* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-/* needed by inet_aton() */
-#define _SVID_SOURCE
-
#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 <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 "rtp_asf.h"
-#include "rtp_vorbis.h"
+#include "rtpdec_asf.h"
//#define DEBUG
//#define DEBUG_RTP_TCP
int rtsp_default_protocols = (1 << RTSP_LOWER_TRANSPORT_UDP);
#endif
+/* 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 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'. */
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:
hex_to_data(codec->extradata, value);
}
break;
- case CODEC_ID_VORBIS:
- ff_vorbis_parse_fmtp_config(codec, ctx, attr, value);
- break;
default:
break;
}
uint32_t offset;
} AttrNameMap;
-/* All known fmtp parmeters and the corresping RTPAttrTypeEnum */
+/* All known fmtp parameters and the corresponding RTPAttrTypeEnum */
#define ATTR_NAME_TYPE_INT 0
#define ATTR_NAME_TYPE_STR 1
static const AttrNameMap attr_names[]=
{ 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 rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
- char *value, int value_size)
+int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
+ char *value, int value_size)
{
skip_spaces(p);
if (**p) {
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];
+ char value[4096];
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 (rtsp_next_attr_and_value(&p, attr, sizeof(attr),
- value, sizeof(value))) {
+ 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,
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;
if (strcmp(buf1, "IP4") != 0)
return;
get_word_sep(buf1, sizeof(buf1), "/", &p);
- if (inet_aton(buf1, &sdp_ip) == 0)
+ if (ff_inet_aton(buf1, &sdp_ip) == 0)
return;
ttl = 16;
if (*p == '/') {
}
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 */
- url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
- NULL, NULL, 0, p);
+ ff_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]!='/')
* "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;
}
/* close and free RTSP streams */
-static void rtsp_close_streams(AVFormatContext *s)
+void ff_rtsp_close_streams(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
int i;
if (s->oformat) {
AVFormatContext *rtpctx = rtsp_st->transport_priv;
av_write_trailer(rtpctx);
- url_fclose(rtpctx->pb);
+ 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)
av_close_input_stream (rt->asf_ctx);
rt->asf_ctx = NULL;
}
- av_freep(&rt->auth_b64);
}
-static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st, URLContext *handle) {
+static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st,
+ URLContext *handle)
+{
+ RTSPState *rt = s->priv_data;
AVFormatContext *rtpctx;
int ret;
AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
/* Copy other stream parameters. */
rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio;
+ /* Set the synchronized start time. */
+ rtpctx->start_time_realtime = rt->start_time;
+
/* Remove the local codec, link to the original codec
* context instead, to give the rtp muxer access to
* codec parameters. */
av_free(rtpctx->streams[0]->codec);
rtpctx->streams[0]->codec = st->codec;
- url_fdopen(&rtpctx->pb, handle);
+ if (handle) {
+ url_fdopen(&rtpctx->pb, handle);
+ } else
+ url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
ret = av_write_header(rtpctx);
if (ret) {
- url_fclose(rtpctx->pb);
+ if (handle) {
+ url_fclose(rtpctx->pb);
+ } else {
+ uint8_t *ptr;
+ url_close_dyn_buf(rtpctx->pb, &ptr);
+ av_free(ptr);
+ }
av_free(rtpctx->streams[0]);
av_free(rtpctx);
return NULL;
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,
return 0;
}
-#if CONFIG_RTSP_DEMUXER
+#if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
static int rtsp_probe(AVProbeData *p)
{
if (av_strstart(p->filename, "rtsp:", NULL))
if (*p == '=') {
p++;
get_word_sep(buf, sizeof(buf), ";,", &p);
- if (inet_aton(buf, &ipaddr))
+ if (ff_inet_aton(buf, &ipaddr))
th->destination = ntohl(ipaddr.s_addr);
}
}
}
}
-void rtsp_parse_line(RTSPMessageHeader *reply, const char *buf)
+void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
+ HTTPAuthState *auth_state)
{
const char *p;
} else if (av_stristart(p, "Location:", &p)) {
skip_spaces(&p);
av_strlcpy(reply->location, p , sizeof(reply->location));
+ } else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) {
+ skip_spaces(&p);
+ ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p);
+ } else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) {
+ skip_spaces(&p);
+ ff_http_auth_handle_header(auth_state, "Authentication-Info", p);
}
}
/* skip a RTP/TCP interleaved packet */
-static void rtsp_skip_packet(AVFormatContext *s)
+void ff_rtsp_skip_packet(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
int ret, len, len1;
}
}
-/**
- * Read a RTSP message from the server, or prepare to read data
- * packets if we're reading data interleaved over the TCP/RTSP
- * connection as well.
