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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 "avformat.h"
#include <unistd.h> /* for select() prototype */
#include <sys/time.h>
#include <netinet/in.h>
#include <sys/socket.h>
#include <arpa/inet.h>
#include "rtp_internal.h"
//#define DEBUG
//#define DEBUG_RTP_TCP
enum RTSPClientState {
RTSP_STATE_IDLE,
RTSP_STATE_PLAYING,
RTSP_STATE_PAUSED,
};
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;
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 */
int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_UDP);
FFRTSPCallback *ff_rtsp_callback = NULL;
static int rtsp_probe(AVProbeData *p)
{
if (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[] */
for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec->codec_type == AVRtpPayloadTypes[i].codec_type)){
codec->codec_id = AVRtpPayloadTypes[i].codec_id;
break;
}
}
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 (!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':
pstrcpy(s->title, sizeof(s->title), p);
break;
case 'i':
if (s->nb_streams == 0) {
pstrcpy(s->comment, sizeof(s->comment), p);
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(AVRtpPayloadTypes[rtsp_st->sdp_payload_type].enc_name, "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 */
pstrcpy(rtsp_st->control_url, sizeof(rtsp_st->control_url), s->filename);
break;
case 'a':
if (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 */
pstrcat(rtsp_st->control_url, sizeof(rtsp_st->control_url), "/");
pstrcat(rtsp_st->control_url, sizeof(rtsp_st->control_url), p);
} else {
pstrcpy(rtsp_st->control_url, sizeof(rtsp_st->control_url), p);
}
} else if (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 (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(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(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)
}
break;
}
}
static int sdp_parse(AVFormatContext *s, const char *content)
{
const char *p;
int letter;
char buf[1024], *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++;
get_word_sep(profile, sizeof(profile), "/;,", &p);
lower_transport[0] = '\0';
if (*p == '/') {
p++;
get_word_sep(lower_transport, sizeof(lower_transport),
";,", &p);
}
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 (stristart(p, "Session:", &p)) {
get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
} else if (stristart(p, "Content-Length:", &p)) {
reply->content_length = strtol(p, NULL, 10);
} else if (stristart(p, "Transport:", &p)) {
rtsp_parse_transport(reply, p);
} else if (stristart(p, "CSeq:", &p)) {
reply->seq = strtol(p, NULL, 10);
} else if (stristart(p, "Range:", &p)) {
rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
}
}
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 = (buf[1] << 8) | buf[2];
#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++;
pstrcpy(buf, sizeof(buf), cmd);
snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq);
pstrcat(buf, sizeof(buf), buf1);
if (rt->session_id[0] != '\0' && !strstr(cmd, "\nIf-Match:")) {
snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id);
pstrcat(buf, sizeof(buf), buf1);
}
pstrcat(buf, sizeof(buf), "\r\n");
#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);
pstrcat(rt->last_reply, sizeof(rt->last_reply), p);
pstrcat(rt->last_reply, sizeof(rt->last_reply), "\n");
}
line_count++;
}
if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
pstrcpy(rt->session_id, sizeof(rt->session_id), reply->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;
}
/* useful for modules: set RTSP callback function */
void rtsp_set_callback(FFRTSPCallback *rtsp_cb)
{
ff_rtsp_callback = rtsp_cb;
}
/* 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(rtsp_st);
}
av_free(rt->rtsp_streams);
}
static int rtsp_read_header(AVFormatContext *s,
AVFormatParameters *ap)
{
RTSPState *rt = s->priv_data;
char host[1024], path[1024], tcpname[1024], cmd[2048];
URLContext *rtsp_hd;
int port, i, j, ret, err;
RTSPHeader reply1, *reply = &reply1;
unsigned char *content = NULL;
RTSPStream *rtsp_st;
int protocol_mask;
AVStream *st;
/* 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;
/* open the tcp connexion */
snprintf(tcpname, sizeof(tcpname), "tcp://%s:%d", host, port);
if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0)
return AVERROR_IO;
rt->rtsp_hd = rtsp_hd;
rt->seq = 0;
/* describe the stream */
snprintf(cmd, sizeof(cmd),
"DESCRIBE %s RTSP/1.0\r\n"
"Accept: application/sdp\r\n",
s->filename);
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;
}
protocol_mask = rtsp_default_protocols;
/* 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];
/* compute available transports */
transport[0] = '\0';
/* RTP/UDP */
if (protocol_mask & (1 << 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://?localport=%d", j);
if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0) {
j += 2; /* we will use two port by rtp stream (rtp and rtcp) */
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);
if (transport[0] != '\0')
pstrcat(transport, sizeof(transport), ",");
snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1,
"RTP/AVP/UDP;unicast;client_port=%d-%d",
port, port + 1);
}
/* RTP/TCP */
else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_TCP)) {
if (transport[0] != '\0')
pstrcat(transport, sizeof(transport), ",");
snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1,
"RTP/AVP/TCP");
}
else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST)) {
if (transport[0] != '\0')
pstrcat(transport, sizeof(transport), ",");
snprintf(transport + strlen(transport),
sizeof(transport) - strlen(transport) - 1,
"RTP/AVP/UDP;multicast");
}
snprintf(cmd, sizeof(cmd),
"SETUP %s RTSP/1.0\r\n"
"Transport: %s\r\n",
rtsp_st->control_url, transport);
rtsp_send_cmd(s, cmd, reply, NULL);
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_mask & (1 << 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];
int ttl;
ttl = reply->transports[0].ttl;
if (!ttl)
ttl = 16;
snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d",
host,
reply->transports[0].server_port_min,
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_NOMEM;
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;
}
}
}
/* use callback if available to extend setup */
if (ff_rtsp_callback) {
if (ff_rtsp_callback(RTSP_ACTION_CLIENT_SETUP, rt->session_id,
NULL, 0, rt->last_reply) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
}
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 = (buf[1] << 8) | buf[2];
#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 -1;
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 (rtsp_st->rtp_ctx)
rtp_check_and_send_back_rr(rtsp_st->rtp_ctx, len);
break;
}
if (len < 0)
return AVERROR_IO;
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;
} else {
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;
} else {
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 = timestamp;
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);
if (ff_rtsp_callback) {
ff_rtsp_callback(RTSP_ACTION_CLIENT_TEARDOWN, rt->session_id,
NULL, 0, NULL);
}
rtsp_close_streams(rt);
url_close(rt->rtsp_hd);
return 0;
}
AVInputFormat rtsp_demuxer = {
"rtsp",
"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,
};
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 && 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?multicast=1&ttl=%d",
inet_ntoa(rtsp_st->sdp_ip),
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_NOMEM;
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",
"SDP",
sizeof(RTSPState),
sdp_probe,
sdp_read_header,
sdp_read_packet,
sdp_read_close,
};
#endif
/* 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 (strstart(p, "http://", NULL) ||
strstart(p, "rtsp://", NULL))
return AVPROBE_SCORE_MAX;
return 0;
}
/* called from utils.c */
int redir_open(AVFormatContext **ic_ptr, ByteIOContext *f)
{
char buf[4096], *q;
int c;
AVFormatContext *ic = NULL;
/* 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;
}
*ic_ptr = ic;
if (!ic)
return AVERROR_IO;
else
return 0;
}
AVInputFormat redir_demuxer = {
"redir",
"Redirector format",
0,
redir_probe,
NULL,
NULL,
NULL,
};
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