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-rw-r--r--ffserver.c4042
1 files changed, 4042 insertions, 0 deletions
diff --git a/ffserver.c b/ffserver.c
new file mode 100644
index 0000000000..1a27583677
--- /dev/null
+++ b/ffserver.c
@@ -0,0 +1,4042 @@
+/*
+ * Copyright (c) 2000, 2001, 2002 Fabrice Bellard
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * multiple format streaming server based on the FFmpeg libraries
+ */
+
+#include "config.h"
+#if !HAVE_CLOSESOCKET
+#define closesocket close
+#endif
+#include <string.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include "libavformat/avformat.h"
+/* FIXME: those are internal headers, ffserver _really_ shouldn't use them */
+#include "libavformat/ffm.h"
+#include "libavformat/network.h"
+#include "libavformat/os_support.h"
+#include "libavformat/rtpdec.h"
+#include "libavformat/rtpproto.h"
+#include "libavformat/rtsp.h"
+#include "libavformat/rtspcodes.h"
+#include "libavformat/avio_internal.h"
+#include "libavformat/internal.h"
+#include "libavformat/url.h"
+
+#include "libavutil/avassert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/lfg.h"
+#include "libavutil/dict.h"
+#include "libavutil/intreadwrite.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/random_seed.h"
+#include "libavutil/parseutils.h"
+#include "libavutil/opt.h"
+#include "libavutil/time.h"
+
+#include <stdarg.h>
+#if HAVE_UNISTD_H
+#include <unistd.h>
+#endif
+#include <fcntl.h>
+#include <sys/ioctl.h>
+#if HAVE_POLL_H
+#include <poll.h>
+#endif
+#include <errno.h>
+#include <time.h>
+#include <sys/wait.h>
+#include <signal.h>
+
+#include "cmdutils.h"
+#include "ffserver_config.h"
+
+#define PATH_LENGTH 1024
+
+const char program_name[] = "ffserver";
+const int program_birth_year = 2000;
+
+static const OptionDef options[];
+
+enum HTTPState {
+ HTTPSTATE_WAIT_REQUEST,
+ HTTPSTATE_SEND_HEADER,
+ HTTPSTATE_SEND_DATA_HEADER,
+ HTTPSTATE_SEND_DATA, /* sending TCP or UDP data */
+ HTTPSTATE_SEND_DATA_TRAILER,
+ HTTPSTATE_RECEIVE_DATA,
+ HTTPSTATE_WAIT_FEED, /* wait for data from the feed */
+ HTTPSTATE_READY,
+
+ RTSPSTATE_WAIT_REQUEST,
+ RTSPSTATE_SEND_REPLY,
+ RTSPSTATE_SEND_PACKET,
+};
+
+static const char * const http_state[] = {
+ "HTTP_WAIT_REQUEST",
+ "HTTP_SEND_HEADER",
+
+ "SEND_DATA_HEADER",
+ "SEND_DATA",
+ "SEND_DATA_TRAILER",
+ "RECEIVE_DATA",
+ "WAIT_FEED",
+ "READY",
+
+ "RTSP_WAIT_REQUEST",
+ "RTSP_SEND_REPLY",
+ "RTSP_SEND_PACKET",
+};
+
+#define IOBUFFER_INIT_SIZE 8192
+
+/* timeouts are in ms */
+#define HTTP_REQUEST_TIMEOUT (15 * 1000)
+#define RTSP_REQUEST_TIMEOUT (3600 * 24 * 1000)
+
+#define SYNC_TIMEOUT (10 * 1000)
+
+typedef struct RTSPActionServerSetup {
+ uint32_t ipaddr;
+ char transport_option[512];
+} RTSPActionServerSetup;
+
+typedef struct {
+ int64_t count1, count2;
+ int64_t time1, time2;
+} DataRateData;
+
+/* context associated with one connection */
+typedef struct HTTPContext {
+ enum HTTPState state;
+ int fd; /* socket file descriptor */
+ struct sockaddr_in from_addr; /* origin */
+ struct pollfd *poll_entry; /* used when polling */
+ int64_t timeout;
+ uint8_t *buffer_ptr, *buffer_end;
+ int http_error;
+ int post;
+ int chunked_encoding;
+ int chunk_size; /* 0 if it needs to be read */
+ struct HTTPContext *next;
+ int got_key_frame; /* stream 0 => 1, stream 1 => 2, stream 2=> 4 */
+ int64_t data_count;
+ /* feed input */
+ int feed_fd;
+ /* input format handling */
+ AVFormatContext *fmt_in;
+ int64_t start_time; /* In milliseconds - this wraps fairly often */
+ int64_t first_pts; /* initial pts value */
+ int64_t cur_pts; /* current pts value from the stream in us */
+ int64_t cur_frame_duration; /* duration of the current frame in us */
+ int cur_frame_bytes; /* output frame size, needed to compute
+ the time at which we send each
+ packet */
+ int pts_stream_index; /* stream we choose as clock reference */
+ int64_t cur_clock; /* current clock reference value in us */
+ /* output format handling */
+ struct FFServerStream *stream;
+ /* -1 is invalid stream */
+ int feed_streams[FFSERVER_MAX_STREAMS]; /* index of streams in the feed */
+ int switch_feed_streams[FFSERVER_MAX_STREAMS]; /* index of streams in the feed */
+ int switch_pending;
+ AVFormatContext fmt_ctx; /* instance of FFServerStream for one user */
+ int last_packet_sent; /* true if last data packet was sent */
+ int suppress_log;
+ DataRateData datarate;
+ int wmp_client_id;
+ char protocol[16];
+ char method[16];
+ char url[128];
+ int buffer_size;
+ uint8_t *buffer;
+ int is_packetized; /* if true, the stream is packetized */
+ int packet_stream_index; /* current stream for output in state machine */
+
+ /* RTSP state specific */
+ uint8_t *pb_buffer; /* XXX: use that in all the code */
+ AVIOContext *pb;
+ int seq; /* RTSP sequence number */
+
+ /* RTP state specific */
+ enum RTSPLowerTransport rtp_protocol;
+ char session_id[32]; /* session id */
+ AVFormatContext *rtp_ctx[FFSERVER_MAX_STREAMS];
+
+ /* RTP/UDP specific */
+ URLContext *rtp_handles[FFSERVER_MAX_STREAMS];
+
+ /* RTP/TCP specific */
+ struct HTTPContext *rtsp_c;
+ uint8_t *packet_buffer, *packet_buffer_ptr, *packet_buffer_end;
+} HTTPContext;
+
+typedef struct FeedData {
+ long long data_count;
+ float avg_frame_size; /* frame size averaged over last frames with exponential mean */
+} FeedData;
+
+static HTTPContext *first_http_ctx;
+
+static FFServerConfig config = {
+ .nb_max_http_connections = 2000,
+ .nb_max_connections = 5,
+ .max_bandwidth = 1000,
+ .use_defaults = 1,
+};
+
+static void new_connection(int server_fd, int is_rtsp);
+static void close_connection(HTTPContext *c);
+
+/* HTTP handling */
+static int handle_connection(HTTPContext *c);
+static inline void print_stream_params(AVIOContext *pb, FFServerStream *stream);
+static void compute_status(HTTPContext *c);
+static int open_input_stream(HTTPContext *c, const char *info);
+static int http_parse_request(HTTPContext *c);
+static int http_send_data(HTTPContext *c);
+static int http_start_receive_data(HTTPContext *c);
+static int http_receive_data(HTTPContext *c);
+
+/* RTSP handling */
+static int rtsp_parse_request(HTTPContext *c);
+static void rtsp_cmd_describe(HTTPContext *c, const char *url);
+static void rtsp_cmd_options(HTTPContext *c, const char *url);
+static void rtsp_cmd_setup(HTTPContext *c, const char *url,
+ RTSPMessageHeader *h);
+static void rtsp_cmd_play(HTTPContext *c, const char *url,
+ RTSPMessageHeader *h);
+static void rtsp_cmd_interrupt(HTTPContext *c, const char *url,
+ RTSPMessageHeader *h, int pause_only);
+
+/* SDP handling */
+static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
+ struct in_addr my_ip);
+
+/* RTP handling */
+static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
+ FFServerStream *stream,
+ const char *session_id,
+ enum RTSPLowerTransport rtp_protocol);
+static int rtp_new_av_stream(HTTPContext *c,
+ int stream_index, struct sockaddr_in *dest_addr,
+ HTTPContext *rtsp_c);
+/* utils */
+static size_t htmlencode (const char *src, char **dest);
+static inline void cp_html_entity (char *buffer, const char *entity);
+static inline int check_codec_match(AVCodecContext *ccf, AVCodecContext *ccs,
+ int stream);
+
+static const char *my_program_name;
+
+static int no_launch;
+static int need_to_start_children;
+
+/* maximum number of simultaneous HTTP connections */
+static unsigned int nb_connections;
+
+static uint64_t current_bandwidth;
+
+/* Making this global saves on passing it around everywhere */
+static int64_t cur_time;
+
+static AVLFG random_state;
+
+static FILE *logfile = NULL;
+
+static inline void cp_html_entity (char *buffer, const char *entity) {
+ if (!buffer || !entity)
+ return;
+ while (*entity)
+ *buffer++ = *entity++;
+}
+
+/**
+ * Substitutes known conflicting chars on a text string with
+ * their corresponding HTML entities.
+ *
+ * Returns the number of bytes in the 'encoded' representation
+ * not including the terminating NUL.
+ */
+static size_t htmlencode (const char *src, char **dest) {
+ const char *amp = "&amp;";
+ const char *lt = "&lt;";
+ const char *gt = "&gt;";
+ const char *start;
+ char *tmp;
+ size_t final_size = 0;
+
+ if (!src)
+ return 0;
+
+ start = src;
+
+ /* Compute needed dest size */
+ while (*src != '\0') {
+ switch(*src) {
+ case 38: /* & */
+ final_size += 5;
+ break;
+ case 60: /* < */
+ case 62: /* > */
+ final_size += 4;
+ break;
+ default:
+ final_size++;
+ }
+ src++;
+ }
+
+ src = start;
+ *dest = av_mallocz(final_size + 1);
+ if (!*dest)
+ return 0;
+
+ /* Build dest */
+ tmp = *dest;
+ while (*src != '\0') {
+ switch(*src) {
+ case 38: /* & */
+ cp_html_entity (tmp, amp);
+ tmp += 5;
+ break;
+ case 60: /* < */
+ cp_html_entity (tmp, lt);
+ tmp += 4;
+ break;
+ case 62: /* > */
+ cp_html_entity (tmp, gt);
+ tmp += 4;
+ break;
+ default:
+ *tmp = *src;
+ tmp += 1;
+ }
+ src++;
+ }
+ *tmp = '\0';
+
+ return final_size;
+}
+
+static int64_t ffm_read_write_index(int fd)
+{
+ uint8_t buf[8];
+
+ if (lseek(fd, 8, SEEK_SET) < 0)
+ return AVERROR(EIO);
+ if (read(fd, buf, 8) != 8)
+ return AVERROR(EIO);
+ return AV_RB64(buf);
+}
+
+static int ffm_write_write_index(int fd, int64_t pos)
+{
+ uint8_t buf[8];
+ int i;
+
+ for(i=0;i<8;i++)
+ buf[i] = (pos >> (56 - i * 8)) & 0xff;
+ if (lseek(fd, 8, SEEK_SET) < 0)
+ goto bail_eio;
+ if (write(fd, buf, 8) != 8)
+ goto bail_eio;
+
+ return 8;
+
+bail_eio:
+ return AVERROR(EIO);
+}
+
+static void ffm_set_write_index(AVFormatContext *s, int64_t pos,
+ int64_t file_size)
+{
+ av_opt_set_int(s, "server_attached", 1, AV_OPT_SEARCH_CHILDREN);
+ av_opt_set_int(s, "ffm_write_index", pos, AV_OPT_SEARCH_CHILDREN);
+ av_opt_set_int(s, "ffm_file_size", file_size, AV_OPT_SEARCH_CHILDREN);
+}
+
+static char *ctime1(char *buf2, size_t buf_size)
+{
+ time_t ti;
+ char *p;
+
+ ti = time(NULL);
+ p = ctime(&ti);
+ if (!p || !*p) {
+ *buf2 = '\0';
+ return buf2;
+ }
+ av_strlcpy(buf2, p, buf_size);
+ p = buf2 + strlen(buf2) - 1;
+ if (*p == '\n')
+ *p = '\0';
+ return buf2;
+}
+
+static void http_vlog(const char *fmt, va_list vargs)
+{
+ static int print_prefix = 1;
+ char buf[32];
+
+ if (!logfile)
+ return;
+
+ if (print_prefix) {
+ ctime1(buf, sizeof(buf));
+ fprintf(logfile, "%s ", buf);
+ }
+ print_prefix = strstr(fmt, "\n") != NULL;
+ vfprintf(logfile, fmt, vargs);
+ fflush(logfile);
+}
+
+#ifdef __GNUC__
+__attribute__ ((format (printf, 1, 2)))
+#endif
+static void http_log(const char *fmt, ...)
