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added RTP/TCP protocol support 

Originally committed as revision 2063 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Fabrice Bellard 2003-07-17 10:31:23 +00:00
parent 416e35081a
commit bc3513865a
1 changed files with 225 additions and 106 deletions
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327
ffserver.c
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@ -60,6 +60,7 @@ enum HTTPState {
RTSPSTATE_WAIT_REQUEST,
RTSPSTATE_SEND_REPLY,
RTSPSTATE_SEND_PACKET,
};
const char *http_state[] = {
@ -77,6 +78,7 @@ const char *http_state[] = {
"RTSP_WAIT_REQUEST",
"RTSP_SEND_REPLY",
"RTSP_SEND_PACKET",
};
#define IOBUFFER_INIT_SIZE 8192
@ -143,11 +145,16 @@ typedef struct HTTPContext {
enum RTSPProtocol rtp_protocol;
char session_id[32]; /* session id */
AVFormatContext *rtp_ctx[MAX_STREAMS];
URLContext *rtp_handles[MAX_STREAMS];
/* RTP short term bandwidth limitation */
int packet_byte_count;
int packet_start_time_us; /* used for short durations (a few
seconds max) */
/* RTP/UDP specific */
URLContext *rtp_handles[MAX_STREAMS];
/* RTP/TCP specific */
struct HTTPContext *rtsp_c;
uint8_t *packet_buffer, *packet_buffer_ptr, *packet_buffer_end;
} HTTPContext;
static AVFrame dummy_frame;
@ -259,9 +266,11 @@ static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
/* RTP handling */
static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
FFStream *stream, const char *session_id);
FFStream *stream, const char *session_id,
enum RTSPProtocol rtp_protocol);
static int rtp_new_av_stream(HTTPContext *c,
int stream_index, struct sockaddr_in *dest_addr);
int stream_index, struct sockaddr_in *dest_addr,
HTTPContext *rtsp_c);
static const char *my_program_name;
static const char *my_program_dir;
@ -289,7 +298,7 @@ static long gettime_ms(void)
static FILE *logfile = NULL;
static void http_log(const char *fmt, ...)
static void __attribute__ ((format (printf, 1, 2))) http_log(const char *fmt, ...)
{
va_list ap;
va_start(ap, fmt);
@ -477,7 +486,8 @@ static void start_multicast(void)
dest_addr.sin_addr = stream->multicast_ip;
dest_addr.sin_port = htons(stream->multicast_port);
rtp_c = rtp_new_connection(&dest_addr, stream, session_id);
rtp_c = rtp_new_connection(&dest_addr, stream, session_id,
RTSP_PROTOCOL_RTP_UDP_MULTICAST);
if (!rtp_c) {
continue;
}
@ -487,14 +497,12 @@ static void start_multicast(void)
continue;
}
rtp_c->rtp_protocol = RTSP_PROTOCOL_RTP_UDP_MULTICAST;
/* open each RTP stream */
for(stream_index = 0; stream_index < stream->nb_streams;
stream_index++) {
dest_addr.sin_port = htons(stream->multicast_port +
2 * stream_index);
if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr) < 0) {
if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr, NULL) < 0) {
fprintf(stderr, "Could not open output stream '%s/streamid=%d'\n",
stream->filename, stream_index);
exit(1);
@ -551,6 +559,7 @@ static int http_server(void)
switch(c->state) {
case HTTPSTATE_SEND_HEADER:
case RTSPSTATE_SEND_REPLY:
case RTSPSTATE_SEND_PACKET:
c->poll_entry = poll_entry;
poll_entry->fd = fd;
poll_entry->events = POLLOUT;
@ -716,6 +725,12 @@ static void close_connection(HTTPContext *c)
}
}
/* remove references, if any (XXX: do it faster) */
for(c1 = first_http_ctx; c1 != NULL; c1 = c1->next) {
if (c1->rtsp_c == c)
c1->rtsp_c = NULL;
}
/* remove connection associated resources */
if (c->fd >= 0)
close(c->fd);
@ -754,7 +769,7 @@ static void close_connection(HTTPContext *c)
/* prepare header */
if (url_open_dyn_buf(&ctx->pb) >= 0) {
av_write_trailer(ctx);
(void) url_close_dyn_buf(&ctx->pb, &c->pb_buffer);
url_close_dyn_buf(&ctx->pb, &c->pb_buffer);
}
}
}
@ -765,6 +780,7 @@ static void close_connection(HTTPContext *c)
if (c->stream)
current_bandwidth -= c->stream->bandwidth;
av_freep(&c->pb_buffer);
av_freep(&c->packet_buffer);
av_free(c->buffer);
av_free(c);
