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FFmpeg/libavformat/rtsp.c
Ronald S. Bultje 0ad306bc81 Remove unused code that used to handle protocol concatenation, i.e. trying
multiple protocols at the same time. We now cycle protocols individually
to autodetect, making this code no longer needed, and thus the support code
for it in make_setup_request() can be removed. See "[PATCH] remove transport
concatenation dead code" on mailinglist.

Originally committed as revision 15172 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-09-03 04:47:44 +00:00

1616 lines
49 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
*/
/* needed by inet_aton() */
#define _SVID_SOURCE
#include "libavutil/avstring.h"
#include "avformat.h"
#include <sys/time.h>
#include <unistd.h> /* for select() prototype */
#include <strings.h>
#include "network.h"
#include "rtsp.h"
#include "rtp_internal.h"
#include "rdt.h"
//#define DEBUG
//#define DEBUG_RTP_TCP
enum RTSPClientState {
RTSP_STATE_IDLE,
RTSP_STATE_PLAYING,
RTSP_STATE_PAUSED,
};
enum RTSPServerType {
RTSP_SERVER_RTP, /*< Standard-compliant RTP-server */
RTSP_SERVER_RDT, /*< Realmedia-style server */
RTSP_SERVER_LAST
};
typedef struct RTSPState {
URLContext *rtsp_hd; /* RTSP TCP connexion handle */
int nb_rtsp_streams;
struct RTSPStream **rtsp_streams;
enum RTSPClientState state;
int64_t seek_timestamp;
/* XXX: currently we use unbuffered input */
// ByteIOContext rtsp_gb;
int seq; /* RTSP command sequence number */
char session_id[512];
enum RTSPProtocol protocol;
enum RTSPServerType server_type;
char last_reply[2048]; /* XXX: allocate ? */
RTPDemuxContext *cur_rtp;
} RTSPState;
typedef struct RTSPStream {
URLContext *rtp_handle; /* RTP stream handle */
RTPDemuxContext *rtp_ctx; /* RTP parse context */
int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */
int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */
char control_url[1024]; /* url for this stream (from SDP) */
int sdp_port; /* port (from SDP content - not used in RTSP) */
struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */
int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */
int sdp_payload_type; /* payload type - only used in SDP */
rtp_payload_data_t rtp_payload_data; /* rtp payload parsing infos from SDP */
RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure)
void *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol)
} RTSPStream;
static int rtsp_read_play(AVFormatContext *s);
/* XXX: currently, the only way to change the protocols consists in
changing this variable */
#if LIBAVFORMAT_VERSION_INT < (53 << 16)
int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_UDP);
#endif
static int rtsp_probe(AVProbeData *p)
{
if (av_strstart(p->filename, "rtsp:", NULL))
return AVPROBE_SCORE_MAX;
return 0;
}
static int redir_isspace(int c)
{
return c == ' ' || c == '\t' || c == '\n' || c == '\r';
}
static void skip_spaces(const char **pp)
{
const char *p;
p = *pp;
while (redir_isspace(*p))
p++;
*pp = p;
}
static void get_word_sep(char *buf, int buf_size, const char *sep,
const char **pp)
{
const char *p;
char *q;
p = *pp;
if (*p == '/')
p++;
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(char *buf, int buf_size, const char **pp)
{
const char *p;
char *q;
p = *pp;
skip_spaces(&p);
q = buf;
while (!redir_isspace(*p) && *p != '\0') {
if ((q - buf) < buf_size - 1)
*q++ = *p;
p++;
}
if (buf_size > 0)
*q = '\0';
*pp = p;
}
/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other
params>] */
static int sdp_parse_rtpmap(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 */
get_word_sep(buf, sizeof(buf), "/", &p);
if (payload_type >= RTP_PT_PRIVATE) {
RTPDynamicProtocolHandler *handler= RTPFirstDynamicPayloadHandler;
while(handler) {
if (!strcmp(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;
}
handler= handler->next;
}
} 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 = (char *)NULL;
if (c_name) {
get_word_sep(buf, sizeof(buf), "/", &p);
i = atoi(buf);
switch (codec->codec_type) {
case CODEC_TYPE_AUDIO:
av_log(codec, 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. Upcoming patch forthcoming. (rdm)
}
av_log(codec, AV_LOG_DEBUG, " audio samplerate set to : %i\n", codec->sample_rate);
av_log(codec, AV_LOG_DEBUG, " audio channels set to : %i\n", codec->channels);
break;
case CODEC_TYPE_VIDEO:
av_log(codec, AV_LOG_DEBUG, " video codec set to : %s\n", c_name);
break;
default:
break;
}
return 0;
}
return -1;
}
/* return the length and optionnaly 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, 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);
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;
default:
break;
}
return;
}
typedef struct attrname_map
{
const char *str;
uint16_t type;
uint32_t offset;
} attrname_map_t;
/* All known fmtp parmeters and the corresping RTPAttrTypeEnum */
#define ATTR_NAME_TYPE_INT 0
#define ATTR_NAME_TYPE_STR 1
static attrname_map_t attr_names[]=
{
{"SizeLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, sizelength)},
{"IndexLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexlength)},
{"IndexDeltaLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexdeltalength)},
{"profile-level-id", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, profile_level_id)},
{"StreamType", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, streamtype)},
{"mode", ATTR_NAME_TYPE_STR, offsetof(rtp_payload_data_t, 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 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];
char value[4096];
int i;
RTSPStream *rtsp_st = st->priv_data;
AVCodecContext *codec = st->codec;
rtp_payload_data_t *rtp_payload_data = &rtsp_st->rtp_payload_data;
/* loop on each attribute */
while(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, 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;
} 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;
int codec_type, payload_type, i;
AVStream *st;
RTSPStream *rtsp_st;
struct in_addr sdp_ip;
int ttl;
#ifdef DEBUG
printf("sdp: %c='%s'\n", letter, buf);
#endif
p = buf;
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 (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_strlcpy(s->title, p, sizeof(s->title));
break;
case 'i':
if (s->nb_streams == 0) {
av_strlcpy(s->comment, p, sizeof(s->comment));
break;
}
break;
case 'm':
/* new stream */
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 {
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 */
rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
}
}
/* put a default control url */
av_strlcpy(rtsp_st->control_url, s->filename, sizeof(rtsp_st->control_url));
break;
case 'a':
if (av_strstart(p, "control:", &p) && s->nb_streams > 0) {
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 */
url_split(proto, sizeof(proto), NULL, 0, NULL, 0, NULL, NULL, 0, p);
if (proto[0] == '\0') {
/* relative control URL */
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)) {
/* NOTE: rtpmap is only supported AFTER the 'm=' 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) {
sdp_parse_rtpmap(st->codec, rtsp_st, payload_type, p);
}
}
} else if (av_strstart(p, "fmtp:", &p)) {
/* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
get_word(buf1, sizeof(buf1), &p);
payload_type = atoi(buf1);
for(i = 0; i < s->nb_streams;i++) {
st = s->streams[i];
rtsp_st = st->priv_data;
if (rtsp_st->sdp_payload_type == payload_type) {
if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) {
if(!rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf)) {
sdp_parse_fmtp(st, p);
}
} else {
sdp_parse_fmtp(st, p);
}
}
}
} else if(av_strstart(p, "framesize:", &p)) {
// let dynamic protocol handlers have a stab at the line.
get_word(buf1, sizeof(buf1), &p);
payload_type = atoi(buf1);
for(i = 0; i < s->nb_streams;i++) {
st = s->streams[i];
rtsp_st = st->priv_data;
if (rtsp_st->sdp_payload_type == payload_type) {
if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) {
rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf);
}
}
}
} else if(av_strstart(p, "range:", &p)) {
int64_t start, end;
// this is so that seeking on a streamed file can work.