+{
+ va_list vargs;
+ va_start(vargs, fmt);
+ http_vlog(fmt, vargs);
+ va_end(vargs);
+}
+
+static void http_av_log(void *ptr, int level, const char *fmt, va_list vargs)
+{
+ static int print_prefix = 1;
+ AVClass *avc = ptr ? *(AVClass**)ptr : NULL;
+ if (level > av_log_get_level())
+ return;
+ if (print_prefix && avc)
+ http_log("[%s @ %p]", avc->item_name(ptr), ptr);
+ print_prefix = strstr(fmt, "\n") != NULL;
+ http_vlog(fmt, vargs);
+}
+
+static void log_connection(HTTPContext *c)
+{
+ if (c->suppress_log)
+ return;
+
+ http_log("%s - - [%s] \"%s %s\" %d %"PRId64"\n",
+ inet_ntoa(c->from_addr.sin_addr), c->method, c->url,
+ c->protocol, (c->http_error ? c->http_error : 200), c->data_count);
+}
+
+static void update_datarate(DataRateData *drd, int64_t count)
+{
+ if (!drd->time1 && !drd->count1) {
+ drd->time1 = drd->time2 = cur_time;
+ drd->count1 = drd->count2 = count;
+ } else if (cur_time - drd->time2 > 5000) {
+ drd->time1 = drd->time2;
+ drd->count1 = drd->count2;
+ drd->time2 = cur_time;
+ drd->count2 = count;
+ }
+}
+
+/* In bytes per second */
+static int compute_datarate(DataRateData *drd, int64_t count)
+{
+ if (cur_time == drd->time1)
+ return 0;
+
+ return ((count - drd->count1) * 1000) / (cur_time - drd->time1);
+}
+
+
+static void start_children(FFServerStream *feed)
+{
+ char *pathname;
+ char *slash;
+ int i;
+ size_t cmd_length;
+
+ if (no_launch)
+ return;
+
+ cmd_length = strlen(my_program_name);
+
+ /**
+ * FIXME: WIP Safeguard. Remove after clearing all harcoded
+ * '1024' path lengths
+ */
+ if (cmd_length > PATH_LENGTH - 1) {
+ http_log("Could not start children. Command line: '%s' exceeds "
+ "path length limit (%d)\n", my_program_name, PATH_LENGTH);
+ return;
+ }
+
+ pathname = av_strdup (my_program_name);
+ if (!pathname) {
+ http_log("Could not allocate memory for children cmd line\n");
+ return;
+ }
+ /* replace "ffserver" with "ffmpeg" in the path of current
+ * program. Ignore user provided path */
+
+ slash = strrchr(pathname, '/');
+ if (!slash)
+ slash = pathname;
+ else
+ slash++;
+ strcpy(slash, "ffmpeg");
+
+ for (; feed; feed = feed->next) {
+
+ if (!feed->child_argv || feed->pid)
+ continue;
+
+ feed->pid_start = time(0);
+
+ feed->pid = fork();
+ if (feed->pid < 0) {
+ http_log("Unable to create children: %s\n", strerror(errno));
+ av_free (pathname);
+ exit(EXIT_FAILURE);
+ }
+
+ if (feed->pid)
+ continue;
+
+ /* In child */
+
+ http_log("Launch command line: ");
+ http_log("%s ", pathname);
+
+ for (i = 1; feed->child_argv[i] && feed->child_argv[i][0]; i++)
+ http_log("%s ", feed->child_argv[i]);
+ http_log("\n");
+
+ for (i = 3; i < 256; i++)
+ close(i);
+
+ if (!config.debug) {
+ if (!freopen("/dev/null", "r", stdin))
+ http_log("failed to redirect STDIN to /dev/null\n;");
+ if (!freopen("/dev/null", "w", stdout))
+ http_log("failed to redirect STDOUT to /dev/null\n;");
+ if (!freopen("/dev/null", "w", stderr))
+ http_log("failed to redirect STDERR to /dev/null\n;");
+ }
+
+ signal(SIGPIPE, SIG_DFL);
+ execvp(pathname, feed->child_argv);
+ av_free (pathname);
+ _exit(1);
+ }
+ av_free (pathname);
+}
+
+/* open a listening socket */
+static int socket_open_listen(struct sockaddr_in *my_addr)
+{
+ int server_fd, tmp;
+
+ server_fd = socket(AF_INET,SOCK_STREAM,0);
+ if (server_fd < 0) {
+ perror ("socket");
+ return -1;
+ }
+
+ tmp = 1;
+ if (setsockopt(server_fd, SOL_SOCKET, SO_REUSEADDR, &tmp, sizeof(tmp)))
+ av_log(NULL, AV_LOG_WARNING, "setsockopt SO_REUSEADDR failed\n");
+
+ my_addr->sin_family = AF_INET;
+ if (bind (server_fd, (struct sockaddr *) my_addr, sizeof (*my_addr)) < 0) {
+ char bindmsg[32];
+ snprintf(bindmsg, sizeof(bindmsg), "bind(port %d)",
+ ntohs(my_addr->sin_port));
+ perror (bindmsg);
+ goto fail;
+ }
+
+ if (listen (server_fd, 5) < 0) {
+ perror ("listen");
+ goto fail;
+ }
+
+ if (ff_socket_nonblock(server_fd, 1) < 0)
+ av_log(NULL, AV_LOG_WARNING, "ff_socket_nonblock failed\n");
+
+ return server_fd;
+
+fail:
+ closesocket(server_fd);
+ return -1;
+}
+
+/* start all multicast streams */
+static void start_multicast(void)
+{
+ FFServerStream *stream;
+ char session_id[32];
+ HTTPContext *rtp_c;
+ struct sockaddr_in dest_addr = {0};
+ int default_port, stream_index;
+ unsigned int random0, random1;
+
+ default_port = 6000;
+ for(stream = config.first_stream; stream; stream = stream->next) {
+
+ if (!stream->is_multicast)
+ continue;
+
+ random0 = av_lfg_get(&random_state);
+ random1 = av_lfg_get(&random_state);
+
+ /* open the RTP connection */
+ snprintf(session_id, sizeof(session_id), "%08x%08x", random0, random1);
+
+ /* choose a port if none given */
+ if (stream->multicast_port == 0) {
+ stream->multicast_port = default_port;
+ default_port += 100;
+ }
+
+ dest_addr.sin_family = AF_INET;
+ 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,
+ RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
+ if (!rtp_c)
+ continue;
+
+ if (open_input_stream(rtp_c, "") < 0) {
+ http_log("Could not open input stream for stream '%s'\n",
+ stream->filename);
+ continue;
+ }
+
+ /* 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, NULL) >= 0)
+ continue;
+
+ http_log("Could not open output stream '%s/streamid=%d'\n",
+ stream->filename, stream_index);
+ exit(1);
+ }
+
+ rtp_c->state = HTTPSTATE_SEND_DATA;
+ }
+}
+
+/* main loop of the HTTP server */
+static int http_server(void)
+{
+ int server_fd = 0, rtsp_server_fd = 0;
+ int ret, delay;
+ struct pollfd *poll_table, *poll_entry;
+ HTTPContext *c, *c_next;
+
+ poll_table = av_mallocz_array(config.nb_max_http_connections + 2,
+ sizeof(*poll_table));
+ if(!poll_table) {
+ http_log("Impossible to allocate a poll table handling %d "
+ "connections.\n", config.nb_max_http_connections);
+ return -1;
+ }
+
+ if (config.http_addr.sin_port) {
+ server_fd = socket_open_listen(&config.http_addr);
+ if (server_fd < 0)
+ goto quit;
+ }
+
+ if (config.rtsp_addr.sin_port) {
+ rtsp_server_fd = socket_open_listen(&config.rtsp_addr);
+ if (rtsp_server_fd < 0) {
+ closesocket(server_fd);
+ goto quit;
+ }
+ }
+
+ if (!rtsp_server_fd && !server_fd) {
+ http_log("HTTP and RTSP disabled.\n");
+ goto quit;
+ }
+
+ http_log("FFserver started.\n");
+
+ start_children(config.first_feed);
+
+ start_multicast();
+
+ for(;;) {
+ poll_entry = poll_table;
+ if (server_fd) {
+ poll_entry->fd = server_fd;
+ poll_entry->events = POLLIN;
+ poll_entry++;
+ }
+ if (rtsp_server_fd) {
+ poll_entry->fd = rtsp_server_fd;
+ poll_entry->events = POLLIN;
+ poll_entry++;
+ }
+
+ /* wait for events on each HTTP handle */
+ c = first_http_ctx;
+ delay = 1000;
+ while (c) {
+ int fd;
+ fd = c->fd;
+ 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;
+ poll_entry++;
+ break;
+ case HTTPSTATE_SEND_DATA_HEADER:
+ case HTTPSTATE_SEND_DATA:
+ case HTTPSTATE_SEND_DATA_TRAILER:
+ if (!c->is_packetized) {
+ /* for TCP, we output as much as we can
+ * (may need to put a limit) */
+ c->poll_entry = poll_entry;
+ poll_entry->fd = fd;
+ poll_entry->events = POLLOUT;
+ 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) */
+ if (delay > 10)
+ delay = 10;
+ }
+ break;
+ case HTTPSTATE_WAIT_REQUEST:
+ case HTTPSTATE_RECEIVE_DATA:
+ case HTTPSTATE_WAIT_FEED:
+ case RTSPSTATE_WAIT_REQUEST:
+ /* need to catch errors */
+ c->poll_entry = poll_entry;
+ poll_entry->fd = fd;
+ poll_entry->events = POLLIN;/* Maybe this will work */
+ poll_entry++;
+ break;
+ default:
+ c->poll_entry = NULL;
+ break;
+ }
+ c = c->next;
+ }
+
+ /* wait for an event on one connection. We poll at least every
+ * second to handle timeouts */
+ do {
+ ret = poll(poll_table, poll_entry - poll_table, delay);
+ if (ret < 0 && ff_neterrno() != AVERROR(EAGAIN) &&
+ ff_neterrno() != AVERROR(EINTR)) {
+ goto quit;
+ }
+ } while (ret < 0);
+
+ cur_time = av_gettime() / 1000;
+
+ if (need_to_start_children) {
+ need_to_start_children = 0;
+ start_children(config.first_feed);
+ }
+
+ /* now handle the events */
+ for(c = first_http_ctx; c; c = c_next) {
+ c_next = c->next;
+ if (handle_connection(c) < 0) {
+ log_connection(c);
+ /* close and free the connection */
+ close_connection(c);
+ }
+ }
+
+ poll_entry = poll_table;
+ if (server_fd) {
+ /* new HTTP connection request ? */
+ if (poll_entry->revents & POLLIN)
+ new_connection(server_fd, 0);
+ poll_entry++;
+ }
+ if (rtsp_server_fd) {
+ /* new RTSP connection request ? */
+ if (poll_entry->revents & POLLIN)
+ new_connection(rtsp_server_fd, 1);
+ }
+ }
+
+quit:
+ av_free(poll_table);
+ return -1;
+}
+
+/* start waiting for a new HTTP/RTSP request */
+static void start_wait_request(HTTPContext *c, int is_rtsp)
+{
+ c->buffer_ptr = c->buffer;
+ c->buffer_end = c->buffer + c->buffer_size - 1; /* leave room for '\0' */
+
+ c->state = is_rtsp ? RTSPSTATE_WAIT_REQUEST : HTTPSTATE_WAIT_REQUEST;
+ c->timeout = cur_time +
+ (is_rtsp ? RTSP_REQUEST_TIMEOUT : HTTP_REQUEST_TIMEOUT);
+}
+
+static void http_send_too_busy_reply(int fd)
+{
+ char buffer[400];
+ int len = snprintf(buffer, sizeof(buffer),
+ "HTTP/1.0 503 Server too busy\r\n"
+ "Content-type: text/html\r\n"
+ "\r\n"
+ "<!DOCTYPE html>\n"
+ "<html><head><title>Too busy</title></head><body>\r\n"
+ "<p>The server is too busy to serve your request at "
+ "this time.</p>\r\n"
+ "<p>The number of current connections is %u, and this "
+ "exceeds the limit of %u.</p>\r\n"
+ "</body></html>\r\n",
+ nb_connections, config.nb_max_connections);
+ av_assert0(len < sizeof(buffer));
+ if (send(fd, buffer, len, 0) < len)
+ av_log(NULL, AV_LOG_WARNING,
+ "Could not send too-busy reply, send() failed\n");
+}
+
+
+static void new_connection(int server_fd, int is_rtsp)
+{
+ struct sockaddr_in from_addr;
+ socklen_t len;
+ int fd;
+ HTTPContext *c = NULL;
+
+ len = sizeof(from_addr);
+ fd = accept(server_fd, (struct sockaddr *)&from_addr,
+ &len);
+ if (fd < 0) {
+ http_log("error during accept %s\n", strerror(errno));
+ return;
+ }
+ if (ff_socket_nonblock(fd, 1) < 0)
+ av_log(NULL, AV_LOG_WARNING, "ff_socket_nonblock failed\n");
+
+ if (nb_connections >= config.nb_max_connections) {
+ http_send_too_busy_reply(fd);
+ goto fail;
+ }
+
+ /* add a new connection */
+ c = av_mallocz(sizeof(HTTPContext));
+ if (!c)
+ goto fail;
+
+ c->fd = fd;
+ c->poll_entry = NULL;
+ c->from_addr = from_addr;
+ c->buffer_size = IOBUFFER_INIT_SIZE;
+ c->buffer = av_malloc(c->buffer_size);
+ if (!c->buffer)
+ goto fail;
+
+ c->next = first_http_ctx;
+ first_http_ctx = c;
+ nb_connections++;
+
+ start_wait_request(c, is_rtsp);
+
+ return;
+
+ fail:
+ if (c) {
+ av_freep(&c->buffer);
+ av_free(c);
+ }
+ closesocket(fd);
+}
+
+static void close_connection(HTTPContext *c)
+{
+ HTTPContext **cp, *c1;
+ int i, nb_streams;
+ AVFormatContext *ctx;
+ AVStream *st;
+
+ /* remove connection from list */
+ cp = &first_http_ctx;
+ while (*cp) {
+ c1 = *cp;
+ if (c1 == c)
+ *cp = c->next;
+ else
+ cp = &c1->next;
+ }
+
+ /* remove references, if any (XXX: do it faster) */
+ for(c1 = first_http_ctx; c1; c1 = c1->next) {
+ if (c1->rtsp_c == c)
+ c1->rtsp_c = NULL;
+ }
+
+ /* remove connection associated resources */
+ if (c->fd >= 0)
+ closesocket(c->fd);
+ if (c->fmt_in) {
+ /* close each frame parser */
+ for(i=0;i<c->fmt_in->nb_streams;i++) {
+ st = c->fmt_in->streams[i];
+ if (st->codec->codec)
+ avcodec_close(st->codec);
+ }
+ avformat_close_input(&c->fmt_in);
+ }
+
+ /* free RTP output streams if any */
+ nb_streams = 0;
+ if (c->stream)
+ nb_streams = c->stream->nb_streams;
+
+ for(i=0;i<nb_streams;i++) {
+ ctx = c->rtp_ctx[i];
+ if (ctx) {
+ av_write_trailer(ctx);
+ av_dict_free(&ctx->metadata);
+ av_freep(&ctx->streams[0]);
+ av_freep(&ctx);
+ }
+ ffurl_close(c->rtp_handles[i]);
+ }
+
+ ctx = &c->fmt_ctx;
+
+ if (!c->last_packet_sent && c->state == HTTPSTATE_SEND_DATA_TRAILER) {
+ /* prepare header */
+ if (ctx->oformat && avio_open_dyn_buf(&ctx->pb) >= 0) {
+ av_write_trailer(ctx);
+ av_freep(&c->pb_buffer);
+ avio_close_dyn_buf(ctx->pb, &c->pb_buffer);
+ }
+ }
+
+ for(i=0; i<ctx->nb_streams; i++)
+ av_freep(&ctx->streams[i]);
+ av_freep(&ctx->streams);
