mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-21 10:55:51 +02:00
03f8fc0897
instead of using the original host name, since the RTP (and UDP) protocols may choose another IP address if the host name resolves into several different addresses. Originally committed as revision 22398 to svn://svn.ffmpeg.org/ffmpeg/trunk
2000 lines
65 KiB
C
2000 lines
65 KiB
C
/*
|
|
* RTSP/SDP client
|
|
* Copyright (c) 2002 Fabrice Bellard
|
|
*
|
|
* This file is part of FFmpeg.
|
|
*
|
|
* FFmpeg is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Lesser General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2.1 of the License, or (at your option) any later version.
|
|
*
|
|
* FFmpeg is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Lesser General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Lesser General Public
|
|
* License along with FFmpeg; if not, write to the Free Software
|
|
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
|
*/
|
|
|
|
#include "libavutil/base64.h"
|
|
#include "libavutil/avstring.h"
|
|
#include "libavutil/intreadwrite.h"
|
|
#include "avformat.h"
|
|
|
|
#include <sys/time.h>
|
|
#if HAVE_SYS_SELECT_H
|
|
#include <sys/select.h>
|
|
#endif
|
|
#include <strings.h>
|
|
#include "network.h"
|
|
#include "rtsp.h"
|
|
|
|
#include "rtpdec.h"
|
|
#include "rdt.h"
|
|
#include "rtpdec_asf.h"
|
|
#include "rtpdec_vorbis.h"
|
|
|
|
//#define DEBUG
|
|
//#define DEBUG_RTP_TCP
|
|
|
|
#if LIBAVFORMAT_VERSION_INT < (53 << 16)
|
|
int rtsp_default_protocols = (1 << RTSP_LOWER_TRANSPORT_UDP);
|
|
#endif
|
|
|
|
#define SPACE_CHARS " \t\r\n"
|
|
/* we use memchr() instead of strchr() here because strchr() will return
|
|
* the terminating '\0' of SPACE_CHARS instead of NULL if c is '\0'. */
|
|
#define redir_isspace(c) memchr(SPACE_CHARS, c, 4)
|
|
static void skip_spaces(const char **pp)
|
|
{
|
|
const char *p;
|
|
p = *pp;
|
|
while (redir_isspace(*p))
|
|
p++;
|
|
*pp = p;
|
|
}
|
|
|
|
static void get_word_until_chars(char *buf, int buf_size,
|
|
const char *sep, const char **pp)
|
|
{
|
|
const char *p;
|
|
char *q;
|
|
|
|
p = *pp;
|
|
skip_spaces(&p);
|
|
q = buf;
|
|
while (!strchr(sep, *p) && *p != '\0') {
|
|
if ((q - buf) < buf_size - 1)
|
|
*q++ = *p;
|
|
p++;
|
|
}
|
|
if (buf_size > 0)
|
|
*q = '\0';
|
|
*pp = p;
|
|
}
|
|
|
|
static void get_word_sep(char *buf, int buf_size, const char *sep,
|
|
const char **pp)
|
|
{
|
|
if (**pp == '/') (*pp)++;
|
|
get_word_until_chars(buf, buf_size, sep, pp);
|
|
}
|
|
|
|
static void get_word(char *buf, int buf_size, const char **pp)
|
|
{
|
|
get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
|
|
}
|
|
|
|
/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
|
|
static int sdp_parse_rtpmap(AVFormatContext *s,
|
|
AVCodecContext *codec, RTSPStream *rtsp_st,
|
|
int payload_type, const char *p)
|
|
{
|
|
char buf[256];
|
|
int i;
|
|
AVCodec *c;
|
|
const char *c_name;
|
|
|
|
/* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
|
|
* see if we can handle this kind of payload.
