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ffmpeg/libavformat/rtsp.c

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/*
* 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 "libavutil/random_seed.h"
#include "avformat.h"
#include <sys/time.h>
#if HAVE_SYS_SELECT_H
#include <sys/select.h>
#endif
#include <strings.h>
#include "internal.h"
#include "network.h"
#include "os_support.h"
#include "http.h"
#include "rtsp.h"
#include "rtpdec.h"
#include "rdt.h"
#include "rtpdec_formats.h"
#include "rtpenc_chain.h"
//#define DEBUG
//#define DEBUG_RTP_TCP
/* Timeout values for socket select, in ms,
* and read_packet(), in seconds */
#define SELECT_TIMEOUT_MS 100
#define READ_PACKET_TIMEOUT_S 10
#define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / SELECT_TIMEOUT_MS
#define SDP_MAX_SIZE 16384
#define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
static void get_word_until_chars(char *buf, int buf_size,
const char *sep, const char **pp)
{
const char *p;
char *q;
p = *pp;
p += strspn(p, SPACE_CHARS);
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 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];
p += strspn(p, SPACE_CHARS);
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);
}
static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
{
struct addrinfo hints, *ai = NULL;
memset(&hints, 0, sizeof(hints));
hints.ai_flags = AI_NUMERICHOST;
if (getaddrinfo(buf, NULL, &hints, &ai))
return -1;
memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
freeaddrinfo(ai);
return 0;
}
#if CONFIG_RTPDEC
static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
RTSPStream *rtsp_st, AVCodecContext *codec)
{
if (!handler)
return;
codec->codec_id = handler->codec_id;
rtsp_st->dynamic_handler = handler;
if (handler->open)
rtsp_st->dynamic_protocol_context = handler->open();
}
/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
static int sdp_parse_rtpmap(AVFormatContext *s,
AVStream *st, RTSPStream *rtsp_st,
int payload_type, const char *p)
{
AVCodecContext *codec = st->codec;
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 =
ff_rtp_handler_find_by_name(buf, codec->codec_type);
init_rtp_handler(handler, rtsp_st, codec);
/* If no dynamic handler was found, check with the list of standard
* allocated types, if such a stream for some reason happens to
* use a private payload type. This isn't handled in rtpdec.c, since
* the format name from the rtpmap line never is passed into rtpdec. */
if (!rtsp_st->dynamic_handler)
codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
} 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 AVMEDIA_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;
av_set_pts_info(st, 32, 1, codec->sample_rate);
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 AVMEDIA_TYPE_VIDEO:
av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
if (i > 0)
av_set_pts_info(st, 32, 1, i);
break;
default:
break;
}
return 0;
}
/* parse the attribute line from the fmtp a line of an sdp response. 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)
{
*p += strspn(*p, SPACE_CHARS);
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;
}
typedef struct SDPParseState {
/* SDP only */
struct sockaddr_storage 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 AVMediaType codec_type;
int payload_type, i;
AVStream *st;
RTSPStream *rtsp_st;
struct sockaddr_storage 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") && strcmp(buf1, "IP6"))
return;
get_word_sep(buf1, sizeof(buf1), "/", &p);
if (get_sockaddr(buf1, &sdp_ip))
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_set2(&s->metadata, "title", p, 0);
break;
case 'i':
if (s->nb_streams == 0) {
av_metadata_set2(&s->metadata, "comment", p, 0);
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 = AVMEDIA_TYPE_AUDIO;
} else if (!strcmp(st_type, "video")) {
codec_type = AVMEDIA_TYPE_VIDEO;
} else if (!strcmp(st_type, "application")) {
codec_type = AVMEDIA_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) {
RTPDynamicProtocolHandler *handler;
/* if standard payload type, we can find the codec right now */
ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
st->codec->sample_rate > 0)
av_set_pts_info(st, 32, 1, st->codec->sample_rate);
/* Even static payload types may need a custom depacketizer */
handler = ff_rtp_handler_find_by_id(
rtsp_st->sdp_payload_type, st->codec->codec_type);
init_rtp_handler(handler, rtsp_st, st->codec);
}
}
/* 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 */
av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
NULL, NULL, 0, p);
if (proto[0] == '\0') {
/* relative control URL */
if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
av_strlcat(rtsp_st->control_url, "/",
sizeof(rtsp_st->control_url));
av_strlcat(rtsp_st->control_url, p,
sizeof(rtsp_st->control_url));
} else
av_strlcpy(rtsp_st->control_url, p,
sizeof(rtsp_st->control_url));
}
} else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
/* NOTE: rtpmap is only supported AFTER the 'm=' tag */
get_word(buf1, sizeof(buf1), &p);
payload_type = atoi(buf1);
st = s->streams[s->nb_streams - 1];
rtsp_st = st->priv_data;
sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
} else if (av_strstart(p, "fmtp:", &p) ||
av_strstart(p, "framesize:", &p)) {
/* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
// let dynamic protocol handlers have a stab at the line.