rtsp_parse_range_npt(p, &start, &end);
s->start_time= start;
s->duration= (end==AV_NOPTS_VALUE)?AV_NOPTS_VALUE:end-start; // AV_NOPTS_VALUE means live broadcast (and can't seek)
} else if (s->nb_streams > 0) {
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->streams[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;
char buf[2048], *q;
SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
memset(s1, 0, sizeof(SDPParseState));
p = content;
for(;;) {
skip_spaces(&p);
letter = *p;
if (letter == '\0')
break;
p++;
if (*p != '=')
goto next_line;
p++;
/* get the content */
q = buf;
while (*p != '\n' && *p != '\r' && *p != '\0') {
if ((q - buf) < sizeof(buf) - 1)
*q++ = *p;
p++;
}
*q = '\0';
sdp_parse_line(s, s1, letter, buf);
next_line:
while (*p != '\n' && *p != '\0')
p++;
if (*p == '\n')
p++;
}
return 0;
}
static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
{
const char *p;
int v;
p = *pp;
skip_spaces(&p);
v = strtol(p, (char **)&p, 10);
if (*p == '-') {
p++;
*min_ptr = v;
v = strtol(p, (char **)&p, 10);
*max_ptr = v;
} else {
*min_ptr = v;
*max_ptr = v;
}
*pp = p;
}
/* XXX: only one transport specification is parsed */
static void rtsp_parse_transport(RTSPHeader *reply, const char *p)
{
char transport_protocol[16];
char profile[16];
char lower_transport[16];
char parameter[16];
RTSPTransportField *th;
char buf[256];
reply->nb_transports = 0;
for(;;) {
skip_spaces(&p);
if (*p == '\0')
break;
th = &reply->transports[reply->nb_transports];
get_word_sep(transport_protocol, sizeof(transport_protocol),
"/", &p);
if (*p == '/')
p++;
if (!strcasecmp (transport_protocol, "rtp")) {
get_word_sep(profile, sizeof(profile), "/;,", &p);
lower_transport[0] = '\0';
/* rtp/avp/<protocol> */
if (*p == '/') {
p++;
get_word_sep(lower_transport, sizeof(lower_transport),
";,", &p);
}
} else if (!strcasecmp (transport_protocol, "x-pn-tng")) {
/* x-pn-tng/<protocol> */
get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
profile[0] = '\0';
}
if (!strcasecmp(lower_transport, "TCP"))
th->protocol = RTSP_PROTOCOL_RTP_TCP;
else
th->protocol = RTSP_PROTOCOL_RTP_UDP;
if (*p == ';')
p++;
/* get each parameter */
while (*p != '\0' && *p != ',') {
get_word_sep(parameter, sizeof(parameter), "=;,", &p);
if (!strcmp(parameter, "port")) {
if (*p == '=') {
p++;
rtsp_parse_range(&th->port_min, &th->port_max, &p);
}
} else if (!strcmp(parameter, "client_port")) {
if (*p == '=') {
p++;
rtsp_parse_range(&th->client_port_min,
&th->client_port_max, &p);
}
} else if (!strcmp(parameter, "server_port")) {
if (*p == '=') {
p++;
rtsp_parse_range(&th->server_port_min,
&th->server_port_max, &p);
}
} else if (!strcmp(parameter, "interleaved")) {
if (*p == '=') {
p++;
rtsp_parse_range(&th->interleaved_min,
&th->interleaved_max, &p);
}
} else if (!strcmp(parameter, "multicast")) {
if (th->protocol == RTSP_PROTOCOL_RTP_UDP)
th->protocol = RTSP_PROTOCOL_RTP_UDP_MULTICAST;
} else if (!strcmp(parameter, "ttl")) {
if (*p == '=') {
p++;
th->ttl = strtol(p, (char **)&p, 10);
}
} else if (!strcmp(parameter, "destination")) {
struct in_addr ipaddr;
if (*p == '=') {
p++;
get_word_sep(buf, sizeof(buf), ";,", &p);
if (inet_aton(buf, &ipaddr))
th->destination = ntohl(ipaddr.s_addr);
}
}
while (*p != ';' && *p != '\0' && *p != ',')
p++;
if (*p == ';')
p++;
}
if (*p == ',')
p++;
reply->nb_transports++;
}
}
void rtsp_parse_line(RTSPHeader *reply, const char *buf)
{
