#include "libavutil/base64.h"
#include "libavutil/avstring.h"
#include "libavutil/intreadwrite.h"
+#include "libavutil/random_seed.h"
#include "avformat.h"
#include <sys/time.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_asf.h"
-#include "rtpdec_vorbis.h"
+#include "rtpdec_formats.h"
//#define DEBUG
//#define DEBUG_RTP_TCP
int rtsp_default_protocols = (1 << RTSP_LOWER_TRANSPORT_UDP);
#endif
-#define SPACE_CHARS " \t\r\n"
-/* we use memchr() instead of strchr() here because strchr() will return
- * the terminating '\0' of SPACE_CHARS instead of NULL if c is '\0'. */
-#define redir_isspace(c) memchr(SPACE_CHARS, c, 4)
-static void skip_spaces(const char **pp)
-{
- const char *p;
- p = *pp;
- while (redir_isspace(*p))
- p++;
- *pp = p;
-}
+/* 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
static void get_word_until_chars(char *buf, int buf_size,
const char *sep, const char **pp)
char *q;
p = *pp;
- skip_spaces(&p);
+ p += strspn(p, SPACE_CHARS);
q = buf;
while (!strchr(sep, *p) && *p != '\0') {
if ((q - buf) < buf_size - 1)
get_word_sep(buf, sizeof(buf), "/", &p);
i = atoi(buf);
switch (codec->codec_type) {
- case CODEC_TYPE_AUDIO:
+ 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;
av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
codec->channels);
break;
- case CODEC_TYPE_VIDEO:
+ case AVMEDIA_TYPE_VIDEO:
av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
break;
default:
return 0;
}
-/* return the length and optionally the data */
-static int hex_to_data(uint8_t *data, const char *p)
-{
- int c, len, v;
-
- len = 0;
- v = 1;
- for (;;) {
- skip_spaces(&p);
- if (*p == '\0')
- break;
- c = toupper((unsigned char) *p++);
- if (c >= '0' && c <= '9')
- c = c - '0';
- else if (c >= 'A' && c <= 'F')
- c = c - 'A' + 10;
- else
- break;
- v = (v << 4) | c;
- if (v & 0x100) {
- if (data)
- data[len] = v;
- len++;
- v = 1;
- }
- }
- return len;
-}
-
-static void sdp_parse_fmtp_config(AVCodecContext * codec, void *ctx,
- char *attr, char *value)
-{
- switch (codec->codec_id) {
- case CODEC_ID_MPEG4:
- case CODEC_ID_AAC:
- if (!strcmp(attr, "config")) {
- /* decode the hexa encoded parameter */
- int len = hex_to_data(NULL, value);
- if (codec->extradata)
- av_free(codec->extradata);
- codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
- if (!codec->extradata)
- return;
- codec->extradata_size = len;
- hex_to_data(codec->extradata, value);
- }
- break;
- case CODEC_ID_VORBIS:
- ff_vorbis_parse_fmtp_config(codec, ctx, attr, value);
- break;
- default:
- break;
- }
- return;
-}
-
-typedef struct {
- const char *str;
- uint16_t type;
- uint32_t offset;
-} AttrNameMap;
-
-/* All known fmtp parmeters and the corresping RTPAttrTypeEnum */
-#define ATTR_NAME_TYPE_INT 0
-#define ATTR_NAME_TYPE_STR 1
-static const AttrNameMap attr_names[]=
-{
- { "SizeLength", ATTR_NAME_TYPE_INT,
- offsetof(RTPPayloadData, sizelength) },
- { "IndexLength", ATTR_NAME_TYPE_INT,
- offsetof(RTPPayloadData, indexlength) },
- { "IndexDeltaLength", ATTR_NAME_TYPE_INT,
- offsetof(RTPPayloadData, indexdeltalength) },
- { "profile-level-id", ATTR_NAME_TYPE_INT,
- offsetof(RTPPayloadData, profile_level_id) },
- { "StreamType", ATTR_NAME_TYPE_INT,
- offsetof(RTPPayloadData, streamtype) },
- { "mode", ATTR_NAME_TYPE_STR,
- offsetof(RTPPayloadData, mode) },
- { NULL, -1, -1 },
-};
-
-/* parse the attribute line from the fmtp a line of an sdp resonse. This
+/* 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)
