unsigned char ch;
|
|
const char *p;
|
|
int ret, content_length, line_count = 0;
|
|
unsigned char *content = NULL;
|
|
|
|
memset(reply, 0, sizeof(*reply));
|
|
|
|
/* parse reply (XXX: use buffers) */
|
|
rt->last_reply[0] = '\0';
|
|
for (;;) {
|
|
q = buf;
|
|
for (;;) {
|
|
ret = url_read_complete(rt->rtsp_hd, &ch, 1);
|
|
#ifdef DEBUG_RTP_TCP
|
|
dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
|
|
#endif
|
|
if (ret != 1)
|
|
return -1;
|
|
if (ch == '\n')
|
|
break;
|
|
if (ch == '$') {
|
|
/* XXX: only parse it if first char on line ? */
|
|
if (return_on_interleaved_data) {
|
|
return 1;
|
|
} else
|
|
ff_rtsp_skip_packet(s);
|
|
} else if (ch != '\r') {
|
|
if ((q - buf) < sizeof(buf) - 1)
|
|
*q++ = ch;
|
|
}
|
|
}
|
|
*q = '\0';
|
|
|
|
dprintf(s, "line='%s'\n", buf);
|
|
|
|
/* test if last line */
|
|
if (buf[0] == '\0')
|
|
break;
|
|
p = buf;
|
|
if (line_count == 0) {
|
|
/* get reply code */
|
|
get_word(buf1, sizeof(buf1), &p);
|
|
get_word(buf1, sizeof(buf1), &p);
|
|
reply->status_code = atoi(buf1);
|
|
} else {
|
|
ff_rtsp_parse_line(reply, p, &rt->auth_state);
|
|
av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
|
|
av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
|
|
}
|
|
line_count++;
|
|
}
|
|
|
|
if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
|
|
av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
|
|
|
|
content_length = reply->content_length;
|
|
if (content_length > 0) {
|
|
/* leave some room for a trailing '\0' (useful for simple parsing) */
|
|
content = av_malloc(content_length + 1);
|
|
(void)url_read_complete(rt->rtsp_hd, content, content_length);
|
|
content[content_length] = '\0';
|
|
}
|
|
if (content_ptr)
|
|
*content_ptr = content;
|
|
else
|
|
av_free(content);
|
|
|
|
if (rt->seq != reply->seq) {
|
|
av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
|
|
rt->seq, reply->seq);
|
|
}
|
|
|
|
/* EOS */
|
|
if (reply->notice == 2101 /* End-of-Stream Reached */ ||
|
|
reply->notice == 2104 /* Start-of-Stream Reached */ ||
|
|
reply->notice == 2306 /* Continuous Feed Terminated */) {
|
|
rt->state = RTSP_STATE_IDLE;
|
|
} else if (reply->notice >= 4400 && reply->notice < 5500) {
|
|
return AVERROR(EIO); /* data or server error */
|
|
} else if (reply->notice == 2401 /* Ticket Expired */ ||
|
|
(reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
|
|
return AVERROR(EPERM);
|
|
|
|
return 0;
|
|
}
|
|
|
|
void ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
|
|
const char *method, const char *url,
|
|
const char *headers,
|
|
const unsigned char *send_content,
|
|
int send_content_length)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
char buf[4096];
|
|
|
|
rt->seq++;
|
|
snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
|
|
if (headers)
|
|
av_strlcat(buf, headers, sizeof(buf));
|
|
av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
|
|
if (rt->session_id[0] != '\0' && (!headers ||
|
|
!strstr(headers, "\nIf-Match:"))) {
|
|
av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
|
|
}
|
|
if (rt->auth[0]) {
|
|
char *str = ff_http_auth_create_response(&rt->auth_state,
|
|
rt->auth, url, method);
|
|
if (str)
|
|
av_strlcat(buf, str, sizeof(buf));
|
|
av_free(str);
|
|
}
|
|
if (send_content_length > 0 && send_content)
|
|
av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
|
|
av_strlcat(buf, "\r\n", sizeof(buf));
|
|
|
|
dprintf(s, "Sending:\n%s--\n", buf);
|
|
|
|
url_write(rt->rtsp_hd, buf, strlen(buf));
|
|
if (send_content_length > 0 && send_content)
|
|
url_write(rt->rtsp_hd, send_content, send_content_length);
|
|
rt->last_cmd_time = av_gettime();
|
|
}
|
|
|
|
void ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
|
|
const char *url, const char *headers)
|
|
{
|
|
ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
|
|
}
|
|
|
|
void ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
|
|
const char *headers, RTSPMessageHeader *reply,
|
|
unsigned char **content_ptr)
|
|
{
|
|
ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
|
|
content_ptr, NULL, 0);
|
|
}
|
|
|
|
void ff_rtsp_send_cmd_with_content(AVFormatContext *s,
|
|
const char *method, const char *url,
|
|
const char *header,
|
|
RTSPMessageHeader *reply,
|
|
unsigned char **content_ptr,
|
|
const unsigned char *send_content,
|
|
int send_content_length)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
HTTPAuthType cur_auth_type;
|
|
|
|
retry:
|
|
cur_auth_type = rt->auth_state.auth_type;
|
|
ff_rtsp_send_cmd_with_content_async(s, method, url, header,
|
|
send_content, send_content_length);
|
|
|
|
ff_rtsp_read_reply(s, reply, content_ptr, 0);
|
|
|
|
if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
|
|
rt->auth_state.auth_type != HTTP_AUTH_NONE)
|
|
goto retry;
|
|
}
|
|
|
|
/**
|
|
* @return 0 on success, <0 on error, 1 if protocol is unavailable.