const char *p;
/* NOTE: we do case independent match for broken servers */
p = buf;
if (av_stristart(p, "Session:", &p)) {
get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
} else if (av_stristart(p, "Content-Length:", &p)) {
reply->content_length = strtol(p, NULL, 10);
} else if (av_stristart(p, "Transport:", &p)) {
rtsp_parse_transport(reply, p);
} else if (av_stristart(p, "CSeq:", &p)) {
reply->seq = strtol(p, NULL, 10);
} else if (av_stristart(p, "Range:", &p)) {
rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
} else if (av_stristart(p, "RealChallenge1:", &p)) {
skip_spaces(&p);
av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
}
}
static int url_readbuf(URLContext *h, unsigned char *buf, int size)
{
int ret, len;
len = 0;
while (len < size) {
ret = url_read(h, buf+len, size-len);
if (ret < 1)
return ret;
len += ret;
}
return len;
}
/* skip a RTP/TCP interleaved packet */
static void rtsp_skip_packet(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
int ret, len, len1;
uint8_t buf[1024];
ret = url_readbuf(rt->rtsp_hd, buf, 3);
if (ret != 3)
return;
len = AV_RB16(buf + 1);
#ifdef DEBUG
printf("skipping RTP packet len=%d\n", len);
#endif
/* skip payload */
while (len > 0) {
len1 = len;
if (len1 > sizeof(buf))
len1 = sizeof(buf);
ret = url_readbuf(rt->rtsp_hd, buf, len1);
if (ret != len1)
return;
len -= len1;
}
}
static void rtsp_send_cmd(AVFormatContext *s,
const char *cmd, RTSPHeader *reply,
unsigned char **content_ptr)
{
RTSPState *rt = s->priv_data;
char buf[4096], buf1[1024], *q;
unsigned char ch;
const char *p;
int content_length, line_count;
unsigned char *content = NULL;
memset(reply, 0, sizeof(RTSPHeader));
rt->seq++;
av_strlcpy(buf, cmd, sizeof(buf));
snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq);
av_strlcat(buf, buf1, sizeof(buf));
if (rt->session_id[0] != '\0' && !strstr(cmd, "\nIf-Match:")) {
snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id);
av_strlcat(buf, buf1, sizeof(buf));
}
av_strlcat(buf, "\r\n", sizeof(buf));
#ifdef DEBUG
printf("Sending:\n%s--\n", buf);
#endif
url_write(rt->rtsp_hd, buf, strlen(buf));
/* parse reply (XXX: use buffers) */
line_count = 0;
rt->last_reply[0] = '\0';
for(;;) {
q = buf;
for(;;) {
if (url_readbuf(rt->rtsp_hd, &ch, 1) != 1)
break;
if (ch == '\n')
break;
if (ch == '$') {
/* XXX: only parse it if first char on line ? */
rtsp_skip_packet(s);
} else if (ch != '\r') {
if ((q - buf) < sizeof(buf) - 1)
*q++ = ch;
}
}
*q = '\0';
#ifdef DEBUG
printf("line='%s'\n", buf);
#endif
/* test if last line */
if (buf[0] == '\0')
break;
p = buf;
if (line_count == 0) {
/* get reply code */
get_word(buf1, sizeof(buf1), &p);
get_word(buf1, sizeof(buf1), &p);
reply->status_code = atoi(buf1);
} else {
rtsp_parse_line(reply, p);
av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
}
line_count++;
}
if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
content_length = reply->content_length;
if (content_length > 0) {
/* leave some room for a trailing '\0' (useful for simple parsing) */
content = av_malloc(content_length + 1);
(void)url_readbuf(rt->rtsp_hd, content, content_length);
content[content_length] = '\0';
}
if (content_ptr)
*content_ptr = content;
else
av_free(content);
}
/* close and free RTSP streams */
static void rtsp_close_streams(RTSPState *rt)
{
int i;
RTSPStream *rtsp_st;
for(i=0;i<rt->nb_rtsp_streams;i++) {
rtsp_st = rt->rtsp_streams[i];
