From bc3513865a10f0f070bdc4eafd4e0df2b2ca3f6d Mon Sep 17 00:00:00 2001 From: Fabrice Bellard Date: Thu, 17 Jul 2003 10:31:23 +0000 Subject: [PATCH] added RTP/TCP protocol support Originally committed as revision 2063 to svn://svn.ffmpeg.org/ffmpeg/trunk --- ffserver.c | 331 ++++++++++++++++++++++++++++++++++++----------------- 1 file changed, 225 insertions(+), 106 deletions(-) diff --git a/ffserver.c b/ffserver.c index a0ded91e6d..d48c7c7a71 100644 --- a/ffserver.c +++ b/ffserver.c @@ -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); @@ -746,7 +761,7 @@ static void close_connection(HTTPContext *c) url_close(h); } } - + ctx = &c->fmt_ctx; if (!c->last_packet_sent) { @@ -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; } @@ -3096,10 +3183,12 @@ 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 */ 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;inb_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;