|
|
* The space should normally not be there but some Real streams or
|
|
* particular servers ("RealServer Version 6.1.3.970", see issue 1658)
|
|
* have a trailing space. */
|
|
get_word_sep(buf, sizeof(buf), "/ ", &p);
|
|
if (payload_type >= RTP_PT_PRIVATE) {
|
|
RTPDynamicProtocolHandler *handler;
|
|
for (handler = RTPFirstDynamicPayloadHandler;
|
|
handler; handler = handler->next) {
|
|
if (!strcasecmp(buf, handler->enc_name) &&
|
|
codec->codec_type == handler->codec_type) {
|
|
codec->codec_id = handler->codec_id;
|
|
rtsp_st->dynamic_handler = handler;
|
|
if (handler->open)
|
|
rtsp_st->dynamic_protocol_context = handler->open();
|
|
break;
|
|
}
|
|
}
|
|
} else {
|
|
/* We are in a standard case
|
|
* (from http://www.iana.org/assignments/rtp-parameters). */
|
|
/* search into AVRtpPayloadTypes[] */
|
|
codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
|
|
}
|
|
|
|
c = avcodec_find_decoder(codec->codec_id);
|
|
if (c && c->name)
|
|
c_name = c->name;
|
|
else
|
|
c_name = "(null)";
|
|
|
|
get_word_sep(buf, sizeof(buf), "/", &p);
|
|
i = atoi(buf);
|
|
switch (codec->codec_type) {
|
|
case CODEC_TYPE_AUDIO:
|
|
av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
|
|
codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
|
|
codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
|
|
if (i > 0) {
|
|
codec->sample_rate = i;
|
|
get_word_sep(buf, sizeof(buf), "/", &p);
|
|
i = atoi(buf);
|
|
if (i > 0)
|
|
codec->channels = i;
|
|
// TODO: there is a bug here; if it is a mono stream, and
|
|
// less than 22000Hz, faad upconverts to stereo and twice
|
|
// the frequency. No problem, but the sample rate is being
|
|
// set here by the sdp line. Patch on its way. (rdm)
|
|
}
|
|
av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
|
|
codec->sample_rate);
|
|
av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
|
|
codec->channels);
|
|
break;
|
|
case CODEC_TYPE_VIDEO:
|
|
av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/* return the length and optionally the data */
|
|
static int hex_to_data(uint8_t *data, const char *p)
|
|
{
|
|
int c, len, v;
|
|
|
|
len = 0;
|
|
v = 1;
|
|
for (;;) {
|
|
skip_spaces(&p);
|
|
if (*p == '\0')
|
|
break;
|
|
c = toupper((unsigned char) *p++);
|
|
if (c >= '0' && c <= '9')
|
|
c = c - '0';
|
|
else if (c >= 'A' && c <= 'F')
|
|
c = c - 'A' + 10;
|
|
else
|
|
break;
|
|
v = (v << 4) | c;
|
|
if (v & 0x100) {
|
|
if (data)
|
|
data[len] = v;
|
|
len++;
|
|
v = 1;
|
|
}
|
|
}
|
|
return len;
|
|
}
|
|
|
|
static void sdp_parse_fmtp_config(AVCodecContext * codec, void *ctx,
|
|
char *attr, char *value)
|
|
{
|
|
switch (codec->codec_id) {
|
|
case CODEC_ID_MPEG4:
|
|
case CODEC_ID_AAC:
|
|
if (!strcmp(attr, "config")) {
|
|
/* decode the hexa encoded parameter */
|
|
int len = hex_to_data(NULL, value);
|
|
if (codec->extradata)
|
|
av_free(codec->extradata);
|
|
codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
|
|
if (!codec->extradata)
|
|
return;
|
|
codec->extradata_size = len;
|
|
hex_to_data(codec->extradata, value);
|
|
}
|
|
break;
|
|
case CODEC_ID_VORBIS:
|
|
ff_vorbis_parse_fmtp_config(codec, ctx, attr, value);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return;
|
|
}
|
|
|
|
typedef struct {
|
|
const char *str;
|
|
uint16_t type;
|
|
uint32_t offset;
|
|
} AttrNameMap;
|
|
|
|
/* All known fmtp parmeters and the corresping RTPAttrTypeEnum */
|
|
#define ATTR_NAME_TYPE_INT 0
|
|
#define ATTR_NAME_TYPE_STR 1
|
|
static const AttrNameMap attr_names[]=
|
|
{
|
|