+ av_freep(&ctx->priv_data);
+
+ if (c->stream && !c->post && c->stream->stream_type == STREAM_TYPE_LIVE)
+ current_bandwidth -= c->stream->bandwidth;
+
+ /* signal that there is no feed if we are the feeder socket */
+ if (c->state == HTTPSTATE_RECEIVE_DATA && c->stream) {
+ c->stream->feed_opened = 0;
+ close(c->feed_fd);
+ }
+
+ av_freep(&c->pb_buffer);
+ av_freep(&c->packet_buffer);
+ av_freep(&c->buffer);
+ av_free(c);
+ nb_connections--;
+}
+
+static int handle_connection(HTTPContext *c)
+{
+ int len, ret;
+ uint8_t *ptr;
+
+ switch(c->state) {
+ case HTTPSTATE_WAIT_REQUEST:
+ case RTSPSTATE_WAIT_REQUEST:
+ /* timeout ? */
+ if ((c->timeout - cur_time) < 0)
+ return -1;
+ if (c->poll_entry->revents & (POLLERR | POLLHUP))
+ return -1;
+
+ /* no need to read if no events */
+ if (!(c->poll_entry->revents & POLLIN))
+ return 0;
+ /* read the data */
+ read_loop:
+ if (!(len = recv(c->fd, c->buffer_ptr, 1, 0)))
+ return -1;
+
+ if (len < 0) {
+ if (ff_neterrno() != AVERROR(EAGAIN) &&
+ ff_neterrno() != AVERROR(EINTR))
+ return -1;
+ break;
+ }
+ /* search for end of request. */
+ c->buffer_ptr += len;
+ ptr = c->buffer_ptr;
+ 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)
+ ret = http_parse_request(c);
+ else
+ ret = rtsp_parse_request(c);
+
+ if (ret < 0)
+ return -1;
+ } else if (ptr >= c->buffer_end) {
+ /* request too long: cannot do anything */
+ return -1;
+ } else goto read_loop;
+
+ break;
+
+ case HTTPSTATE_SEND_HEADER:
+ if (c->poll_entry->revents & (POLLERR | POLLHUP))
+ return -1;
+
+ /* no need to write if no events */
+ if (!(c->poll_entry->revents & POLLOUT))
+ return 0;
+ len = send(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr, 0);
+ if (len < 0) {
+ if (ff_neterrno() != AVERROR(EAGAIN) &&
+ ff_neterrno() != AVERROR(EINTR)) {
+ goto close_connection;
+ }
+ break;
+ }
+ c->buffer_ptr += len;
+ if (c->stream)
+ c->stream->bytes_served += len;
+ c->data_count += len;
+ if (c->buffer_ptr >= c->buffer_end) {
+ av_freep(&c->pb_buffer);
+ /* if error, exit */
+ if (c->http_error)
+ return -1;
+ /* all the buffer was sent : synchronize to the incoming
+ * stream */
+ c->state = HTTPSTATE_SEND_DATA_HEADER;
+ c->buffer_ptr = c->buffer_end = c->buffer;
+ }
+ break;
+
+ case HTTPSTATE_SEND_DATA:
+ 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 */
+ if (!c->is_packetized) {
+ if (c->poll_entry->revents & (POLLERR | POLLHUP))
+ return -1;
+
+ /* no need to read if no events */
+ if (!(c->poll_entry->revents & POLLOUT))
+ return 0;
+ }
+ if (http_send_data(c) < 0)
+ return -1;
+ /* close connection if trailer sent */
+ if (c->state == HTTPSTATE_SEND_DATA_TRAILER)
+ return -1;
+ /* Check if it is a single jpeg frame 123 */
+ if (c->stream->single_frame && c->data_count > c->cur_frame_bytes && c->cur_frame_bytes > 0) {
+ close_connection(c);
+ }
+ break;
+ case HTTPSTATE_RECEIVE_DATA:
+ /* no need to read if no events */
+ if (c->poll_entry->revents & (POLLERR | POLLHUP))
+ return -1;
+ if (!(c->poll_entry->revents & POLLIN))
+ return 0;
+ if (http_receive_data(c) < 0)
+ return -1;
+ break;
+ case HTTPSTATE_WAIT_FEED:
+ /* no need to read if no events */
+ if (c->poll_entry->revents & (POLLIN | POLLERR | POLLHUP))
+ return -1;
+
+ /* nothing to do, we'll be waken up by incoming feed packets */
+ break;
+
+ case RTSPSTATE_SEND_REPLY:
+ if (c->poll_entry->revents & (POLLERR | POLLHUP))
+ goto close_connection;
+ /* no need to write if no events */
+ if (!(c->poll_entry->revents & POLLOUT))
+ return 0;
+ len = send(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr, 0);
+ if (len < 0) {
+ if (ff_neterrno() != AVERROR(EAGAIN) &&
+ ff_neterrno() != AVERROR(EINTR)) {
+ goto close_connection;
+ }
+ break;
+ }
+ c->buffer_ptr += len;
+ c->data_count += len;
+ if (c->buffer_ptr >= c->buffer_end) {
+ /* all the buffer was sent : wait for a new request */
+ av_freep(&c->pb_buffer);
+ start_wait_request(c, 1);
+ }
+ 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 = send(c->fd, c->packet_buffer_ptr,
+ c->packet_buffer_end - c->packet_buffer_ptr, 0);
+ if (len < 0) {
+ if (ff_neterrno() != AVERROR(EAGAIN) &&
+ ff_neterrno() != AVERROR(EINTR)) {
+ /* error : close connection */
+ av_freep(&c->packet_buffer);
+ return -1;
+ }
+ break;
+ }
+ 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;
+ default:
+ return -1;
+ }
+ return 0;
+
+close_connection:
+ av_freep(&c->pb_buffer);
+ return -1;
+}
+
+static int extract_rates(char *rates, int ratelen, const char *request)
+{
+ const char *p;
+
+ for (p = request; *p && *p != '\r' && *p != '\n'; ) {
+ if (av_strncasecmp(p, "Pragma:", 7) == 0) {
+ const char *q = p + 7;
+
+ while (*q && *q != '\n' && av_isspace(*q))
+ q++;
+
+ if (av_strncasecmp(q, "stream-switch-entry=", 20) == 0) {
+ int stream_no;
+ int rate_no;
+
+ q += 20;
+
+ memset(rates, 0xff, ratelen);
+
+ while (1) {
+ while (*q && *q != '\n' && *q != ':')
+ q++;
+
+ if (sscanf(q, ":%d:%d", &stream_no, &rate_no) != 2)
+ break;
+
+ stream_no--;
+ if (stream_no < ratelen && stream_no >= 0)
+ rates[stream_no] = rate_no;
+
+ while (*q && *q != '\n' && !av_isspace(*q))
+ q++;
+ }
+
+ return 1;
+ }
+ }
+ p = strchr(p, '\n');
+ if (!p)
+ break;
+
+ p++;
+ }
+
+ return 0;
+}
+
+static int find_stream_in_feed(FFServerStream *feed, AVCodecContext *codec,
+ int bit_rate)
+{
+ int i;
+ int best_bitrate = 100000000;
+ int best = -1;
+
+ for (i = 0; i < feed->nb_streams; i++) {
+ AVCodecContext *feed_codec = feed->streams[i]->codec;
+
+ if (feed_codec->codec_id != codec->codec_id ||
+ feed_codec->sample_rate != codec->sample_rate ||
+ feed_codec->width != codec->width ||
+ feed_codec->height != codec->height)
+ continue;
+
+ /* Potential stream */
+
+ /* We want the fastest stream less than bit_rate, or the slowest
+ * faster than bit_rate
+ */
+
+ if (feed_codec->bit_rate <= bit_rate) {
+ if (best_bitrate > bit_rate ||
+ feed_codec->bit_rate > best_bitrate) {
+ best_bitrate = feed_codec->bit_rate;
+ best = i;
+ }
+ continue;
+ }
+ if (feed_codec->bit_rate < best_bitrate) {
+ best_bitrate = feed_codec->bit_rate;
+ best = i;
+ }
+ }
+ return best;
+}
+
+static int modify_current_stream(HTTPContext *c, char *rates)
+{
+ int i;
+ FFServerStream *req = c->stream;
+ int action_required = 0;
+
+ /* Not much we can do for a feed */
+ if (!req->feed)
+ return 0;
+
+ for (i = 0; i < req->nb_streams; i++) {
+ AVCodecContext *codec = req->streams[i]->codec;
+
+ switch(rates[i]) {
+ case 0:
+ c->switch_feed_streams[i] = req->feed_streams[i];
+ break;
+ case 1:
+ c->switch_feed_streams[i] = find_stream_in_feed(req->feed, codec, codec->bit_rate / 2);
+ break;
+ case 2:
+ /* Wants off or slow */
+ c->switch_feed_streams[i] = find_stream_in_feed(req->feed, codec, codec->bit_rate / 4);
+#ifdef WANTS_OFF
+ /* This doesn't work well when it turns off the only stream! */
+ c->switch_feed_streams[i] = -2;
+ c->feed_streams[i] = -2;
+#endif
+ break;
+ }
+
+ if (c->switch_feed_streams[i] >= 0 &&
+ c->switch_feed_streams[i] != c->feed_streams[i]) {
+ action_required = 1;
+ }
+ }
+
+ return action_required;
+}
+
+static void get_word(char *buf, int buf_size, const char **pp)
+{
+ const char *p;
+ char *q;
+
+ p = *pp;
+ p += strspn(p, SPACE_CHARS);
+ q = buf;
+ while (!av_isspace(*p) && *p != '\0') {
+ if ((q - buf) < buf_size - 1)
+ *q++ = *p;
+ p++;
+ }
+ if (buf_size > 0)
+ *q = '\0';
+ *pp = p;
+}
+
+static FFServerIPAddressACL* parse_dynamic_acl(FFServerStream *stream,
+ HTTPContext *c)
+{
+ FILE* f;
+ char line[1024];
+ char cmd[1024];
+ FFServerIPAddressACL *acl = NULL;
+ int line_num = 0;
+ const char *p;
+
+ f = fopen(stream->dynamic_acl, "r");
+ if (!f) {
+ perror(stream->dynamic_acl);
+ return NULL;
+ }
+
+ acl = av_mallocz(sizeof(FFServerIPAddressACL));
+ if (!acl) {
+ fclose(f);
+ return NULL;
+ }
+
+ /* Build ACL */
+ while (fgets(line, sizeof(line), f)) {
+ line_num++;
+ p = line;
+ while (av_isspace(*p))
+ p++;
+ if (*p == '\0' || *p == '#')
+ continue;
+ ffserver_get_arg(cmd, sizeof(cmd), &p);
+
+ if (!av_strcasecmp(cmd, "ACL"))
+ ffserver_parse_acl_row(NULL, NULL, acl, p, stream->dynamic_acl,
+ line_num);
+ }
+ fclose(f);
+ return acl;
+}
+
+
+static void free_acl_list(FFServerIPAddressACL *in_acl)
+{
+ FFServerIPAddressACL *pacl, *pacl2;
+
+ pacl = in_acl;
+ while(pacl) {
+ pacl2 = pacl;
+ pacl = pacl->next;
+ av_freep(pacl2);
+ }
+}
+
+static int validate_acl_list(FFServerIPAddressACL *in_acl, HTTPContext *c)
+{
+ enum FFServerIPAddressAction last_action = IP_DENY;
+ FFServerIPAddressACL *acl;
+ struct in_addr *src = &c->from_addr.sin_addr;
+ unsigned long src_addr = src->s_addr;
+
+ for (acl = in_acl; acl; acl = acl->next) {
+ if (src_addr >= acl->first.s_addr && src_addr <= acl->last.s_addr)
+ return (acl->action == IP_ALLOW) ? 1 : 0;
+ last_action = acl->action;
+ }
+
+ /* Nothing matched, so return not the last action */
+ return (last_action == IP_DENY) ? 1 : 0;
+}
+
+static int validate_acl(FFServerStream *stream, HTTPContext *c)
+{
+ int ret = 0;
+ FFServerIPAddressACL *acl;
+
+ /* if stream->acl is null validate_acl_list will return 1 */
+ ret = validate_acl_list(stream->acl, c);
+
+ if (stream->dynamic_acl[0]) {
+ acl = parse_dynamic_acl(stream, c);
+ ret = validate_acl_list(acl, c);
+ free_acl_list(acl);
+ }
+
+ return ret;
+}
+
+/**
+ * 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];
+ char file2[1024];
+ char *p;
+ FFServerStream *stream;
+
+ av_strlcpy(file1, filename, sizeof(file1));
+ p = strrchr(file1, '.');
+ if (p)
+ *p = '\0';
+ for(stream = config.first_stream; stream; stream = stream->next) {
+ av_strlcpy(file2, stream->filename, sizeof(file2));
+ p = strrchr(file2, '.');
+ if (p)
+ *p = '\0';
+ if (!strcmp(file1, file2)) {
+ av_strlcpy(filename, stream->filename, max_size);
+ break;
+ }
+ }
+}
+
+enum RedirType {
+ REDIR_NONE,
+ REDIR_ASX,
+ REDIR_RAM,
+ REDIR_ASF,
+ REDIR_RTSP,
+ REDIR_SDP,
+};
+
+/* parse HTTP request and prepare header */
+static int http_parse_request(HTTPContext *c)
+{
+ const char *p;
+ char *p1;
+ enum RedirType redir_type;
+ char cmd[32];
+ char info[1024], filename[1024];
+ char url[1024], *q;
+ char protocol[32];
+ char msg[1024];
+ char *encoded_msg = NULL;
+ const char *mime_type;
+ FFServerStream *stream;
+ int i;
+ char ratebuf[32];
+ const char *useragent = 0;
+
+ p = c->buffer;
+ get_word(cmd, sizeof(cmd), &p);
+ av_strlcpy(c->method, cmd, sizeof(c->method));
+
+ if (!strcmp(cmd, "GET"))
+ c->post = 0;
+ else if (!strcmp(cmd, "POST"))
+ c->post = 1;
+ else
+ return -1;
+
+ get_word(url, sizeof(url), &p);
+ av_strlcpy(c->url, url, sizeof(c->url));
+
+ get_word(protocol, sizeof(protocol), (const char **)&p);
+ if (strcmp(protocol, "HTTP/1.0") && strcmp(protocol, "HTTP/1.1"))
+ return -1;
+
+ av_strlcpy(c->protocol, protocol, sizeof(c->protocol));
+
+ if (config.debug)
+ http_log("%s - - New connection: %s %s\n",
+ inet_ntoa(c->from_addr.sin_addr), cmd, url);
+
+ /* find the filename and the optional info string in the request */
+ p1 = strchr(url, '?');
+ if (p1) {
+ av_strlcpy(info, p1, sizeof(info));
+ *p1 = '\0';
+ } else
+ info[0] = '\0';
+
+ av_strlcpy(filename, url + ((*url == '/') ? 1 : 0), sizeof(filename)-1);
+
+ for (p = c->buffer; *p && *p != '\r' && *p != '\n'; ) {
+ if (av_strncasecmp(p, "User-Agent:", 11) == 0) {
+ useragent = p + 11;
+ if (*useragent && *useragent != '\n' && av_isspace(*useragent))
+ useragent++;
+ break;
+ }
+ p = strchr(p, '\n');
+ if (!p)
+ break;
+
+ p++;
+ }
+
+ redir_type = REDIR_NONE;
+ if (av_match_ext(filename, "asx")) {
+ redir_type = REDIR_ASX;
+ filename[strlen(filename)-1] = 'f';
+ } else if (av_match_ext(filename, "asf") &&
+ (!useragent || av_strncasecmp(useragent, "NSPlayer", 8))) {
+ /* if this isn't WMP or lookalike, return the redirector file */
+ redir_type = REDIR_ASF;
+ } else if (av_match_ext(filename, "rpm,ram")) {
+ redir_type = REDIR_RAM;
+ strcpy(filename + strlen(filename)-2, "m");
+ } else if (av_match_ext(filename, "rtsp")) {
+ redir_type = REDIR_RTSP;
+ compute_real_filename(filename, sizeof(filename) - 1);