- *
- * @param s RTSP demuxer context
- * @param reply pointer where the RTSP message header will be stored
- * @param content_ptr pointer where the RTSP message body, if any, will
- * be stored (length is in reply)
- * @param return_on_interleaved_data whether the function may return if we
- * encounter a data marker ('$'), which precedes data
- * packets over interleaved TCP/RTSP connections. If this
- * is set, this function will return 1 after encountering
- * a '$'. If it is not set, the function will skip any
- * data packets (if they are encountered), until a reply
- * has been fully parsed. If no more data is available
- * without parsing a reply, it will return an error.
- *
- * @returns 1 if a data packets is ready to be received, -1 on error,
- * and 0 on success.
- */
-static int rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
- unsigned char **content_ptr,
- int return_on_interleaved_data)
+int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
+ unsigned char **content_ptr,
+ int return_on_interleaved_data)
{
RTSPState *rt = s->priv_data;
char buf[4096], buf1[1024], *q;
if (return_on_interleaved_data) {
return 1;
} else
- rtsp_skip_packet(s);
+ ff_rtsp_skip_packet(s);
} else if (ch != '\r') {
if ((q - buf) < sizeof(buf) - 1)
*q++ = ch;
get_word(buf1, sizeof(buf1), &p);
reply->status_code = atoi(buf1);
} else {
- rtsp_parse_line(reply, p);
+ ff_rtsp_parse_line(reply, p, &rt->auth_state);
av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
}
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 */ ||
return 0;
}
-static void rtsp_send_cmd_with_content_async(AVFormatContext *s,
- const char *cmd,
- 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++;
- av_strlcpy(buf, cmd, sizeof(buf));
- snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq);
- av_strlcat(buf, buf1, sizeof(buf));
- if (rt->session_id[0] != '\0' && !strstr(cmd, "\nIf-Match:")) {
- snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id);
- av_strlcat(buf, buf1, sizeof(buf));
+ 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 (rt->auth_b64)
- av_strlcatf(buf, sizeof(buf),
- "Authorization: Basic %s\r\n",
- rt->auth_b64);
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, 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;
}
-static void rtsp_send_cmd_async(AVFormatContext *s, const char *cmd)
+int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
+ const char *url, const char *headers)
{
- rtsp_send_cmd_with_content_async(s, cmd, NULL, 0);
+ return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
}
-static void rtsp_send_cmd(AVFormatContext *s,
- const char *cmd, 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)
{
- rtsp_send_cmd_async(s, cmd);
-
- rtsp_read_reply(s, reply, content_ptr, 0);
+ return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
+ content_ptr, NULL, 0);
}
-static void rtsp_send_cmd_with_content(AVFormatContext *s,
- const char *cmd,
- 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)
{
- rtsp_send_cmd_with_content_async(s, cmd, send_content, send_content_length);
+ RTSPState *rt = s->priv_data;
+ HTTPAuthType cur_auth_type;
+ int ret;
- rtsp_read_reply(s, reply, content_ptr, 0);
+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) ) < 0)
+ return ret;
+
+ if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
+ rt->auth_state.auth_type != HTTP_AUTH_NONE)
+ goto retry;
+
+ 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)
/* first try in specified port range */
if (RTSP_RTP_PORT_MIN != 0) {
while (j <= RTSP_RTP_PORT_MAX) {
- snprintf(buf, sizeof(buf), "rtp://%s?localport=%d",
- host, j);
+ 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)
* 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);