if (rtsp_st) {
if (rtsp_st->rtp_ctx)
rtp_parse_close(rtsp_st->rtp_ctx);
if (rtsp_st->rtp_handle)
url_close(rtsp_st->rtp_handle);
if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
rtsp_st->dynamic_handler->close(rtsp_st->dynamic_protocol_context);
}
}
av_free(rt->rtsp_streams);
}
/**
* @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 protocol, const char *real_challenge)
{
RTSPState *rt = s->priv_data;
int j, i, err;
RTSPStream *rtsp_st;
AVStream *st;
RTSPHeader reply1, *reply = &reply1;
char cmd[2048];
const char *trans_pref;
if (rt->server_type == RTSP_SERVER_RDT)
trans_pref = "x-pn-tng";
else
trans_pref = "RTP/AVP";
/* for each stream, make the setup request */
/* XXX: we assume the same server is used for the control of each
RTSP stream */
for(j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
char transport[2048];
rtsp_st = rt->rtsp_streams[i];
/* RTP/UDP */
if (protocol == RTSP_PROTOCOL_RTP_UDP) {
char buf[256];
/* first try in specified port range */
if (RTSP_RTP_PORT_MIN != 0) {
while(j <= RTSP_RTP_PORT_MAX) {
snprintf(buf, sizeof(buf), "rtp://%s?localport=%d", host, j);
j += 2; /* we will use two port by rtp stream (rtp and rtcp) */
if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0) {
goto rtp_opened;
}
}
}
/* then try on any port
** if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
** err = AVERROR_INVALIDDATA;
** goto fail;
** }
*/
rtp_opened:
port = rtp_get_local_port(rtsp_st->rtp_handle);
snprintf(transport, sizeof(transport) - 1,
"%s/UDP;unicast;client_port=%d",
trans_pref, port);
if (rt->server_type == RTSP_SERVER_RTP)
av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
}
/* RTP/TCP */
else if (protocol == RTSP_PROTOCOL_RTP_TCP) {
snprintf(transport, sizeof(transport) - 1,
"%s/TCP", trans_pref);
}
else if (protocol == RTSP_PROTOCOL_RTP_UDP_MULTICAST) {
snprintf(transport, sizeof(transport) - 1,
"%s/UDP;multicast", trans_pref);
}
if (rt->server_type == RTSP_SERVER_RDT)
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_RDT) {
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);
}
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].protocol != rt->protocol) {
err = AVERROR_INVALIDDATA;
goto fail;
}
} else {
rt->protocol = reply->transports[0].protocol;
}
/* close RTP connection if not choosen */
if (reply->transports[0].protocol != RTSP_PROTOCOL_RTP_UDP &&
(protocol == RTSP_PROTOCOL_RTP_UDP)) {
url_close(rtsp_st->rtp_handle);
rtsp_st->rtp_handle = NULL;
}
switch(reply->transports[0].protocol) {
case RTSP_PROTOCOL_RTP_TCP:
rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
break;
case RTSP_PROTOCOL_RTP_UDP:
{
char url[1024];
/* XXX: also use address if specified */
snprintf(url, sizeof(url), "rtp://%s:%d",
host, reply->transports[0].server_port_min);
if (rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
}
break;
case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
{
char url[1024];
struct in_addr in;
in.s_addr = htonl(reply->transports[0].destination);
snprintf(url, sizeof(url), "rtp://%s:%d?ttl=%d",
inet_ntoa(in),
reply->transports[0].port_min,
reply->transports[0].ttl);
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
}
break;
}
/* open the RTP context */
st = NULL;
if (rtsp_st->stream_index >= 0)
st = s->streams[rtsp_st->stream_index];
if (!st)
s->ctx_flags |= AVFMTCTX_NOHEADER;
rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
if (!rtsp_st->rtp_ctx) {