{ "SizeLength", ATTR_NAME_TYPE_INT,
|
|
offsetof(RTPPayloadData, sizelength) },
|
|
{ "IndexLength", ATTR_NAME_TYPE_INT,
|
|
offsetof(RTPPayloadData, indexlength) },
|
|
{ "IndexDeltaLength", ATTR_NAME_TYPE_INT,
|
|
offsetof(RTPPayloadData, indexdeltalength) },
|
|
{ "profile-level-id", ATTR_NAME_TYPE_INT,
|
|
offsetof(RTPPayloadData, profile_level_id) },
|
|
{ "StreamType", ATTR_NAME_TYPE_INT,
|
|
offsetof(RTPPayloadData, streamtype) },
|
|
{ "mode", ATTR_NAME_TYPE_STR,
|
|
offsetof(RTPPayloadData, mode) },
|
|
{ NULL, -1, -1 },
|
|
};
|
|
|
|
/* parse the attribute line from the fmtp a line of an sdp resonse. This
|
|
* is broken out as a function because it is used in rtp_h264.c, which is
|
|
* forthcoming. */
|
|
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
|
|
char *value, int value_size)
|
|
{
|
|
skip_spaces(p);
|
|
if (**p) {
|
|
get_word_sep(attr, attr_size, "=", p);
|
|
if (**p == '=')
|
|
(*p)++;
|
|
get_word_sep(value, value_size, ";", p);
|
|
if (**p == ';')
|
|
(*p)++;
|
|
return 1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/* parse a SDP line and save stream attributes */
|
|
static void sdp_parse_fmtp(AVStream *st, const char *p)
|
|
{
|
|
char attr[256];
|
|
/* Vorbis setup headers can be up to 12KB and are sent base64
|
|
* encoded, giving a 12KB * (4/3) = 16KB FMTP line. */
|
|
char value[16384];
|
|
int i;
|
|
RTSPStream *rtsp_st = st->priv_data;
|
|
AVCodecContext *codec = st->codec;
|
|
RTPPayloadData *rtp_payload_data = &rtsp_st->rtp_payload_data;
|
|
|
|
/* loop on each attribute */
|
|
while (ff_rtsp_next_attr_and_value(&p, attr, sizeof(attr),
|
|
value, sizeof(value))) {
|
|
/* grab the codec extra_data from the config parameter of the fmtp
|
|
* line */
|
|
sdp_parse_fmtp_config(codec, rtsp_st->dynamic_protocol_context,
|
|
attr, value);
|
|
/* Looking for a known attribute */
|
|
for (i = 0; attr_names[i].str; ++i) {
|
|
if (!strcasecmp(attr, attr_names[i].str)) {
|
|
if (attr_names[i].type == ATTR_NAME_TYPE_INT) {
|
|
*(int *)((char *)rtp_payload_data +
|
|
attr_names[i].offset) = atoi(value);
|
|
} else if (attr_names[i].type == ATTR_NAME_TYPE_STR)
|
|
*(char **)((char *)rtp_payload_data +
|
|
attr_names[i].offset) = av_strdup(value);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/** Parse a string p in the form of Range:npt=xx-xx, and determine the start
|
|
* and end time.
|
|
* Used for seeking in the rtp stream.
|
|
*/
|
|
static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
|
|
{
|
|
char buf[256];
|
|
|
|
skip_spaces(&p);
|
|
if (!av_stristart(p, "npt=", &p))
|
|
return;
|
|
|
|
*start = AV_NOPTS_VALUE;
|
|
*end = AV_NOPTS_VALUE;
|
|
|
|
get_word_sep(buf, sizeof(buf), "-", &p);
|
|
*start = parse_date(buf, 1);
|
|
if (*p == '-') {
|
|
p++;
|
|
get_word_sep(buf, sizeof(buf), "-", &p);
|
|
*end = parse_date(buf, 1);
|
|
}
|
|
// av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
|
|
// av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
|
|
}
|
|
|
|
typedef struct SDPParseState {
|
|
/* SDP only */
|
|
struct in_addr default_ip;
|
|
int default_ttl;
|
|
int skip_media; ///< set if an unknown m= line occurs
|
|
} SDPParseState;
|
|
|
|
static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
|
|
int letter, const char *buf)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
char buf1[64], st_type[64];
|
|
const char *p;
|
|
enum CodecType codec_type;
|
|
int payload_type, i;
|
|
AVStream *st;
|
|
RTSPStream *rtsp_st;
|
|
struct in_addr sdp_ip;
|
|
int ttl;
|
|
|
|