|
|
*/
|
|
static int make_setup_request(AVFormatContext *s, const char *host, int port,
|
|
int lower_transport, const char *real_challenge)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int rtx, j, i, err, interleave = 0;
|
|
RTSPStream *rtsp_st;
|
|
RTSPMessageHeader reply1, *reply = &reply1;
|
|
char cmd[2048];
|
|
const char *trans_pref;
|
|
|
|
if (rt->transport == RTSP_TRANSPORT_RDT)
|
|
trans_pref = "x-pn-tng";
|
|
else
|
|
trans_pref = "RTP/AVP";
|
|
|
|
/* default timeout: 1 minute */
|
|
rt->timeout = 60;
|
|
|
|
/* for each stream, make the setup request */
|
|
/* XXX: we assume the same server is used for the control of each
|
|
* RTSP stream */
|
|
|
|
for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
|
|
char transport[2048];
|
|
|
|
/**
|
|
* WMS serves all UDP data over a single connection, the RTX, which
|
|
* isn't necessarily the first in the SDP but has to be the first
|
|
* to be set up, else the second/third SETUP will fail with a 461.
|
|
*/
|
|
if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
|
|
rt->server_type == RTSP_SERVER_WMS) {
|
|
if (i == 0) {
|
|
/* rtx first */
|
|
for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
|
|
int len = strlen(rt->rtsp_streams[rtx]->control_url);
|
|
if (len >= 4 &&
|
|
!strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
|
|
"/rtx"))
|
|
break;
|
|
}
|
|
if (rtx == rt->nb_rtsp_streams)
|
|
return -1; /* no RTX found */
|
|
rtsp_st = rt->rtsp_streams[rtx];
|
|
} else
|
|
rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
|
|
} else
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
|
|
/* RTP/UDP */
|
|
if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
|
|
char buf[256];
|
|
|
|
if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
|
|
port = reply->transports[0].client_port_min;
|
|
goto have_port;
|
|
}
|
|
|
|
/* first try in specified port range */
|
|
if (RTSP_RTP_PORT_MIN != 0) {
|
|
while (j <= RTSP_RTP_PORT_MAX) {
|
|
ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
|
|
"?localport=%d", j);
|
|
/* we will use two ports per rtp stream (rtp and rtcp) */
|
|
j += 2;
|
|
if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
|
|
goto rtp_opened;
|
|
}
|
|
}
|
|
|
|
#if 0
|
|
/* then try on any port */
|
|
if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
|
|
err = AVERROR_INVALIDDATA;
|
|
goto fail;
|
|
}
|
|
#endif
|
|
|
|
rtp_opened:
|
|
port = rtp_get_local_port(rtsp_st->rtp_handle);
|
|
have_port:
|
|
snprintf(transport, sizeof(transport) - 1,
|
|
"%s/UDP;", trans_pref);
|
|
if (rt->server_type != RTSP_SERVER_REAL)
|
|
av_strlcat(transport, "unicast;", sizeof(transport));
|
|
av_strlcatf(transport, sizeof(transport),
|
|
"client_port=%d", port);
|
|
if (rt->transport == RTSP_TRANSPORT_RTP &&
|
|
!(rt->server_type == RTSP_SERVER_WMS && i > 0))
|
|
av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
|
|
}
|
|
|
|
/* RTP/TCP */
|
|
else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
|
|
/** For WMS streams, the application streams are only used for
|
|
* UDP. When trying to set it up for TCP streams, the server
|
|
* will return an error. Therefore, we skip those streams. */
|
|
if (rt->server_type == RTSP_SERVER_WMS &&
|
|
s->streams[rtsp_st->stream_index]->codec->codec_type ==
|
|
AVMEDIA_TYPE_DATA)
|
|
continue;
|
|
snprintf(transport, sizeof(transport) - 1,
|
|
"%s/TCP;", trans_pref);
|
|
if (rt->server_type == RTSP_SERVER_WMS)
|
|
av_strlcat(transport, "unicast;", sizeof(transport));
|
|
av_strlcatf(transport, sizeof(transport),
|
|
"interleaved=%d-%d",
|
|
interleave, interleave + 1);
|
|
interleave += 2;
|
|
}
|
|
|
|
else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
|
|
snprintf(transport, sizeof(transport) - 1,
|
|
"%s/UDP;multicast", trans_pref);
|
|
}
|
|
if (s->oformat) {
|
|
av_strlcat(transport, ";mode=receive", sizeof(transport));
|
|
} else if (rt->server_type == RTSP_SERVER_REAL ||
|
|
rt->server_type == RTSP_SERVER_WMS)
|
|
av_strlcat(transport, ";mode=play", sizeof(transport));
|
|
snprintf(cmd, sizeof(cmd),
|
|
"Transport: %s\r\n",
|
|
transport);
|
|
if (i == 0 && rt->server_type == RTSP_SERVER_REAL) {
|
|
char real_res[41], real_csum[9];