err = AVERROR(ENOMEM);
goto fail;
} else {
if(rtsp_st->dynamic_handler) {
rtsp_st->rtp_ctx->dynamic_protocol_context= rtsp_st->dynamic_protocol_context;
rtsp_st->rtp_ctx->parse_packet= rtsp_st->dynamic_handler->parse_packet;
}
}
}
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_header(AVFormatContext *s,
AVFormatParameters *ap)
{
RTSPState *rt = s->priv_data;
char host[1024], path[1024], tcpname[1024], cmd[2048], *option_list, *option;
URLContext *rtsp_hd;
int port, ret, err;
RTSPHeader reply1, *reply = &reply1;
unsigned char *content = NULL;
int protocol_mask = 0;
char real_challenge[64];
/* extract hostname and port */
url_split(NULL, 0, NULL, 0,
host, sizeof(host), &port, path, sizeof(path), s->filename);
if (port < 0)
port = RTSP_DEFAULT_PORT;
/* search for options */
option_list = strchr(path, '?');
if (option_list) {
/* remove the options from the path */
*option_list++ = 0;
while(option_list) {
/* move the option pointer */
option = option_list;
option_list = strchr(option_list, '&');
if (option_list)
*(option_list++) = 0;
/* handle the options */
if (strcmp(option, "udp") == 0)
protocol_mask = (1<< RTSP_PROTOCOL_RTP_UDP);
else if (strcmp(option, "multicast") == 0)
protocol_mask = (1<< RTSP_PROTOCOL_RTP_UDP_MULTICAST);
else if (strcmp(option, "tcp") == 0)
protocol_mask = (1<< RTSP_PROTOCOL_RTP_TCP);
}
}
if (!protocol_mask)
protocol_mask = (1 << RTSP_PROTOCOL_RTP_LAST) - 1;
/* open the tcp connexion */
snprintf(tcpname, sizeof(tcpname), "tcp://%s:%d", host, port);
if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0)
return AVERROR(EIO);
rt->rtsp_hd = rtsp_hd;
rt->seq = 0;
/* request options supported by the server; this also detects server type */
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_RDT)
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));
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_RDT && reply->real_challenge[0]) {
rt->server_type = RTSP_SERVER_RDT;
continue;
} else if (rt->server_type == RTSP_SERVER_RDT) {
strcpy(real_challenge, reply->real_challenge);
}
break;
}
/* 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_RDT) {
/**
* 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));
}
rtsp_send_cmd(s, cmd, reply, &content);
if (!content) {
err = AVERROR_INVALIDDATA;
goto fail;
}
if (reply->status_code != RTSP_STATUS_OK) {
err = AVERROR_INVALIDDATA;
goto fail;
}
/* now we got the SDP description, we parse it */
ret = sdp_parse(s, (const char *)content);
av_freep(&content);
if (ret < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
do {
int protocol = ff_log2_tab[protocol_mask & ~(protocol_mask - 1)];
err = make_setup_request(s, host, port, protocol,
rt->server_type == RTSP_SERVER_RDT ?
real_challenge : NULL);
if (err < 0)
goto fail;
protocol_mask &= ~(1 << protocol);
if (protocol_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 */
if (ap->initial_pause) {
/* do not start immediately */
} else {
if (rtsp_read_play(s) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
}
return 0;
fail:
rtsp_close_streams(rt);
av_freep(&content);
url_close(rt->rtsp_hd);
return err;
}
static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
uint8_t *buf, int buf_size)
{
RTSPState *rt = s->priv_data;
int id, len, i, ret;
RTSPStream *rtsp_st;
#ifdef DEBUG_RTP_TCP
printf("tcp_read_packet:\n");
#endif
redo:
for(;;) {
ret = url_readbuf(rt->rtsp_hd, buf, 1);
#ifdef DEBUG_RTP_TCP