+ } else if (av_match_ext(filename, "sdp")) {
+ redir_type = REDIR_SDP;
+ compute_real_filename(filename, sizeof(filename) - 1);
+ }
+
+ /* "redirect" request to index.html */
+ if (!strlen(filename))
+ av_strlcpy(filename, "index.html", sizeof(filename) - 1);
+
+ stream = config.first_stream;
+ while (stream) {
+ if (!strcmp(stream->filename, filename) && validate_acl(stream, c))
+ break;
+ stream = stream->next;
+ }
+ if (!stream) {
+ snprintf(msg, sizeof(msg), "File '%s' not found", url);
+ http_log("File '%s' not found\n", url);
+ goto send_error;
+ }
+
+ c->stream = stream;
+ memcpy(c->feed_streams, stream->feed_streams, sizeof(c->feed_streams));
+ memset(c->switch_feed_streams, -1, sizeof(c->switch_feed_streams));
+
+ if (stream->stream_type == STREAM_TYPE_REDIRECT) {
+ c->http_error = 301;
+ q = c->buffer;
+ snprintf(q, c->buffer_size,
+ "HTTP/1.0 301 Moved\r\n"
+ "Location: %s\r\n"
+ "Content-type: text/html\r\n"
+ "\r\n"
+ "<!DOCTYPE html>\n"
+ "<html><head><title>Moved</title></head><body>\r\n"
+ "You should be <a href=\"%s\">redirected</a>.\r\n"
+ "</body></html>\r\n",
+ stream->feed_filename, stream->feed_filename);
+ q += strlen(q);
+ /* prepare output buffer */
+ c->buffer_ptr = c->buffer;
+ c->buffer_end = q;
+ c->state = HTTPSTATE_SEND_HEADER;
+ return 0;
+ }
+
+ /* If this is WMP, get the rate information */
+ if (extract_rates(ratebuf, sizeof(ratebuf), c->buffer)) {
+ if (modify_current_stream(c, ratebuf)) {
+ for (i = 0; i < FF_ARRAY_ELEMS(c->feed_streams); i++) {
+ if (c->switch_feed_streams[i] >= 0)
+ c->switch_feed_streams[i] = -1;
+ }
+ }
+ }
+
+ if (c->post == 0 && stream->stream_type == STREAM_TYPE_LIVE)
+ current_bandwidth += stream->bandwidth;
+
+ /* If already streaming this feed, do not let another feeder start */
+ if (stream->feed_opened) {
+ snprintf(msg, sizeof(msg), "This feed is already being received.");
+ http_log("Feed '%s' already being received\n", stream->feed_filename);
+ goto send_error;
+ }
+
+ if (c->post == 0 && config.max_bandwidth < current_bandwidth) {
+ c->http_error = 503;
+ q = c->buffer;
+ snprintf(q, c->buffer_size,
+ "HTTP/1.0 503 Server too busy\r\n"
+ "Content-type: text/html\r\n"
+ "\r\n"
+ "<!DOCTYPE html>\n"
+ "<html><head><title>Too busy</title></head><body>\r\n"
+ "<p>The server is too busy to serve your request at "
+ "this time.</p>\r\n"
+ "<p>The bandwidth being served (including your stream) "
+ "is %"PRIu64"kbit/s, and this exceeds the limit of "
+ "%"PRIu64"kbit/s.</p>\r\n"
+ "</body></html>\r\n",
+ current_bandwidth, config.max_bandwidth);
+ q += strlen(q);
+ /* prepare output buffer */
+ c->buffer_ptr = c->buffer;
+ c->buffer_end = q;
+ c->state = HTTPSTATE_SEND_HEADER;
+ return 0;
+ }
+
+ if (redir_type != REDIR_NONE) {
+ const char *hostinfo = 0;
+
+ for (p = c->buffer; *p && *p != '\r' && *p != '\n'; ) {
+ if (av_strncasecmp(p, "Host:", 5) == 0) {
+ hostinfo = p + 5;
+ break;
+ }
+ p = strchr(p, '\n');
+ if (!p)
+ break;
+
+ p++;
+ }
+
+ if (hostinfo) {
+ char *eoh;
+ char hostbuf[260];
+
+ while (av_isspace(*hostinfo))
+ hostinfo++;
+
+ eoh = strchr(hostinfo, '\n');
+ if (eoh) {
+ if (eoh[-1] == '\r')
+ eoh--;
+
+ if (eoh - hostinfo < sizeof(hostbuf) - 1) {
+ memcpy(hostbuf, hostinfo, eoh - hostinfo);
+ hostbuf[eoh - hostinfo] = 0;
+
+ c->http_error = 200;
+ q = c->buffer;
+ switch(redir_type) {
+ case REDIR_ASX:
+ snprintf(q, c->buffer_size,
+ "HTTP/1.0 200 ASX Follows\r\n"
+ "Content-type: video/x-ms-asf\r\n"
+ "\r\n"
+ "<ASX Version=\"3\">\r\n"
+ //"<!-- Autogenerated by ffserver -->\r\n"
+ "<ENTRY><REF HREF=\"http://%s/%s%s\"/></ENTRY>\r\n"
+ "</ASX>\r\n", hostbuf, filename, info);
+ q += strlen(q);
+ break;
+ case REDIR_RAM:
+ snprintf(q, c->buffer_size,
+ "HTTP/1.0 200 RAM Follows\r\n"
+ "Content-type: audio/x-pn-realaudio\r\n"
+ "\r\n"
+ "# Autogenerated by ffserver\r\n"
+ "http://%s/%s%s\r\n", hostbuf, filename, info);
+ q += strlen(q);
+ break;
+ case REDIR_ASF:
+ snprintf(q, c->buffer_size,
+ "HTTP/1.0 200 ASF Redirect follows\r\n"
+ "Content-type: video/x-ms-asf\r\n"
+ "\r\n"
+ "[Reference]\r\n"
+ "Ref1=http://%s/%s%s\r\n", hostbuf, filename, info);
+ q += strlen(q);
+ break;
+ case REDIR_RTSP:
+ {
+ char hostname[256], *p;
+ /* extract only hostname */
+ av_strlcpy(hostname, hostbuf, sizeof(hostname));
+ p = strrchr(hostname, ':');
+ if (p)
+ *p = '\0';
+ snprintf(q, c->buffer_size,
+ "HTTP/1.0 200 RTSP Redirect follows\r\n"
+ /* XXX: incorrect MIME type ? */
+ "Content-type: application/x-rtsp\r\n"
+ "\r\n"
+ "rtsp://%s:%d/%s\r\n", hostname, ntohs(config.rtsp_addr.sin_port), filename);
+ q += strlen(q);
+ }
+ break;
+ case REDIR_SDP:
+ {
+ uint8_t *sdp_data;
+ int sdp_data_size;
+ socklen_t len;
+ struct sockaddr_in my_addr;
+
+ snprintf(q, c->buffer_size,
+ "HTTP/1.0 200 OK\r\n"
+ "Content-type: application/sdp\r\n"
+ "\r\n");
+ q += strlen(q);
+
+ len = sizeof(my_addr);
+
+ /* XXX: Should probably fail? */
+ if (getsockname(c->fd, (struct sockaddr *)&my_addr, &len))
+ http_log("getsockname() failed\n");
+
+ /* XXX: should use a dynamic buffer */
+ sdp_data_size = prepare_sdp_description(stream,
+ &sdp_data,
+ my_addr.sin_addr);
+ if (sdp_data_size > 0) {
+ memcpy(q, sdp_data, sdp_data_size);
+ q += sdp_data_size;
+ *q = '\0';
+ av_free(sdp_data);
+ }
+ }
+ break;
+ default:
+ abort();
+ break;
+ }
+
+ /* prepare output buffer */
+ c->buffer_ptr = c->buffer;
+ c->buffer_end = q;
+ c->state = HTTPSTATE_SEND_HEADER;
+ return 0;
+ }
+ }
+ }
+
+ snprintf(msg, sizeof(msg), "ASX/RAM file not handled");
+ goto send_error;
+ }
+
+ stream->conns_served++;
+
+ /* XXX: add there authenticate and IP match */
+
+ if (c->post) {
+ /* if post, it means a feed is being sent */
+ if (!stream->is_feed) {
+ /* However it might be a status report from WMP! Let us log the
+ * data as it might come handy one day. */
+ const char *logline = 0;
+ int client_id = 0;
+
+ for (p = c->buffer; *p && *p != '\r' && *p != '\n'; ) {
+ if (av_strncasecmp(p, "Pragma: log-line=", 17) == 0) {
+ logline = p;
+ break;
+ }
+ if (av_strncasecmp(p, "Pragma: client-id=", 18) == 0)
+ client_id = strtol(p + 18, 0, 10);
+ p = strchr(p, '\n');
+ if (!p)
+ break;
+
+ p++;
+ }
+
+ if (logline) {
+ char *eol = strchr(logline, '\n');
+
+ logline += 17;
+
+ if (eol) {
+ if (eol[-1] == '\r')
+ eol--;
+ http_log("%.*s\n", (int) (eol - logline), logline);
+ c->suppress_log = 1;
+ }
+ }
+
+#ifdef DEBUG
+ http_log("\nGot request:\n%s\n", c->buffer);
+#endif
+
+ if (client_id && extract_rates(ratebuf, sizeof(ratebuf), c->buffer)) {
+ HTTPContext *wmpc;
+
+ /* Now we have to find the client_id */
+ for (wmpc = first_http_ctx; wmpc; wmpc = wmpc->next) {
+ if (wmpc->wmp_client_id == client_id)
+ break;
+ }
+
+ if (wmpc && modify_current_stream(wmpc, ratebuf))
+ wmpc->switch_pending = 1;
+ }
+
+ snprintf(msg, sizeof(msg), "POST command not handled");
+ c->stream = 0;
+ goto send_error;
+ }
+ if (http_start_receive_data(c) < 0) {
+ snprintf(msg, sizeof(msg), "could not open feed");
+ goto send_error;
+ }
+ c->http_error = 0;
+ c->state = HTTPSTATE_RECEIVE_DATA;
+ return 0;
+ }
+
+#ifdef DEBUG
+ if (strcmp(stream->filename + strlen(stream->filename) - 4, ".asf") == 0)
+ http_log("\nGot request:\n%s\n", c->buffer);
+#endif
+
+ if (c->stream->stream_type == STREAM_TYPE_STATUS)
+ goto send_status;
+
+ /* open input stream */
+ if (open_input_stream(c, info) < 0) {
+ snprintf(msg, sizeof(msg), "Input stream corresponding to '%s' not found", url);
+ goto send_error;
+ }
+
+ /* prepare HTTP header */
+ c->buffer[0] = 0;
+ av_strlcatf(c->buffer, c->buffer_size, "HTTP/1.0 200 OK\r\n");
+ mime_type = c->stream->fmt->mime_type;
+ if (!mime_type)
+ mime_type = "application/x-octet-stream";
+ av_strlcatf(c->buffer, c->buffer_size, "Pragma: no-cache\r\n");
+
+ /* for asf, we need extra headers */
+ if (!strcmp(c->stream->fmt->name,"asf_stream")) {
+ /* Need to allocate a client id */
+
+ c->wmp_client_id = av_lfg_get(&random_state);
+
+ av_strlcatf(c->buffer, c->buffer_size, "Server: Cougar 4.1.0.3923\r\nCache-Control: no-cache\r\nPragma: client-id=%d\r\nPragma: features=\"broadcast\"\r\n", c->wmp_client_id);
+ }
+ av_strlcatf(c->buffer, c->buffer_size, "Content-Type: %s\r\n", mime_type);
+ av_strlcatf(c->buffer, c->buffer_size, "\r\n");
+ q = c->buffer + strlen(c->buffer);
+
+ /* prepare output buffer */
+ c->http_error = 0;
+ c->buffer_ptr = c->buffer;
+ c->buffer_end = q;
+ c->state = HTTPSTATE_SEND_HEADER;
+ return 0;
+ send_error:
+ c->http_error = 404;
+ q = c->buffer;
+ if (!htmlencode(msg, &encoded_msg)) {
+ http_log("Could not encode filename '%s' as HTML\n", msg);
+ }
+ snprintf(q, c->buffer_size,
+ "HTTP/1.0 404 Not Found\r\n"
+ "Content-type: text/html\r\n"
+ "\r\n"
+ "<!DOCTYPE html>\n"
+ "<html>\n"
+ "<head>\n"
+ "<meta charset=\"UTF-8\">\n"
+ "<title>404 Not Found</title>\n"
+ "</head>\n"
+ "<body>%s</body>\n"
+ "</html>\n", encoded_msg? encoded_msg : "File not found");
+ q += strlen(q);
+ /* prepare output buffer */
+ c->buffer_ptr = c->buffer;
+ c->buffer_end = q;
+ c->state = HTTPSTATE_SEND_HEADER;
+ av_freep(&encoded_msg);
+ return 0;
+ send_status:
+ compute_status(c);
+ /* horrible: we use this value to avoid
+ * going to the send data state */
+ c->http_error = 200;
+ c->state = HTTPSTATE_SEND_HEADER;
+ return 0;
+}
+
+static void fmt_bytecount(AVIOContext *pb, int64_t count)
+{
+ static const char suffix[] = " kMGTP";
+ const char *s;
+
+ for (s = suffix; count >= 100000 && s[1]; count /= 1000, s++);
+
+ avio_printf(pb, "%"PRId64"%c", count, *s);
+}
+
+static inline void print_stream_params(AVIOContext *pb, FFServerStream *stream)
+{
+ int i, stream_no;
+ const char *type = "unknown";
+ char parameters[64];
+ AVStream *st;
+ AVCodec *codec;
+
+ stream_no = stream->nb_streams;
+
+ avio_printf(pb, "<table cellspacing=0 cellpadding=4><tr><th>Stream<th>"
+ "type<th>kbit/s<th align=left>codec<th align=left>"
+ "Parameters\n");
+
+ for (i = 0; i < stream_no; i++) {
+ st = stream->streams[i];
+ codec = avcodec_find_encoder(st->codec->codec_id);
+
+ parameters[0] = 0;
+
+ switch(st->codec->codec_type) {
+ case AVMEDIA_TYPE_AUDIO:
+ type = "audio";
+ snprintf(parameters, sizeof(parameters), "%d channel(s), %d Hz",
+ st->codec->channels, st->codec->sample_rate);
+ break;
+ case AVMEDIA_TYPE_VIDEO:
+ type = "video";
+ snprintf(parameters, sizeof(parameters),
+ "%dx%d, q=%d-%d, fps=%d", st->codec->width,
+ st->codec->height, st->codec->qmin, st->codec->qmax,
+ st->codec->time_base.den / st->codec->time_base.num);
+ break;
+ default:
+ abort();
+ }
+
+ avio_printf(pb, "<tr><td align=right>%d<td>%s<td align=right>%"PRId64
+ "<td>%s<td>%s\n",
+ i, type, (int64_t)st->codec->bit_rate/1000,
+ codec ? codec->name : "", parameters);
+ }
+
+ avio_printf(pb, "</table>\n");
+}
+
+static void compute_status(HTTPContext *c)
+{
+ HTTPContext *c1;
+ FFServerStream *stream;
+ char *p;
+ time_t ti;
+ int i, len;
+ AVIOContext *pb;
+
+ if (avio_open_dyn_buf(&pb) < 0) {
+ /* XXX: return an error ? */
+ c->buffer_ptr = c->buffer;
+ c->buffer_end = c->buffer;
+ return;
+ }
+
+ avio_printf(pb, "HTTP/1.0 200 OK\r\n");
+ avio_printf(pb, "Content-type: text/html\r\n");
+ avio_printf(pb, "Pragma: no-cache\r\n");
+ avio_printf(pb, "\r\n");
+
+ avio_printf(pb, "<!DOCTYPE html>\n");
+ avio_printf(pb, "<html><head><title>%s Status</title>\n", program_name);
+ if (c->stream->feed_filename[0])
+ avio_printf(pb, "<link rel=\"shortcut icon\" href=\"%s\">\n",
+ c->stream->feed_filename);
+ avio_printf(pb, "</head>\n<body>");
+ avio_printf(pb, "<h1>%s Status</h1>\n", program_name);
+ /* format status */
+ avio_printf(pb, "<h2>Available Streams</h2>\n");
+ avio_printf(pb, "<table cellspacing=0 cellpadding=4>\n");
+ avio_printf(pb, "<tr><th valign=top>Path<th align=left>Served<br>Conns<th><br>bytes<th valign=top>Format<th>Bit rate<br>kbit/s<th align=left>Video<br>kbit/s<th><br>Codec<th align=left>Audio<br>kbit/s<th><br>Codec<th align=left valign=top>Feed\n");
+ stream = config.first_stream;
+ while (stream) {
