#include "rtpdec.h"
#include "rdt.h"
#include "rtp_asf.h"
+#include "rtp_vorbis.h"
//#define DEBUG
//#define DEBUG_RTP_TCP
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_LOWER_TRANSPORT_UDP);
#endif
}
#define SPACE_CHARS " \t\r\n"
-#define redir_isspace(c) strchr(SPACE_CHARS, c)
+/* we use memchr() instead of strchr() here because strchr() will return
+ * the terminating '\0' of SPACE_CHARS instead of NULL if c is '\0'. */
+#define redir_isspace(c) memchr(SPACE_CHARS, c, 4)
static void skip_spaces(const char **pp)
{
const char *p;
v = 1;
for(;;) {
skip_spaces(&p);
- if (p == '\0')
+ if (*p == '\0')
break;
c = toupper((unsigned char)*p++);
if (c >= '0' && c <= '9')
return len;
}
-static void sdp_parse_fmtp_config(AVCodecContext *codec, char *attr, char *value)
+static void sdp_parse_fmtp_config(AVCodecContext * codec, void *ctx,
+ char *attr, char *value)
{
switch (codec->codec_id) {
case CODEC_ID_MPEG4:
if (!strcmp(attr, "config")) {
/* decode the hexa encoded parameter */
int len = hex_to_data(NULL, value);
+ if (codec->extradata)
+ av_free(codec->extradata);
codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
if (!codec->extradata)
return;
hex_to_data(codec->extradata, value);
}
break;
+ case CODEC_ID_VORBIS:
+ ff_vorbis_parse_fmtp_config(codec, ctx, attr, value);
+ break;
default:
break;
}
int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
{
skip_spaces(p);
- if(**p)
- {
+ if(**p) {
get_word_sep(attr, attr_size, "=", p);
if (**p == '=')
(*p)++;
static void sdp_parse_fmtp(AVStream *st, const char *p)
{
char attr[256];
- char value[4096];
+ /* Vorbis setup headers can be up to 12KB and are sent base64
+ * encoded, giving a 12KB * (4/3) = 16KB FMTP line. */
+ char value[16384];
int i;
RTSPStream *rtsp_st = st->priv_data;
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);
+ sdp_parse_fmtp_config(codec, rtsp_st->dynamic_protocol_context,
+ attr, value);
/* Looking for a known attribute */
for (i = 0; attr_names[i].str; ++i) {
if (!strcasecmp(attr, attr_names[i].str)) {
* 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. */
- char buf[8192], *q;
+ * buffer is large.
+ *
+ * The Vorbis FMTP line can be up to 16KB - see sdp_parse_fmtp. */
+ char buf[16384], *q;
SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
memset(s1, 0, sizeof(SDPParseState));
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);
}
/* 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)) {
return 0;
}
-static void rtsp_send_cmd(AVFormatContext *s,
+static void rtsp_send_cmd_async (AVFormatContext *s,
const char *cmd, RTSPMessageHeader *reply,
unsigned char **content_ptr)
{
printf("Sending:\n%s--\n", buf);
#endif
url_write(rt->rtsp_hd, buf, strlen(buf));
+ rt->last_cmd_time = av_gettime();
+}
+
+static void rtsp_send_cmd (AVFormatContext *s,
+ const char *cmd, RTSPMessageHeader *reply,
+ unsigned char **content_ptr)
+{
+ rtsp_send_cmd_async(s, cmd, reply, content_ptr);
rtsp_read_reply(s, reply, content_ptr, 0);
}
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 */
goto fail;
}
+ if (reply->timeout > 0)
+ rt->timeout = reply->timeout;
+
if (rt->server_type == RTSP_SERVER_REAL)
rt->need_subscription = 1;
RTSPStream *rtsp_st;
int ret, len;
uint8_t buf[10 * RTP_MAX_PACKET_LENGTH];
+ RTSPMessageHeader reply1, *reply = &reply1;
+ char cmd[1024];
if (rt->server_type == RTSP_SERVER_REAL) {
int i;
- RTSPMessageHeader reply1, *reply = &reply1;
enum AVDiscard cache[MAX_STREAMS];
- char cmd[1024];
for (i = 0; i < s->nb_streams; i++)
cache[i] = s->streams[i]->discard;
/* more packets may follow, so we save the RTP context */
rt->cur_transport_priv = rtsp_st->transport_priv;
}
+
+ /* send dummy request to keep TCP connection alive */
+ if ((rt->server_type == RTSP_SERVER_WMS ||
+ rt->server_type == RTSP_SERVER_REAL) &&
+ (av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
+ if (rt->server_type == RTSP_SERVER_WMS) {
+ snprintf(cmd, sizeof(cmd) - 1,
+ "GET_PARAMETER %s RTSP/1.0\r\n",
+ s->filename);
+ rtsp_send_cmd_async(s, cmd, reply, NULL);
+ } else {
+ rtsp_send_cmd_async(s, "OPTIONS * RTSP/1.0\r\n",
+ reply, NULL);
+ }
+ }
+
return 0;
}
{
const char *p;
p = pd->buf;
- while (redir_isspace(*p))
- p++;
+ skip_spaces(&p);
if (av_strstart(p, "http://", NULL) ||
av_strstart(p, "rtsp://", NULL))
return AVPROBE_SCORE_MAX;
projects / frescor / ffmpeg.git / blobdiff