nb_connections--;
@ -917,6 +933,31 @@ static int handle_connection(HTTPContext *c)
}
}
break;
case RTSPSTATE_SEND_PACKET:
if (c->poll_entry->revents & (POLLERR | POLLHUP)) {
av_freep(&c->packet_buffer);
return -1;
}
/* no need to write if no events */
if (!(c->poll_entry->revents & POLLOUT))
return 0;
len = write(c->fd, c->packet_buffer_ptr,
c->packet_buffer_end - c->packet_buffer_ptr);
if (len < 0) {
if (errno != EAGAIN && errno != EINTR) {
/* error : close connection */
av_freep(&c->packet_buffer);
return -1;
}
} else {
c->packet_buffer_ptr += len;
if (c->packet_buffer_ptr >= c->packet_buffer_end) {
/* all the buffer was sent : wait for a new request */
av_freep(&c->packet_buffer);
c->state = RTSPSTATE_WAIT_REQUEST;
}
}
break;
case HTTPSTATE_READY:
/* nothing to do */
break;
@ -2087,13 +2128,15 @@ static int compute_send_delay(HTTPContext *c)
if (datarate > c->stream->bandwidth * 2000) {
return 1000;
}
if(!c->stream->feed && c->first_pts!=AV_NOPTS_VALUE) {
time_pts = ((int64_t)(cur_time - c->start_time) * c->fmt_in->pts_den) /
((int64_t) c->fmt_in->pts_num*1000);
delta_pts = c->cur_pts - time_pts;
m_delay = (delta_pts * 1000 * c->fmt_in->pts_num) / c->fmt_in->pts_den;
return m_delay>0 ? m_delay : 0;
} else return 0;
if (!c->stream->feed && c->first_pts!=AV_NOPTS_VALUE) {
time_pts = ((int64_t)(cur_time - c->start_time) * c->fmt_in->pts_den) /
((int64_t) c->fmt_in->pts_num*1000);
delta_pts = c->cur_pts - time_pts;
m_delay = (delta_pts * 1000 * c->fmt_in->pts_num) / c->fmt_in->pts_den;
return m_delay>0 ? m_delay : 0;
} else {
return 0;
}
}
#endif
@ -2103,6 +2146,7 @@ static int http_prepare_data(HTTPContext *c)
int i, len, ret;
AVFormatContext *ctx;
av_freep(&c->pb_buffer);
switch(c->state) {
case HTTPSTATE_SEND_DATA_HEADER:
memset(&c->fmt_ctx, 0, sizeof(c->fmt_ctx));
@ -2273,8 +2317,12 @@ static int http_prepare_data(HTTPContext *c)
#endif
if (c->is_packetized) {
ret = url_open_dyn_packet_buf(&ctx->pb,
url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]));
int max_packet_size;
if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP)
max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
else
max_packet_size = url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]);
ret = url_open_dyn_packet_buf(&ctx->pb, max_packet_size);
c->packet_byte_count = 0;
c->packet_start_time_us = av_gettime();
} else {
@ -2327,76 +2375,115 @@ static int http_prepare_data(HTTPContext *c)
#define SHORT_TERM_BANDWIDTH 8000000
/* should convert the format at the same time */
/* send data starting at c->buffer_ptr to the output connection
(either UDP or TCP connection) */
static int http_send_data(HTTPContext *c)
{
int len, ret, dt;
while (c->buffer_ptr >= c->buffer_end) {
av_freep(&c->pb_buffer);
ret = http_prepare_data(c);
if (ret < 0)
return -1;
else if (ret == 0) {
continue;
} else {
/* state change requested */
return 0;
}
}
if (c->buffer_ptr < c->buffer_end) {
if (c->is_packetized) {
/* RTP/UDP data output */
len = c->buffer_end - c->buffer_ptr;
if (len < 4) {
/* fail safe - should never happen */
fail1:
c->buffer_ptr = c->buffer_end;
return 0;
for(;;) {
if (c->buffer_ptr >= c->buffer_end) {
ret = http_prepare_data(c);
if (ret < 0)
return -1;
else if (ret != 0) {
/* state change requested */
break;
}
len = (c->buffer_ptr[0] << 24) |
(c->buffer_ptr[1] << 16) |
(c->buffer_ptr[2] << 8) |
(c->buffer_ptr[3]);
if (len > (c->buffer_end - c->buffer_ptr))
goto fail1;
/* short term bandwidth limitation */
dt = av_gettime() - c->packet_start_time_us;
if (dt < 1)
dt = 1;
if ((c->packet_byte_count + len) * (int64_t)1000000 >=
(SHORT_TERM_BANDWIDTH / 8) * (int64_t)dt) {
/* bandwidth overflow : wait at most one tick and retry */
c->state = HTTPSTATE_WAIT_SHORT;
return 0;
}
c->buffer_ptr += 4;
url_write(c->rtp_handles[c->packet_stream_index],
c->buffer_ptr, len);
c->buffer_ptr += len;
c->packet_byte_count += len;
} else {
