mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2024-12-14 19:25:01 +00:00
RTP multicast begins to work in MPEG1 - simplified stream bandwidth computation (no need to recompute it at each request)
Originally committed as revision 1260 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Fabrice Bellard
parent
d2f48f3555
commit
6edd6884b5
212
ffserver.c
212
ffserver.c
@ -120,7 +120,6 @@ typedef struct HTTPContext {
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AVFormatContext fmt_ctx; /* instance of FFStream for one user */
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int last_packet_sent; /* true if last data packet was sent */
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int suppress_log;
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int bandwidth;
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DataRateData datarate;
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int wmp_client_id;
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char protocol[16];
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@ -190,12 +189,15 @@ typedef struct FFStream {
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time_t pid_start; /* Of ffmpeg process */
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char **child_argv;
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struct FFStream *next;
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int bandwidth; /* bandwidth, in kbits/s */
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/* RTSP options */
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char *rtsp_option;
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/* multicast specific */
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int is_multicast;
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struct in_addr multicast_ip;
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int multicast_port; /* first port used for multicast */
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int multicast_ttl;
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int loop; /* if true, send the stream in loops (only meaningful if file) */
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/* feed specific */
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int feed_opened; /* true if someone is writing to the feed */
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@ -247,7 +249,7 @@ static int prepare_sdp_description(FFStream *stream, UINT8 **pbuffer,
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struct in_addr my_ip);
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/* RTP handling */
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static HTTPContext *rtp_new_connection(HTTPContext *rtsp_c,
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static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
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FFStream *stream, const char *session_id);
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static int rtp_new_av_stream(HTTPContext *c,
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int stream_index, struct sockaddr_in *dest_addr);
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@ -263,8 +265,8 @@ static int need_to_start_children;
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int nb_max_connections;
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int nb_connections;
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int nb_max_bandwidth;
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int nb_bandwidth;
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int max_bandwidth;
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int current_bandwidth;
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static long cur_time; // Making this global saves on passing it around everywhere
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@ -290,25 +292,31 @@ static void http_log(char *fmt, ...)
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va_end(ap);
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}
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static void log_connection(HTTPContext *c)
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static char *ctime1(char *buf2)
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{
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char buf1[32], buf2[32], *p;
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time_t ti;
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char *p;
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if (c->suppress_log)
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return;
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/* XXX: reentrant function ? */
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p = inet_ntoa(c->from_addr.sin_addr);
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strcpy(buf1, p);
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ti = time(NULL);
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p = ctime(&ti);
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strcpy(buf2, p);
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p = buf2 + strlen(p) - 1;
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if (*p == '\n')
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*p = '\0';
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return buf2;
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}
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static void log_connection(HTTPContext *c)
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{
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char buf2[32];
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if (c->suppress_log)
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return;
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http_log("%s - - [%s] \"%s %s %s\" %d %lld\n",
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buf1, buf2, c->method, c->url, c->protocol, (c->http_error ? c->http_error : 200), c->data_count);
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inet_ntoa(c->from_addr.sin_addr),
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ctime1(buf2), c->method, c->url,
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c->protocol, (c->http_error ? c->http_error : 200), c->data_count);
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}
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static void update_datarate(DataRateData *drd, INT64 count)
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@ -340,7 +348,6 @@ static int get_longterm_datarate(DataRateData *drd, INT64 count)
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/* You get the first 3 seconds flat out */
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if (cur_time - drd->time1 < 3000)
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return 0;
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return compute_datarate(drd, count);
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}
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@ -431,6 +438,61 @@ static int socket_open_listen(struct sockaddr_in *my_addr)
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return server_fd;
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}
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/* start all multicast streams */
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static void start_multicast(void)
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{
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FFStream *stream;
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char session_id[32];
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HTTPContext *rtp_c;
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struct sockaddr_in dest_addr;
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int default_port, stream_index;
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default_port = 6000;
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for(stream = first_stream; stream != NULL; stream = stream->next) {
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if (stream->is_multicast) {
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/* open the RTP connection */
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snprintf(session_id, sizeof(session_id),