|
|
ff_rdt_calc_response_and_checksum(real_res, real_csum,
|
|
real_challenge);
|
|
av_strlcatf(cmd, sizeof(cmd),
|
|
"If-Match: %s\r\n"
|
|
"RealChallenge2: %s, sd=%s\r\n",
|
|
rt->session_id, real_res, real_csum);
|
|
}
|
|
ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
|
|
if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
|
|
err = 1;
|
|
goto fail;
|
|
} else if (reply->status_code != RTSP_STATUS_OK ||
|
|
reply->nb_transports != 1) {
|
|
err = AVERROR_INVALIDDATA;
|
|
goto fail;
|
|
}
|
|
|
|
/* XXX: same protocol for all streams is required */
|
|
if (i > 0) {
|
|
if (reply->transports[0].lower_transport != rt->lower_transport ||
|
|
reply->transports[0].transport != rt->transport) {
|
|
err = AVERROR_INVALIDDATA;
|
|
goto fail;
|
|
}
|
|
} else {
|
|
rt->lower_transport = reply->transports[0].lower_transport;
|
|
rt->transport = reply->transports[0].transport;
|
|
}
|
|
|
|
/* close RTP connection if not choosen */
|
|
if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&
|
|
(lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {
|
|
url_close(rtsp_st->rtp_handle);
|
|
rtsp_st->rtp_handle = NULL;
|
|
}
|
|
|
|
switch(reply->transports[0].lower_transport) {
|
|
case RTSP_LOWER_TRANSPORT_TCP:
|
|
rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
|
|
rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
|
|
break;
|
|
|
|
case RTSP_LOWER_TRANSPORT_UDP: {
|
|
char url[1024];
|
|
|
|
/* XXX: also use address if specified */
|
|
ff_url_join(url, sizeof(url), "rtp", NULL, host,
|
|
reply->transports[0].server_port_min, NULL);
|
|
if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
|
|
rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
|
|
err = AVERROR_INVALIDDATA;
|
|
goto fail;
|
|
}
|
|
/* Try to initialize the connection state in a
|
|
* potential NAT router by sending dummy packets.
|
|
* RTP/RTCP dummy packets are used for RDT, too.
|
|
*/
|
|
if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat)
|
|
rtp_send_punch_packets(rtsp_st->rtp_handle);
|
|
break;
|
|
}
|
|
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
|
|
char url[1024];
|
|
struct in_addr in;
|
|
int port, ttl;
|
|
|
|
if (reply->transports[0].destination) {
|
|
in.s_addr = htonl(reply->transports[0].destination);
|
|
port = reply->transports[0].port_min;
|
|
ttl = reply->transports[0].ttl;
|
|
} else {
|
|
in = rtsp_st->sdp_ip;
|
|
port = rtsp_st->sdp_port;
|
|
ttl = rtsp_st->sdp_ttl;
|
|
}
|
|
ff_url_join(url, sizeof(url), "rtp", NULL, inet_ntoa(in),
|
|
port, "?ttl=%d", ttl);
|
|
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
|
|
err = AVERROR_INVALIDDATA;
|
|
goto fail;
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
|
|
goto fail;
|
|
}
|
|
|
|
if (reply->timeout > 0)
|
|
rt->timeout = reply->timeout;
|
|
|
|
if (rt->server_type == RTSP_SERVER_REAL)
|
|
rt->need_subscription = 1;
|
|
|
|
return 0;
|
|
|
|
fail:
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
if (rt->rtsp_streams[i]->rtp_handle) {
|
|
url_close(rt->rtsp_streams[i]->rtp_handle);
|
|
rt->rtsp_streams[i]->rtp_handle = NULL;
|
|
}
|
|
}
|
|
return err;
|
|
}
|
|
|
|
static int rtsp_read_play(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
RTSPMessageHeader reply1, *reply = &reply1;
|
|
char cmd[1024];
|
|
|
|
av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
|
|
|
|
if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
|
|
if (rt->state == RTSP_STATE_PAUSED) {
|
|
cmd[0] = 0;
|
|
} else {
|
|
snprintf(cmd, sizeof(cmd),
|
|
"Range: npt=%0.3f-\r\n",
|
|
(double)rt->seek_timestamp / AV_TIME_BASE);
|
|
}
|
|
ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL);
|
|
if (reply->status_code != RTSP_STATUS_OK) {
|
|
return -1;
|
|
}
|
|
}
|
|
rt->state = RTSP_STATE_STREAMING;
|
|
return 0;
|
|
}
|
|
|
|
static int rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
char cmd[1024];
|
|
unsigned char *content = NULL;
|
|
int ret;
|
|
|
|
/* describe the stream */
|
|
snprintf(cmd, sizeof(cmd),
|
|
"Accept: application/sdp\r\n");
|
|
if (rt->server_type == RTSP_SERVER_REAL) {
|
|
/**
|
|
* The Require: attribute is needed for proper streaming from
|
|
* Realmedia servers.