{
- skip_spaces(p);
+ *p += strspn(*p, SPACE_CHARS);
if (**p) {
get_word_sep(attr, attr_size, "=", p);
if (**p == '=')
return 0;
}
-/* parse a SDP line and save stream attributes */
-static void sdp_parse_fmtp(AVStream *st, const char *p)
-{
- char attr[256];
- /* Vorbis setup headers can be up to 12KB and are sent base64
- * encoded, giving a 12KB * (4/3) = 16KB FMTP line. */
- char value[16384];
- int i;
- RTSPStream *rtsp_st = st->priv_data;
- AVCodecContext *codec = st->codec;
- RTPPayloadData *rtp_payload_data = &rtsp_st->rtp_payload_data;
-
- /* loop on each attribute */
- while (ff_rtsp_next_attr_and_value(&p, attr, sizeof(attr),
- value, sizeof(value))) {
- /* grab the codec extra_data from the config parameter of the fmtp
- * line */
- sdp_parse_fmtp_config(codec, rtsp_st->dynamic_protocol_context,
- attr, value);
- /* Looking for a known attribute */
- for (i = 0; attr_names[i].str; ++i) {
- if (!strcasecmp(attr, attr_names[i].str)) {
- if (attr_names[i].type == ATTR_NAME_TYPE_INT) {
- *(int *)((char *)rtp_payload_data +
- attr_names[i].offset) = atoi(value);
- } else if (attr_names[i].type == ATTR_NAME_TYPE_STR)
- *(char **)((char *)rtp_payload_data +
- attr_names[i].offset) = av_strdup(value);
- }
- }
- }
-}
-
/** Parse a string p in the form of Range:npt=xx-xx, and determine the start
* and end time.
* Used for seeking in the rtp stream.
{
char buf[256];
- skip_spaces(&p);
+ p += strspn(p, SPACE_CHARS);
if (!av_stristart(p, "npt=", &p))
return;
RTSPState *rt = s->priv_data;
char buf1[64], st_type[64];
const char *p;
- enum CodecType codec_type;
+ enum AVMediaType codec_type;
int payload_type, i;
AVStream *st;
RTSPStream *rtsp_st;
}
break;
case 's':
- av_metadata_set(&s->metadata, "title", p);
+ av_metadata_set2(&s->metadata, "title", p, 0);
break;
case 'i':
if (s->nb_streams == 0) {
- av_metadata_set(&s->metadata, "comment", p);
+ av_metadata_set2(&s->metadata, "comment", p, 0);
break;
}
break;
s1->skip_media = 0;
get_word(st_type, sizeof(st_type), &p);
if (!strcmp(st_type, "audio")) {
- codec_type = CODEC_TYPE_AUDIO;
+ codec_type = AVMEDIA_TYPE_AUDIO;
} else if (!strcmp(st_type, "video")) {
- codec_type = CODEC_TYPE_VIDEO;
+ codec_type = AVMEDIA_TYPE_VIDEO;
} else if (!strcmp(st_type, "application")) {
- codec_type = CODEC_TYPE_DATA;
+ codec_type = AVMEDIA_TYPE_DATA;
} else {
s1->skip_media = 1;
return;
rtsp_st = st->priv_data;
/* XXX: may need to add full url resolution */
- ff_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
+ av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
NULL, NULL, 0, p);
if (proto[0] == '\0') {
/* relative control URL */
st = s->streams[s->nb_streams - 1];
rtsp_st = st->priv_data;
sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p);
- } else if (av_strstart(p, "fmtp:", &p)) {
+ } else if (av_strstart(p, "fmtp:", &p) ||
+ av_strstart(p, "framesize:", &p)) {
/* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
- get_word(buf1, sizeof(buf1), &p);
- payload_type = atoi(buf1);
- for (i = 0; i < s->nb_streams; i++) {
- st = s->streams[i];
- rtsp_st = st->priv_data;
- if (rtsp_st->sdp_payload_type == payload_type) {
- if (!(rtsp_st->dynamic_handler &&
- rtsp_st->dynamic_handler->parse_sdp_a_line &&
- rtsp_st->dynamic_handler->parse_sdp_a_line(s,
- i, rtsp_st->dynamic_protocol_context, buf)))
- sdp_parse_fmtp(st, p);
- }
- }
- } else if (av_strstart(p, "framesize:", &p)) {
// let dynamic protocol handlers have a stab at the line.