printf("ret=%d c=%02x [%c]\n", ret, buf[0], buf[0]);
#endif
if (ret != 1)
return -1;
if (buf[0] == '$')
break;
}
ret = url_readbuf(rt->rtsp_hd, buf, 3);
if (ret != 3)
return -1;
id = buf[0];
len = AV_RB16(buf + 1);
#ifdef DEBUG_RTP_TCP
printf("id=%d len=%d\n", id, len);
#endif
if (len > buf_size || len < 12)
goto redo;
/* get the data */
ret = url_readbuf(rt->rtsp_hd, buf, len);
if (ret != len)
return -1;
/* find the matching stream */
for(i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
if (id >= rtsp_st->interleaved_min &&
id <= rtsp_st->interleaved_max)
goto found;
}
goto redo;
found:
*prtsp_st = rtsp_st;
return len;
}
static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
uint8_t *buf, int buf_size)
{
RTSPState *rt = s->priv_data;
RTSPStream *rtsp_st;
fd_set rfds;
int fd1, fd2, fd_max, n, i, ret;
struct timeval tv;
for(;;) {
if (url_interrupt_cb())
return AVERROR(EINTR);
FD_ZERO(&rfds);
fd_max = -1;
for(i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
/* currently, we cannot probe RTCP handle because of blocking restrictions */
rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2);
if (fd1 > fd_max)
fd_max = fd1;
FD_SET(fd1, &rfds);
}
tv.tv_sec = 0;
tv.tv_usec = 100 * 1000;
n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
if (n > 0) {
for(i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2);
if (FD_ISSET(fd1, &rfds)) {
ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
if (ret > 0) {
*prtsp_st = rtsp_st;
return ret;
}
}
}
}
}
}
static int rtsp_read_packet(AVFormatContext *s,
AVPacket *pkt)
{
RTSPState *rt = s->priv_data;
RTSPStream *rtsp_st;
int ret, len;
uint8_t buf[RTP_MAX_PACKET_LENGTH];
/* get next frames from the same RTP packet */
if (rt->cur_rtp) {
ret = rtp_parse_packet(rt->cur_rtp, pkt, NULL, 0);
if (ret == 0) {
rt->cur_rtp = NULL;
return 0;
} else if (ret == 1) {
return 0;
} else {
rt->cur_rtp = NULL;
}
}
/* read next RTP packet */
redo:
switch(rt->protocol) {
default:
case RTSP_PROTOCOL_RTP_TCP:
len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
break;
case RTSP_PROTOCOL_RTP_UDP:
case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
if (len >=0 && rtsp_st->rtp_ctx)
rtp_check_and_send_back_rr(rtsp_st->rtp_ctx, len);
break;
}
if (len < 0)
return len;
ret = rtp_parse_packet(rtsp_st->rtp_ctx, pkt, buf, len);
if (ret < 0)
goto redo;
if (ret == 1) {
/* more packets may follow, so we save the RTP context */
rt->cur_rtp = rtsp_st->rtp_ctx;
}
return 0;
}
static int rtsp_read_play(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
RTSPHeader reply1, *reply = &reply1;
char cmd[1024];
av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
if (rt->state == RTSP_STATE_PAUSED) {
snprintf(cmd, sizeof(cmd),
"PLAY %s RTSP/1.0\r\n",
s->filename);
} else {
snprintf(cmd, sizeof(cmd),
"PLAY %s RTSP/1.0\r\n"
"Range: npt=%0.3f-\r\n",
s->filename,
(double)rt->seek_timestamp / AV_TIME_BASE);
}
rtsp_send_cmd(s, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
return -1;
}
rt->state = RTSP_STATE_PLAYING;
return 0;
}
/* pause the stream */
static int rtsp_read_pause(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
RTSPHeader reply1, *reply = &reply1;
char cmd[1024];
rt = s->priv_data;
if (rt->state != RTSP_STATE_PLAYING)
return 0;
snprintf(cmd, sizeof(cmd),
"PAUSE %s RTSP/1.0\r\n",
s->filename);
rtsp_send_cmd(s, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
return -1;