dprintf(s, "sdp: %c='%s'\n", letter, buf);
|
|
|
|
p = buf;
|
|
if (s1->skip_media && letter != 'm')
|
|
return;
|
|
switch (letter) {
|
|
case 'c':
|
|
get_word(buf1, sizeof(buf1), &p);
|
|
if (strcmp(buf1, "IN") != 0)
|
|
return;
|
|
get_word(buf1, sizeof(buf1), &p);
|
|
if (strcmp(buf1, "IP4") != 0)
|
|
return;
|
|
get_word_sep(buf1, sizeof(buf1), "/", &p);
|
|
if (ff_inet_aton(buf1, &sdp_ip) == 0)
|
|
return;
|
|
ttl = 16;
|
|
if (*p == '/') {
|
|
p++;
|
|
get_word_sep(buf1, sizeof(buf1), "/", &p);
|
|
ttl = atoi(buf1);
|
|
}
|
|
if (s->nb_streams == 0) {
|
|
s1->default_ip = sdp_ip;
|
|
s1->default_ttl = ttl;
|
|
} else {
|
|
st = s->streams[s->nb_streams - 1];
|
|
rtsp_st = st->priv_data;
|
|
rtsp_st->sdp_ip = sdp_ip;
|
|
rtsp_st->sdp_ttl = ttl;
|
|
}
|
|
break;
|
|
case 's':
|
|
av_metadata_set(&s->metadata, "title", p);
|
|
break;
|
|
case 'i':
|
|
if (s->nb_streams == 0) {
|
|
av_metadata_set(&s->metadata, "comment", p);
|
|
break;
|
|
}
|
|
break;
|
|
case 'm':
|
|
/* new stream */
|
|
s1->skip_media = 0;
|
|
get_word(st_type, sizeof(st_type), &p);
|
|
if (!strcmp(st_type, "audio")) {
|
|
codec_type = CODEC_TYPE_AUDIO;
|
|
} else if (!strcmp(st_type, "video")) {
|
|
codec_type = CODEC_TYPE_VIDEO;
|
|
} else if (!strcmp(st_type, "application")) {
|
|
codec_type = CODEC_TYPE_DATA;
|
|
} else {
|
|
s1->skip_media = 1;
|
|
return;
|
|
}
|
|
rtsp_st = av_mallocz(sizeof(RTSPStream));
|
|
if (!rtsp_st)
|
|
return;
|
|
rtsp_st->stream_index = -1;
|
|
dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
|
|
|
|
rtsp_st->sdp_ip = s1->default_ip;
|
|
rtsp_st->sdp_ttl = s1->default_ttl;
|
|
|
|
get_word(buf1, sizeof(buf1), &p); /* port */
|
|
rtsp_st->sdp_port = atoi(buf1);
|
|
|
|
get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
|
|
|
|
/* XXX: handle list of formats */
|
|
get_word(buf1, sizeof(buf1), &p); /* format list */
|
|
rtsp_st->sdp_payload_type = atoi(buf1);
|
|
|
|
if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
|
|
/* no corresponding stream */
|
|
} else {
|
|
st = av_new_stream(s, 0);
|
|
if (!st)
|
|
return;
|
|
st->priv_data = rtsp_st;
|
|
rtsp_st->stream_index = st->index;
|
|
st->codec->codec_type = codec_type;
|
|
if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
|
|
/* if standard payload type, we can find the codec right now */
|
|
ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
|
|
}
|
|
}
|
|
/* put a default control url */
|
|
av_strlcpy(rtsp_st->control_url, rt->control_uri,
|
|
sizeof(rtsp_st->control_url));
|
|
break;
|
|
case 'a':
|
|
if (av_strstart(p, "control:", &p)) {
|
|
if (s->nb_streams == 0) {
|
|
if (!strncmp(p, "rtsp://", 7))
|
|
av_strlcpy(rt->control_uri, p,
|
|
sizeof(rt->control_uri));
|
|
} else {
|
|
char proto[32];
|
|
/* get the control url */
|
|
st = s->streams[s->nb_streams - 1];
|
|
rtsp_st = st->priv_data;
|
|
|
|
/* XXX: may need to add full url resolution */
|
|
ff_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 = st->priv_data;
|
|
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);
|
|
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;
|
|
}
|
|
av_freep(&rt->auth_b64);
|
|
}
|
|
|
|
static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st, URLContext *handle) {
|
|
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;
|
|
|
|
/* 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;
|
|
|
|
url_fdopen(&rtpctx->pb, handle);
|
|
ret = av_write_header(rtpctx);
|
|
|
|
if (ret) {
|
|
url_fclose(rtpctx->pb);
|
|
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);
|
|