get_word(buf1, sizeof(buf1), &p);
payload_type = atoi(buf1);
for (i = 0; i < 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 (av_strstart(p, "SampleRate:integer;", &p) &&
s->nb_streams > 0) {
st = s->streams[s->nb_streams - 1];
st->codec->sample_rate = atoi(p);
} else {
if (rt->server_type == RTSP_SERVER_WMS)
ff_wms_parse_sdp_a_line(s, p);
if (s->nb_streams > 0) {
if (rt->server_type == RTSP_SERVER_REAL)
ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
rtsp_st = 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;
}
}
int ff_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 xiph_parse_sdp_line
* in rtpdec_xiph.c. */
char buf[16384], *q;
SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
memset(s1, 0, sizeof(SDPParseState));
p = content;
for (;;) {
p += strspn(p, SPACE_CHARS);
letter = *p;
if (letter == '\0')
break;
p++;
if (*p != '=')
goto next_line;
p++;
/* get the content */
q = buf;
while (*p != '\n' && *p != '\r' && *p != '\0') {
if ((q - buf) < sizeof(buf) - 1)
*q++ = *p;
p++;
}
*q = '\0';
sdp_parse_line(s, s1, letter, buf);
next_line:
while (*p != '\n' && *p != '\0')
p++;
if (*p == '\n')
p++;
}
return 0;
}
#endif /* CONFIG_RTPDEC */
/* close and free RTSP streams */
void ff_rtsp_close_streams(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
int i;
RTSPStream *rtsp_st;
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
if (rtsp_st) {
if (rtsp_st->transport_priv) {
if (s->oformat) {
AVFormatContext *rtpctx = rtsp_st->transport_priv;
av_write_trailer(rtpctx);
if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
uint8_t *ptr;
url_close_dyn_buf(rtpctx->pb, &ptr);
av_free(ptr);
} else {
url_fclose(rtpctx->pb);
}
av_metadata_free(&rtpctx->streams[0]->metadata);
av_metadata_free(&rtpctx->metadata);
av_free(rtpctx->streams[0]);
av_free(rtpctx);
} else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
ff_rdt_parse_close(rtsp_st->transport_priv);
else if (CONFIG_RTPDEC)
rtp_parse_close(rtsp_st->transport_priv);
}
if (rtsp_st->rtp_handle)
url_close(rtsp_st->rtp_handle);
if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
rtsp_st->dynamic_handler->close(
rtsp_st->dynamic_protocol_context);
}
}
av_free(rt->rtsp_streams);
if (rt->asf_ctx) {
av_close_input_stream (rt->asf_ctx);
rt->asf_ctx = NULL;
}
av_free(rt->recvbuf);
}
static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
{
RTSPState *rt = s->priv_data;
AVStream *st = NULL;
/* open the RTP context */
if (rtsp_st->stream_index >= 0)
st = s->streams[rtsp_st->stream_index];
if (!st)
s->ctx_flags |= AVFMTCTX_NOHEADER;
if (s->oformat && CONFIG_RTSP_MUXER) {
rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
rtsp_st->rtp_handle,
RTSP_TCP_MAX_PACKET_SIZE);
/* Ownership of rtp_handle is passed to the rtp mux context */
rtsp_st->rtp_handle = NULL;
} else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
rtsp_st->dynamic_protocol_context,
rtsp_st->dynamic_handler);
else if (CONFIG_RTPDEC)
rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
rtsp_st->sdp_payload_type,
(rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
if (!rtsp_st->transport_priv) {
return AVERROR(ENOMEM);
} else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
if (rtsp_st->dynamic_handler) {
rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
rtsp_st->dynamic_protocol_context,
rtsp_st->dynamic_handler);
}
}
return 0;
}
#if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
{
const char *p;
int v;
p = *pp;
p += strspn(p, SPACE_CHARS);
v = strtol(p, (char **)&p, 10);
if (*p == '-') {
p++;
*min_ptr = v;
v = strtol(p, (char **)&p, 10);
*max_ptr = v;
} else {
*min_ptr = v;
*max_ptr = v;
}
*pp = p;
}
/* XXX: only one transport specification is parsed */
static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
{
char transport_protocol[16];
char profile[16];
char lower_transport[16];
char parameter[16];
RTSPTransportField *th;
char buf[256];
reply->nb_transports = 0;
for (;;) {
p += strspn(p, SPACE_CHARS);
if (*p == '\0')
break;
th = &reply->transports[reply->nb_transports];
get_word_sep(transport_protocol, sizeof(transport_protocol),
"/", &p);
if (!strcasecmp (transport_protocol, "rtp")) {
get_word_sep(profile, sizeof(profile), "/;,", &p);
lower_transport[0] = '\0';
/* rtp/avp/<protocol> */
if (*p == '/') {
get_word_sep(lower_transport, sizeof(lower_transport),
";,", &p);
}
th->transport = RTSP_TRANSPORT_RTP;
} else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
!strcasecmp (transport_protocol, "x-real-rdt")) {
/* x-pn-tng/<protocol> */
get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
profile[0] = '\0';
th->transport = RTSP_TRANSPORT_RDT;
}
if (!strcasecmp(lower_transport, "TCP"))
th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
else
th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
if (*p == ';')
p++;
/* get each parameter */
while (*p != '\0' && *p != ',') {
get_word_sep(parameter, sizeof(parameter), "=;,", &p);
if (!strcmp(parameter, "port")) {
if (*p == '=') {
p++;
rtsp_parse_range(&th->port_min, &th->port_max, &p);
}
} else if (!strcmp(parameter, "client_port")) {
if (*p == '=') {
p++;
rtsp_parse_range(&th->client_port_min,
&th->client_port_max, &p);