+ char sfilename[1024];
+ char *eosf;
+
+ if (stream->feed == stream) {
+ stream = stream->next;
+ continue;
+ }
+
+ av_strlcpy(sfilename, stream->filename, sizeof(sfilename) - 10);
+ eosf = sfilename + strlen(sfilename);
+ if (eosf - sfilename >= 4) {
+ if (strcmp(eosf - 4, ".asf") == 0)
+ strcpy(eosf - 4, ".asx");
+ else if (strcmp(eosf - 3, ".rm") == 0)
+ 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 */
+ eosf = strrchr(sfilename, '.');
+ if (!eosf)
+ eosf = sfilename + strlen(sfilename);
+ if (stream->is_multicast)
+ strcpy(eosf, ".sdp");
+ else
+ strcpy(eosf, ".rtsp");
+ }
+ }
+
+ avio_printf(pb, "<tr><td><a href=\"/%s\">%s</a> ",
+ sfilename, stream->filename);
+ avio_printf(pb, "<td align=right> %d <td align=right> ",
+ stream->conns_served);
+ fmt_bytecount(pb, stream->bytes_served);
+
+ switch(stream->stream_type) {
+ case STREAM_TYPE_LIVE: {
+ int audio_bit_rate = 0;
+ int video_bit_rate = 0;
+ const char *audio_codec_name = "";
+ const char *video_codec_name = "";
+ const char *audio_codec_name_extra = "";
+ const char *video_codec_name_extra = "";
+
+ for(i=0;i<stream->nb_streams;i++) {
+ AVStream *st = stream->streams[i];
+ AVCodec *codec = avcodec_find_encoder(st->codec->codec_id);
+
+ switch(st->codec->codec_type) {
+ case AVMEDIA_TYPE_AUDIO:
+ audio_bit_rate += st->codec->bit_rate;
+ if (codec) {
+ if (*audio_codec_name)
+ audio_codec_name_extra = "...";
+ audio_codec_name = codec->name;
+ }
+ break;
+ case AVMEDIA_TYPE_VIDEO:
+ video_bit_rate += st->codec->bit_rate;
+ if (codec) {
+ if (*video_codec_name)
+ video_codec_name_extra = "...";
+ video_codec_name = codec->name;
+ }
+ break;
+ case AVMEDIA_TYPE_DATA:
+ video_bit_rate += st->codec->bit_rate;
+ break;
+ default:
+ abort();
+ }
+ }
+
+ avio_printf(pb, "<td align=center> %s <td align=right> %d "
+ "<td align=right> %d <td> %s %s <td align=right> "
+ "%d <td> %s %s",
+ stream->fmt->name, stream->bandwidth,
+ video_bit_rate / 1000, video_codec_name,
+ video_codec_name_extra, audio_bit_rate / 1000,
+ audio_codec_name, audio_codec_name_extra);
+
+ if (stream->feed)
+ avio_printf(pb, "<td>%s", stream->feed->filename);
+ else
+ avio_printf(pb, "<td>%s", stream->feed_filename);
+ avio_printf(pb, "\n");
+ }
+ break;
+ default:
+ avio_printf(pb, "<td align=center> - <td align=right> - "
+ "<td align=right> - <td><td align=right> - <td>\n");
+ break;
+ }
+ stream = stream->next;
+ }
+ avio_printf(pb, "</table>\n");
+
+ stream = config.first_stream;
+ while (stream) {
+
+ if (stream->feed != stream) {
+ stream = stream->next;
+ continue;
+ }
+
+ avio_printf(pb, "<h2>Feed %s</h2>", stream->filename);
+ if (stream->pid) {
+ avio_printf(pb, "Running as pid %"PRId64".\n", (int64_t) stream->pid);
+
+#if defined(linux)
+ {
+ FILE *pid_stat;
+ char ps_cmd[64];
+
+ /* This is somewhat linux specific I guess */
+ snprintf(ps_cmd, sizeof(ps_cmd),
+ "ps -o \"%%cpu,cputime\" --no-headers %"PRId64"",
+ (int64_t) stream->pid);
+
+ pid_stat = popen(ps_cmd, "r");
+ if (pid_stat) {
+ char cpuperc[10];
+ char cpuused[64];
+
+ if (fscanf(pid_stat, "%9s %63s", cpuperc, cpuused) == 2) {
+ avio_printf(pb, "Currently using %s%% of the cpu. "
+ "Total time used %s.\n",
+ cpuperc, cpuused);
+ }
+ fclose(pid_stat);
+ }
+ }
+#endif
+
+ avio_printf(pb, "<p>");
+ }
+
+ print_stream_params(pb, stream);
+ stream = stream->next;
+ }
+
+ /* connection status */
+ avio_printf(pb, "<h2>Connection Status</h2>\n");
+
+ avio_printf(pb, "Number of connections: %d / %d<br>\n",
+ nb_connections, config.nb_max_connections);
+
+ avio_printf(pb, "Bandwidth in use: %"PRIu64"k / %"PRIu64"k<br>\n",
+ current_bandwidth, config.max_bandwidth);
+
+ avio_printf(pb, "<table>\n");
+ avio_printf(pb, "<tr><th>#<th>File<th>IP<th>Proto<th>State<th>Target "
+ "bit/s<th>Actual bit/s<th>Bytes transferred\n");
+ c1 = first_http_ctx;
+ i = 0;
+ while (c1) {
+ int bitrate;
+ int j;
+
+ bitrate = 0;
+ if (c1->stream) {
+ for (j = 0; j < c1->stream->nb_streams; j++) {
+ if (!c1->stream->feed)
+ bitrate += c1->stream->streams[j]->codec->bit_rate;
+ else if (c1->feed_streams[j] >= 0)
+ bitrate += c1->stream->feed->streams[c1->feed_streams[j]]->codec->bit_rate;
+ }
+ }
+
+ i++;
+ p = inet_ntoa(c1->from_addr.sin_addr);
+ avio_printf(pb, "<tr><td><b>%d</b><td>%s%s<td>%s<td>%s<td>%s"
+ "<td align=right>",
+ i, c1->stream ? c1->stream->filename : "",
+ c1->state == HTTPSTATE_RECEIVE_DATA ? "(input)" : "", p,
+ c1->protocol, http_state[c1->state]);
+ fmt_bytecount(pb, bitrate);
+ avio_printf(pb, "<td align=right>");
+ fmt_bytecount(pb, compute_datarate(&c1->datarate, c1->data_count) * 8);
+ avio_printf(pb, "<td align=right>");
+ fmt_bytecount(pb, c1->data_count);
+ avio_printf(pb, "\n");
+ c1 = c1->next;
+ }
+ avio_printf(pb, "</table>\n");
+
+ /* date */
+ ti = time(NULL);
+ p = ctime(&ti);
+ avio_printf(pb, "<hr size=1 noshade>Generated at %s", p);
+ avio_printf(pb, "</body>\n</html>\n");
+
+ len = avio_close_dyn_buf(pb, &c->pb_buffer);
+ c->buffer_ptr = c->pb_buffer;
+ c->buffer_end = c->pb_buffer + len;
+}
+
+static int open_input_stream(HTTPContext *c, const char *info)
+{
+ char buf[128];
+ char input_filename[1024];
+ AVFormatContext *s = NULL;
+ int buf_size, i, ret;
+ int64_t stream_pos;
+
+ /* find file name */
+ if (c->stream->feed) {
+ strcpy(input_filename, c->stream->feed->feed_filename);
+ buf_size = FFM_PACKET_SIZE;
+ /* compute position (absolute time) */
+ if (av_find_info_tag(buf, sizeof(buf), "date", info)) {
+ if ((ret = av_parse_time(&stream_pos, buf, 0)) < 0) {
+ http_log("Invalid date specification '%s' for stream\n", buf);
+ return ret;
+ }
+ } else if (av_find_info_tag(buf, sizeof(buf), "buffer", info)) {
+ int prebuffer = strtol(buf, 0, 10);
+ stream_pos = av_gettime() - prebuffer * (int64_t)1000000;
+ } else
+ stream_pos = av_gettime() - c->stream->prebuffer * (int64_t)1000;
+ } else {
+ strcpy(input_filename, c->stream->feed_filename);
+ buf_size = 0;
+ /* compute position (relative time) */
+ if (av_find_info_tag(buf, sizeof(buf), "date", info)) {
+ if ((ret = av_parse_time(&stream_pos, buf, 1)) < 0) {
+ http_log("Invalid date specification '%s' for stream\n", buf);
+ return ret;
+ }
+ } else
+ stream_pos = 0;
+ }
+ if (!input_filename[0]) {
+ http_log("No filename was specified for stream\n");
+ return AVERROR(EINVAL);
+ }
+
+ /* open stream */
+ ret = avformat_open_input(&s, input_filename, c->stream->ifmt,
+ &c->stream->in_opts);
+ if (ret < 0) {
+ http_log("Could not open input '%s': %s\n",
+ input_filename, av_err2str(ret));
+ return ret;
+ }
+
+ /* set buffer size */
+ if (buf_size > 0) {
+ ret = ffio_set_buf_size(s->pb, buf_size);
+ if (ret < 0) {
+ http_log("Failed to set buffer size\n");
+ return ret;
+ }
+ }
+
+ s->flags |= AVFMT_FLAG_GENPTS;
+ c->fmt_in = s;
+ if (strcmp(s->iformat->name, "ffm") &&
+ (ret = avformat_find_stream_info(c->fmt_in, NULL)) < 0) {
+ http_log("Could not find stream info for input '%s'\n", input_filename);
+ avformat_close_input(&s);
+ return ret;
+ }
+
+ /* choose stream as clock source (we favor the video stream if
+ * present) for packet sending */
+ c->pts_stream_index = 0;
+ for(i=0;i<c->stream->nb_streams;i++) {
+ if (c->pts_stream_index == 0 &&
+ c->stream->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
+ c->pts_stream_index = i;
+ }
+ }
+
+ if (c->fmt_in->iformat->read_seek)
+ av_seek_frame(c->fmt_in, -1, stream_pos, 0);
+ /* set the start time (needed for maxtime and RTP packet timing) */
+ c->start_time = cur_time;
+ c->first_pts = AV_NOPTS_VALUE;
+ return 0;
+}
+
+/* return the server clock (in us) */
+static int64_t get_server_clock(HTTPContext *c)
+{
+ /* compute current pts value from system time */
+ return (cur_time - c->start_time) * 1000;
+}
+
+/* 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;
+
+ frame_bytes = c->cur_frame_bytes;
+ if (frame_bytes <= 0)
+ return c->cur_pts;
+
+ bytes_left = c->buffer_end - c->buffer_ptr;
+ bytes_sent = frame_bytes - bytes_left;
+ return c->cur_pts + (c->cur_frame_duration * bytes_sent) / frame_bytes;
+}
+
+
+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:
+ ctx = avformat_alloc_context();
+ if (!ctx)
+ return AVERROR(ENOMEM);
+ c->fmt_ctx = *ctx;
+ av_freep(&ctx);
+ av_dict_copy(&(c->fmt_ctx.metadata), c->stream->metadata, 0);
+ c->fmt_ctx.streams = av_mallocz_array(c->stream->nb_streams,
+ sizeof(AVStream *));
+ if (!c->fmt_ctx.streams)
+ return AVERROR(ENOMEM);
+
+ for(i=0;i<c->stream->nb_streams;i++) {
+ AVStream *src;
+ c->fmt_ctx.streams[i] = av_mallocz(sizeof(AVStream));
+
+ /* 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];
+ else
+ src = c->stream->feed->streams[c->stream->feed_streams[i]];
+
+ *(c->fmt_ctx.streams[i]) = *src;
+ c->fmt_ctx.streams[i]->priv_data = 0;
+ /* XXX: should be done in AVStream, not in codec */
+ c->fmt_ctx.streams[i]->codec->frame_number = 0;
+ }
+ /* set output format parameters */
+ c->fmt_ctx.oformat = c->stream->fmt;
+ c->fmt_ctx.nb_streams = c->stream->nb_streams;
+
+ c->got_key_frame = 0;
+
+ /* prepare header and save header data in a stream */
+ if (avio_open_dyn_buf(&c->fmt_ctx.pb) < 0) {
+ /* XXX: potential leak */
+ return -1;
+ }
+ c->fmt_ctx.pb->seekable = 0;
+
+ /*
+ * HACK to avoid MPEG-PS muxer to spit many underflow errors
+ * Default value from FFmpeg
+ * Try to set it using configuration option
+ */
+ c->fmt_ctx.max_delay = (int)(0.7*AV_TIME_BASE);
+
+ if ((ret = avformat_write_header(&c->fmt_ctx, NULL)) < 0) {
+ http_log("Error writing output header for stream '%s': %s\n",
+ c->stream->filename, av_err2str(ret));
+ return ret;
+ }
+ av_dict_free(&c->fmt_ctx.metadata);
+
+ len = avio_close_dyn_buf(c->fmt_ctx.pb, &c->pb_buffer);
+ c->buffer_ptr = c->pb_buffer;
+ c->buffer_end = c->pb_buffer + len;
+
+ c->state = HTTPSTATE_SEND_DATA;
+ c->last_packet_sent = 0;
+ break;
+ case HTTPSTATE_SEND_DATA:
+ /* find a new packet */
+ /* read a packet from the input stream */
+ if (c->stream->feed)
+ ffm_set_write_index(c->fmt_in,
+ c->stream->feed->feed_write_index,
+ c->stream->feed->feed_size);
+
+ if (c->stream->max_time &&
+ c->stream->max_time + c->start_time - cur_time < 0)
+ /* We have timed out */
+ c->state = HTTPSTATE_SEND_DATA_TRAILER;
+ else {
+ AVPacket pkt;
+ redo:
+ ret = av_read_frame(c->fmt_in, &pkt);
+ 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 */
+ c->state = HTTPSTATE_WAIT_FEED;
+ return 1; /* state changed */
+ }
+ if (ret == AVERROR(EAGAIN)) {
+ /* input not ready, come back later */
+ return 0;
+ }
+ if (c->stream->loop) {
+ avformat_close_input(&c->fmt_in);
+ if (open_input_stream(c, "") < 0)
+ goto no_loop;
+ goto redo;
+ } else {
+ no_loop:
+ /* must send trailer now because EOF or error */
+ c->state = HTTPSTATE_SEND_DATA_TRAILER;
+ }
+ } else {
+ int source_index = pkt.stream_index;
+ /* update first pts if needed */
+ if (c->first_pts == AV_NOPTS_VALUE && pkt.dts != AV_NOPTS_VALUE) {
+ c->first_pts = av_rescale_q(pkt.dts, c->fmt_in->streams[pkt.stream_index]->time_base, AV_TIME_BASE_Q);
+ c->start_time = cur_time;
+ }
+ /* send it to the appropriate stream */
+ if (c->stream->feed) {
+ /* if coming from a feed, select the right stream */
+ if (c->switch_pending) {
+ c->switch_pending = 0;
+ for(i=0;i<c->stream->nb_streams;i++) {
+ if (c->switch_feed_streams[i] == pkt.stream_index)
+ if (pkt.flags & AV_PKT_FLAG_KEY)
+ c->switch_feed_streams[i] = -1;
+ if (c->switch_feed_streams[i] >= 0)
+ c->switch_pending = 1;
+ }
+ }
+ for(i=0;i<c->stream->nb_streams;i++) {
+ if (c->stream->feed_streams[i] == pkt.stream_index) {
+ AVStream *st = c->fmt_in->streams[source_index];
+ pkt.stream_index = i;
+ if (pkt.flags & AV_PKT_FLAG_KEY &&
+ (st->codec->codec_type == AVMEDIA_TYPE_VIDEO ||
+ c->stream->nb_streams == 1))
+ c->got_key_frame = 1;
+ if (!c->stream->send_on_key || c->got_key_frame)
+ goto send_it;
+ }
+ }
+ } else {
+ AVCodecContext *codec;
+ AVStream *ist, *ost;
+ send_it:
+ ist = c->fmt_in->streams[source_index];
+ /* specific handling for RTP: we use several
+ * output streams (one for each RTP connection).