/* Ownage 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)
|
|
{
|
|
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));
|
|
}
|
|
}
|
|
|
|
/* skip a RTP/TCP interleaved packet */
|
|
static void rtsp_skip_packet(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int ret, len, len1;
|
|
uint8_t buf[1024];
|
|
|
|
ret = url_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
|
|
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);
|
|
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);
|
|
|
|
/* 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 *cmd,
|
|
const unsigned char *send_content,
|
|
int send_content_length)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
char buf[4096], buf1[1024];
|
|
|
|
rt->seq++;
|
|
av_strlcpy(buf, cmd, sizeof(buf));
|
|
snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq);
|
|
av_strlcat(buf, buf1, sizeof(buf));
|
|
if (rt->session_id[0] != '\0' && !strstr(cmd, "\nIf-Match:")) {
|
|
snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id);
|
|
av_strlcat(buf, buf1, sizeof(buf));
|
|
}
|
|
if (rt->auth_b64)
|
|
av_strlcatf(buf, sizeof(buf),
|
|
"Authorization: Basic %s\r\n",
|
|
rt->auth_b64);
|
|
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 *cmd)
|
|
{
|
|
ff_rtsp_send_cmd_with_content_async(s, cmd, NULL, 0);
|
|
}
|
|
|
|
void ff_rtsp_send_cmd(AVFormatContext *s,
|
|
const char *cmd, RTSPMessageHeader *reply,
|
|
unsigned char **content_ptr)
|
|
{
|
|
ff_rtsp_send_cmd_async(s, cmd);
|
|
|
|
ff_rtsp_read_reply(s, reply, content_ptr, 0);
|
|
}
|
|
|
|
void ff_rtsp_send_cmd_with_content(AVFormatContext *s,
|
|
const char *cmd,
|
|
RTSPMessageHeader *reply,
|
|
unsigned char **content_ptr,
|
|
const unsigned char *send_content,
|
|
int send_content_length)
|
|
{
|
|
ff_rtsp_send_cmd_with_content_async(s, cmd, send_content, send_content_length);
|
|
|
|
ff_rtsp_read_reply(s, reply, content_ptr, 0);
|
|
}
|
|
|
|
/**
|
|
* @returns 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 ==
|
|
CODEC_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),
|
|
"SETUP %s RTSP/1.0\r\n"
|
|
"Transport: %s\r\n",
|
|
rtsp_st->control_url, 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, 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) {
|
|
snprintf(cmd, sizeof(cmd),
|
|
"PLAY %s RTSP/1.0\r\n",
|
|
rt->control_uri);
|
|
} else {
|
|
snprintf(cmd, sizeof(cmd),
|
|
"PLAY %s RTSP/1.0\r\n"
|
|
"Range: npt=%0.3f-\r\n",
|
|
rt->control_uri,
|
|
(double)rt->seek_timestamp / AV_TIME_BASE);
|
|
}
|
|
ff_rtsp_send_cmd(s, 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)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
RTSPMessageHeader reply1, *reply = &reply1;
|
|
char cmd[1024];
|
|
unsigned char *content = NULL;
|
|
int ret;
|
|
|
|
/* describe the stream */
|
|
snprintf(cmd, sizeof(cmd),
|
|
"DESCRIBE %s RTSP/1.0\r\n"
|
|
"Accept: application/sdp\r\n",
|
|
s->filename);
|
|
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, 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)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
RTSPMessageHeader reply1, *reply = &reply1;
|
|
char cmd[1024];
|
|
int i;
|
|
char *sdp;
|
|
|
|
/* Announce the stream */
|
|
snprintf(cmd, sizeof(cmd),
|
|
"ANNOUNCE %s RTSP/1.0\r\n"
|
|
"Content-Type: application/sdp\r\n",
|
|
s->filename);
|
|
sdp = av_mallocz(8192);
|
|
if (sdp == NULL)
|
|
return AVERROR(ENOMEM);
|
|
if (avf_sdp_create(&s, 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, cmd, 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, s->filename, 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) {