}
} else if (!strcmp(parameter, "server_port")) {
if (*p == '=') {
p++;
rtsp_parse_range(&th->server_port_min,
&th->server_port_max, &p);
}
} else if (!strcmp(parameter, "interleaved")) {
if (*p == '=') {
p++;
rtsp_parse_range(&th->interleaved_min,
&th->interleaved_max, &p);
}
} else if (!strcmp(parameter, "multicast")) {
if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
} else if (!strcmp(parameter, "ttl")) {
if (*p == '=') {
p++;
th->ttl = strtol(p, (char **)&p, 10);
}
} else if (!strcmp(parameter, "destination")) {
if (*p == '=') {
p++;
get_word_sep(buf, sizeof(buf), ";,", &p);
get_sockaddr(buf, &th->destination);
}
} else if (!strcmp(parameter, "source")) {
if (*p == '=') {
p++;
get_word_sep(buf, sizeof(buf), ";,", &p);
av_strlcpy(th->source, buf, sizeof(th->source));
}
}
while (*p != ';' && *p != '\0' && *p != ',')
p++;
if (*p == ';')
p++;
}
if (*p == ',')
p++;
reply->nb_transports++;
}
}
static void handle_rtp_info(RTSPState *rt, const char *url,
uint32_t seq, uint32_t rtptime)
{
int i;
if (!rtptime || !url[0])
return;
if (rt->transport != RTSP_TRANSPORT_RTP)
return;
for (i = 0; i < rt->nb_rtsp_streams; i++) {
RTSPStream *rtsp_st = rt->rtsp_streams[i];
RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
if (!rtpctx)
continue;
if (!strcmp(rtsp_st->control_url, url)) {
rtpctx->base_timestamp = rtptime;
break;
}
}
}
static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
{
int read = 0;
char key[20], value[1024], url[1024] = "";
uint32_t seq = 0, rtptime = 0;
for (;;) {
p += strspn(p, SPACE_CHARS);
if (!*p)
break;
get_word_sep(key, sizeof(key), "=", &p);
if (*p != '=')
break;
p++;
get_word_sep(value, sizeof(value), ";, ", &p);
read++;
if (!strcmp(key, "url"))
av_strlcpy(url, value, sizeof(url));
else if (!strcmp(key, "seq"))
seq = strtol(value, NULL, 10);
else if (!strcmp(key, "rtptime"))
rtptime = strtol(value, NULL, 10);
if (*p == ',') {
handle_rtp_info(rt, url, seq, rtptime);
url[0] = '\0';
seq = rtptime = 0;
read = 0;
}
if (*p)
p++;
}
if (read > 0)
handle_rtp_info(rt, url, seq, rtptime);
}
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
RTSPState *rt, const char *method)
{
const char *p;
/* NOTE: we do case independent match for broken servers */
p = buf;
if (av_stristart(p, "Session:", &p)) {
int t;
get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
if (av_stristart(p, ";timeout=", &p) &&
(t = strtol(p, NULL, 10)) > 0) {
reply->timeout = t;
}
} else if (av_stristart(p, "Content-Length:", &p)) {
reply->content_length = strtol(p, NULL, 10);
} else if (av_stristart(p, "Transport:", &p)) {
rtsp_parse_transport(reply, p);
} else if (av_stristart(p, "CSeq:", &p)) {
reply->seq = strtol(p, NULL, 10);
} else if (av_stristart(p, "Range:", &p)) {
rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
} else if (av_stristart(p, "RealChallenge1:", &p)) {
p += strspn(p, SPACE_CHARS);
av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
} else if (av_stristart(p, "Server:", &p)) {
p += strspn(p, SPACE_CHARS);
av_strlcpy(reply->server, p, sizeof(reply->server));
} else if (av_stristart(p, "Notice:", &p) ||
av_stristart(p, "X-Notice:", &p)) {
reply->notice = strtol(p, NULL, 10);
} else if (av_stristart(p, "Location:", &p)) {
p += strspn(p, SPACE_CHARS);
av_strlcpy(reply->location, p , sizeof(reply->location));
} else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
p += strspn(p, SPACE_CHARS);
ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
} else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
p += strspn(p, SPACE_CHARS);
ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
} else if (av_stristart(p, "Content-Base:", &p) && rt) {
p += strspn(p, SPACE_CHARS);
if (method && !strcmp(method, "DESCRIBE"))
av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
} else if (av_stristart(p, "RTP-Info:", &p) && rt) {
p += strspn(p, SPACE_CHARS);
if (method && !strcmp(method, "PLAY"))
rtsp_parse_rtp_info(rt, p);
}
}
/* skip a RTP/TCP interleaved packet */
void ff_rtsp_skip_packet(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
int ret, len, len1;
uint8_t buf[1024];
ret = url_read_complete(rt->rtsp_hd, buf, 3);
if (ret != 3)
return;
len = AV_RB16(buf + 1);
dprintf(s, "skipping RTP packet len=%d\n", len);
/* skip payload */
while (len > 0) {
len1 = len;
if (len1 > sizeof(buf))
len1 = sizeof(buf);
ret = url_read_complete(rt->rtsp_hd, buf, len1);
if (ret != len1)
return;
len -= len1;
}
}
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
unsigned char **content_ptr,
int return_on_interleaved_data, const char *method)
{
RTSPState *rt = s->priv_data;
char buf[4096], buf1[1024], *q;
unsigned char ch;
const char *p;
int ret, content_length, line_count = 0;
unsigned char *content = NULL;
memset(reply, 0, sizeof(*reply));
/* parse reply (XXX: use buffers) */
rt->last_reply[0] = '\0';
for (;;) {
q = buf;
for (;;) {
ret = url_read_complete(rt->rtsp_hd, &ch, 1);
#ifdef DEBUG_RTP_TCP
dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
#endif
if (ret != 1)
return AVERROR_EOF;
if (ch == '\n')
break;
if (ch == '$') {
/* XXX: only parse it if first char on line ? */
if (return_on_interleaved_data) {
return 1;
} else
ff_rtsp_skip_packet(s);