get_word(buf1, sizeof(buf1), &p);
payload_type = atoi(buf1);
* "rulebooks" describing their properties. Therefore, the SDP line
* buffer is large.
*
- * The Vorbis FMTP line can be up to 16KB - see sdp_parse_fmtp. */
+ * 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 (;;) {
- skip_spaces(&p);
+ p += strspn(p, SPACE_CHARS);
letter = *p;
if (letter == '\0')
break;
if (s->oformat) {
rtsp_st->transport_priv = rtsp_rtp_mux_open(s, st, rtsp_st->rtp_handle);
- /* Ownage of rtp_handle is passed to the rtp mux context */
+ /* Ownership of rtp_handle is passed to the rtp mux context */
rtsp_st->rtp_handle = NULL;
} else if (rt->transport == RTSP_TRANSPORT_RDT)
rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
rtsp_st->dynamic_handler);
else
rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
- rtsp_st->sdp_payload_type,
- &rtsp_st->rtp_payload_data);
+ rtsp_st->sdp_payload_type);
if (!rtsp_st->transport_priv) {
return AVERROR(ENOMEM);
int v;
p = *pp;
- skip_spaces(&p);
+ p += strspn(p, SPACE_CHARS);
v = strtol(p, (char **)&p, 10);
if (*p == '-') {
p++;
reply->nb_transports = 0;
for (;;) {
- skip_spaces(&p);
+ p += strspn(p, SPACE_CHARS);
if (*p == '\0')
break;
} 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);
+ p += strspn(p, SPACE_CHARS);
av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
} else if (av_stristart(p, "Server:", &p)) {
- skip_spaces(&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)) {
- skip_spaces(&p);
+ p += strspn(p, SPACE_CHARS);
av_strlcpy(reply->location, p , sizeof(reply->location));
} else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) {
- skip_spaces(&p);
+ p += strspn(p, SPACE_CHARS);
ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p);
} else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) {
- skip_spaces(&p);
+ p += strspn(p, SPACE_CHARS);
ff_http_auth_handle_header(auth_state, "Authentication-Info", p);
}
}
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->auth_state);
av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
return 0;
}
-void 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)
+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], buf1[1024];
+ 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));
- snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq);
- av_strlcat(buf, buf1, sizeof(buf));
+ av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
if (rt->session_id[0] != '\0' && (!headers ||
!strstr(headers, "\nIf-Match:"))) {
- snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id);
- av_strlcat(buf, buf1, sizeof(buf));
+ 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,
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, buf, strlen(buf));
- if (send_content_length > 0 && send_content)
- url_write(rt->rtsp_hd, send_content, send_content_length);
+ 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;
}
-void ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
- const char *url, const char *headers)
+int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
+ const char *url, const char *headers)
{
- ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
+ return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
}
-void ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
- const char *headers, RTSPMessageHeader *reply,
- unsigned char **content_ptr)
+int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
+ const char *headers, RTSPMessageHeader *reply,
+ unsigned char **content_ptr)
{
- ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
- content_ptr, NULL, 0);
+ return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
+ content_ptr, NULL, 0);
}
-void 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)
+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;
- ff_rtsp_send_cmd_with_content_async(s, method, url, header,
- send_content, send_content_length);
+ if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
+ send_content,
+ send_content_length)))
+ return ret;
- ff_rtsp_read_reply(s, reply, content_ptr, 0);
+ if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0) ) < 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;
}
/**
- * @returns 0 on success, <0 on error, 1 if protocol is unavailable.