|
|
int auth_len = strlen(auth), b64_len = ((auth_len + 2) / 3) * 4 + 1;
|
|
|
|
if (!(rt->auth_b64 = av_malloc(b64_len)))
|
|
return AVERROR(ENOMEM);
|
|
if (!av_base64_encode(rt->auth_b64, b64_len, auth, auth_len)) {
|
|
err = AVERROR(EINVAL);
|
|
goto fail;
|
|
}
|
|
}
|
|
if (port < 0)
|
|
port = RTSP_DEFAULT_PORT;
|
|
|
|
/* search for options */
|
|
option_list = strchr(path, '?');
|
|
if (option_list) {
|
|
filename = strchr(s->filename, '?');
|
|
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 {
|
|
strcpy(++filename, option);
|
|
filename += strlen(option);
|
|
if (option_list) *filename = '&';
|
|
}
|
|
}
|
|
*filename = 0;
|
|
}
|
|
|
|
if (!lower_transport_mask)
|
|
lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
|
|
|
|
if (s->oformat) {
|
|
/* Only UDP output is supported at the moment. */
|
|
lower_transport_mask &= 1 << RTSP_LOWER_TRANSPORT_UDP;
|
|
if (!lower_transport_mask) {
|
|
av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
|
|
"only UDP is supported for output.\n");
|
|
err = AVERROR(EINVAL);
|
|
goto fail;
|
|
}
|
|
}
|
|
|
|
/* 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 */
|
|
av_strlcpy(rt->control_uri, s->filename,
|
|
sizeof(rt->control_uri));
|
|
for (rt->server_type = RTSP_SERVER_RTP;;) {
|
|
snprintf(cmd, sizeof(cmd),
|
|
"OPTIONS %s RTSP/1.0\r\n", s->filename);
|
|
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, 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);
|
|
else
|
|
err = rtsp_setup_output_streams(s);
|
|
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;
|
|
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 = 100 * 1000;
|
|
n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
|
|
if (n > 0) {
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
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;
|
|
|
|
ff_rtsp_read_reply(s, &reply, NULL, 0);
|
|
/* XXX: parse message */
|
|
if (rt->state != RTSP_STATE_STREAMING)
|
|
return 0;
|
|
}
|
|
#endif
|
|
}
|
|
}
|
|
}
|
|
|
|
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),
|
|
"SET_PARAMETER %s RTSP/1.0\r\n"
|
|
"Unsubscribe: %s\r\n",
|
|
rt->control_uri, rt->last_subscription);
|
|
ff_rtsp_send_cmd(s, 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),
|
|
"SET_PARAMETER %s RTSP/1.0\r\n"
|
|
"Subscribe: ",
|
|
rt->control_uri);
|
|
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, 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) {
|
|
snprintf(cmd, sizeof(cmd) - 1,
|
|
"GET_PARAMETER %s RTSP/1.0\r\n",
|
|
rt->control_uri);
|
|
ff_rtsp_send_cmd_async(s, cmd);
|
|
} else {
|
|
ff_rtsp_send_cmd_async(s, "OPTIONS * RTSP/1.0\r\n");
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* pause the stream */
|
|
static int rtsp_read_pause(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
RTSPMessageHeader reply1, *reply = &reply1;
|
|
char cmd[1024];
|
|
|
|
rt = s->priv_data;
|
|
|
|
if (rt->state != RTSP_STATE_STREAMING)
|
|
return 0;
|
|
else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
|
|
snprintf(cmd, sizeof(cmd),
|
|
"PAUSE %s RTSP/1.0\r\n",
|
|
rt->control_uri);
|
|
ff_rtsp_send_cmd(s, cmd, 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;
|
|
char cmd[1024];
|
|
|
|
#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
|
|
snprintf(cmd, sizeof(cmd),
|
|
"TEARDOWN %s RTSP/1.0\r\n",
|
|
s->filename);
|
|
ff_rtsp_send_cmd_async(s, cmd);
|
|
|
|
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,
|
|
};
|