+ * XXX: need more abstract handling */
+ if (c->is_packetized) {
+ /* compute send time and duration */
+ if (pkt.dts != AV_NOPTS_VALUE) {
+ c->cur_pts = av_rescale_q(pkt.dts, ist->time_base, AV_TIME_BASE_Q);
+ c->cur_pts -= c->first_pts;
+ }
+ c->cur_frame_duration = av_rescale_q(pkt.duration, ist->time_base, AV_TIME_BASE_Q);
+ /* find RTP context */
+ c->packet_stream_index = pkt.stream_index;
+ ctx = c->rtp_ctx[c->packet_stream_index];
+ if(!ctx) {
+ av_packet_unref(&pkt);
+ break;
+ }
+ codec = ctx->streams[0]->codec;
+ /* only one stream per RTP connection */
+ pkt.stream_index = 0;
+ } else {
+ ctx = &c->fmt_ctx;
+ /* Fudge here */
+ codec = ctx->streams[pkt.stream_index]->codec;
+ }
+
+ if (c->is_packetized) {
+ int max_packet_size;
+ if (c->rtp_protocol == RTSP_LOWER_TRANSPORT_TCP)
+ max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
+ else
+ 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
+ ret = avio_open_dyn_buf(&ctx->pb);
+
+ if (ret < 0) {
+ /* XXX: potential leak */
+ return -1;
+ }
+ ost = ctx->streams[pkt.stream_index];
+
+ ctx->pb->seekable = 0;
+ if (pkt.dts != AV_NOPTS_VALUE)
+ pkt.dts = av_rescale_q(pkt.dts, ist->time_base,
+ ost->time_base);
+ if (pkt.pts != AV_NOPTS_VALUE)
+ pkt.pts = av_rescale_q(pkt.pts, ist->time_base,
+ ost->time_base);
+ pkt.duration = av_rescale_q(pkt.duration, ist->time_base,
+ ost->time_base);
+ if ((ret = av_write_frame(ctx, &pkt)) < 0) {
+ http_log("Error writing frame to output for stream '%s': %s\n",
+ c->stream->filename, av_err2str(ret));
+ c->state = HTTPSTATE_SEND_DATA_TRAILER;
+ }
+
+ av_freep(&c->pb_buffer);
+ len = avio_close_dyn_buf(ctx->pb, &c->pb_buffer);
+ ctx->pb = NULL;
+ c->cur_frame_bytes = len;
+ c->buffer_ptr = c->pb_buffer;
+ c->buffer_end = c->pb_buffer + len;
+
+ codec->frame_number++;
+ if (len == 0) {
+ av_packet_unref(&pkt);
+ goto redo;
+ }
+ }
+ av_packet_unref(&pkt);
+ }
+ }
+ break;
+ default:
+ case HTTPSTATE_SEND_DATA_TRAILER:
+ /* last packet test ? */
+ if (c->last_packet_sent || c->is_packetized)
+ return -1;
+ ctx = &c->fmt_ctx;
+ /* prepare header */
+ if (avio_open_dyn_buf(&ctx->pb) < 0) {
+ /* XXX: potential leak */
+ return -1;
+ }
+ c->fmt_ctx.pb->seekable = 0;
+ av_write_trailer(ctx);
+ len = avio_close_dyn_buf(ctx->pb, &c->pb_buffer);
+ c->buffer_ptr = c->pb_buffer;
+ c->buffer_end = c->pb_buffer + len;
+
+ c->last_packet_sent = 1;
+ break;
+ }
+ return 0;
+}
+
+/* should convert the format at the same time */
+/* send data starting at c->buffer_ptr to the output connection
+ * (either UDP or TCP)
+ */
+static int http_send_data(HTTPContext *c)
+{
+ int len, ret;
+
+ for(;;) {
+ if (c->buffer_ptr >= c->buffer_end) {
+ ret = http_prepare_data(c);
+ if (ret < 0)
+ return -1;
+ else if (ret)
+ /* state change requested */
+ break;
+ } 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;
+ }
+ 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 ((get_packet_send_clock(c) - get_server_clock(c)) > 0) {
+ /* nothing to send yet: we can wait */
+ return 0;
+ }
+
+ c->data_count += len;
+ update_datarate(&c->datarate, c->data_count);
+ if (c->stream)
+ c->stream->bytes_served += len;
+
+ if (c->rtp_protocol == RTSP_LOWER_TRANSPORT_TCP) {
+ /* RTP packets are sent inside the RTSP TCP connection */
+ AVIOContext *pb;
+ 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 (avio_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;
+ avio_write(pb, header, 4);
+ /* write RTP packet data */
+ c->buffer_ptr += 4;
+ avio_write(pb, c->buffer_ptr, len);
+ size = avio_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;
+ c->buffer_ptr += len;
+
+ /* send everything we can NOW */
+ len = send(rtsp_c->fd, rtsp_c->packet_buffer_ptr,
+ rtsp_c->packet_buffer_end - rtsp_c->packet_buffer_ptr, 0);
+ if (len > 0)
+ 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 */
+ rtsp_c->state = RTSPSTATE_SEND_PACKET;
+ break;
+ } else
+ /* all data has been sent */
+ av_freep(&c->packet_buffer);
+ } else {
+ /* send RTP packet directly in UDP */
+ c->buffer_ptr += 4;
+ ffurl_write(c->rtp_handles[c->packet_stream_index],
+ c->buffer_ptr, len);
+ c->buffer_ptr += len;
+ /* here we continue as we can send several packets
+ * per 10 ms slot */
+ }
+ } else {
+ /* TCP data output */
+ len = send(c->fd, c->buffer_ptr,
+ c->buffer_end - c->buffer_ptr, 0);
+ if (len < 0) {
+ if (ff_neterrno() != AVERROR(EAGAIN) &&
+ ff_neterrno() != AVERROR(EINTR))
+ /* error : close connection */
+ return -1;
+ else
+ return 0;
+ }
+ c->buffer_ptr += len;
+
+ c->data_count += len;
+ update_datarate(&c->datarate, c->data_count);
+ if (c->stream)
+ c->stream->bytes_served += len;
+ break;
+ }
+ }
+ } /* for(;;) */
+ return 0;
+}
+
+static int http_start_receive_data(HTTPContext *c)
+{
+ int fd;
+ int ret;
+ int64_t ret64;
+
+ if (c->stream->feed_opened) {
+ http_log("Stream feed '%s' was not opened\n",
+ c->stream->feed_filename);
+ return AVERROR(EINVAL);
+ }
+
+ /* Don't permit writing to this one */
+ if (c->stream->readonly) {
+ http_log("Cannot write to read-only file '%s'\n",
+ c->stream->feed_filename);
+ return AVERROR(EINVAL);
+ }
+
+ /* open feed */
+ fd = open(c->stream->feed_filename, O_RDWR);
+ if (fd < 0) {
+ ret = AVERROR(errno);
+ http_log("Could not open feed file '%s': %s\n",
+ c->stream->feed_filename, strerror(errno));
+ return ret;
+ }
+ c->feed_fd = fd;
+
+ if (c->stream->truncate) {
+ /* truncate feed file */
+ ffm_write_write_index(c->feed_fd, FFM_PACKET_SIZE);
+ http_log("Truncating feed file '%s'\n", c->stream->feed_filename);
+ if (ftruncate(c->feed_fd, FFM_PACKET_SIZE) < 0) {
+ ret = AVERROR(errno);
+ http_log("Error truncating feed file '%s': %s\n",
+ c->stream->feed_filename, strerror(errno));
+ return ret;
+ }
+ } else {
+ ret64 = ffm_read_write_index(fd);
+ if (ret64 < 0) {
+ http_log("Error reading write index from feed file '%s': %s\n",
+ c->stream->feed_filename, strerror(errno));
+ return ret64;
+ }
+ c->stream->feed_write_index = ret64;
+ }
+
+ c->stream->feed_write_index = FFMAX(ffm_read_write_index(fd),
+ FFM_PACKET_SIZE);
+ c->stream->feed_size = lseek(fd, 0, SEEK_END);
+ lseek(fd, 0, SEEK_SET);
+
+ /* init buffer input */
+ c->buffer_ptr = c->buffer;
+ c->buffer_end = c->buffer + FFM_PACKET_SIZE;
+ c->stream->feed_opened = 1;
+ c->chunked_encoding = !!av_stristr(c->buffer, "Transfer-Encoding: chunked");
+ return 0;
+}
+
+static int http_receive_data(HTTPContext *c)
+{
+ HTTPContext *c1;
+ int len, loop_run = 0;
+
+ while (c->chunked_encoding && !c->chunk_size &&
+ c->buffer_end > c->buffer_ptr) {
+ /* read chunk header, if present */
+ len = recv(c->fd, c->buffer_ptr, 1, 0);
+
+ if (len < 0) {
+ if (ff_neterrno() != AVERROR(EAGAIN) &&
+ ff_neterrno() != AVERROR(EINTR))
+ /* error : close connection */
+ goto fail;
+ return 0;
+ } else if (len == 0) {
+ /* end of connection : close it */
+ goto fail;
+ } else if (c->buffer_ptr - c->buffer >= 2 &&
+ !memcmp(c->buffer_ptr - 1, "\r\n", 2)) {
+ c->chunk_size = strtol(c->buffer, 0, 16);
+ if (c->chunk_size == 0) // end of stream
+ goto fail;
+ c->buffer_ptr = c->buffer;
+ break;
+ } else if (++loop_run > 10)
+ /* no chunk header, abort */
+ goto fail;
+ else
+ c->buffer_ptr++;
+ }
+
+ if (c->buffer_end > c->buffer_ptr) {
+ len = recv(c->fd, c->buffer_ptr,
+ FFMIN(c->chunk_size, c->buffer_end - c->buffer_ptr), 0);
+ if (len < 0) {
+ if (ff_neterrno() != AVERROR(EAGAIN) &&
+ ff_neterrno() != AVERROR(EINTR))
+ /* error : close connection */
+ goto fail;
+ } else if (len == 0)
+ /* end of connection : close it */
+ goto fail;
+ else {
+ c->chunk_size -= len;
+ c->buffer_ptr += len;
+ c->data_count += len;
+ update_datarate(&c->datarate, c->data_count);
+ }
+ }
+
+ if (c->buffer_ptr - c->buffer >= 2 && c->data_count > FFM_PACKET_SIZE) {
+ if (c->buffer[0] != 'f' ||
+ c->buffer[1] != 'm') {
+ http_log("Feed stream has become desynchronized -- disconnecting\n");
+ goto fail;
+ }
+ }
+
+ 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 (c->data_count > FFM_PACKET_SIZE) {
+ /* XXX: use llseek or url_seek
+ * XXX: Should probably fail? */
+ if (lseek(c->feed_fd, feed->feed_write_index, SEEK_SET) == -1)
+ http_log("Seek to %"PRId64" failed\n", feed->feed_write_index);
+
+ if (write(c->feed_fd, c->buffer, FFM_PACKET_SIZE) < 0) {
+ http_log("Error writing to feed file: %s\n", strerror(errno));
+ goto fail;
+ }
+
+ feed->feed_write_index += FFM_PACKET_SIZE;
+ /* update file size */
+ if (feed->feed_write_index > c->stream->feed_size)
+ feed->feed_size = feed->feed_write_index;
+
+ /* handle wrap around if max file size reached */
+ if (c->stream->feed_max_size &&
+ feed->feed_write_index >= c->stream->feed_max_size)
+ feed->feed_write_index = FFM_PACKET_SIZE;
+
+ /* write index */
+ if (ffm_write_write_index(c->feed_fd, feed->feed_write_index) < 0) {
+ http_log("Error writing index to feed file: %s\n",
+ strerror(errno));
+ goto fail;
+ }
+
+ /* wake up any waiting connections */
+ for(c1 = first_http_ctx; c1; c1 = c1->next) {
+ if (c1->state == HTTPSTATE_WAIT_FEED &&
+ c1->stream->feed == c->stream->feed)
+ c1->state = HTTPSTATE_SEND_DATA;
+ }
+ } else {
+ /* We have a header in our hands that contains useful data */
+ AVFormatContext *s = avformat_alloc_context();
+ AVIOContext *pb;
+ AVInputFormat *fmt_in;
+ int i;
+
+ if (!s)
+ goto fail;
+
+ /* use feed output format name to find corresponding input format */
+ fmt_in = av_find_input_format(feed->fmt->name);
+ if (!fmt_in)
+ goto fail;
+
+ pb = avio_alloc_context(c->buffer, c->buffer_end - c->buffer,
+ 0, NULL, NULL, NULL, NULL);
+ if (!pb)
+ goto fail;
+
+ pb->seekable = 0;
+
+ s->pb = pb;
+ if (avformat_open_input(&s, c->stream->feed_filename, fmt_in, NULL) < 0) {
+ av_freep(&pb);
+ goto fail;
+ }
+
+ /* Now we have the actual streams */
+ if (s->nb_streams != feed->nb_streams) {
+ avformat_close_input(&s);
+ av_freep(&pb);
+ http_log("Feed '%s' stream number does not match registered feed\n",
+ c->stream->feed_filename);
+ goto fail;
+ }
+
+ for (i = 0; i < s->nb_streams; i++) {
+ AVStream *fst = feed->streams[i];
+ AVStream *st = s->streams[i];
+ avcodec_copy_context(fst->codec, st->codec);
+ }
+
+ avformat_close_input(&s);
+ av_freep(&pb);
+ }
+ c->buffer_ptr = c->buffer;
+ }
+
+ return 0;
+ fail:
+ c->stream->feed_opened = 0;
+ close(c->feed_fd);
+ /* wake up any waiting connections to stop waiting for feed */
+ for(c1 = first_http_ctx; c1; c1 = c1->next) {
+ if (c1->state == HTTPSTATE_WAIT_FEED &&
+ c1->stream->feed == c->stream->feed)
+ c1->state = HTTPSTATE_SEND_DATA_TRAILER;
+ }
+ return -1;
+}
+
+/********************************************************************/
+/* RTSP handling */
+
+static void rtsp_reply_header(HTTPContext *c, enum RTSPStatusCode error_number)
+{
+ const char *str;
+ time_t ti;
+ struct tm *tm;
+ char buf2[32];
+
+ str = RTSP_STATUS_CODE2STRING(error_number);
+ if (!str)
+ str = "Unknown Error";
+
+ avio_printf(c->pb, "RTSP/1.0 %d %s\r\n", error_number, str);
+ avio_printf(c->pb, "CSeq: %d\r\n", c->seq);
+
+ /* output GMT time */
+ ti = time(NULL);
+ tm = gmtime(&ti);
+ strftime(buf2, sizeof(buf2), "%a, %d %b %Y %H:%M:%S", tm);
+ avio_printf(c->pb, "Date: %s GMT\r\n", buf2);
+}
+
+static void rtsp_reply_error(HTTPContext *c, enum RTSPStatusCode error_number)
+{
+ rtsp_reply_header(c, error_number);
+ avio_printf(c->pb, "\r\n");
+}
+
+static int rtsp_parse_request(HTTPContext *c)
+{
+ const char *p, *p1, *p2;
+ char cmd[32];
+ char url[1024];
+ char protocol[32];
+ char line[1024];
+ int len;
+ RTSPMessageHeader header1 = { 0 }, *header = &header1;
+
+ c->buffer_ptr[0] = '\0';
+ p = c->buffer;
+
+ get_word(cmd, sizeof(cmd), &p);
+ get_word(url, sizeof(url), &p);
+ get_word(protocol, sizeof(protocol), &p);
+
+ av_strlcpy(c->method, cmd, sizeof(c->method));
+ av_strlcpy(c->url, url, sizeof(c->url));
+ av_strlcpy(c->protocol, protocol, sizeof(c->protocol));
+
+ if (avio_open_dyn_buf(&c->pb) < 0) {
+ /* XXX: cannot do more */
+ c->pb = NULL; /* safety */
+ return -1;
+ }
+
+ /* check version name */
+ if (strcmp(protocol, "RTSP/1.0")) {
+ rtsp_reply_error(c, RTSP_STATUS_VERSION);
+ goto the_end;
+ }
+
+ /* parse each header line */
+ /* skip to next line */
+ while (*p != '\n' && *p != '\0')
+ p++;
+ if (*p == '\n')
+ p++;
+ while (*p != '\0') {
+ p1 = memchr(p, '\n', (char *)c->buffer_ptr - p);