}
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_PLAYING:
if (rtsp_read_play(s) != 0)
return -1;
break;
case RTSP_STATE_PAUSED:
rt->state = RTSP_STATE_IDLE;
break;
}
return 0;
}
static int rtsp_read_close(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
RTSPHeader reply1, *reply = &reply1;
char cmd[1024];
#if 0
/* NOTE: it is valid to flush the buffer here */
if (rt->protocol == RTSP_PROTOCOL_RTP_TCP) {
url_fclose(&rt->rtsp_gb);
}
#endif
snprintf(cmd, sizeof(cmd),
"TEARDOWN %s RTSP/1.0\r\n",
s->filename);
rtsp_send_cmd(s, cmd, reply, NULL);
rtsp_close_streams(rt);
url_close(rt->rtsp_hd);
return 0;
}
#ifdef CONFIG_RTSP_DEMUXER
AVInputFormat rtsp_demuxer = {
"rtsp",
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];
AVStream *st;
/* read the whole sdp file */
/* XXX: better loading */
content = av_malloc(SDP_MAX_SIZE);
size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
if (size <= 0) {
av_free(content);
return AVERROR_INVALIDDATA;
}
content[size] ='\0';
sdp_parse(s, content);
av_free(content);
/* open each RTP stream */
for(i=0;i<rt->nb_rtsp_streams;i++) {
rtsp_st = rt->rtsp_streams[i];
snprintf(url, sizeof(url), "rtp://%s:%d?localport=%d&ttl=%d",
inet_ntoa(rtsp_st->sdp_ip),
rtsp_st->sdp_port,
rtsp_st->sdp_port,
rtsp_st->sdp_ttl);
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
/* open the RTP context */
st = NULL;
if (rtsp_st->stream_index >= 0)
st = s->streams[rtsp_st->stream_index];
if (!st)
s->ctx_flags |= AVFMTCTX_NOHEADER;
rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
if (!rtsp_st->rtp_ctx) {
err = AVERROR(ENOMEM);
goto fail;
} else {
if(rtsp_st->dynamic_handler) {
rtsp_st->rtp_ctx->dynamic_protocol_context= rtsp_st->dynamic_protocol_context;
rtsp_st->rtp_ctx->parse_packet= rtsp_st->dynamic_handler->parse_packet;
}
}
}
return 0;
fail:
rtsp_close_streams(rt);
return err;
}
static int sdp_read_packet(AVFormatContext *s,
AVPacket *pkt)
{
return rtsp_read_packet(s, pkt);
}
static int sdp_read_close(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
rtsp_close_streams(rt);
return 0;
}
#ifdef CONFIG_SDP_DEMUXER
AVInputFormat sdp_demuxer = {
"sdp",
NULL_IF_CONFIG_SMALL("SDP"),
sizeof(RTSPState),
sdp_probe,
sdp_read_header,
sdp_read_packet,
sdp_read_close,
};
#endif
#ifdef CONFIG_REDIR_DEMUXER
/* dummy redirector format (used directly in av_open_input_file now) */
static int redir_probe(AVProbeData *pd)
{
const char *p;
p = pd->buf;
while (redir_isspace(*p))
p++;
if (av_strstart(p, "http://", NULL) ||
av_strstart(p, "rtsp://", NULL))
return AVPROBE_SCORE_MAX;
return 0;
}
static int redir_read_header(AVFormatContext *s, AVFormatParameters *ap)
{
char buf[4096], *q;
int c;
AVFormatContext *ic = NULL;
ByteIOContext *f = s->pb;
/* parse each URL and try to open it */
c = url_fgetc(f);
while (c != URL_EOF) {
/* skip spaces */
for(;;) {
if (!redir_isspace(c))
break;
c = url_fgetc(f);
}
if (c == URL_EOF)
break;
/* record url */
q = buf;
for(;;) {
if (c == URL_EOF || redir_isspace(c))
break;
if ((q - buf) < sizeof(buf) - 1)
*q++ = c;
c = url_fgetc(f);
}
*q = '\0';
//printf("URL='%s'\n", buf);
/* try to open the media file */
if (av_open_input_file(&ic, buf, NULL, 0, NULL) == 0)
break;
}
if (!ic)
return AVERROR(EIO);
*s = *ic;
url_fclose(f);
return 0;
}
AVInputFormat redir_demuxer = {
"redir",
NULL_IF_CONFIG_SMALL("Redirector format"),
0,
redir_probe,
redir_read_header,
NULL,
NULL,
};
#endif