From 469c335c55674b31069aaadaaae014b33def1dff Mon Sep 17 00:00:00 2001 From: "Reynaldo H. Verdejo Pinochet" Date: Wed, 24 Jun 2015 18:49:38 -0300 Subject: [PATCH] ffserver: unify comment formating & drop unneeded braces Signed-off-by: Reynaldo H. Verdejo Pinochet --- ffserver.c | 114 ++++++++++++++++++++++++++++------------------------- 1 file changed, 60 insertions(+), 54 deletions(-) diff --git a/ffserver.c b/ffserver.c index 6b537b3304..1be545f909 100644 --- a/ffserver.c +++ b/ffserver.c @@ -31,7 +31,7 @@ #include #include #include "libavformat/avformat.h" -// FIXME those are internal headers, ffserver _really_ shouldn't use them +/* FIXME: those are internal headers, ffserver _really_ shouldn't use them */ #include "libavformat/ffm.h" #include "libavformat/network.h" #include "libavformat/os_support.h" @@ -251,7 +251,8 @@ static unsigned int nb_connections; static uint64_t current_bandwidth; -static int64_t cur_time; // Making this global saves on passing it around everywhere +/* Making this global saves on passing it around everywhere */ +static int64_t cur_time; static AVLFG random_state; @@ -630,9 +631,8 @@ static int http_server(void) poll_entry++; } else { /* when ffserver is doing the timing, we work by - looking at which packet needs to be sent every - 10 ms */ - /* one tick wait XXX: 10 ms assumed */ + * looking at which packet needs to be sent every + * 10 ms (one tick wait XXX: 10 ms assumed) */ if (delay > 10) delay = 10; } @@ -655,7 +655,7 @@ static int http_server(void) } /* wait for an event on one connection. We poll at least every - second to handle timeouts */ + * second to handle timeouts */ do { ret = poll(poll_table, poll_entry - poll_table, delay); if (ret < 0 && ff_neterrno() != AVERROR(EAGAIN) && @@ -900,11 +900,11 @@ static int handle_connection(HTTPContext *c) if ((ptr >= c->buffer + 2 && !memcmp(ptr-2, "\n\n", 2)) || (ptr >= c->buffer + 4 && !memcmp(ptr-4, "\r\n\r\n", 4))) { /* request found : parse it and reply */ - if (c->state == HTTPSTATE_WAIT_REQUEST) { + if (c->state == HTTPSTATE_WAIT_REQUEST) ret = http_parse_request(c); - } else { + else ret = rtsp_parse_request(c); - } + if (ret < 0) return -1; } else if (ptr >= c->buffer_end) { @@ -949,8 +949,8 @@ static int handle_connection(HTTPContext *c) case HTTPSTATE_SEND_DATA_HEADER: case HTTPSTATE_SEND_DATA_TRAILER: /* for packetized output, we consider we can always write (the - input streams set the speed). It may be better to verify - that we do not rely too much on the kernel queues */ + * input streams set the speed). It may be better to verify + * that we do not rely too much on the kernel queues */ if (!c->is_packetized) { if (c->poll_entry->revents & (POLLERR | POLLHUP)) return -1; @@ -1277,8 +1277,10 @@ static int validate_acl(FFServerStream *stream, HTTPContext *c) return ret; } -/* compute the real filename of a file by matching it without its - extensions to all the stream's filenames */ +/** + * compute the real filename of a file by matching it without its + * extensions to all the stream's filenames + */ static void compute_real_filename(char *filename, int max_size) { char file1[1024]; @@ -1396,7 +1398,7 @@ static int http_parse_request(HTTPContext *c) compute_real_filename(filename, sizeof(filename) - 1); } - // "redirect" / request to index.html + /* "redirect" request to index.html */ if (!strlen(filename)) av_strlcpy(filename, "index.html", sizeof(filename) - 1); @@ -1735,8 +1737,9 @@ static int http_parse_request(HTTPContext *c) return 0; send_status: compute_status(c); - c->http_error = 200; /* horrible : we use this value to avoid - going to the send data state */ + /* horrible: we use this value to avoid + * going to the send data state */ + c->http_error = 200; c->state = HTTPSTATE_SEND_HEADER; return 0; } @@ -1847,8 +1850,8 @@ static void compute_status(HTTPContext *c) strcpy(eosf - 