+ * @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)
* will return an error. Therefore, we skip those streams. */
if (rt->server_type == RTSP_SERVER_WMS &&
s->streams[rtsp_st->stream_index]->codec->codec_type ==
- CODEC_TYPE_DATA)
+ AVMEDIA_TYPE_DATA)
continue;
snprintf(transport, sizeof(transport) - 1,
"%s/TCP;", trans_pref);
rt->transport = reply->transports[0].transport;
}
- /* close RTP connection if not choosen */
+ /* close RTP connection if not chosen */
if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&
(lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {
url_close(rtsp_st->rtp_handle);
{
RTSPState *rt = s->priv_data;
RTSPMessageHeader reply1, *reply = &reply1;
+ int i;
char cmd[1024];
av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
if (reply->status_code != RTSP_STATUS_OK) {
return -1;
}
+ if (reply->range_start != AV_NOPTS_VALUE &&
+ rt->transport == RTSP_TRANSPORT_RTP) {
+ for (i = 0; i < rt->nb_rtsp_streams; i++) {
+ RTSPStream *rtsp_st = rt->rtsp_streams[i];
+ RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
+ AVStream *st = NULL;
+ if (!rtpctx)
+ continue;
+ if (rtsp_st->stream_index >= 0)
+ st = s->streams[rtsp_st->stream_index];
+ rtpctx->last_rtcp_ntp_time = AV_NOPTS_VALUE;
+ rtpctx->first_rtcp_ntp_time = AV_NOPTS_VALUE;
+ if (st)
+ rtpctx->range_start_offset = av_rescale_q(reply->range_start,
+ AV_TIME_BASE_Q,
+ st->time_base);
+ }
+ }
}
rt->state = RTSP_STATE_STREAMING;
return 0;
rt->start_time = av_gettime();
/* Announce the stream */
- sdp = av_mallocz(8192);
+ sdp = av_mallocz(SDP_MAX_SIZE);
if (sdp == NULL)
return AVERROR(ENOMEM);
/* We create the SDP based on the RTSP AVFormatContext where we
ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
"rtsp", NULL, addr, -1, NULL);
ctx_array[0] = &sdp_ctx;
- if (avf_sdp_create(ctx_array, 1, sdp, 8192)) {
+ if (avf_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
av_free(sdp);
return AVERROR_INVALIDDATA;
}
return 0;
}
+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;
- URLContext *rtsp_hd;
int port, err, tcp_fd;
- RTSPMessageHeader reply1, *reply = &reply1;
+ RTSPMessageHeader reply1 = {0}, *reply = &reply1;
int lower_transport_mask = 0;
char real_challenge[64];
struct sockaddr_storage peer;
if (!ff_network_init())
return AVERROR(EIO);
redirect:
+ rt->control_transport = RTSP_MODE_PLAIN;
/* extract hostname and port */
- ff_url_split(NULL, 0, auth, sizeof(auth),
+ 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));
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 {
/* Write options back into the buffer, using memmove instead
* of strcpy since the strings may overlap. */
/* 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) {
+ 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);
}
}
- /* open the tcp connexion */
- ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
- if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0) {
- err = AVERROR(EIO);
- goto fail;
+ /* 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->rtsp_hd = rtsp_hd;
rt->seq = 0;
- tcp_fd = url_get_file_handle(rtsp_hd);
+ 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);
}
- /* 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);
/* request options supported by the server; this also detects server
* type */
for (rt->server_type = RTSP_SERVER_RTP;;) {
goto fail;