if (s->oformat) {
av_strlcat(transport, ";mode=receive", sizeof(transport));
} else if (rt->server_type == RTSP_SERVER_REAL ||
- rt->server_type == RTSP_SERVER_WMS)
+ rt->server_type == RTSP_SERVER_WMS)
av_strlcat(transport, ";mode=play", sizeof(transport));
snprintf(cmd, sizeof(cmd),
- "SETUP %s RTSP/1.0\r\n"
"Transport: %s\r\n",
- rtsp_st->control_url, transport);
+ transport);
if (i == 0 && rt->server_type == RTSP_SERVER_REAL) {
char real_res[41], real_csum[9];
ff_rdt_calc_response_and_checksum(real_res, real_csum,
"RealChallenge2: %s, sd=%s\r\n",
rt->session_id, real_res, real_csum);
}
- rtsp_send_cmd(s, cmd, reply, NULL);
+ 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;
char url[1024];
/* XXX: also use address if specified */
- snprintf(url, sizeof(url), "rtp://%s:%d",
- host, reply->transports[0].server_port_min);
+ ff_url_join(url, sizeof(url), "rtp", NULL, host,
+ reply->transports[0].server_port_min, NULL);
if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
err = AVERROR_INVALIDDATA;
port = rtsp_st->sdp_port;
ttl = rtsp_st->sdp_ttl;
}
- snprintf(url, sizeof(url), "rtp://%s:%d?ttl=%d",
- inet_ntoa(in), port, ttl);
+ ff_url_join(url, sizeof(url), "rtp", NULL, inet_ntoa(in),
+ port, "?ttl=%d", ttl);
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
{
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 (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
if (rt->state == RTSP_STATE_PAUSED) {
- snprintf(cmd, sizeof(cmd),
- "PLAY %s RTSP/1.0\r\n",
- rt->control_uri);
+ cmd[0] = 0;
} else {
snprintf(cmd, sizeof(cmd),
- "PLAY %s RTSP/1.0\r\n"
"Range: npt=%0.3f-\r\n",
- rt->control_uri,
(double)rt->seek_timestamp / AV_TIME_BASE);
}
- rtsp_send_cmd(s, cmd, reply, NULL);
+ ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL);
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;
}
-static int rtsp_setup_input_streams(AVFormatContext *s)
+static int rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
{
RTSPState *rt = s->priv_data;
- RTSPMessageHeader reply1, *reply = &reply1;
char cmd[1024];
unsigned char *content = NULL;
int ret;
/* describe the stream */
snprintf(cmd, sizeof(cmd),
- "DESCRIBE %s RTSP/1.0\r\n"
- "Accept: application/sdp\r\n",
- s->filename);
+ "Accept: application/sdp\r\n");
if (rt->server_type == RTSP_SERVER_REAL) {
/**
* The Require: attribute is needed for proper streaming from
"Require: com.real.retain-entity-for-setup\r\n",
sizeof(cmd));
}
- rtsp_send_cmd(s, cmd, reply, &content);
+ ff_rtsp_send_cmd(s, "DESCRIBE", rt->control_uri, cmd, reply, &content);
if (!content)
return AVERROR_INVALIDDATA;
if (reply->status_code != RTSP_STATUS_OK) {
return 0;
}
-static int rtsp_connect(AVFormatContext *s)
+static int rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
+{
+ RTSPState *rt = s->priv_data;
+ RTSPMessageHeader reply1, *reply = &reply1;
+ int i;
+ char *sdp;
+ AVFormatContext sdp_ctx, *ctx_array[1];
+
+ rt->start_time = av_gettime();
+
+ /* Announce the stream */
+ sdp = av_mallocz(8192);
+ if (sdp == NULL)
+ return AVERROR(ENOMEM);
+ /* We create the SDP based on the RTSP AVFormatContext where we
+ * aren't allowed to change the filename field. (We create the SDP
+ * based on the RTSP context since the contexts for the RTP streams
+ * don't exist yet.) In order to specify a custom URL with the actual
+ * peer IP instead of the originally specified hostname, we create
+ * a temporary copy of the AVFormatContext, where the custom URL is set.
+ *
+ * FIXME: Create the SDP without copying the AVFormatContext.
+ * This either requires setting up the RTP stream AVFormatContexts
+ * already here (complicating things immensely) or getting a more
+ * flexible SDP creation interface.