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"%08x%08x", (int)random(), (int)random());
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/* choose a port if none given */
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if (stream->multicast_port == 0) {
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stream->multicast_port = default_port;
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default_port += 100;
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}
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dest_addr.sin_family = AF_INET;
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dest_addr.sin_addr = stream->multicast_ip;
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dest_addr.sin_port = htons(stream->multicast_port);
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rtp_c = rtp_new_connection(&dest_addr, stream, session_id);
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if (!rtp_c) {
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continue;
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}
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if (open_input_stream(rtp_c, "") < 0) {
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fprintf(stderr, "Could not open input stream for stream '%s'\n",
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stream->filename);
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continue;
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}
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rtp_c->rtp_protocol = RTSP_PROTOCOL_RTP_UDP_MULTICAST;
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/* open each RTP stream */
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for(stream_index = 0; stream_index < stream->nb_streams;
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stream_index++) {
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dest_addr.sin_port = htons(stream->multicast_port +
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2 * stream_index);
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if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr) < 0) {
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fprintf(stderr, "Could not open input stream %d for stream '%s'\n",
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stream_index, stream->filename);
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continue;
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}
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}
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/* change state to send data */
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rtp_c->state = HTTPSTATE_SEND_DATA;
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}
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}
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}
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/* main loop of the http server */
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static int http_server(void)
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@ -454,6 +516,9 @@ static int http_server(void)
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first_http_ctx = NULL;
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nb_connections = 0;
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first_http_ctx = NULL;
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start_multicast();
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for(;;) {
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poll_entry = poll_table;
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poll_entry->fd = server_fd;
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@ -669,7 +734,8 @@ static void close_connection(HTTPContext *c)
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}
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}
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nb_bandwidth -= c->bandwidth;
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if (c->stream)
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current_bandwidth -= c->stream->bandwidth;
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av_freep(&c->pb_buffer);
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av_free(c->buffer);
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av_free(c);
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@ -1125,9 +1191,7 @@ static int http_parse_request(HTTPContext *c)
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compute_real_filename(filename, sizeof(url) - 1);
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} else if (match_ext(filename, "sdp")) {
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redir_type = REDIR_SDP;
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printf("before %s\n", filename);
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compute_real_filename(filename, sizeof(url) - 1);
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printf("after %s\n", filename);
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}
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stream = first_stream;
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@ -1174,26 +1238,10 @@ static int http_parse_request(HTTPContext *c)
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}
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if (post == 0 && stream->stream_type == STREAM_TYPE_LIVE) {
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/* See if we meet the bandwidth requirements */
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for(i=0;i<stream->nb_streams;i++) {
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AVStream *st = stream->streams[i];
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switch(st->codec.codec_type) {
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case CODEC_TYPE_AUDIO:
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c->bandwidth += st->codec.bit_rate;
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break;
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case CODEC_TYPE_VIDEO:
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c->bandwidth += st->codec.bit_rate;
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break;
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default:
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av_abort();
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}
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}
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current_bandwidth += stream->bandwidth;
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}
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c->bandwidth /= 1000;
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nb_bandwidth += c->bandwidth;
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if (post == 0 && nb_max_bandwidth < nb_bandwidth) {
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if (post == 0 && max_bandwidth < current_bandwidth) {
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c->http_error = 200;
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q = c->buffer;
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q += sprintf(q, "HTTP/1.0 200 Server too busy\r\n");
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@ -1202,7 +1250,7 @@ static int http_parse_request(HTTPContext *c)
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q += sprintf(q, "<html><head><title>Too busy</title></head><body>\r\n");
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q += sprintf(q, "The server is too busy to serve your request at this time.<p>\r\n");
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q += sprintf(q, "The bandwidth being served (including your stream) is %dkbit/sec, and this exceeds the limit of %dkbit/sec\r\n",
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nb_bandwidth, nb_max_bandwidth);
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current_bandwidth, max_bandwidth);
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q += sprintf(q, "</body></html>\r\n");
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/* prepare output buffer */
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@ -1580,7 +1628,7 @@ static void compute_stats(HTTPContext *c)
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}