|
|
*/
|
|
av_strlcat(cmd,
|
|
"Require: com.real.retain-entity-for-setup\r\n",
|
|
sizeof(cmd));
|
|
}
|
|
ff_rtsp_send_cmd(s, "DESCRIBE", rt->control_uri, cmd, reply, &content);
|
|
if (!content)
|
|
return AVERROR_INVALIDDATA;
|
|
if (reply->status_code != RTSP_STATUS_OK) {
|
|
av_freep(&content);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
/* now we got the SDP description, we parse it */
|
|
ret = sdp_parse(s, (const char *)content);
|
|
av_freep(&content);
|
|
if (ret < 0)
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
RTSPMessageHeader reply1, *reply = &reply1;
|
|
int i;
|
|
char *sdp;
|
|
AVFormatContext sdp_ctx, *ctx_array[1];
|
|
|
|
rt->start_time = av_gettime();
|
|
|
|
/* Announce the stream */
|
|
sdp = av_mallocz(8192);
|
|
if (sdp == NULL)
|
|
return AVERROR(ENOMEM);
|
|
/* We create the SDP based on the RTSP AVFormatContext where we
|
|
* aren't allowed to change the filename field. (We create the SDP
|
|
* based on the RTSP context since the contexts for the RTP streams
|
|
* don't exist yet.) In order to specify a custom URL with the actual
|
|
* peer IP instead of the originally specified hostname, we create
|
|
* a temporary copy of the AVFormatContext, where the custom URL is set.
|
|
*
|
|
* FIXME: Create the SDP without copying the AVFormatContext.
|
|
* This either requires setting up the RTP stream AVFormatContexts
|
|
* already here (complicating things immensely) or getting a more
|
|
* flexible SDP creation interface.
|
|
*/
|
|
sdp_ctx = *s;
|
|
ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
|
|
"rtsp", NULL, addr, -1, NULL);
|
|
ctx_array[0] = &sdp_ctx;
|
|
if (avf_sdp_create(ctx_array, 1, sdp, 8192)) {
|
|
av_free(sdp);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
av_log(s, AV_LOG_INFO, "SDP:\n%s\n", sdp);
|
|
ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
|
|
"Content-Type: application/sdp\r\n",
|
|
reply, NULL, sdp, strlen(sdp));
|
|
av_free(sdp);
|
|
if (reply->status_code != RTSP_STATUS_OK)
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
/* Set up the RTSPStreams for each AVStream */
|
|
for (i = 0; i < s->nb_streams; i++) {
|
|
RTSPStream *rtsp_st;
|
|
AVStream *st = s->streams[i];
|
|
|
|
rtsp_st = av_mallocz(sizeof(RTSPStream));
|
|
if (!rtsp_st)
|
|
return AVERROR(ENOMEM);
|
|
dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
|
|
|
|
st->priv_data = rtsp_st;
|
|
rtsp_st->stream_index = i;
|
|
|
|
av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
|
|
/* Note, this must match the relative uri set in the sdp content */
|
|
av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
|
|
"/streamid=%d", i);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int ff_rtsp_connect(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
|
|
char *option_list, *option, *filename;
|
|
URLContext *rtsp_hd;
|
|
int port, err, tcp_fd;
|
|
RTSPMessageHeader reply1, *reply = &reply1;
|
|
int lower_transport_mask = 0;
|
|
char real_challenge[64];
|
|
struct sockaddr_storage peer;
|
|
socklen_t peer_len = sizeof(peer);
|
|
|
|
if (!ff_network_init())
|
|
return AVERROR(EIO);
|
|
redirect:
|
|
/* extract hostname and port */
|
|
ff_url_split(NULL, 0, auth, sizeof(auth),
|
|
host, sizeof(host), &port, path, sizeof(path), s->filename);
|
|
if (*auth) {
|
|
av_strlcpy(rt->auth, auth, sizeof(rt->auth));
|
|
}
|
|
if (port < 0)
|
|
port = RTSP_DEFAULT_PORT;
|
|
|
|
/* search for options */
|
|
option_list = strrchr(path, '?');
|
|
if (option_list) {
|
|
/* Strip out the RTSP specific options, write out the rest of
|
|
* the options back into the same string. */
|
|
filename = option_list;
|
|
while (option_list) {
|
|
/* move the option pointer */