} else if (ch != '\r') {
if ((q - buf) < sizeof(buf) - 1)
*q++ = ch;
}
}
*q = '\0';
dprintf(s, "line='%s'\n", buf);
/* test if last line */
if (buf[0] == '\0')
break;
p = buf;
if (line_count == 0) {
/* get reply code */
get_word(buf1, sizeof(buf1), &p);
get_word(buf1, sizeof(buf1), &p);
reply->status_code = atoi(buf1);
av_strlcpy(reply->reason, p, sizeof(reply->reason));
} else {
ff_rtsp_parse_line(reply, p, rt, method);
av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
}
line_count++;
}
if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
content_length = reply->content_length;
if (content_length > 0) {
/* leave some room for a trailing '\0' (useful for simple parsing) */
content = av_malloc(content_length + 1);
(void)url_read_complete(rt->rtsp_hd, content, content_length);
content[content_length] = '\0';
}
if (content_ptr)
*content_ptr = content;
else
av_free(content);
if (rt->seq != reply->seq) {
av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
rt->seq, reply->seq);
}
/* EOS */
if (reply->notice == 2101 /* End-of-Stream Reached */ ||
reply->notice == 2104 /* Start-of-Stream Reached */ ||
reply->notice == 2306 /* Continuous Feed Terminated */) {
rt->state = RTSP_STATE_IDLE;
} else if (reply->notice >= 4400 && reply->notice < 5500) {
return AVERROR(EIO); /* data or server error */
} else if (reply->notice == 2401 /* Ticket Expired */ ||
(reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
return AVERROR(EPERM);
return 0;
}
int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
const char *method, const char *url,
const char *headers,
const unsigned char *send_content,
int send_content_length)
{
RTSPState *rt = s->priv_data;
char buf[4096], *out_buf;
char base64buf[AV_BASE64_SIZE(sizeof(buf))];
/* Add in RTSP headers */
out_buf = buf;
rt->seq++;
snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
if (headers)
av_strlcat(buf, headers, sizeof(buf));
av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
if (rt->session_id[0] != '\0' && (!headers ||
!strstr(headers, "\nIf-Match:"))) {
av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
}
if (rt->auth[0]) {
char *str = ff_http_auth_create_response(&rt->auth_state,
rt->auth, url, method);
if (str)
av_strlcat(buf, str, sizeof(buf));
av_free(str);
}
if (send_content_length > 0 && send_content)
av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
av_strlcat(buf, "\r\n", sizeof(buf));
/* base64 encode rtsp if tunneling */
if (rt->control_transport == RTSP_MODE_TUNNEL) {
av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
out_buf = base64buf;
}
dprintf(s, "Sending:\n%s--\n", buf);
url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
if (send_content_length > 0 && send_content) {
if (rt->control_transport == RTSP_MODE_TUNNEL) {
av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
"with content data not supported\n");
return AVERROR_PATCHWELCOME;
}
url_write(rt->rtsp_hd_out, send_content, send_content_length);
}
rt->last_cmd_time = av_gettime();
return 0;
}
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
const char *url, const char *headers)
{
return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
}
int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
const char *headers, RTSPMessageHeader *reply,
unsigned char **content_ptr)
{
return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
content_ptr, NULL, 0);
}
int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
const char *method, const char *url,
const char *header,
RTSPMessageHeader *reply,
unsigned char **content_ptr,
const unsigned char *send_content,
int send_content_length)
{
RTSPState *rt = s->priv_data;
HTTPAuthType cur_auth_type;
int ret;
retry:
cur_auth_type = rt->auth_state.auth_type;
if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
send_content,
send_content_length)))
return ret;
if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
return ret;
if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
rt->auth_state.auth_type != HTTP_AUTH_NONE)
goto retry;
if (reply->status_code > 400){
av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
method,
reply->status_code,
reply->reason);
av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
}
return 0;
}
/**
* @return 0 on success, <0 on error, 1 if protocol is unavailable.
*/
static int make_setup_request(AVFormatContext *s, const char *host, int port,
int lower_transport, const char *real_challenge)
{
RTSPState *rt = s->priv_data;
int rtx, j, i, err, interleave = 0;
RTSPStream *rtsp_st;
RTSPMessageHeader reply1, *reply = &reply1;
char cmd[2048];
const char *trans_pref;
if (rt->transport == RTSP_TRANSPORT_RDT)
trans_pref = "x-pn-tng";
else
trans_pref = "RTP/AVP";
/* default timeout: 1 minute */
rt->timeout = 60;
/* for each stream, make the setup request */
/* XXX: we assume the same server is used for the control of each
* RTSP stream */
for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
char transport[2048];
/**
* WMS serves all UDP data over a single connection, the RTX, which
* isn't necessarily the first in the SDP but has to be the first
* to be set up, else the second/third SETUP will fail with a 461.