3, ".ram"); else if (stream->fmt && !strcmp(stream->fmt->name, "rtp")) { /* generate a sample RTSP director if - unicast. Generate an SDP redirector if - multicast */ + * unicast. Generate an SDP redirector if + * multicast */ eosf = strrchr(sfilename, '.'); if (!eosf) eosf = sfilename + strlen(sfilename); @@ -2119,8 +2122,7 @@ static int64_t get_server_clock(HTTPContext *c) return (cur_time - c->start_time) * 1000; } -/* return the estimated time at which the current packet must be sent - (in us) */ +/* return the estimated time (in us) at which the current packet must be sent */ static int64_t get_packet_send_clock(HTTPContext *c) { int bytes_left, bytes_sent, frame_bytes; @@ -2158,7 +2160,8 @@ static int http_prepare_data(HTTPContext *c) AVStream *src; c->fmt_ctx.streams[i] = av_mallocz(sizeof(AVStream)); - /* if file or feed, then just take streams from FFServerStream struct */ + /* if file or feed, then just take streams from FFServerStream + * struct */ if (!c->stream->feed || c->stream->feed == c->stream) src = c->stream->streams[i]; @@ -2223,7 +2226,7 @@ static int http_prepare_data(HTTPContext *c) if (ret < 0) { if (c->stream->feed) { /* if coming from feed, it means we reached the end of the - ffm file, so must wait for more data */ + * ffm file, so must wait for more data */ c->state = HTTPSTATE_WAIT_FEED; return 1; /* state changed */ } @@ -2310,9 +2313,9 @@ static int http_prepare_data(HTTPContext *c) max_packet_size = c->rtp_handles[c->packet_stream_index]->max_packet_size; ret = ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size); - } else { + } else ret = avio_open_dyn_buf(&ctx->pb); - } + if (ret < 0) { /* XXX: potential leak */ return -1; @@ -2375,7 +2378,8 @@ static int http_prepare_data(HTTPContext *c) /* should convert the format at the same time */ /* send data starting at c->buffer_ptr to the output connection - * (either UDP or TCP) */ + * (either UDP or TCP) + */ static int http_send_data(HTTPContext *c) { int len, ret; @@ -2456,8 +2460,8 @@ static int http_send_data(HTTPContext *c) rtsp_c->packet_buffer_ptr += len; if (rtsp_c->packet_buffer_ptr < rtsp_c->packet_buffer_end) { /* if we could not send all the data, we will - send it later, so a new state is needed to - "lock" the RTSP TCP connection */ + * send it later, so a new state is needed to + * "lock" the RTSP TCP connection */ rtsp_c->state = RTSPSTATE_SEND_PACKET; break; } else @@ -2585,12 +2589,11 @@ static int http_receive_data(HTTPContext *c) goto fail; c->buffer_ptr = c->buffer; break; - } else if (++loop_run > 10) { + } else if (++loop_run > 10) /* no chunk header, abort */ goto fail; - } else { + else c->buffer_ptr++; - } } if (c->buffer_end > c->buffer_ptr) { @@ -2623,7 +2626,7 @@ static int http_receive_data(HTTPContext *c) if (c->buffer_ptr >= c->buffer_end) { FFServerStream *feed = c->stream; /* a packet has been received : write it in the store, except - if header */ + * if header */ if (c->data_count > FFM_PACKET_SIZE) { /* XXX: use llseek or url_seek * XXX: Should probably fail? */ @@ -2829,10 +2832,10 @@ static int rtsp_parse_request(HTTPContext *c) the_end: len = avio_close_dyn_buf(c->pb, &c->pb_buffer); c->pb = NULL; /* safety */ - if (len < 0) { + if (len < 0) /* XXX: cannot do more */ return -1; - } + c->buffer_ptr = c->pb_buffer; c->buffer_end = c->pb_buffer + len; c->state = RTSPSTATE_SEND_REPLY; @@ -2851,9 +2854,9 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer, *pbuffer = NULL; avc = avformat_alloc_context(); - if (!avc || !rtp_format) { + if (!avc || !rtp_format) return -1; - } + avc->oformat = rtp_format; av_dict_set(&avc->metadata, "title", entry ? entry->value : "No Title", 0); @@ -2862,9 +2865,8 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer, snprintf(avc->filename, 1024, "rtp://%s:%d?multicast=1?ttl=%d", inet_ntoa(stream->multicast_ip), stream->multicast_port, stream->multicast_ttl); - } else { + } else snprintf(avc->filename, 1024, "rtp://0.0.0.0"); - } avc->streams = av_malloc_array(avc->nb_streams, sizeof(*avc->streams)); if (!avc->streams) @@ -2894,7 +2896,7 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer, static void rtsp_cmd_options(HTTPContext *c, const char *url) { -// rtsp_reply_header(c, RTSP_STATUS_OK); + /* rtsp_reply_header(c, RTSP_STATUS_OK); */ avio_printf(c->pb, "RTSP/1.0 %d %s\r\n", RTSP_STATUS_OK, "OK"); avio_printf(c->pb, "CSeq: %d\r\n", c->seq); avio_printf(c->pb, "Public: %s\r\n", @@ -3061,7 +3063,7 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url, } /* test if stream is OK (test needed because several SETUP needs - to be done for a given file) */ + * to be done for a given file) */ if (rtp_c->stream != stream) { rtsp_reply_error(c, RTSP_STATUS_SERVICE); return; @@ -3122,8 +3124,10 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url, } -/* find an RTP connection by using the session ID. Check consistency - with filename */ +/** + * find an RTP connection by using the session ID. Check consistency + * with filename + */ static HTTPContext *find_rtp_session_with_url(const char *url, const char *session_id) { @@ -3146,10 +3150,10 @@ static HTTPContext *find_rtp_session_with_url(const char *url, for(s=0; sstream->nb_streams; ++s) { snprintf(buf, sizeof(buf), "%s/streamid=%d", rtp_c->stream->filename, s); - if(!strncmp(path, buf, sizeof(buf))) { - // XXX: Should we reply with RTSP_STATUS_ONLY_AGGREGATE if nb_streams>1? + if(!strncmp(path, buf, sizeof(buf))) + /* XXX: Should we reply with RTSP_STATUS_ONLY_AGGREGATE + * if nb_streams>1? */ return rtp_c; - } } len = strlen(path); if (len > 0 && path[len - 1] == '/' && @@ -3227,7 +3231,7 @@ static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr, const char *proto_str; /* XXX: should output a warning page when coming - close to the connection limit */ + * close to the connection limit */ if (nb_connections >= config.nb_max_connections) goto fail; @@ -3282,9 +3286,11 @@ static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr, return NULL; } -/* add a new RTP stream in an RTP connection (used in RTSP SETUP - command). If RTP/TCP protocol is used, TCP connection 'rtsp_c' is - used. */ +/** + * add a new RTP stream in an RTP connection (used in RTSP SETUP + * 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, HTTPContext *rtsp_c) @@ -3362,10 +3368,10 @@ static int rtp_new_av_stream(HTTPContext *c, /* normally, no packets should be output here, but the packet size may * be checked */ - if (ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0) { + if (ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0) /* XXX: close stream */ goto fail; - } + if (avformat_write_header(ctx, NULL) < 0) { fail: if (h) @@ -3402,12 +3408,12 @@ static AVStream *add_av_stream1(FFServerStream *stream, return NULL; } avcodec_copy_context(fst->codec, codec); - } else { + } else /* live streams must use the actual feed's codec since it may be * updated later to carry extradata needed by them. */ fst->codec = codec; - } + fst->priv_data = av_mallocz(sizeof(FeedData)); fst->index = stream->nb_streams; avpriv_set_pts_info(fst, 33, 1, 90000); @@ -3539,7 +3545,7 @@ static void build_file_streams(void) /* open stream */ if (stream->fmt && !strcmp(stream->fmt->name, "rtp")) { /* specific case : if transport stream output to RTP, - we use a raw transport stream reader */ + * we use a raw transport stream reader */ av_dict_set(&stream->in_opts, "mpeg2ts_compute_pcr", "1", 0); } @@ -3561,7 +3567,7 @@ static void build_file_streams(void) remove_stream(stream); } else { /* find all the AVStreams inside and reference them in - 'stream' */ + * 'stream' */ if (avformat_find_stream_info(infile, NULL) < 0) { http_log("Could not find codec parameters from '%s'\n", stream->feed_filename);