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url_fprintf(pb, "<TD align=center> %s <TD align=right> %d <TD align=right> %d <TD> %s %s <TD align=right> %d <TD> %s %s",
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stream->fmt->name,
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(audio_bit_rate + video_bit_rate) / 1000,
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stream->bandwidth,
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video_bit_rate / 1000, video_codec_name, video_codec_name_extra,
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audio_bit_rate / 1000, audio_codec_name, audio_codec_name_extra);
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if (stream->feed) {
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@ -1702,7 +1750,7 @@ static void compute_stats(HTTPContext *c)
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nb_connections, nb_max_connections);
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url_fprintf(pb, "Bandwidth in use: %dk / %dk<BR>\n",
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nb_bandwidth, nb_max_bandwidth);
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current_bandwidth, max_bandwidth);
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url_fprintf(pb, "<TABLE>\n");
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url_fprintf(pb, "<TR><th>#<th>File<th>IP<th>Proto<th>State<th>Target bits/sec<th>Actual bits/sec<th>Bytes transferred\n");
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@ -1999,7 +2047,7 @@ static int compute_send_delay(HTTPContext *c)
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{
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int datarate = 8 * get_longterm_datarate(&c->datarate, c->data_count);
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if (datarate > c->bandwidth * 2000) {
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if (datarate > c->stream->bandwidth * 2000) {
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return 1000;
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}
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return 0;
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@ -2082,6 +2130,7 @@ static int http_prepare_data(HTTPContext *c)
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return 1; /* state changed */
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}
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}
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redo:
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if (av_read_frame(c->fmt_in, &pkt) < 0) {
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if (c->stream->feed && c->stream->feed->feed_opened) {
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/* if coming from feed, it means we reached the end of the
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@ -2089,8 +2138,17 @@ static int http_prepare_data(HTTPContext *c)
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c->state = HTTPSTATE_WAIT_FEED;
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return 1; /* state changed */
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} else {
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/* must send trailer now because eof or error */
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c->state = HTTPSTATE_SEND_DATA_TRAILER;
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if (c->stream->loop) {
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av_close_input_file(c->fmt_in);
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c->fmt_in = NULL;
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if (open_input_stream(c, "") < 0)
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goto no_loop;
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goto redo;
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} else {
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no_loop:
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/* must send trailer now because eof or error */
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c->state = HTTPSTATE_SEND_DATA_TRAILER;
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}
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}
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} else {
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/* update first pts if needed */
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@ -2143,6 +2201,8 @@ static int http_prepare_data(HTTPContext *c)
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c->packet_stream_index = pkt.stream_index;
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ctx = c->rtp_ctx[c->packet_stream_index];
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codec = &ctx->streams[0]->codec;
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/* only one stream per RTP connection */
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pkt.stream_index = 0;
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} else {
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ctx = &c->fmt_ctx;
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/* Fudge here */
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@ -2721,7 +2781,7 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
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/* find rtp session, and create it if none found */
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rtp_c = find_rtp_session(h->session_id);
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if (!rtp_c) {
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rtp_c = rtp_new_connection(c, stream, h->session_id);
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rtp_c = rtp_new_connection(&c->from_addr, stream, h->session_id);
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if (!rtp_c) {
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rtsp_reply_error(c, RTSP_STATUS_BANDWIDTH);
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return;
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@ -2923,7 +2983,7 @@ static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPHeader *h)
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/********************************************************************/
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/* RTP handling */
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static HTTPContext *rtp_new_connection(HTTPContext *rtsp_c,
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static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
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FFStream *stream, const char *session_id)
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{
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HTTPContext *c = NULL;
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@ -2940,7 +3000,7 @@ static HTTPContext *rtp_new_connection(HTTPContext *rtsp_c,
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c->fd = -1;
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c->poll_entry = NULL;
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c->from_addr = rtsp_c->from_addr;
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c->from_addr = *from_addr;
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c->buffer_size = IOBUFFER_INIT_SIZE;
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c->buffer = av_malloc(c->buffer_size);
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if (!c->buffer)
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@ -2953,6 +3013,8 @@ static HTTPContext *rtp_new_connection(HTTPContext *rtsp_c,
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/* protocol is shown in statistics */
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pstrcpy(c->protocol, sizeof(c->protocol), "RTP");
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current_bandwidth += stream->bandwidth;
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c->next = first_http_ctx;
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first_http_ctx = c;
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return c;
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@ -2976,6 +3038,7 @@ static int rtp_new_av_stream(HTTPContext *c,
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char *ipaddr;
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URLContext *h;