*/
if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
rt->server_type == RTSP_SERVER_WMS) {
if (i == 0) {
/* rtx first */
for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
int len = strlen(rt->rtsp_streams[rtx]->control_url);
if (len >= 4 &&
!strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
"/rtx"))
break;
}
if (rtx == rt->nb_rtsp_streams)
return -1; /* no RTX found */
rtsp_st = rt->rtsp_streams[rtx];
} else
rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
} else
rtsp_st = rt->rtsp_streams[i];
/* RTP/UDP */
if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
char buf[256];
if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
port = reply->transports[0].client_port_min;
goto have_port;
}
/* first try in specified port range */
if (RTSP_RTP_PORT_MIN != 0) {
while (j <= RTSP_RTP_PORT_MAX) {
ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
"?localport=%d", j);
/* we will use two ports per rtp stream (rtp and rtcp) */
j += 2;
if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
goto rtp_opened;
}
}
#if 0
/* then try on any port */
if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
#else
av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
err = AVERROR(EIO);
goto fail;
#endif
rtp_opened:
port = rtp_get_local_rtp_port(rtsp_st->rtp_handle);
have_port:
snprintf(transport, sizeof(transport) - 1,
"%s/UDP;", trans_pref);
if (rt->server_type != RTSP_SERVER_REAL)
av_strlcat(transport, "unicast;", sizeof(transport));
av_strlcatf(transport, sizeof(transport),
"client_port=%d", port);
if (rt->transport == RTSP_TRANSPORT_RTP &&
!(rt->server_type == RTSP_SERVER_WMS && i > 0))
av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
}
/* RTP/TCP */
else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
/** For WMS streams, the application streams are only used for
* UDP. When trying to set it up for TCP streams, the server
* will return an error. Therefore, we skip those streams. */
if (rt->server_type == RTSP_SERVER_WMS &&
s->streams[rtsp_st->stream_index]->codec->codec_type ==
AVMEDIA_TYPE_DATA)
continue;
snprintf(transport, sizeof(transport) - 1,
"%s/TCP;", trans_pref);
if (rt->server_type == RTSP_SERVER_WMS)
av_strlcat(transport, "unicast;", sizeof(transport));
av_strlcatf(transport, sizeof(transport),
"interleaved=%d-%d",
interleave, interleave + 1);
interleave += 2;
}
else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
snprintf(transport, sizeof(transport) - 1,
"%s/UDP;multicast", trans_pref);
}
if (s->oformat) {
av_strlcat(transport, ";mode=receive", sizeof(transport));
} else if (rt->server_type == RTSP_SERVER_REAL ||
rt->server_type == RTSP_SERVER_WMS)
av_strlcat(transport, ";mode=play", sizeof(transport));
snprintf(cmd, sizeof(cmd),
"Transport: %s\r\n",
transport);
if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
char real_res[41], real_csum[9];
ff_rdt_calc_response_and_checksum(real_res, real_csum,
real_challenge);
av_strlcatf(cmd, sizeof(cmd),
"If-Match: %s\r\n"
"RealChallenge2: %s, sd=%s\r\n",
rt->session_id, real_res, real_csum);
}
ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
err = 1;
goto fail;
} else if (reply->status_code != RTSP_STATUS_OK ||
reply->nb_transports != 1) {
err = AVERROR_INVALIDDATA;
goto fail;
}
/* XXX: same protocol for all streams is required */
if (i > 0) {
if (reply->transports[0].lower_transport != rt->lower_transport ||
reply->transports[0].transport != rt->transport) {
err = AVERROR_INVALIDDATA;
goto fail;
}
} else {
rt->lower_transport = reply->transports[0].lower_transport;
rt->transport = reply->transports[0].transport;
}
/* Fail if the server responded with another lower transport mode
* than what we requested. */
if (reply->transports[0].lower_transport != lower_transport) {
av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
err = AVERROR_INVALIDDATA;
goto fail;
}
switch(reply->transports[0].lower_transport) {
case RTSP_LOWER_TRANSPORT_TCP:
rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
break;
case RTSP_LOWER_TRANSPORT_UDP: {
char url[1024], options[30] = "";
if (rt->filter_source)
av_strlcpy(options, "?connect=1", sizeof(options));
/* Use source address if specified */
if (reply->transports[0].source[0]) {
ff_url_join(url, sizeof(url), "rtp", NULL,
reply->transports[0].source,
reply->transports[0].server_port_min, options);
} else {
ff_url_join(url, sizeof(url), "rtp", NULL, host,
reply->transports[0].server_port_min, options);
}
if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
/* Try to initialize the connection state in a
* potential NAT router by sending dummy packets.
* RTP/RTCP dummy packets are used for RDT, too.