lower_transport_mask &= ~(1 << lower_transport);
if (lower_transport_mask == 0 && err == 1) {
- err = AVERROR(FF_NETERROR(EPROTONOSUPPORT));
+ err = FF_NETERROR(EPROTONOSUPPORT);
goto fail;
}
} while (err);
return 0;
fail:
ff_rtsp_close_streams(s);
- url_close(rt->rtsp_hd);
+ 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",
if (ret)
return ret;
+ rt->real_setup_cache = av_mallocz(2 * s->nb_streams * sizeof(*rt->real_setup_cache));
+ if (!rt->real_setup_cache)
+ return AVERROR(ENOMEM);
+ rt->real_setup = rt->real_setup_cache + s->nb_streams * sizeof(*rt->real_setup);
+
if (ap->initial_pause) {
/* do not start immediately */
} else {
if (rtsp_read_play(s) < 0) {
ff_rtsp_close_streams(s);
- url_close(rt->rtsp_hd);
+ ff_rtsp_close_connections(s);
return AVERROR_INVALIDDATA;
}
}
RTSPState *rt = s->priv_data;
RTSPStream *rtsp_st;
fd_set rfds;
- int fd, fd_max, n, i, ret, tcp_fd;
+ int fd, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
struct timeval tv;
for (;;) {
}
}
tv.tv_sec = 0;
- tv.tv_usec = 100 * 1000;
+ 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) {
return 0;
}
#endif
- }
+ } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
+ return FF_NETERROR(ETIMEDOUT);
+ } else if (n < 0 && errno != EINTR)
+ return AVERROR(errno);
}
}
return AVERROR_EOF;
if (rt->transport == RTSP_TRANSPORT_RDT) {
ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
- } else
+ } else {
ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, buf, 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;
+ for (i = 0; i < rt->nb_rtsp_streams; i++) {
+ RTPDemuxContext *rtpctx2 = rtsp_st->transport_priv;
+ if (rtpctx2 &&
+ rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE)
+ rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
+ }
+ }
+ }
+ }
if (ret < 0)
goto redo;
if (ret == 1)
if (rt->server_type == RTSP_SERVER_REAL) {
int i;
- enum AVDiscard cache[MAX_STREAMS];
for (i = 0; i < s->nb_streams; i++)
- cache[i] = s->streams[i]->discard;
+ rt->real_setup[i] = s->streams[i]->discard;
if (!rt->need_subscription) {
- if (memcmp (cache, rt->real_setup_cache,
+ if (memcmp (rt->real_setup, rt->real_setup_cache,
sizeof(enum AVDiscard) * s->nb_streams)) {
snprintf(cmd, sizeof(cmd),
"Unsubscribe: %s\r\n",
if (rt->need_subscription) {
int r, rule_nr, first = 1;
- memcpy(rt->real_setup_cache, cache,
+ memcpy(rt->real_setup_cache, rt->real_setup,
sizeof(enum AVDiscard) * s->nb_streams);
rt->last_subscription[0] = 0;
return ret;
/* send dummy request to keep TCP connection alive */
- if ((rt->server_type == RTSP_SERVER_WMS ||
- rt->server_type == RTSP_SERVER_REAL) &&
- (av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
+ if ((av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
if (rt->server_type == RTSP_SERVER_WMS) {
ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
} else {
RTSPState *rt = s->priv_data;
RTSPMessageHeader reply1, *reply = &reply1;
- rt = s->priv_data;
-
if (rt->state != RTSP_STATE_STREAMING)
return 0;
else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
ff_rtsp_close_streams(s);
- url_close(rt->rtsp_hd);
+ ff_rtsp_close_connections(s);
ff_network_close();
+ rt->real_setup = NULL;
+ av_freep(&rt->real_setup_cache);
return 0;
}
return 0;
}
-#define SDP_MAX_SIZE 8192
-
static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
{
RTSPState *rt = s->priv_data;
projects / ffmpeg / blobdiff