/* TCP data output */
len = write(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
if (len < 0) {
if (errno != EAGAIN && errno != EINTR) {
/* error : close connection */
return -1;
} else {
if (c->is_packetized) {
/* RTP data output */
len = c->buffer_end - c->buffer_ptr;
if (len < 4) {
/* fail safe - should never happen */
fail1:
c->buffer_ptr = c->buffer_end;
return 0;
}
} else {
len = (c->buffer_ptr[0] << 24) |
(c->buffer_ptr[1] << 16) |
(c->buffer_ptr[2] << 8) |
(c->buffer_ptr[3]);
if (len > (c->buffer_end - c->buffer_ptr))
goto fail1;
if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP) {
/* RTP packets are sent inside the RTSP TCP connection */
ByteIOContext pb1, *pb = &pb1;
int interleaved_index, size;
uint8_t header[4];
HTTPContext *rtsp_c;
rtsp_c = c->rtsp_c;
/* if no RTSP connection left, error */
if (!rtsp_c)
return -1;
/* if already sending something, then wait. */
if (rtsp_c->state != RTSPSTATE_WAIT_REQUEST) {
break;
}
if (url_open_dyn_buf(pb) < 0)
goto fail1;
interleaved_index = c->packet_stream_index * 2;
/* RTCP packets are sent at odd indexes */
if (c->buffer_ptr[1] == 200)
interleaved_index++;
/* write RTSP TCP header */
header[0] = '$';
header[1] = interleaved_index;
header[2] = len >> 8;
header[3] = len;
put_buffer(pb, header, 4);
/* write RTP packet data */
c->buffer_ptr += 4;
put_buffer(pb, c->buffer_ptr, len);
size = url_close_dyn_buf(pb, &c->packet_buffer);
/* prepare asynchronous TCP sending */
rtsp_c->packet_buffer_ptr = c->packet_buffer;
rtsp_c->packet_buffer_end = c->packet_buffer + size;
rtsp_c->state = RTSPSTATE_SEND_PACKET;
} else {
/* send RTP packet directly in UDP */
/* short term bandwidth limitation */
dt = av_gettime() - c->packet_start_time_us;
if (dt < 1)
dt = 1;
if ((c->packet_byte_count + len) * (int64_t)1000000 >=
(SHORT_TERM_BANDWIDTH / 8) * (int64_t)dt) {
/* bandwidth overflow : wait at most one tick and retry */
c->state = HTTPSTATE_WAIT_SHORT;
return 0;
}
c->buffer_ptr += 4;
url_write(c->rtp_handles[c->packet_stream_index],
c->buffer_ptr, len);
}
c->buffer_ptr += len;
c->packet_byte_count += len;
} else {
/* TCP data output */
len = write(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
if (len < 0) {
if (errno != EAGAIN && errno != EINTR) {
/* error : close connection */
return -1;
} else {
return 0;
}
} else {
c->buffer_ptr += len;
}
}
c->data_count += len;
update_datarate(&c->datarate, c->data_count);
if (c->stream)
c->stream->bytes_served += len;
break;
}
c->data_count += len;
update_datarate(&c->datarate, c->data_count);
if (c->stream)
c->stream->bytes_served += len;
}
} /* for(;;) */
return 0;
}
@ -2884,7 +2971,18 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
/* find rtp session, and create it if none found */
rtp_c = find_rtp_session(h->session_id);
if (!rtp_c) {
rtp_c = rtp_new_connection(&c->from_addr, stream, h->session_id);
/* always prefer UDP */
th = find_transport(h, RTSP_PROTOCOL_RTP_UDP);
if (!th) {
th = find_transport(h, RTSP_PROTOCOL_RTP_TCP);
if (!th) {
rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
return;
}
}
rtp_c = rtp_new_connection(&c->from_addr, stream, h->session_id,
th->protocol);
if (!rtp_c) {
rtsp_reply_error(c, RTSP_STATUS_BANDWIDTH);
return;
@ -2895,17 +2993,6 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
return;
}
/* always prefer UDP */
th = find_transport(h, RTSP_PROTOCOL_RTP_UDP);
if (!th) {
th = find_transport(h, RTSP_PROTOCOL_RTP_TCP);
if (!th) {
rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
return;
}
}
rtp_c->rtp_protocol = th->protocol;
}
/* test if stream is OK (test needed because several SETUP needs
@ -2947,7 +3034,7 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
}
/* setup stream */
if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr) < 0) {
if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr, c) < 0) {
rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