*/
if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
CONFIG_RTPDEC)
rtp_send_punch_packets(rtsp_st->rtp_handle);
break;
}
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
char url[1024], namebuf[50];
struct sockaddr_storage addr;
int port, ttl;
if (reply->transports[0].destination.ss_family) {
addr = reply->transports[0].destination;
port = reply->transports[0].port_min;
ttl = reply->transports[0].ttl;
} else {
addr = rtsp_st->sdp_ip;
port = rtsp_st->sdp_port;
ttl = rtsp_st->sdp_ttl;
}
getnameinfo((struct sockaddr*) &addr, sizeof(addr),
namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
port, "?ttl=%d", ttl);
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
break;
}
}
if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
goto fail;
}
if (reply->timeout > 0)
rt->timeout = reply->timeout;
if (rt->server_type == RTSP_SERVER_REAL)
rt->need_subscription = 1;
return 0;
fail:
for (i = 0; i < rt->nb_rtsp_streams; i++) {
if (rt->rtsp_streams[i]->rtp_handle) {
url_close(rt->rtsp_streams[i]->rtp_handle);
rt->rtsp_streams[i]->rtp_handle = NULL;
}
}
return err;
}
void ff_rtsp_close_connections(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
url_close(rt->rtsp_hd);
rt->rtsp_hd = rt->rtsp_hd_out = NULL;
}
int ff_rtsp_connect(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
char *option_list, *option, *filename;
int port, err, tcp_fd;
RTSPMessageHeader reply1 = {0}, *reply = &reply1;
int lower_transport_mask = 0;
char real_challenge[64];
struct sockaddr_storage peer;
socklen_t peer_len = sizeof(peer);
if (!ff_network_init())
return AVERROR(EIO);
redirect:
rt->control_transport = RTSP_MODE_PLAIN;
/* extract hostname and port */
av_url_split(NULL, 0, auth, sizeof(auth),
host, sizeof(host), &port, path, sizeof(path), s->filename);
if (*auth) {
av_strlcpy(rt->auth, auth, sizeof(rt->auth));
}
if (port < 0)
port = RTSP_DEFAULT_PORT;
/* search for options */
option_list = strrchr(path, '?');
if (option_list) {
/* Strip out the RTSP specific options, write out the rest of
* the options back into the same string. */
filename = option_list;
while (option_list) {
/* move the option pointer */
option = ++option_list;
option_list = strchr(option_list, '&');
if (option_list)
*option_list = 0;
/* handle the options */
if (!strcmp(option, "udp")) {
lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
} else if (!strcmp(option, "multicast")) {
lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
} else if (!strcmp(option, "tcp")) {
lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
} else if(!strcmp(option, "http")) {
lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
rt->control_transport = RTSP_MODE_TUNNEL;
} else if (!strcmp(option, "filter_src")) {
rt->filter_source = 1;
} else {
/* Write options back into the buffer, using memmove instead
* of strcpy since the strings may overlap. */
int len = strlen(option);
memmove(++filename, option, len);
filename += len;
if (option_list) *filename = '&';
}
}
*filename = 0;
}
if (!lower_transport_mask)
lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
if (s->oformat) {
/* Only UDP or TCP - UDP multicast isn't supported. */
lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
(1 << RTSP_LOWER_TRANSPORT_TCP);
if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
"only UDP and TCP are supported for output.\n");
err = AVERROR(EINVAL);
goto fail;
}
}
/* Construct the URI used in request; this is similar to s->filename,
* but with authentication credentials removed and RTSP specific options
* stripped out. */
ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
host, port, "%s", path);
if (rt->control_transport == RTSP_MODE_TUNNEL) {
/* set up initial handshake for tunneling */
char httpname[1024];
char sessioncookie[17];
char headers[1024];
ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
av_get_random_seed(), av_get_random_seed());
/* GET requests */
if (url_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) {
err = AVERROR(EIO);
goto fail;
}
/* generate GET headers */
snprintf(headers, sizeof(headers),
"x-sessioncookie: %s\r\n"
"Accept: application/x-rtsp-tunnelled\r\n"
"Pragma: no-cache\r\n"
"Cache-Control: no-cache\r\n",
sessioncookie);
ff_http_set_headers(rt->rtsp_hd, headers);
/* complete the connection */
if (url_connect(rt->rtsp_hd)) {
err = AVERROR(EIO);
goto fail;
}
/* POST requests */
if (url_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
err = AVERROR(EIO);
goto fail;
}
/* generate POST headers */
snprintf(headers, sizeof(headers),
"x-sessioncookie: %s\r\n"
"Content-Type: application/x-rtsp-tunnelled\r\n"
"Pragma: no-cache\r\n"
"Cache-Control: no-cache\r\n"
"Content-Length: 32767\r\n"
"Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
sessioncookie);
ff_http_set_headers(rt->rtsp_hd_out, headers);
ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
/* Initialize the authentication state for the POST session. The HTTP
* protocol implementation doesn't properly handle multi-pass
* authentication for POST requests, since it would require one of
* the following:
* - implementing Expect: 100-continue, which many HTTP servers
* don't support anyway, even less the RTSP servers that do HTTP
* tunneling
* - sending the whole POST data until getting a 401 reply specifying
* what authentication method to use, then resending all that data
* - waiting for potential 401 replies directly after sending the
* POST header (waiting for some unspecified time)
* Therefore, we copy the full auth state, which works for both basic
* and digest. (For digest, we would have to synchronize the nonce
* count variable between the two sessions, if we'd do more requests
* with the original session, though.)
*/
ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
/* complete the connection */
if (url_connect(rt->rtsp_hd_out)) {
err = AVERROR(EIO);
goto fail;
}
} else {
/* open the tcp connection */
ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
err = AVERROR(EIO);
goto fail;
}
rt->rtsp_hd_out = rt->rtsp_hd;
}
rt->seq = 0;
tcp_fd = url_get_file_handle(rt->rtsp_hd);
if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
NULL, 0, NI_NUMERICHOST);
}
/* request options supported by the server; this also detects server
* type */
for (rt->server_type = RTSP_SERVER_RTP;;) {
cmd[0] = 0;
if (rt->server_type == RTSP_SERVER_REAL)
av_strlcat(cmd,
/**
* The following entries are required for proper
* streaming from a Realmedia server. They are
* interdependent in some way although we currently
* don't quite understand how. Values were copied
* from mplayer SVN r23589.