+ if (!p1)
+ break;
+ p2 = p1;
+ if (p2 > p && p2[-1] == '\r')
+ p2--;
+ /* skip empty line */
+ if (p2 == p)
+ break;
+ len = p2 - p;
+ if (len > sizeof(line) - 1)
+ len = sizeof(line) - 1;
+ memcpy(line, p, len);
+ line[len] = '\0';
+ ff_rtsp_parse_line(NULL, header, line, NULL, NULL);
+ p = p1 + 1;
+ }
+
+ /* handle sequence number */
+ c->seq = header->seq;
+
+ if (!strcmp(cmd, "DESCRIBE"))
+ rtsp_cmd_describe(c, url);
+ else if (!strcmp(cmd, "OPTIONS"))
+ rtsp_cmd_options(c, url);
+ else if (!strcmp(cmd, "SETUP"))
+ rtsp_cmd_setup(c, url, header);
+ else if (!strcmp(cmd, "PLAY"))
+ rtsp_cmd_play(c, url, header);
+ else if (!strcmp(cmd, "PAUSE"))
+ rtsp_cmd_interrupt(c, url, header, 1);
+ else if (!strcmp(cmd, "TEARDOWN"))
+ rtsp_cmd_interrupt(c, url, header, 0);
+ else
+ rtsp_reply_error(c, RTSP_STATUS_METHOD);
+
+ the_end:
+ len = avio_close_dyn_buf(c->pb, &c->pb_buffer);
+ c->pb = NULL; /* safety */
+ 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;
+ return 0;
+}
+
+static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
+ struct in_addr my_ip)
+{
+ AVFormatContext *avc;
+ AVStream *avs = NULL;
+ AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
+ AVDictionaryEntry *entry = av_dict_get(stream->metadata, "title", NULL, 0);
+ int i;
+
+ *pbuffer = NULL;
+
+ avc = avformat_alloc_context();
+ if (!avc || !rtp_format)
+ return -1;
+
+ avc->oformat = rtp_format;
+ av_dict_set(&avc->metadata, "title",
+ entry ? entry->value : "No Title", 0);
+ avc->nb_streams = stream->nb_streams;
+ if (stream->is_multicast) {
+ snprintf(avc->filename, 1024, "rtp://%s:%d?multicast=1?ttl=%d",
+ inet_ntoa(stream->multicast_ip),
+ stream->multicast_port, stream->multicast_ttl);
+ } else
+ snprintf(avc->filename, 1024, "rtp://0.0.0.0");
+
+ avc->streams = av_malloc_array(avc->nb_streams, sizeof(*avc->streams));
+ if (!avc->streams)
+ goto sdp_done;
+
+ avs = av_malloc_array(avc->nb_streams, sizeof(*avs));
+ if (!avs)
+ goto sdp_done;
+
+ for(i = 0; i < stream->nb_streams; i++) {
+ avc->streams[i] = &avs[i];
+ avc->streams[i]->codec = stream->streams[i]->codec;
+ avcodec_parameters_from_context(stream->streams[i]->codecpar, stream->streams[i]->codec);
+ avc->streams[i]->codecpar = stream->streams[i]->codecpar;
+ }
+ *pbuffer = av_mallocz(2048);
+ if (!*pbuffer)
+ goto sdp_done;
+ av_sdp_create(&avc, 1, *pbuffer, 2048);
+
+ sdp_done:
+ av_freep(&avc->streams);
+ av_dict_free(&avc->metadata);
+ av_free(avc);
+ av_free(avs);
+
+ return *pbuffer ? strlen(*pbuffer) : AVERROR(ENOMEM);
+}
+
+static void rtsp_cmd_options(HTTPContext *c, const char *url)
+{
+ /* 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",
+ "OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE");
+ avio_printf(c->pb, "\r\n");
+}
+
+static void rtsp_cmd_describe(HTTPContext *c, const char *url)
+{
+ FFServerStream *stream;
+ char path1[1024];
+ const char *path;
+ uint8_t *content;
+ int content_length;
+ socklen_t len;
+ struct sockaddr_in my_addr;
+
+ /* find which URL is asked */
+ av_url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url);
+ path = path1;
+ if (*path == '/')
+ path++;
+
+ for(stream = config.first_stream; stream; stream = stream->next) {
+ if (!stream->is_feed &&
+ stream->fmt && !strcmp(stream->fmt->name, "rtp") &&
+ !strcmp(path, stream->filename)) {
+ goto found;
+ }
+ }
+ /* no stream found */
+ rtsp_reply_error(c, RTSP_STATUS_NOT_FOUND);
+ return;
+
+ found:
+ /* prepare the media description in SDP format */
+
+ /* get the host IP */
+ len = sizeof(my_addr);
+ getsockname(c->fd, (struct sockaddr *)&my_addr, &len);
+ content_length = prepare_sdp_description(stream, &content,
+ my_addr.sin_addr);
+ if (content_length < 0) {
+ rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
+ return;
+ }
+ rtsp_reply_header(c, RTSP_STATUS_OK);
+ avio_printf(c->pb, "Content-Base: %s/\r\n", url);
+ avio_printf(c->pb, "Content-Type: application/sdp\r\n");
+ avio_printf(c->pb, "Content-Length: %d\r\n", content_length);
+ avio_printf(c->pb, "\r\n");
+ avio_write(c->pb, content, content_length);
+ av_free(content);
+}
+
+static HTTPContext *find_rtp_session(const char *session_id)
+{
+ HTTPContext *c;
+
+ if (session_id[0] == '\0')
+ return NULL;
+
+ for(c = first_http_ctx; c; c = c->next) {
+ if (!strcmp(c->session_id, session_id))
+ return c;
+ }
+ return NULL;
+}
+
+static RTSPTransportField *find_transport(RTSPMessageHeader *h, enum RTSPLowerTransport lower_transport)
+{
+ RTSPTransportField *th;
+ int i;
+
+ for(i=0;i<h->nb_transports;i++) {
+ th = &h->transports[i];
+ if (th->lower_transport == lower_transport)
+ return th;
+ }
+ return NULL;
+}
+
+static void rtsp_cmd_setup(HTTPContext *c, const char *url,
+ RTSPMessageHeader *h)
+{
+ FFServerStream *stream;
+ int stream_index, rtp_port, rtcp_port;
+ char buf[1024];
+ char path1[1024];
+ const char *path;
+ HTTPContext *rtp_c;
+ RTSPTransportField *th;
+ struct sockaddr_in dest_addr;
+ RTSPActionServerSetup setup;
+
+ /* find which URL is asked */
+ av_url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url);
+ path = path1;
+ if (*path == '/')
+ path++;
+
+ /* now check each stream */
+ for(stream = config.first_stream; stream; stream = stream->next) {
+ if (stream->is_feed || !stream->fmt ||
+ strcmp(stream->fmt->name, "rtp")) {
+ continue;
+ }
+ /* accept aggregate filenames only if single stream */
+ if (!strcmp(path, stream->filename)) {
+ if (stream->nb_streams != 1) {
+ rtsp_reply_error(c, RTSP_STATUS_AGGREGATE);
+ return;
+ }
+ stream_index = 0;
+ goto found;
+ }
+
+ for(stream_index = 0; stream_index < stream->nb_streams;
+ stream_index++) {
+ snprintf(buf, sizeof(buf), "%s/streamid=%d",
+ stream->filename, stream_index);
+ if (!strcmp(path, buf))
+ goto found;
+ }
+ }
+ /* no stream found */
+ rtsp_reply_error(c, RTSP_STATUS_SERVICE); /* XXX: right error ? */
+ return;
+ found:
+
+ /* generate session id if needed */
+ if (h->session_id[0] == '\0') {
+ unsigned random0 = av_lfg_get(&random_state);
+ unsigned random1 = av_lfg_get(&random_state);
+ snprintf(h->session_id, sizeof(h->session_id), "%08x%08x",
+ random0, random1);
+ }
+
+ /* find RTP session, and create it if none found */
+ rtp_c = find_rtp_session(h->session_id);
+ if (!rtp_c) {
+ /* always prefer UDP */
+ th = find_transport(h, RTSP_LOWER_TRANSPORT_UDP);
+ if (!th) {
+ th = find_transport(h, RTSP_LOWER_TRANSPORT_TCP);
+ if (!th) {
+ rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
+ return;
+ }
+ }
+
+ rtp_c = rtp_new_connection(&c->from_addr, stream, h->session_id,
+ th->lower_transport);
+ if (!rtp_c) {
+ rtsp_reply_error(c, RTSP_STATUS_BANDWIDTH);
+ return;
+ }
+
+ /* open input stream */
+ if (open_input_stream(rtp_c, "") < 0) {
+ rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
+ return;
+ }
+ }
+
+ /* test if stream is OK (test needed because several SETUP needs
+ * to be done for a given file) */
+ if (rtp_c->stream != stream) {
+ rtsp_reply_error(c, RTSP_STATUS_SERVICE);
+ return;
+ }
+
+ /* test if stream is already set up */
+ if (rtp_c->rtp_ctx[stream_index]) {
+ rtsp_reply_error(c, RTSP_STATUS_STATE);
+ return;
+ }
+
+ /* check transport */
+ th = find_transport(h, rtp_c->rtp_protocol);
+ if (!th || (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
+ th->client_port_min <= 0)) {
+ rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
+ return;
+ }
+
+ /* setup default options */
+ setup.transport_option[0] = '\0';
+ dest_addr = rtp_c->from_addr;
+ dest_addr.sin_port = htons(th->client_port_min);
+
+ /* setup stream */
+ if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr, c) < 0) {
+ rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
+ return;
+ }
+
+ /* now everything is OK, so we can send the connection parameters */
+ rtsp_reply_header(c, RTSP_STATUS_OK);
+ /* session ID */
+ avio_printf(c->pb, "Session: %s\r\n", rtp_c->session_id);
+
+ switch(rtp_c->rtp_protocol) {
+ case RTSP_LOWER_TRANSPORT_UDP:
+ rtp_port = ff_rtp_get_local_rtp_port(rtp_c->rtp_handles[stream_index]);
+ rtcp_port = ff_rtp_get_local_rtcp_port(rtp_c->rtp_handles[stream_index]);
+ avio_printf(c->pb, "Transport: RTP/AVP/UDP;unicast;"
+ "client_port=%d-%d;server_port=%d-%d",
+ th->client_port_min, th->client_port_max,
+ rtp_port, rtcp_port);
+ break;
+ case RTSP_LOWER_TRANSPORT_TCP:
+ avio_printf(c->pb, "Transport: RTP/AVP/TCP;interleaved=%d-%d",
+ stream_index * 2, stream_index * 2 + 1);
+ break;
+ default:
+ break;
+ }
+ if (setup.transport_option[0] != '\0')
+ avio_printf(c->pb, ";%s", setup.transport_option);
+ avio_printf(c->pb, "\r\n");
+
+
+ avio_printf(c->pb, "\r\n");
+}
+
+
+/**
+ * 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)
+{
+ HTTPContext *rtp_c;
+ char path1[1024];
+ const char *path;
+ char buf[1024];
+ int s, len;
+
+ rtp_c = find_rtp_session(session_id);
+ if (!rtp_c)
+ return NULL;
+
+ /* find which URL is asked */
+ av_url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url);
+ path = path1;
+ if (*path == '/')
+ path++;
+ if(!strcmp(path, rtp_c->stream->filename)) return rtp_c;
+ for(s=0; s<rtp_c->stream->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? */
+ return rtp_c;
+ }
+ len = strlen(path);
+ if (len > 0 && path[len - 1] == '/' &&
+ !strncmp(path, rtp_c->stream->filename, len - 1))
+ return rtp_c;
+ return NULL;
+}
+
+static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPMessageHeader *h)
+{
+ HTTPContext *rtp_c;
+
+ rtp_c = find_rtp_session_with_url(url, h->session_id);
+ if (!rtp_c) {
+ rtsp_reply_error(c, RTSP_STATUS_SESSION);
+ return;
+ }
+
+ if (rtp_c->state != HTTPSTATE_SEND_DATA &&
+ rtp_c->state != HTTPSTATE_WAIT_FEED &&
+ rtp_c->state != HTTPSTATE_READY) {
+ rtsp_reply_error(c, RTSP_STATUS_STATE);
+ return;
+ }
+
+ rtp_c->state = HTTPSTATE_SEND_DATA;
+
+ /* now everything is OK, so we can send the connection parameters */
+ rtsp_reply_header(c, RTSP_STATUS_OK);
+ /* session ID */
+ avio_printf(c->pb, "Session: %s\r\n", rtp_c->session_id);
+ avio_printf(c->pb, "\r\n");
+}
+
+static void rtsp_cmd_interrupt(HTTPContext *c, const char *url,
+ RTSPMessageHeader *h, int pause_only)
+{
+ HTTPContext *rtp_c;
+
+ rtp_c = find_rtp_session_with_url(url, h->session_id);
+ if (!rtp_c) {
+ rtsp_reply_error(c, RTSP_STATUS_SESSION);
+ return;
+ }
+
+ if (pause_only) {
+ if (rtp_c->state != HTTPSTATE_SEND_DATA &&
+ rtp_c->state != HTTPSTATE_WAIT_FEED) {
+ rtsp_reply_error(c, RTSP_STATUS_STATE);
+ return;
+ }
+ rtp_c->state = HTTPSTATE_READY;
+ rtp_c->first_pts = AV_NOPTS_VALUE;
+ }
+
+ /* now everything is OK, so we can send the connection parameters */
+ rtsp_reply_header(c, RTSP_STATUS_OK);
+ /* session ID */
+ avio_printf(c->pb, "Session: %s\r\n", rtp_c->session_id);
+ avio_printf(c->pb, "\r\n");
+
+ if (!pause_only)
+ close_connection(rtp_c);
+}
+
+/********************************************************************/
+/* RTP handling */
+
+static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
+ FFServerStream *stream,
+ const char *session_id,
+ enum RTSPLowerTransport 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 >= config.nb_max_connections)
+ goto fail;
+
+ /* add a new connection */
+ c = av_mallocz(sizeof(HTTPContext));
+ if (!c)
+ goto fail;
+
+ c->fd = -1;
+ c->poll_entry = NULL;
+ c->from_addr = *from_addr;
+ c->buffer_size = IOBUFFER_INIT_SIZE;
+ c->buffer = av_malloc(c->buffer_size);
+ if (!c->buffer)
+ goto fail;
+ nb_connections++;
+ c->stream = stream;
+ av_strlcpy(c->session_id, session_id, sizeof(c->session_id));
+ c->state = HTTPSTATE_READY;
+ c->is_packetized = 1;
+ c->rtp_protocol = rtp_protocol;
+
+ /* protocol is shown in statistics */
+ switch(c->rtp_protocol) {
+ case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
+ proto_str = "MCAST";
+ break;
+ case RTSP_LOWER_TRANSPORT_UDP:
+ proto_str = "UDP";
+ break;
+ case RTSP_LOWER_TRANSPORT_TCP:
+ proto_str = "TCP";
+ break;
+ default:
+ proto_str = "???";
+ break;
+ }
+ av_strlcpy(c->protocol, "RTP/", sizeof(c->protocol));
+ av_strlcat(c->protocol, proto_str, sizeof(c->protocol));
+
+ current_bandwidth += stream->bandwidth;
+
+ c->next = first_http_ctx;
+ first_http_ctx = c;
+ return c;
+
+ fail:
+ if (c) {
+ av_freep(&c->buffer);
+ av_free(c);
+ }
+ 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.
