mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-21 10:55:51 +02:00
added RTP/TCP protocol support
Originally committed as revision 2063 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Fabrice Bellard
parent
416e35081a
commit
bc3513865a
331
ffserver.c
331
ffserver.c
@ -60,6 +60,7 @@ enum HTTPState {
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RTSPSTATE_WAIT_REQUEST,
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RTSPSTATE_SEND_REPLY,
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RTSPSTATE_SEND_PACKET,
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};
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const char *http_state[] = {
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@ -77,6 +78,7 @@ const char *http_state[] = {
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"RTSP_WAIT_REQUEST",
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"RTSP_SEND_REPLY",
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"RTSP_SEND_PACKET",
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};
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#define IOBUFFER_INIT_SIZE 8192
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@ -143,11 +145,16 @@ typedef struct HTTPContext {
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enum RTSPProtocol rtp_protocol;
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char session_id[32]; /* session id */
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AVFormatContext *rtp_ctx[MAX_STREAMS];
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URLContext *rtp_handles[MAX_STREAMS];
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/* RTP short term bandwidth limitation */
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int packet_byte_count;
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int packet_start_time_us; /* used for short durations (a few
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seconds max) */
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/* RTP/UDP specific */
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URLContext *rtp_handles[MAX_STREAMS];
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/* RTP/TCP specific */
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struct HTTPContext *rtsp_c;
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uint8_t *packet_buffer, *packet_buffer_ptr, *packet_buffer_end;
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} HTTPContext;
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static AVFrame dummy_frame;
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@ -259,9 +266,11 @@ static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
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/* RTP handling */
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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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FFStream *stream, const char *session_id,
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enum RTSPProtocol rtp_protocol);
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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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int stream_index, struct sockaddr_in *dest_addr,
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HTTPContext *rtsp_c);
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static const char *my_program_name;
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static const char *my_program_dir;
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@ -289,7 +298,7 @@ static long gettime_ms(void)
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static FILE *logfile = NULL;
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static void http_log(const char *fmt, ...)
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static void __attribute__ ((format (printf, 1, 2))) http_log(const char *fmt, ...)
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{
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va_list ap;
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va_start(ap, fmt);
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@ -477,7 +486,8 @@ static void start_multicast(void)
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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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rtp_c = rtp_new_connection(&dest_addr, stream, session_id,
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RTSP_PROTOCOL_RTP_UDP_MULTICAST);
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if (!rtp_c) {
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continue;
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}
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@ -487,14 +497,12 @@ static void start_multicast(void)
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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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if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr, NULL) < 0) {
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fprintf(stderr, "Could not open output stream '%s/streamid=%d'\n",
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stream->filename, stream_index);
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exit(1);
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@ -551,6 +559,7 @@ static int http_server(void)
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switch(c->state) {
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case HTTPSTATE_SEND_HEADER:
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case RTSPSTATE_SEND_REPLY:
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case RTSPSTATE_SEND_PACKET:
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c->poll_entry = poll_entry;
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poll_entry->fd = fd;
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poll_entry->events = POLLOUT;
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@ -716,6 +725,12 @@ static void close_connection(HTTPContext *c)
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}
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}
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/* remove references, if any (XXX: do it faster) */
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for(c1 = first_http_ctx; c1 != NULL; c1 = c1->next) {
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if (c1->rtsp_c == c)
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c1->rtsp_c = NULL;
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}
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/* remove connection associated resources */
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if (c->fd >= 0)
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close(c->fd);
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@ -746,7 +761,7 @@ static void close_connection(HTTPContext *c)
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url_close(h);
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}
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}
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ctx = &c->fmt_ctx;
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if (!c->last_packet_sent) {
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@ -754,7 +769,7 @@ static void close_connection(HTTPContext *c)
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/* prepare header */
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if (url_open_dyn_buf(&ctx->pb) >= 0) {
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av_write_trailer(ctx);
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(void) url_close_dyn_buf(&ctx->pb, &c->pb_buffer);
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url_close_dyn_buf(&ctx->pb, &c->pb_buffer);
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}
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}
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}
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@ -765,6 +780,7 @@ static void close_connection(HTTPContext *c)
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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_freep(&c->packet_buffer);
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av_free(c->buffer);
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av_free(c);
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nb_connections--;
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@ -917,6 +933,31 @@ static int handle_connection(HTTPContext *c)
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}
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}
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break;
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case RTSPSTATE_SEND_PACKET:
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if (c->poll_entry->revents & (POLLERR | POLLHUP)) {
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av_freep(&c->packet_buffer);
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return -1;
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}
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/* no need to write if no events */
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if (!(c->poll_entry->revents & POLLOUT))
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return 0;
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len = write(c->fd, c->packet_buffer_ptr,
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c->packet_buffer_end - c->packet_buffer_ptr);
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if (len < 0) {
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if (errno != EAGAIN && errno != EINTR) {
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/* error : close connection */
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av_freep(&c->packet_buffer);
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return -1;
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}
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} else {
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c->packet_buffer_ptr += len;
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if (c->packet_buffer_ptr >= c->packet_buffer_end) {
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/* all the buffer was sent : wait for a new request */
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av_freep(&c->packet_buffer);
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c->state = RTSPSTATE_WAIT_REQUEST;