* @param CompanyID is a 16-byte ID in base64
* @param ClientChallenge is a 16-byte ID in hex
*/
"ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
"PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
"CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
"GUID: 00000000-0000-0000-0000-000000000000\r\n",
sizeof(cmd));
ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
err = AVERROR_INVALIDDATA;
goto fail;
}
/* detect server type if not standard-compliant RTP */
if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
rt->server_type = RTSP_SERVER_REAL;
continue;
} else if (!strncasecmp(reply->server, "WMServer/", 9)) {
rt->server_type = RTSP_SERVER_WMS;
} else if (rt->server_type == RTSP_SERVER_REAL)
strcpy(real_challenge, reply->real_challenge);
break;
}
if (s->iformat && CONFIG_RTSP_DEMUXER)
err = ff_rtsp_setup_input_streams(s, reply);
else if (CONFIG_RTSP_MUXER)
err = ff_rtsp_setup_output_streams(s, host);
if (err)
goto fail;
do {
int lower_transport = ff_log2_tab[lower_transport_mask &
~(lower_transport_mask - 1)];
err = make_setup_request(s, host, port, lower_transport,
rt->server_type == RTSP_SERVER_REAL ?
real_challenge : NULL);
if (err < 0)
goto fail;
lower_transport_mask &= ~(1 << lower_transport);
if (lower_transport_mask == 0 && err == 1) {
err = FF_NETERROR(EPROTONOSUPPORT);
goto fail;
}
} while (err);
rt->state = RTSP_STATE_IDLE;
rt->seek_timestamp = 0; /* default is to start stream at position zero */
return 0;
fail:
ff_rtsp_close_streams(s);
ff_rtsp_close_connections(s);
if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
av_strlcpy(s->filename, reply->location, sizeof(s->filename));
av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
reply->status_code,
s->filename);
goto redirect;
}
ff_network_close();
return err;
}
#endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
#if CONFIG_RTPDEC
static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
uint8_t *buf, int buf_size, int64_t wait_end)
{
RTSPState *rt = s->priv_data;
RTSPStream *rtsp_st;
fd_set rfds;
int fd, fd_rtcp, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
struct timeval tv;
for (;;) {
if (url_interrupt_cb())
return AVERROR(EINTR);
if (wait_end && wait_end - av_gettime() < 0)
return AVERROR(EAGAIN);
FD_ZERO(&rfds);
if (rt->rtsp_hd) {
tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
FD_SET(tcp_fd, &rfds);
} else {
fd_max = 0;
tcp_fd = -1;
}
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
if (rtsp_st->rtp_handle) {
fd = url_get_file_handle(rtsp_st->rtp_handle);
fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
if (FFMAX(fd, fd_rtcp) > fd_max)
fd_max = FFMAX(fd, fd_rtcp);
FD_SET(fd, &rfds);
FD_SET(fd_rtcp, &rfds);
}
}
tv.tv_sec = 0;
tv.tv_usec = SELECT_TIMEOUT_MS * 1000;
n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
if (n > 0) {
timeout_cnt = 0;
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
if (rtsp_st->rtp_handle) {
fd = url_get_file_handle(rtsp_st->rtp_handle);
fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
if (FD_ISSET(fd_rtcp, &rfds) || FD_ISSET(fd, &rfds)) {
ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
if (ret > 0) {
*prtsp_st = rtsp_st;
return ret;
}
}
}
}
#if CONFIG_RTSP_DEMUXER
if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
RTSPMessageHeader reply;
ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
if (ret < 0)
return ret;
/* XXX: parse message */
if (rt->state != RTSP_STATE_STREAMING)
return 0;
}
#endif
} else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
return FF_NETERROR(ETIMEDOUT);
} else if (n < 0 && errno != EINTR)
return AVERROR(errno);
}
}
int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
{
RTSPState *rt = s->priv_data;
int ret, len;
RTSPStream *rtsp_st, *first_queue_st = NULL;
int64_t wait_end = 0;
if (rt->nb_byes == rt->nb_rtsp_streams)
return AVERROR_EOF;
/* get next frames from the same RTP packet */
if (rt->cur_transport_priv) {
if (rt->transport == RTSP_TRANSPORT_RDT) {
ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
} else
ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
if (ret == 0) {
rt->cur_transport_priv = NULL;
return 0;
} else if (ret == 1) {
return 0;
} else
rt->cur_transport_priv = NULL;
}
if (rt->transport == RTSP_TRANSPORT_RTP) {
int i;
int64_t first_queue_time = 0;
for (i = 0; i < rt->nb_rtsp_streams; i++) {
RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
int64_t queue_time;
if (!rtpctx)
continue;
queue_time = ff_rtp_queued_packet_time(rtpctx);
if (queue_time && (queue_time - first_queue_time < 0 ||
!first_queue_time)) {
first_queue_time = queue_time;
first_queue_st = rt->rtsp_streams[i];
}
}
if (first_queue_time)
wait_end = first_queue_time + s->max_delay;
}
/* read next RTP packet */
redo:
if (!rt->recvbuf) {
rt->recvbuf = av_malloc(RECVBUF_SIZE);
if (!rt->recvbuf)
return AVERROR(ENOMEM);
}
switch(rt->lower_transport) {
default:
#if CONFIG_RTSP_DEMUXER
case RTSP_LOWER_TRANSPORT_TCP:
len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
break;
#endif
case RTSP_LOWER_TRANSPORT_UDP:
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
break;
}
if (len == AVERROR(EAGAIN) && first_queue_st &&
rt->transport == RTSP_TRANSPORT_RTP) {
rtsp_st = first_queue_st;
ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
goto end;
}
if (len < 0)
return len;
if (len == 0)