+ */
+ sdp_ctx = *s;
+ 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)) {
+ av_free(sdp);
+ return AVERROR_INVALIDDATA;
+ }
+ av_log(s, AV_LOG_INFO, "SDP:\n%s\n", sdp);
+ ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
+ "Content-Type: application/sdp\r\n",
+ reply, NULL, sdp, strlen(sdp));
+ av_free(sdp);
+ if (reply->status_code != RTSP_STATUS_OK)
+ return AVERROR_INVALIDDATA;
+
+ /* Set up the RTSPStreams for each AVStream */
+ for (i = 0; i < s->nb_streams; i++) {
+ RTSPStream *rtsp_st;
+ AVStream *st = s->streams[i];
+
+ rtsp_st = av_mallocz(sizeof(RTSPStream));
+ if (!rtsp_st)
+ return AVERROR(ENOMEM);
+ dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
+
+ st->priv_data = rtsp_st;
+ rtsp_st->stream_index = i;
+
+ av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
+ /* Note, this must match the relative uri set in the sdp content */
+ av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
+ "/streamid=%d", i);
+ }
+
+ 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);
+}
+
+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;
- RTSPMessageHeader reply1, *reply = &reply1;
+ URLContext *rtsp_hd, *rtsp_hd_out;
+ int port, err, tcp_fd;
+ RTSPMessageHeader reply1 = {}, *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 */
- url_split(NULL, 0, auth, sizeof(auth),
- host, sizeof(host), &port, path, sizeof(path), s->filename);
+ ff_url_split(NULL, 0, auth, sizeof(auth),
+ host, sizeof(host), &port, path, sizeof(path), s->filename);
if (*auth) {
- int auth_len = strlen(auth), b64_len = ((auth_len + 2) / 3) * 4 + 1;
-
- if (!(rt->auth_b64 = av_malloc(b64_len)))
- return AVERROR(ENOMEM);
- if (!av_base64_encode(rt->auth_b64, b64_len, auth, auth_len)) {
- err = AVERROR(EINVAL);
- goto fail;
- }
+ av_strlcpy(rt->auth, auth, sizeof(rt->auth));
}
if (port < 0)
port = RTSP_DEFAULT_PORT;
/* search for options */
- option_list = strchr(path, '?');
+ option_list = strrchr(path, '?');
if (option_list) {
- filename = strchr(s->filename, '?');
+ /* 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;
/* handle the options */
if (!strcmp(option, "udp")) {
- lower_transport_mask = (1<< RTSP_LOWER_TRANSPORT_UDP);
+ lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
} else if (!strcmp(option, "multicast")) {
- lower_transport_mask = (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
+ lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
} else if (!strcmp(option, "tcp")) {
- lower_transport_mask = (1<< RTSP_LOWER_TRANSPORT_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 {
- strcpy(++filename, option);
- filename += strlen(option);
+ /* 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 = '&';
}
}
if (!lower_transport_mask)
lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
- /* open the tcp connexion */
- snprintf(tcpname, sizeof(tcpname), "tcp://%s:%d", host, port);
- if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0) {
- err = AVERROR(EIO);
- goto fail;
+ 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", NULL, host, port, "%s", path);
+ snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
+ av_get_random_seed(), av_get_random_seed());
+
+ /* GET requests */
+ if (url_open(&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(rtsp_hd, headers);
+
+ /* complete the connection */
+ if (url_read(rtsp_hd, NULL, 0)) {
+ err = AVERROR(EIO);
+ goto fail;
+ }
+
+ /* POST requests */
+ if (url_open(&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(rtsp_hd_out, headers);
+ ff_http_set_chunked_transfer_encoding(rtsp_hd_out, 0);
+
+ } else {
+ /* open the tcp connection */
+ 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;
+ }
+ rtsp_hd_out = rtsp_hd;
}
rt->rtsp_hd = rtsp_hd;
+ rt->rtsp_hd_out = rtsp_hd_out;
rt->seq = 0;
+ tcp_fd = url_get_file_handle(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 */
- av_strlcpy(rt->control_uri, s->filename,
- sizeof(rt->control_uri));
for (rt->server_type = RTSP_SERVER_RTP;;) {
- snprintf(cmd, sizeof(cmd),
- "OPTIONS %s RTSP/1.0\r\n", s->filename);
+ cmd[0] = 0;
if (rt->server_type == RTSP_SERVER_REAL)
av_strlcat(cmd,
/**
"CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
"GUID: 00000000-0000-0000-0000-000000000000\r\n",
sizeof(cmd));
- rtsp_send_cmd(s, cmd, reply, NULL);
+ ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