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}
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}
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break;
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case HTTPSTATE_READY:
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/* nothing to do */
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break;
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@ -2087,13 +2128,15 @@ static int compute_send_delay(HTTPContext *c)
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if (datarate > c->stream->bandwidth * 2000) {
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return 1000;
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}
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if(!c->stream->feed && c->first_pts!=AV_NOPTS_VALUE) {
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time_pts = ((int64_t)(cur_time - c->start_time) * c->fmt_in->pts_den) /
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((int64_t) c->fmt_in->pts_num*1000);
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delta_pts = c->cur_pts - time_pts;
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m_delay = (delta_pts * 1000 * c->fmt_in->pts_num) / c->fmt_in->pts_den;
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return m_delay>0 ? m_delay : 0;
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} else return 0;
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if (!c->stream->feed && c->first_pts!=AV_NOPTS_VALUE) {
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time_pts = ((int64_t)(cur_time - c->start_time) * c->fmt_in->pts_den) /
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((int64_t) c->fmt_in->pts_num*1000);
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delta_pts = c->cur_pts - time_pts;
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m_delay = (delta_pts * 1000 * c->fmt_in->pts_num) / c->fmt_in->pts_den;
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return m_delay>0 ? m_delay : 0;
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} else {
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return 0;
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}
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}
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#endif
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@ -2103,6 +2146,7 @@ static int http_prepare_data(HTTPContext *c)
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int i, len, ret;
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AVFormatContext *ctx;
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av_freep(&c->pb_buffer);
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switch(c->state) {
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case HTTPSTATE_SEND_DATA_HEADER:
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memset(&c->fmt_ctx, 0, sizeof(c->fmt_ctx));
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@ -2273,8 +2317,12 @@ static int http_prepare_data(HTTPContext *c)
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#endif
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if (c->is_packetized) {
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ret = url_open_dyn_packet_buf(&ctx->pb,
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url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]));
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int max_packet_size;
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if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP)
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max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
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else
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max_packet_size = url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]);
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ret = url_open_dyn_packet_buf(&ctx->pb, max_packet_size);
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c->packet_byte_count = 0;
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c->packet_start_time_us = av_gettime();
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} else {
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@ -2327,76 +2375,115 @@ static int http_prepare_data(HTTPContext *c)
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#define SHORT_TERM_BANDWIDTH 8000000
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/* should convert the format at the same time */
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/* send data starting at c->buffer_ptr to the output connection
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(either UDP or TCP connection) */
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static int http_send_data(HTTPContext *c)
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{
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int len, ret, dt;
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while (c->buffer_ptr >= c->buffer_end) {
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av_freep(&c->pb_buffer);
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ret = http_prepare_data(c);
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if (ret < 0)
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return -1;
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else if (ret == 0) {
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continue;
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} else {
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/* state change requested */
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return 0;
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}
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}
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if (c->buffer_ptr < c->buffer_end) {
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if (c->is_packetized) {
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/* RTP/UDP data output */
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len = c->buffer_end - c->buffer_ptr;
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if (len < 4) {
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/* fail safe - should never happen */
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fail1:
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c->buffer_ptr = c->buffer_end;
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return 0;
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for(;;) {
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if (c->buffer_ptr >= c->buffer_end) {
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ret = http_prepare_data(c);
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if (ret < 0)
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return -1;
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else if (ret != 0) {
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/* state change requested */
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break;
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}
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len = (c->buffer_ptr[0] << 24) |
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(c->buffer_ptr[1] << 16) |
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(c->buffer_ptr[2] << 8) |
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(c->buffer_ptr[3]);
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if (len > (c->buffer_end - c->buffer_ptr))
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goto fail1;
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/* short term bandwidth limitation */
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dt = av_gettime() - c->packet_start_time_us;
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if (dt < 1)
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dt = 1;
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if ((c->packet_byte_count + len) * (int64_t)1000000 >=
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(SHORT_TERM_BANDWIDTH / 8) * (int64_t)dt) {
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/* bandwidth overflow : wait at most one tick and retry */
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c->state = HTTPSTATE_WAIT_SHORT;
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return 0;
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}
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c->buffer_ptr += 4;
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url_write(c->rtp_handles[c->packet_stream_index],
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c->buffer_ptr, len);
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c->buffer_ptr += len;
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c->packet_byte_count += len;
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} else {
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/* TCP data output */
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len = write(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
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if (len < 0) {
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if (errno != EAGAIN && errno != EINTR) {
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/* error : close connection */
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return -1;
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} else {
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if (c->is_packetized) {
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/* RTP data output */
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len = c->buffer_end - c->buffer_ptr;
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if (len < 4) {
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/* fail safe - should never happen */
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fail1:
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c->buffer_ptr = c->buffer_end;
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return 0;
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}
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} else {
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len = (c->buffer_ptr[0] << 24) |
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(c->buffer_ptr[1] << 16) |
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(c->buffer_ptr[2] << 8) |
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(c->buffer_ptr[3]);