return;
}
@ -3096,9 +3183,11 @@ static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPHeader *h)
/* RTP handling */
static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
FFStream *stream, const char *session_id)
FFStream *stream, const char *session_id,
enum RTSPProtocol rtp_protocol)
{
HTTPContext *c = NULL;
const char *proto_str;
/* XXX: should output a warning page when coming
close to the connection limit */
@ -3122,8 +3211,25 @@ static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
pstrcpy(c->session_id, sizeof(c->session_id), session_id);
c->state = HTTPSTATE_READY;
c->is_packetized = 1;
c->rtp_protocol = rtp_protocol;
/* protocol is shown in statistics */
pstrcpy(c->protocol, sizeof(c->protocol), "RTP");
switch(c->rtp_protocol) {
case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
proto_str = "MCAST";
break;
case RTSP_PROTOCOL_RTP_UDP:
proto_str = "UDP";
break;
case RTSP_PROTOCOL_RTP_TCP:
proto_str = "TCP";
break;
default:
proto_str = "???";
break;
}
pstrcpy(c->protocol, sizeof(c->protocol), "RTP/");
pstrcat(c->protocol, sizeof(c->protocol), proto_str);
current_bandwidth += stream->bandwidth;
@ -3140,10 +3246,11 @@ static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
}
/* add a new RTP stream in an RTP connection (used in RTSP SETUP
command). if dest_addr is NULL, then TCP tunneling in RTSP is
command). If RTP/TCP protocol is used, TCP connection 'rtsp_c' is
used. */
static int rtp_new_av_stream(HTTPContext *c,
int stream_index, struct sockaddr_in *dest_addr)
int stream_index, struct sockaddr_in *dest_addr,
HTTPContext *rtsp_c)
{
AVFormatContext *ctx;
AVStream *st;
@ -3151,6 +3258,7 @@ static int rtp_new_av_stream(HTTPContext *c,
URLContext *h;
uint8_t *dummy_buf;
char buf2[32];
int max_packet_size;
/* now we can open the relevant output stream */
ctx = av_mallocz(sizeof(AVFormatContext));
@ -3173,9 +3281,13 @@ static int rtp_new_av_stream(HTTPContext *c,
sizeof(AVStream));
}
if (dest_addr) {
/* build destination RTP address */
ipaddr = inet_ntoa(dest_addr->sin_addr);
/* build destination RTP address */
ipaddr = inet_ntoa(dest_addr->sin_addr);
switch(c->rtp_protocol) {
case RTSP_PROTOCOL_RTP_UDP:
case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
/* RTP/UDP case */
/* XXX: also pass as parameter to function ? */
if (c->stream->is_multicast) {
@ -3194,18 +3306,24 @@ static int rtp_new_av_stream(HTTPContext *c,
if (url_open(&h, ctx->filename, URL_WRONLY) < 0)
goto fail;
c->rtp_handles[stream_index] = h;
} else {
max_packet_size = url_get_max_packet_size(h);
break;
case RTSP_PROTOCOL_RTP_TCP:
/* RTP/TCP case */
c->rtsp_c = rtsp_c;
max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
break;
default:
goto fail;
}
http_log("%s:%d - - [%s] \"RTPSTART %s/streamid=%d\"\n",
http_log("%s:%d - - [%s] \"PLAY %s/streamid=%d %s\"\n",
ipaddr, ntohs(dest_addr->sin_port),
ctime1(buf2),
c->stream->filename, stream_index);
c->stream->filename, stream_index, c->protocol);
/* normally, no packets should be output here, but the packet size may be checked */
if (url_open_dyn_packet_buf(&ctx->pb,
url_get_max_packet_size(h)) < 0) {
if (url_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0) {
/* XXX: close stream */
goto fail;
}
@ -3309,7 +3427,7 @@ static void extract_mpeg4_header(AVFormatContext *infile)
for(i=0;i<infile->nb_streams;i++) {
st = infile->streams[i];
if (st->codec.codec_id == CODEC_ID_MPEG4 &&
st->codec.extradata == NULL) {
st->codec.extradata_size == 0) {
mpeg4_count++;
}
}
@ -3322,7 +3440,8 @@ static void extract_mpeg4_header(AVFormatContext *infile)
break;
st = infile->streams[pkt.stream_index];
if (st->codec.codec_id == CODEC_ID_MPEG4 &&
st->codec.extradata == NULL) {
st->codec.extradata_size == 0) {
av_freep(&st->codec.extradata);
/* fill extradata with the header */
/* XXX: we make hard suppositions here ! */
p = pkt.data;