err = AVERROR_INVALIDDATA;
goto fail;
break;
}
- err = rtsp_setup_input_streams(s);
+ if (s->iformat)
+ err = rtsp_setup_input_streams(s, reply);
+ else
+ err = rtsp_setup_output_streams(s, host);
if (err)
goto fail;
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);
rt->seek_timestamp = 0; /* default is to start stream at position zero */
return 0;
fail:
- rtsp_close_streams(s);
- url_close(rt->rtsp_hd);
- if (reply->status_code >=300 && reply->status_code < 400) {
+ 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
+#if CONFIG_RTSP_DEMUXER
static int rtsp_read_header(AVFormatContext *s,
AVFormatParameters *ap)
{
- RTSPState *rt = s->priv_data;
int ret;
- ret = rtsp_connect(s);
+ ret = ff_rtsp_connect(s);
if (ret)
return ret;
/* do not start immediately */
} else {
if (rtsp_read_play(s) < 0) {
- rtsp_close_streams(s);
- url_close(rt->rtsp_hd);
+ ff_rtsp_close_streams(s);
+ 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) {
if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
RTSPMessageHeader reply;
- rtsp_read_reply(s, &reply, NULL, 0);
+ ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
+ 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);
}
}
for (;;) {
RTSPMessageHeader reply;
- ret = rtsp_read_reply(s, &reply, NULL, 1);
+ ret = ff_rtsp_read_reply(s, &reply, NULL, 1);
if (ret == -1)
return -1;
if (ret == 1) /* received '$' */
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 (memcmp (cache, rt->real_setup_cache,
sizeof(enum AVDiscard) * s->nb_streams)) {
snprintf(cmd, sizeof(cmd),
- "SET_PARAMETER %s RTSP/1.0\r\n"
"Unsubscribe: %s\r\n",
- rt->control_uri, rt->last_subscription);
- rtsp_send_cmd(s, cmd, reply, NULL);
+ rt->last_subscription);
+ ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
+ cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK)
return AVERROR_INVALIDDATA;
rt->need_subscription = 1;
rt->last_subscription[0] = 0;
snprintf(cmd, sizeof(cmd),
- "SET_PARAMETER %s RTSP/1.0\r\n"
- "Subscribe: ",
- rt->control_uri);
+ "Subscribe: ");
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rule_nr = 0;
for (r = 0; r < s->nb_streams; r++) {
}
}
av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription);
- rtsp_send_cmd(s, cmd, reply, NULL);
+ ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
+ cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK)
return AVERROR_INVALIDDATA;
rt->need_subscription = 0;
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",
- rt->control_uri);
- rtsp_send_cmd_async(s, cmd);
+ ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
} else {
- rtsp_send_cmd_async(s, "OPTIONS * RTSP/1.0\r\n");
+ ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL);
}
}
{
RTSPState *rt = s->priv_data;
RTSPMessageHeader reply1, *reply = &reply1;
- char cmd[1024];
-
- rt = s->priv_data;
if (rt->state != RTSP_STATE_STREAMING)
return 0;
else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
- snprintf(cmd, sizeof(cmd),
- "PAUSE %s RTSP/1.0\r\n",
- rt->control_uri);
- rtsp_send_cmd(s, cmd, reply, NULL);
+ ff_rtsp_send_cmd(s, "PAUSE", rt->control_uri, NULL, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
return -1;
}
static int rtsp_read_close(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
- char cmd[1024];
#if 0
/* NOTE: it is valid to flush the buffer here */
url_fclose(&rt->rtsp_gb);
}
#endif
- snprintf(cmd, sizeof(cmd),
- "TEARDOWN %s RTSP/1.0\r\n",
- s->filename);
- rtsp_send_cmd_async(s, cmd);
+ ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
- rtsp_close_streams(s);
- url_close(rt->rtsp_hd);
+ ff_rtsp_close_streams(s);
+ ff_rtsp_close_connections(s);
+ ff_network_close();
return 0;
}
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);
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
- snprintf(url, sizeof(url), "rtp://%s:%d?localport=%d&ttl=%d",
- inet_ntoa(rtsp_st->sdp_ip),
- rtsp_st->sdp_port,
- rtsp_st->sdp_port,
- rtsp_st->sdp_ttl);
+ ff_url_join(url, sizeof(url), "rtp", NULL,
+ inet_ntoa(rtsp_st->sdp_ip), 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;
}
return 0;
fail:
- rtsp_close_streams(s);
+ ff_rtsp_close_streams(s);
+ ff_network_close();
return err;
}
static int sdp_read_close(AVFormatContext *s)
{
- rtsp_close_streams(s);
+ ff_rtsp_close_streams(s);
+ ff_network_close();
return 0;
}
projects / ffmpeg / blobdiff