|
|
option = ++option_list;
|
|
option_list = strchr(option_list, '&');
|
|
if (option_list)
|
|
*option_list = 0;
|
|
|
|
/* handle the options */
|
|
if (!strcmp(option, "udp")) {
|
|
lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
|
|
} else if (!strcmp(option, "multicast")) {
|
|
lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
|
|
} else if (!strcmp(option, "tcp")) {
|
|
lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
|
|
} else {
|
|
/* Write options back into the buffer, using memmove instead
|
|
* of strcpy since the strings may overlap. */
|
|
int len = strlen(option);
|
|
memmove(++filename, option, len);
|
|
filename += len;
|
|
if (option_list) *filename = '&';
|
|
}
|
|
}
|
|
*filename = 0;
|
|
}
|
|
|
|
if (!lower_transport_mask)
|
|
lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
|
|
|
|
if (s->oformat) {
|
|
/* Only UDP or TCP - UDP multicast isn't supported. */
|
|
lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
|
|
(1 << RTSP_LOWER_TRANSPORT_TCP);
|
|
if (!lower_transport_mask) {
|
|
av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
|
|
"only UDP and TCP are supported for output.\n");
|
|
err = AVERROR(EINVAL);
|
|
goto fail;
|
|
}
|
|
}
|
|
|
|
/* Construct the URI used in request; this is similar to s->filename,
|
|
* but with authentication credentials removed and RTSP specific options
|
|
* stripped out. */
|
|
ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
|
|
host, port, "%s", path);
|
|
|
|
/* open the tcp connexion */
|
|
ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
|
|
if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0) {
|
|
err = AVERROR(EIO);
|
|
goto fail;
|
|
}
|
|
rt->rtsp_hd = rtsp_hd;
|
|
rt->seq = 0;
|
|
|
|
tcp_fd = url_get_file_handle(rtsp_hd);
|
|
if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
|
|
getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
|
|
NULL, 0, NI_NUMERICHOST);
|
|
}
|
|
|
|
/* request options supported by the server; this also detects server
|
|
* type */
|
|
for (rt->server_type = RTSP_SERVER_RTP;;) {
|
|
cmd[0] = 0;
|
|
if (rt->server_type == RTSP_SERVER_REAL)
|
|
av_strlcat(cmd,
|
|
/**
|
|
* The following entries are required for proper
|
|
* streaming from a Realmedia server. They are
|
|
* interdependent in some way although we currently
|
|
* don't quite understand how. Values were copied
|
|
* from mplayer SVN r23589.
|
|
* @param CompanyID is a 16-byte ID in base64
|
|
* @param ClientChallenge is a 16-byte ID in hex
|
|
*/
|
|
"ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
|
|
"PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
|
|
"CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
|
|
"GUID: 00000000-0000-0000-0000-000000000000\r\n",
|
|
sizeof(cmd));
|
|
ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
|
|
if (reply->status_code != RTSP_STATUS_OK) {
|
|
err = AVERROR_INVALIDDATA;
|
|
goto fail;
|
|
}
|
|
|
|
/* detect server type if not standard-compliant RTP */
|
|
if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
|
|
rt->server_type = RTSP_SERVER_REAL;
|
|
continue;
|
|
} else if (!strncasecmp(reply->server, "WMServer/", 9)) {
|
|
rt->server_type = RTSP_SERVER_WMS;
|
|
} else if (rt->server_type == RTSP_SERVER_REAL)
|
|
strcpy(real_challenge, reply->real_challenge);
|
|
break;
|
|
}
|
|
|
|
if (s->iformat)
|
|
err = rtsp_setup_input_streams(s, reply);
|
|
else
|
|
err = rtsp_setup_output_streams(s, host);
|
|
if (err)
|
|
goto fail;
|
|
|
|
do {
|
|
int lower_transport = ff_log2_tab[lower_transport_mask &
|
|
~(lower_transport_mask - 1)];
|
|
|
|
err = make_setup_request(s, host, port, lower_transport,
|
|
rt->server_type == RTSP_SERVER_REAL ?