return AVERROR_EOF;
if (rt->transport == RTSP_TRANSPORT_RDT) {
ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
} else {
ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
if (ret < 0) {
/* Either bad packet, or a RTCP packet. Check if the
* first_rtcp_ntp_time field was initialized. */
RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
/* first_rtcp_ntp_time has been initialized for this stream,
* copy the same value to all other uninitialized streams,
* in order to map their timestamp origin to the same ntp time
* as this one. */
int i;
AVStream *st = NULL;
if (rtsp_st->stream_index >= 0)
st = s->streams[rtsp_st->stream_index];
for (i = 0; i < rt->nb_rtsp_streams; i++) {
RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
AVStream *st2 = NULL;
if (rt->rtsp_streams[i]->stream_index >= 0)
st2 = s->streams[rt->rtsp_streams[i]->stream_index];
if (rtpctx2 && st && st2 &&
rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
rtpctx2->rtcp_ts_offset = av_rescale_q(
rtpctx->rtcp_ts_offset, st->time_base,
st2->time_base);
}
}
}
if (ret == -RTCP_BYE) {
rt->nb_byes++;
av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
if (rt->nb_byes == rt->nb_rtsp_streams)
return AVERROR_EOF;
}
}
}
end:
if (ret < 0)
goto redo;
if (ret == 1)
/* more packets may follow, so we save the RTP context */
rt->cur_transport_priv = rtsp_st->transport_priv;
return ret;
}
#endif /* CONFIG_RTPDEC */
#if CONFIG_SDP_DEMUXER
static int sdp_probe(AVProbeData *p1)
{
const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
/* we look for a line beginning "c=IN IP" */
while (p < p_end && *p != '\0') {
if (p + sizeof("c=IN IP") - 1 < p_end &&
av_strstart(p, "c=IN IP", NULL))
return AVPROBE_SCORE_MAX / 2;
while (p < p_end - 1 && *p != '\n') p++;
if (++p >= p_end)
break;
if (*p == '\r')
p++;
}
return 0;
}
static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
{
RTSPState *rt = s->priv_data;
RTSPStream *rtsp_st;
int size, i, err;
char *content;
char url[1024];
if (!ff_network_init())
return AVERROR(EIO);
/* read the whole sdp file */
/* XXX: better loading */
content = av_malloc(SDP_MAX_SIZE);
size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
if (size <= 0) {
av_free(content);
return AVERROR_INVALIDDATA;
}
content[size] ='\0';
ff_sdp_parse(s, content);
av_free(content);
/* open each RTP stream */
for (i = 0; i < rt->nb_rtsp_streams; i++) {
char namebuf[50];
rtsp_st = rt->rtsp_streams[i];
getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
ff_url_join(url, sizeof(url), "rtp", NULL,
namebuf, rtsp_st->sdp_port,
"?localport=%d&ttl=%d", rtsp_st->sdp_port,
rtsp_st->sdp_ttl);
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
goto fail;
}
return 0;
fail:
ff_rtsp_close_streams(s);
ff_network_close();
return err;
}
static int sdp_read_close(AVFormatContext *s)
{
ff_rtsp_close_streams(s);
ff_network_close();
return 0;
}
AVInputFormat sdp_demuxer = {
"sdp",
NULL_IF_CONFIG_SMALL("SDP"),
sizeof(RTSPState),
sdp_probe,
sdp_read_header,
ff_rtsp_fetch_packet,
sdp_read_close,
};
#endif /* CONFIG_SDP_DEMUXER */
#if CONFIG_RTP_DEMUXER
static int rtp_probe(AVProbeData *p)
{
if (av_strstart(p->filename, "rtp:", NULL))
return AVPROBE_SCORE_MAX;
return 0;
}
static int rtp_read_header(AVFormatContext *s,
AVFormatParameters *ap)
{
uint8_t recvbuf[1500];
char host[500], sdp[500];
int ret, port;
URLContext* in = NULL;
int payload_type;
AVCodecContext codec;
struct sockaddr_storage addr;
ByteIOContext pb;
socklen_t addrlen = sizeof(addr);
if (!ff_network_init())
return AVERROR(EIO);
ret = url_open(&in, s->filename, URL_RDONLY);
if (ret)
goto fail;
while (1) {
ret = url_read(in, recvbuf, sizeof(recvbuf));
if (ret == AVERROR(EAGAIN))
continue;
if (ret < 0)
goto fail;
if (ret < 12) {
av_log(s, AV_LOG_WARNING, "Received too short packet\n");
continue;
}
if ((recvbuf[0] & 0xc0) != 0x80) {
av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
"received\n");
continue;
}
payload_type = recvbuf[1] & 0x7f;
break;
}
getsockname(url_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
url_close(in);
in = NULL;
memset(&codec, 0, sizeof(codec));
if (ff_rtp_get_codec_info(&codec, payload_type)) {
av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
"without an SDP file describing it\n",
payload_type);
goto fail;
}
if (codec.codec_type != AVMEDIA_TYPE_DATA) {
av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
"properly you need an SDP file "
"describing it\n");
}
av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
NULL, 0, s->filename);
snprintf(sdp, sizeof(sdp),
"v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
addr.ss_family == AF_INET ? 4 : 6, host,
codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
port, payload_type);
av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
init_put_byte(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
s->pb = &pb;
/* sdp_read_header initializes this again */
ff_network_close();
ret = sdp_read_header(s, ap);
s->pb = NULL;
return ret;
fail:
if (in)
url_close(in);
ff_network_close();
return ret;
}
AVInputFormat rtp_demuxer = {
"rtp",
NULL_IF_CONFIG_SMALL("RTP input format"),
sizeof(RTSPState),
rtp_probe,
rtp_read_header,
ff_rtsp_fetch_packet,
sdp_read_close,
.flags = AVFMT_NOFILE,
};
#endif /* CONFIG_RTP_DEMUXER */
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