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if (len > (c->buffer_end - c->buffer_ptr))
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goto fail1;
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if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP) {
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/* RTP packets are sent inside the RTSP TCP connection */
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ByteIOContext pb1, *pb = &pb1;
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int interleaved_index, size;
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uint8_t header[4];
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HTTPContext *rtsp_c;
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rtsp_c = c->rtsp_c;
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/* if no RTSP connection left, error */
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if (!rtsp_c)
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return -1;
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/* if already sending something, then wait. */
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if (rtsp_c->state != RTSPSTATE_WAIT_REQUEST) {
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break;
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}
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if (url_open_dyn_buf(pb) < 0)
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goto fail1;
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interleaved_index = c->packet_stream_index * 2;
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/* RTCP packets are sent at odd indexes */
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if (c->buffer_ptr[1] == 200)
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interleaved_index++;
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/* write RTSP TCP header */
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header[0] = '$';
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header[1] = interleaved_index;
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header[2] = len >> 8;
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header[3] = len;
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put_buffer(pb, header, 4);
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/* write RTP packet data */
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c->buffer_ptr += 4;
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put_buffer(pb, c->buffer_ptr, len);
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size = url_close_dyn_buf(pb, &c->packet_buffer);
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/* prepare asynchronous TCP sending */
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rtsp_c->packet_buffer_ptr = c->packet_buffer;
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rtsp_c->packet_buffer_end = c->packet_buffer + size;
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rtsp_c->state = RTSPSTATE_SEND_PACKET;
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} else {
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/* send RTP packet directly in UDP */
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/* short term bandwidth limitation */
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dt = av_gettime() - c->packet_start_time_us;
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if (dt < 1)
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dt = 1;
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if ((c->packet_byte_count + len) * (int64_t)1000000 >=
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(SHORT_TERM_BANDWIDTH / 8) * (int64_t)dt) {
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/* bandwidth overflow : wait at most one tick and retry */
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c->state = HTTPSTATE_WAIT_SHORT;
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return 0;
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}
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c->buffer_ptr += 4;
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url_write(c->rtp_handles[c->packet_stream_index],
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c->buffer_ptr, len);
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}
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c->buffer_ptr += len;
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c->packet_byte_count += len;
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} else {
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/* TCP data output */
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len = write(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
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if (len < 0) {
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if (errno != EAGAIN && errno != EINTR) {
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/* error : close connection */
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return -1;
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} else {
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return 0;
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}
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} else {
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c->buffer_ptr += len;
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}
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}
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c->data_count += len;
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update_datarate(&c->datarate, c->data_count);
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if (c->stream)
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c->stream->bytes_served += len;
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break;
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}
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c->data_count += len;
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update_datarate(&c->datarate, c->data_count);
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if (c->stream)
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c->stream->bytes_served += len;
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}
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} /* for(;;) */
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return 0;
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}
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@ -2884,7 +2971,18 @@ 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->from_addr, stream, h->session_id);
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/* always prefer UDP */
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th = find_transport(h, RTSP_PROTOCOL_RTP_UDP);
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if (!th) {
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th = find_transport(h, RTSP_PROTOCOL_RTP_TCP);
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if (!th) {
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rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
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return;
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}
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}
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rtp_c = rtp_new_connection(&c->from_addr, stream, h->session_id,
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th->protocol);
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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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@ -2895,17 +2993,6 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
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rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
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return;
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}
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/* always prefer UDP */
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th = find_transport(h, RTSP_PROTOCOL_RTP_UDP);
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if (!th) {
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th = find_transport(h, RTSP_PROTOCOL_RTP_TCP);
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if (!th) {
|
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rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
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return;
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}
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}
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rtp_c->rtp_protocol = th->protocol;
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}
|
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/* test if stream is OK (test needed because several SETUP needs
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@ -2947,7 +3034,7 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
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}
|
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|
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/* setup stream */
|
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if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr) < 0) {
|
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if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr, c) < 0) {
|
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rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
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return;
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}
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@ -3096,10 +3183,12 @@ static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPHeader *h)
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/* RTP handling */
|
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|
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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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FFStream *stream, const char *session_id,
|
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enum RTSPProtocol rtp_protocol)
|
||||
{
|
||||
HTTPContext *c = NULL;
|
||||
|
||||
const char *proto_str;
|
||||
|
||||
/* XXX: should output a warning page when coming
|
||||
close to the connection limit */
|
||||
if (nb_connections >= nb_max_connections)
|
||||
@ -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;
|
||||
|
||||
Loading…
Reference in New Issue
Block a user