|
|
real_challenge : NULL);
|
|
if (err < 0)
|
|
goto fail;
|
|
lower_transport_mask &= ~(1 << lower_transport);
|
|
if (lower_transport_mask == 0 && err == 1) {
|
|
err = AVERROR(FF_NETERROR(EPROTONOSUPPORT));
|
|
goto fail;
|
|
}
|
|
} while (err);
|
|
|
|
rt->state = RTSP_STATE_IDLE;
|
|
rt->seek_timestamp = 0; /* default is to start stream at position zero */
|
|
return 0;
|
|
fail:
|
|
ff_rtsp_close_streams(s);
|
|
url_close(rt->rtsp_hd);
|
|
if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
|
|
av_strlcpy(s->filename, reply->location, sizeof(s->filename));
|
|
av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
|
|
reply->status_code,
|
|
s->filename);
|
|
goto redirect;
|
|
}
|
|
ff_network_close();
|
|
return err;
|
|
}
|
|
#endif
|
|
|
|
#if CONFIG_RTSP_DEMUXER
|
|
static int rtsp_read_header(AVFormatContext *s,
|
|
AVFormatParameters *ap)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int ret;
|
|
|
|
ret = ff_rtsp_connect(s);
|
|
if (ret)
|
|
return ret;
|
|
|
|
if (ap->initial_pause) {
|
|
/* do not start immediately */
|
|
} else {
|
|
if (rtsp_read_play(s) < 0) {
|
|
ff_rtsp_close_streams(s);
|
|
url_close(rt->rtsp_hd);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
|
|
uint8_t *buf, int buf_size)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
RTSPStream *rtsp_st;
|
|
fd_set rfds;
|
|
int fd, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
|
|
struct timeval tv;
|
|
|
|
for (;;) {
|
|
if (url_interrupt_cb())
|
|
return AVERROR(EINTR);
|
|
FD_ZERO(&rfds);
|
|
if (rt->rtsp_hd) {
|
|
tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
|
|
FD_SET(tcp_fd, &rfds);
|
|
} else {
|
|
fd_max = 0;
|
|
tcp_fd = -1;
|
|
}
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
if (rtsp_st->rtp_handle) {
|
|
/* currently, we cannot probe RTCP handle because of
|
|
* blocking restrictions */
|
|
fd = url_get_file_handle(rtsp_st->rtp_handle);
|
|
if (fd > fd_max)
|
|
fd_max = fd;
|
|
FD_SET(fd, &rfds);
|
|
}
|
|
}
|
|
tv.tv_sec = 0;
|
|
tv.tv_usec = SELECT_TIMEOUT_MS * 1000;
|
|
n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
|
|
if (n > 0) {
|
|
timeout_cnt = 0;
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
if (rtsp_st->rtp_handle) {
|
|
fd = url_get_file_handle(rtsp_st->rtp_handle);
|
|
if (FD_ISSET(fd, &rfds)) {
|
|
ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
|
|
if (ret > 0) {
|
|
*prtsp_st = rtsp_st;
|
|
return ret;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
#if CONFIG_RTSP_DEMUXER
|
|
if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
|
|
RTSPMessageHeader reply;
|
|
|
|
ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
|
|
if (ret < 0)
|
|
return ret;
|
|
/* XXX: parse message */
|
|
if (rt->state != RTSP_STATE_STREAMING)
|
|
return 0;
|
|
}
|
|
#endif
|
|
} else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
|
|
return AVERROR(ETIME);
|
|
} else if (n < 0 && errno != EINTR)
|
|
return AVERROR(errno);
|
|
}
|
|
}
|
|
|
|
static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
|
|
uint8_t *buf, int buf_size)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int id, len, i, ret;
|
|
RTSPStream *rtsp_st;
|
|
|
|
#ifdef DEBUG_RTP_TCP
|
|
dprintf(s, "tcp_read_packet:\n");
|
|
#endif
|
|
redo:
|
|
for (;;) {
|
|
RTSPMessageHeader reply;
|
|
|
|
ret = ff_rtsp_read_reply(s, &reply, NULL, 1);
|
|
if (ret == -1)
|
|
return -1;
|
|
if (ret == 1) /* received '$' */
|
|
break;
|
|
/* XXX: parse message */
|
|
if (rt->state != RTSP_STATE_STREAMING)
|
|
return 0;
|
|
}
|
|
ret = url_read_complete(rt->rtsp_hd, buf, 3);
|
|
if (ret != 3)
|
|
return -1;
|
|
id = buf[0];
|
|
len = AV_RB16(buf + 1);
|
|
#ifdef DEBUG_RTP_TCP
|
|
dprintf(s, "id=%d len=%d\n", id, len);
|
|
#endif
|
|
if (len > buf_size || len < 12)
|
|
goto redo;
|
|
/* get the data */
|
|
ret = url_read_complete(rt->rtsp_hd, buf, len);
|
|
if (ret != len)
|
|
return -1;
|
|
if (rt->transport == RTSP_TRANSPORT_RDT &&
|
|
ff_rdt_parse_header(buf, len, &id, NULL, NULL, NULL, NULL) < 0)