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UINT8 *dummy_buf;
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char buf2[32];
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/* now we can open the relevant output stream */
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ctx = av_mallocz(sizeof(AVFormatContext));
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@ -3002,10 +3065,19 @@ static int rtp_new_av_stream(HTTPContext *c,
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/* build destination RTP address */
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ipaddr = inet_ntoa(dest_addr->sin_addr);
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snprintf(ctx->filename, sizeof(ctx->filename),
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"rtp://%s:%d", ipaddr, ntohs(dest_addr->sin_port));
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printf("open %s\n", ctx->filename);
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/* XXX: also pass as parameter to function ? */
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if (c->stream->is_multicast) {
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int ttl;
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ttl = c->stream->multicast_ttl;
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if (!ttl)
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ttl = 16;
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snprintf(ctx->filename, sizeof(ctx->filename),
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"rtp://%s:%d?multicast=1&ttl=%d",
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ipaddr, ntohs(dest_addr->sin_port), ttl);
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} else {
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snprintf(ctx->filename, sizeof(ctx->filename),
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"rtp://%s:%d", ipaddr, ntohs(dest_addr->sin_port));
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}
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if (url_open(&h, ctx->filename, URL_WRONLY) < 0)
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goto fail;
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@ -3014,6 +3086,11 @@ static int rtp_new_av_stream(HTTPContext *c,
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goto fail;
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}
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http_log("%s:%d - - [%s] \"RTPSTART %s/streamid=%d\"\n",
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ipaddr, ntohs(dest_addr->sin_port),
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ctime1(buf2),
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c->stream->filename, stream_index);
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/* normally, no packets should be output here, but the packet size may be checked */
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if (url_open_dyn_packet_buf(&ctx->pb,
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url_get_max_packet_size(h)) < 0) {
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@ -3286,6 +3363,29 @@ void build_feed_streams(void)
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}
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}
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/* compute the bandwidth used by each stream */
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static void compute_bandwidth(void)
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{
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int bandwidth, i;
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FFStream *stream;
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for(stream = first_stream; stream != NULL; stream = stream->next) {
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bandwidth = 0;
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for(i=0;i<stream->nb_streams;i++) {
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AVStream *st = stream->streams[i];
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switch(st->codec.codec_type) {
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case CODEC_TYPE_AUDIO:
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case CODEC_TYPE_VIDEO:
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bandwidth += st->codec.bit_rate;
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break;
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default:
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break;
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}
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}
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stream->bandwidth = (bandwidth + 999) / 1000;
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}
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}
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static void get_arg(char *buf, int buf_size, const char **pp)
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{
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const char *p;
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@ -3519,7 +3619,7 @@ int parse_ffconfig(const char *filename)
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filename, line_num, arg);
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errors++;
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} else {
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nb_max_bandwidth = val;
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max_bandwidth = val;
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}
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} else if (!strcasecmp(cmd, "CustomLog")) {
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get_arg(logfilename, sizeof(logfilename), &p);
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@ -3952,12 +4052,22 @@ int parse_ffconfig(const char *filename)
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errors++;
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}
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stream->is_multicast = 1;
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stream->loop = 1; /* default is looping */
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}
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} else if (!strcasecmp(cmd, "MulticastPort")) {
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get_arg(arg, sizeof(arg), &p);
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if (stream) {
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stream->multicast_port = atoi(arg);
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}
|
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} else if (!strcasecmp(cmd, "MulticastTTL")) {
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get_arg(arg, sizeof(arg), &p);
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if (stream) {
|
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stream->multicast_ttl = atoi(arg);
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}
|
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} else if (!strcasecmp(cmd, "NoLoop")) {
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if (stream) {
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stream->loop = 0;
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}
|
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} else if (!strcasecmp(cmd, "</Stream>")) {
|
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if (!stream) {
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fprintf(stderr, "%s:%d: No corresponding <Stream> for </Stream>\n",
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@ -4162,7 +4272,7 @@ int main(int argc, char **argv)
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my_rtsp_addr.sin_addr.s_addr = htonl (INADDR_ANY);
|
||||
|
||||
nb_max_connections = 5;
|
||||
nb_max_bandwidth = 1000;
|
||||
max_bandwidth = 1000;
|
||||
first_stream = NULL;
|
||||
logfilename[0] = '\0';
|
||||
|
||||
@ -4180,6 +4290,8 @@ int main(int argc, char **argv)
|
||||
|
||||
build_feed_streams();
|
||||
|
||||
compute_bandwidth();
|
||||
|
||||
/* put the process in background and detach it from its TTY */
|
||||
if (ffserver_daemon) {
|
||||
int pid;
|
||||
|
||||
Loading…
Reference in New Issue
Block a user