|
|
return -1;
|
|
|
|
/* find the matching stream */
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
if (id >= rtsp_st->interleaved_min &&
|
|
id <= rtsp_st->interleaved_max)
|
|
goto found;
|
|
}
|
|
goto redo;
|
|
found:
|
|
*prtsp_st = rtsp_st;
|
|
return len;
|
|
}
|
|
|
|
static int rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int ret, len;
|
|
uint8_t buf[10 * RTP_MAX_PACKET_LENGTH];
|
|
RTSPStream *rtsp_st;
|
|
|
|
/* get next frames from the same RTP packet */
|
|
if (rt->cur_transport_priv) {
|
|
if (rt->transport == RTSP_TRANSPORT_RDT) {
|
|
ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
|
|
} else
|
|
ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
|
|
if (ret == 0) {
|
|
rt->cur_transport_priv = NULL;
|
|
return 0;
|
|
} else if (ret == 1) {
|
|
return 0;
|
|
} else
|
|
rt->cur_transport_priv = NULL;
|
|
}
|
|
|
|
/* read next RTP packet */
|
|
redo:
|
|
switch(rt->lower_transport) {
|
|
default:
|
|
#if CONFIG_RTSP_DEMUXER
|
|
case RTSP_LOWER_TRANSPORT_TCP:
|
|
len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
|
|
break;
|
|
#endif
|
|
case RTSP_LOWER_TRANSPORT_UDP:
|
|
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
|
|
len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
|
|
if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
|
|
rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
|
|
break;
|
|
}
|
|
if (len < 0)
|
|
return len;
|
|
if (len == 0)
|
|
return AVERROR_EOF;
|
|
if (rt->transport == RTSP_TRANSPORT_RDT) {
|
|
ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
|
|
} else
|
|
ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
|
|
if (ret < 0)
|
|
goto redo;
|
|
if (ret == 1)
|
|
/* more packets may follow, so we save the RTP context */
|
|
rt->cur_transport_priv = rtsp_st->transport_priv;
|
|
|
|
return ret;
|
|
}
|
|
|
|
static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int ret;
|
|
RTSPMessageHeader reply1, *reply = &reply1;
|
|
char cmd[1024];
|
|
|
|
if (rt->server_type == RTSP_SERVER_REAL) {
|
|
int i;
|
|
enum AVDiscard cache[MAX_STREAMS];
|
|
|
|
for (i = 0; i < s->nb_streams; i++)
|
|
cache[i] = s->streams[i]->discard;
|
|
|
|
if (!rt->need_subscription) {
|
|
if (memcmp (cache, rt->real_setup_cache,
|
|
sizeof(enum AVDiscard) * s->nb_streams)) {
|
|
snprintf(cmd, sizeof(cmd),
|
|
"Unsubscribe: %s\r\n",
|
|
rt->last_subscription);
|
|
ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
|
|
cmd, reply, NULL);
|
|
if (reply->status_code != RTSP_STATUS_OK)
|
|
return AVERROR_INVALIDDATA;
|
|
rt->need_subscription = 1;
|
|
}
|
|
}
|
|
|
|
if (rt->need_subscription) {
|
|
int r, rule_nr, first = 1;
|
|
|
|
memcpy(rt->real_setup_cache, cache,
|
|
sizeof(enum AVDiscard) * s->nb_streams);
|
|
rt->last_subscription[0] = 0;
|
|
|
|
snprintf(cmd, sizeof(cmd),
|
|
"Subscribe: ");
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
rule_nr = 0;
|
|
for (r = 0; r < s->nb_streams; r++) {
|
|
if (s->streams[r]->priv_data == rt->rtsp_streams[i]) {
|
|
if (s->streams[r]->discard != AVDISCARD_ALL) {
|
|
if (!first)
|
|
av_strlcat(rt->last_subscription, ",",
|
|
sizeof(rt->last_subscription));
|
|
ff_rdt_subscribe_rule(
|
|
rt->last_subscription,
|
|
sizeof(rt->last_subscription), i, rule_nr);
|
|
first = 0;
|
|
}
|
|
rule_nr++;
|
|
}
|
|
}
|
|
}
|
|
av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription);
|
|
ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
|
|
cmd, reply, NULL);
|
|
if (reply->status_code != RTSP_STATUS_OK)
|
|
return AVERROR_INVALIDDATA;
|
|
rt->need_subscription = 0;
|
|
|
|
if (rt->state == RTSP_STATE_STREAMING)
|
|
rtsp_read_play (s);
|
|
}
|
|
}
|
|
|
|
ret = rtsp_fetch_packet(s, pkt);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
/* send dummy request to keep TCP connection alive */
|
|
if ((rt->server_type == RTSP_SERVER_WMS ||
|
|
rt->server_type == RTSP_SERVER_REAL) &&
|
|
(av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
|
|
if (rt->server_type == RTSP_SERVER_WMS) {
|
|
ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
|
|
} else {
|
|
ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL);
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* pause the stream */
|
|
static int rtsp_read_pause(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
RTSPMessageHeader reply1, *reply = &reply1;
|
|
|
|
rt = s->priv_data;
|
|
|
|
if (rt->state != RTSP_STATE_STREAMING)
|
|
return 0;
|
|
else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
|
|
ff_rtsp_send_cmd(s, "PAUSE", rt->control_uri, NULL, reply, NULL);
|
|
if (reply->status_code != RTSP_STATUS_OK) {
|
|
return -1;
|
|
}
|
|
}
|
|
rt->state = RTSP_STATE_PAUSED;
|
|
return 0;
|
|
}
|
|
|
|
static int rtsp_read_seek(AVFormatContext *s, int stream_index,
|
|
int64_t timestamp, int flags)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
|
|
rt->seek_timestamp = av_rescale_q(timestamp,
|
|
s->streams[stream_index]->time_base,
|
|
AV_TIME_BASE_Q);
|
|
switch(rt->state) {
|
|
default:
|
|
case RTSP_STATE_IDLE:
|
|
break;
|
|
case RTSP_STATE_STREAMING:
|
|
if (rtsp_read_pause(s) != 0)
|
|
return -1;
|
|
rt->state = RTSP_STATE_SEEKING;
|
|
if (rtsp_read_play(s) != 0)
|
|
return -1;
|
|
break;
|
|
case RTSP_STATE_PAUSED:
|
|
rt->state = RTSP_STATE_IDLE;
|
|
break;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int rtsp_read_close(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
|
|
#if 0
|
|
/* NOTE: it is valid to flush the buffer here */
|
|
if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
|
|
url_fclose(&rt->rtsp_gb);
|
|
}
|
|
#endif
|
|
ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
|
|
|
|
ff_rtsp_close_streams(s);
|
|
url_close(rt->rtsp_hd);
|
|
ff_network_close();
|
|
return 0;
|
|
}
|
|
|
|
AVInputFormat rtsp_demuxer = {
|
|
"rtsp",
|
|
NULL_IF_CONFIG_SMALL("RTSP input format"),
|
|
sizeof(RTSPState),
|
|
rtsp_probe,
|
|
rtsp_read_header,
|
|
rtsp_read_packet,
|
|
rtsp_read_close,
|
|
rtsp_read_seek,
|
|
.flags = AVFMT_NOFILE,
|
|
.read_play = rtsp_read_play,
|
|
.read_pause = rtsp_read_pause,
|
|
};
|
|
#endif
|
|
|
|
static int sdp_probe(AVProbeData *p1)
|
|
{
|
|
const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
|
|
|
|
/* we look for a line beginning "c=IN IP4" */
|
|
while (p < p_end && *p != '\0') {
|
|
if (p + sizeof("c=IN IP4") - 1 < p_end &&
|
|
av_strstart(p, "c=IN IP4", NULL))
|
|
return AVPROBE_SCORE_MAX / 2;
|
|
|
|
while (p < p_end - 1 && *p != '\n') p++;
|
|
if (++p >= p_end)
|
|
break;
|
|
if (*p == '\r')
|
|
p++;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
#define SDP_MAX_SIZE 8192
|
|
|
|
static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
RTSPStream *rtsp_st;
|
|
int size, i, err;
|
|
char *content;
|
|
char url[1024];
|
|
|
|
if (!ff_network_init())
|
|
return AVERROR(EIO);
|
|
|
|
/* read the whole sdp file */
|
|
/* XXX: better loading */
|
|
content = av_malloc(SDP_MAX_SIZE);
|
|
size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
|
|
if (size <= 0) {
|
|
av_free(content);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
content[size] ='\0';
|
|
|
|
sdp_parse(s, content);
|
|
av_free(content);
|
|
|
|
/* open each RTP stream */
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
|
|
ff_url_join(url, sizeof(url), "rtp", NULL,
|
|
inet_ntoa(rtsp_st->sdp_ip), rtsp_st->sdp_port,
|
|
"?localport=%d&ttl=%d", rtsp_st->sdp_port,
|
|
rtsp_st->sdp_ttl);
|
|
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
|
|
err = AVERROR_INVALIDDATA;
|
|
goto fail;
|
|
}
|
|
if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
|
|
goto fail;
|
|
}
|
|
return 0;
|
|
fail:
|
|
ff_rtsp_close_streams(s);
|
|
ff_network_close();
|
|
return err;
|
|
}
|
|
|
|
static int sdp_read_close(AVFormatContext *s)
|
|
{
|
|
ff_rtsp_close_streams(s);
|
|
ff_network_close();
|
|
return 0;
|
|
}
|
|
|
|
AVInputFormat sdp_demuxer = {
|
|
"sdp",
|
|
NULL_IF_CONFIG_SMALL("SDP"),
|
|
sizeof(RTSPState),
|
|
sdp_probe,
|
|
sdp_read_header,
|
|
rtsp_fetch_packet,
|
|
sdp_read_close,
|
|
};
|