+ */
+static int rtp_new_av_stream(HTTPContext *c,
+ int stream_index, struct sockaddr_in *dest_addr,
+ HTTPContext *rtsp_c)
+{
+ AVFormatContext *ctx;
+ AVStream *st;
+ char *ipaddr;
+ URLContext *h = NULL;
+ uint8_t *dummy_buf;
+ int max_packet_size;
+ void *st_internal;
+
+ /* now we can open the relevant output stream */
+ ctx = avformat_alloc_context();
+ if (!ctx)
+ return -1;
+ ctx->oformat = av_guess_format("rtp", NULL, NULL);
+
+ st = avformat_new_stream(ctx, NULL);
+ if (!st)
+ goto fail;
+
+ av_freep(&st->codec);
+ av_freep(&st->info);
+ st_internal = st->internal;
+
+ if (!c->stream->feed ||
+ c->stream->feed == c->stream)
+ memcpy(st, c->stream->streams[stream_index], sizeof(AVStream));
+ else
+ memcpy(st,
+ c->stream->feed->streams[c->stream->feed_streams[stream_index]],
+ sizeof(AVStream));
+ st->priv_data = NULL;
+ st->internal = st_internal;
+
+ /* build destination RTP address */
+ ipaddr = inet_ntoa(dest_addr->sin_addr);
+
+ switch(c->rtp_protocol) {
+ case RTSP_LOWER_TRANSPORT_UDP:
+ case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
+ /* RTP/UDP case */
+
+ /* XXX: also pass as parameter to function ? */
+ if (c->stream->is_multicast) {
+ int ttl;
+ ttl = c->stream->multicast_ttl;
+ if (!ttl)
+ ttl = 16;
+ snprintf(ctx->filename, sizeof(ctx->filename),
+ "rtp://%s:%d?multicast=1&ttl=%d",
+ ipaddr, ntohs(dest_addr->sin_port), ttl);
+ } else {
+ snprintf(ctx->filename, sizeof(ctx->filename),
+ "rtp://%s:%d", ipaddr, ntohs(dest_addr->sin_port));
+ }
+
+ if (ffurl_open(&h, ctx->filename, AVIO_FLAG_WRITE, NULL, NULL) < 0)
+ goto fail;
+ c->rtp_handles[stream_index] = h;
+ max_packet_size = h->max_packet_size;
+ break;
+ case RTSP_LOWER_TRANSPORT_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 - - \"PLAY %s/streamid=%d %s\"\n",
+ ipaddr, ntohs(dest_addr->sin_port),
+ c->stream->filename, stream_index, c->protocol);
+
+ /* 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)
+ /* XXX: close stream */
+ goto fail;
+
+ if (avformat_write_header(ctx, NULL) < 0) {
+ fail:
+ if (h)
+ ffurl_close(h);
+ av_free(st);
+ av_free(ctx);
+ return -1;
+ }
+ avio_close_dyn_buf(ctx->pb, &dummy_buf);
+ ctx->pb = NULL;
+ av_free(dummy_buf);
+
+ c->rtp_ctx[stream_index] = ctx;
+ return 0;
+}
+
+/********************************************************************/
+/* ffserver initialization */
+
+/* FIXME: This code should use avformat_new_stream() */
+static AVStream *add_av_stream1(FFServerStream *stream,
+ AVCodecContext *codec, int copy)
+{
+ AVStream *fst;
+
+ if(stream->nb_streams >= FF_ARRAY_ELEMS(stream->streams))
+ return NULL;
+
+ fst = av_mallocz(sizeof(AVStream));
+ if (!fst)
+ return NULL;
+ if (copy) {
+ fst->codec = avcodec_alloc_context3(codec->codec);
+ if (!fst->codec) {
+ av_free(fst);
+ return NULL;
+ }
+ avcodec_copy_context(fst->codec, codec);
+ } 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->internal = av_mallocz(sizeof(*fst->internal));
+ fst->internal->avctx = avcodec_alloc_context3(NULL);
+ fst->codecpar = avcodec_parameters_alloc();
+ fst->index = stream->nb_streams;
+ avpriv_set_pts_info(fst, 33, 1, 90000);
+ fst->sample_aspect_ratio = codec->sample_aspect_ratio;
+ stream->streams[stream->nb_streams++] = fst;
+ return fst;
+}
+
+/* return the stream number in the feed */
+static int add_av_stream(FFServerStream *feed, AVStream *st)
+{
+ AVStream *fst;
+ AVCodecContext *av, *av1;
+ int i;
+
+ av = st->codec;
+ for(i=0;i<feed->nb_streams;i++) {
+ av1 = feed->streams[i]->codec;
+ if (av1->codec_id == av->codec_id &&
+ av1->codec_type == av->codec_type &&
+ av1->bit_rate == av->bit_rate) {
+
+ switch(av->codec_type) {
+ case AVMEDIA_TYPE_AUDIO:
+ if (av1->channels == av->channels &&
+ av1->sample_rate == av->sample_rate)
+ return i;
+ break;
+ case AVMEDIA_TYPE_VIDEO:
+ if (av1->width == av->width &&
+ av1->height == av->height &&
+ av1->time_base.den == av->time_base.den &&
+ av1->time_base.num == av->time_base.num &&
+ av1->gop_size == av->gop_size)
+ return i;
+ break;
+ default:
+ abort();
+ }
+ }
+ }
+
+ fst = add_av_stream1(feed, av, 0);
+ if (!fst)
+ return -1;
+ if (av_stream_get_recommended_encoder_configuration(st))
+ av_stream_set_recommended_encoder_configuration(fst,
+ av_strdup(av_stream_get_recommended_encoder_configuration(st)));
+ return feed->nb_streams - 1;
+}
+
+static void remove_stream(FFServerStream *stream)
+{
+ FFServerStream **ps;
+ ps = &config.first_stream;
+ while (*ps) {
+ if (*ps == stream)
+ *ps = (*ps)->next;
+ else
+ ps = &(*ps)->next;
+ }
+}
+
+/* specific MPEG4 handling : we extract the raw parameters */
+static void extract_mpeg4_header(AVFormatContext *infile)
+{
+ int mpeg4_count, i, size;
+ AVPacket pkt;
+ AVStream *st;
+ const uint8_t *p;
+
+ infile->flags |= AVFMT_FLAG_NOFILLIN | AVFMT_FLAG_NOPARSE;
+
+ mpeg4_count = 0;
+ for(i=0;i<infile->nb_streams;i++) {
+ st = infile->streams[i];
+ if (st->codec->codec_id == AV_CODEC_ID_MPEG4 &&
+ st->codec->extradata_size == 0) {
+ mpeg4_count++;
+ }
+ }
+ if (!mpeg4_count)
+ return;
+
+ printf("MPEG4 without extra data: trying to find header in %s\n",
+ infile->filename);
+ while (mpeg4_count > 0) {
+ if (av_read_frame(infile, &pkt) < 0)
+ break;
+ st = infile->streams[pkt.stream_index];
+ if (st->codec->codec_id == AV_CODEC_ID_MPEG4 &&
+ st->codec->extradata_size == 0) {
+ av_freep(&st->codec->extradata);
+ /* fill extradata with the header */
+ /* XXX: we make hard suppositions here ! */
+ p = pkt.data;
+ while (p < pkt.data + pkt.size - 4) {
+ /* stop when vop header is found */
+ if (p[0] == 0x00 && p[1] == 0x00 &&
+ p[2] == 0x01 && p[3] == 0xb6) {
+ size = p - pkt.data;
+ st->codec->extradata = av_mallocz(size + AV_INPUT_BUFFER_PADDING_SIZE);
+ st->codec->extradata_size = size;
+ memcpy(st->codec->extradata, pkt.data, size);
+ break;
+ }
+ p++;
+ }
+ mpeg4_count--;
+ }
+ av_packet_unref(&pkt);
+ }
+}
+
+/* compute the needed AVStream for each file */
+static void build_file_streams(void)
+{
+ FFServerStream *stream;
+ AVFormatContext *infile;
+ int i, ret;
+
+ /* gather all streams */
+ for(stream = config.first_stream; stream; stream = stream->next) {
+ infile = NULL;
+
+ if (stream->stream_type != STREAM_TYPE_LIVE || stream->feed)
+ continue;
+
+ /* the stream comes from a file */
+ /* try to open the file */
+ /* open stream */
+
+
+ /* specific case: if transport stream output to RTP,
+ * we use a raw transport stream reader */
+ if (stream->fmt && !strcmp(stream->fmt->name, "rtp"))
+ av_dict_set(&stream->in_opts, "mpeg2ts_compute_pcr", "1", 0);
+
+ if (!stream->feed_filename[0]) {
+ http_log("Unspecified feed file for stream '%s'\n",
+ stream->filename);
+ goto fail;
+ }
+
+ http_log("Opening feed file '%s' for stream '%s'\n",
+ stream->feed_filename, stream->filename);
+
+ ret = avformat_open_input(&infile, stream->feed_filename,
+ stream->ifmt, &stream->in_opts);
+ if (ret < 0) {
+ http_log("Could not open '%s': %s\n", stream->feed_filename,
+ av_err2str(ret));
+ /* remove stream (no need to spend more time on it) */
+ fail:
+ remove_stream(stream);
+ } else {
+ /* find all the AVStreams inside and reference them in
+ * 'stream' */
+ if (avformat_find_stream_info(infile, NULL) < 0) {
+ http_log("Could not find codec parameters from '%s'\n",
+ stream->feed_filename);
+ avformat_close_input(&infile);
+ goto fail;
+ }
+ extract_mpeg4_header(infile);
+
+ for(i=0;i<infile->nb_streams;i++)
+ add_av_stream1(stream, infile->streams[i]->codec, 1);
+
+ avformat_close_input(&infile);
+ }
+ }
+}
+
+static inline
+int check_codec_match(AVCodecContext *ccf, AVCodecContext *ccs, int stream)
+{
+ int matches = 1;
+
+#define CHECK_CODEC(x) (ccf->x != ccs->x)
+ if (CHECK_CODEC(codec_id) || CHECK_CODEC(codec_type)) {
+ http_log("Codecs do not match for stream %d\n", stream);
+ matches = 0;
+ } else if (CHECK_CODEC(bit_rate) || CHECK_CODEC(flags)) {
+ http_log("Codec bitrates do not match for stream %d\n", stream);
+ matches = 0;
+ } else if (ccf->codec_type == AVMEDIA_TYPE_VIDEO) {
+ if (CHECK_CODEC(time_base.den) ||
+ CHECK_CODEC(time_base.num) ||
+ CHECK_CODEC(width) ||
+ CHECK_CODEC(height)) {
+ http_log("Codec width, height or framerate do not match for stream %d\n", stream);
+ matches = 0;
+ }
+ } else if (ccf->codec_type == AVMEDIA_TYPE_AUDIO) {
+ if (CHECK_CODEC(sample_rate) ||
+ CHECK_CODEC(channels) ||
+ CHECK_CODEC(frame_size)) {
+ http_log("Codec sample_rate, channels, frame_size do not match for stream %d\n", stream);
+ matches = 0;
+ }
+ } else {
+ http_log("Unknown codec type for stream %d\n", stream);
+ matches = 0;
+ }
+
+ return matches;
+}
+
+/* compute the needed AVStream for each feed */
+static int build_feed_streams(void)
+{
+ FFServerStream *stream, *feed;
+ int i, fd;
+
+ /* gather all streams */
+ for(stream = config.first_stream; stream; stream = stream->next) {
+ feed = stream->feed;
+ if (!feed)
+ continue;
+
+ if (stream->is_feed) {
+ for(i=0;i<stream->nb_streams;i++)
+ stream->feed_streams[i] = i;
+ continue;
+ }
+ /* we handle a stream coming from a feed */
+ for(i=0;i<stream->nb_streams;i++)
+ stream->feed_streams[i] = add_av_stream(feed, stream->streams[i]);
+ }
+
+ /* create feed files if needed */
+ for(feed = config.first_feed; feed; feed = feed->next_feed) {
+
+ if (avio_check(feed->feed_filename, AVIO_FLAG_READ) > 0) {
+ AVFormatContext *s = NULL;
+ int matches = 0;
+
+ /* See if it matches */
+
+ if (avformat_open_input(&s, feed->feed_filename, NULL, NULL) < 0) {
+ http_log("Deleting feed file '%s' as it appears "
+ "to be corrupt\n",
+ feed->feed_filename);
+ goto drop;
+ }
+
+ /* set buffer size */
+ if (ffio_set_buf_size(s->pb, FFM_PACKET_SIZE) < 0) {
+ http_log("Failed to set buffer size\n");
+ avformat_close_input(&s);
+ goto bail;
+ }
+
+ /* Now see if it matches */
+ if (s->nb_streams != feed->nb_streams) {
+ http_log("Deleting feed file '%s' as stream counts "
+ "differ (%d != %d)\n",
+ feed->feed_filename, s->nb_streams, feed->nb_streams);
+ goto drop;
+ }
+
+ matches = 1;
+ for(i=0;i<s->nb_streams;i++) {
+ AVStream *sf, *ss;
+
+ sf = feed->streams[i];
+ ss = s->streams[i];
+
+ if (sf->index != ss->index || sf->id != ss->id) {
+ http_log("Index & Id do not match for stream %d (%s)\n",
+ i, feed->feed_filename);
+ matches = 0;
+ break;
+ }
+
+ matches = check_codec_match (sf->codec, ss->codec, i);
+ if (!matches)
+ break;
+ }
+
+drop:
+ if (s)
+ avformat_close_input(&s);
+
+ if (!matches) {
+ if (feed->readonly) {
+ http_log("Unable to delete read-only feed file '%s'\n",
+ feed->feed_filename);
+ goto bail;
+ }
+ unlink(feed->feed_filename);
+ }
+ }
+
+ if (avio_check(feed->feed_filename, AVIO_FLAG_WRITE) <= 0) {
+ AVFormatContext *s = avformat_alloc_context();
+
+ if (!s) {
+ http_log("Failed to allocate context\n");
+ goto bail;
+ }
+
+ if (feed->readonly) {
+ http_log("Unable to create feed file '%s' as it is "
+ "marked readonly\n",
+ feed->feed_filename);
+ avformat_free_context(s);
+ goto bail;
+ }
+
+ /* only write the header of the ffm file */
+ if (avio_open(&s->pb, feed->feed_filename, AVIO_FLAG_WRITE) < 0) {
+ http_log("Could not open output feed file '%s'\n",
+ feed->feed_filename);
+ avformat_free_context(s);
+ goto bail;
+ }
+ s->oformat = feed->fmt;
+ s->nb_streams = feed->nb_streams;
+ s->streams = feed->streams;
+ if (avformat_write_header(s, NULL) < 0) {
+ http_log("Container doesn't support the required parameters\n");
+ avio_closep(&s->pb);
+ s->streams = NULL;
+ s->nb_streams = 0;
+ avformat_free_context(s);
+ goto bail;
+ }
+ /* XXX: need better API */
+ av_freep(&s->priv_data);
+ avio_closep(&s->pb);
+ s->streams = NULL;
+ s->nb_streams = 0;
+ avformat_free_context(s);
+ }
+
+ /* get feed size and write index */
+ fd = open(feed->feed_filename, O_RDONLY);
+ if (fd < 0) {
+ http_log("Could not open output feed file '%s'\n",
+ feed->feed_filename);
+ goto bail;
+ }
+
+ feed->feed_write_index = FFMAX(ffm_read_write_index(fd),
+ FFM_PACKET_SIZE);
+ feed->feed_size = lseek(fd, 0, SEEK_END);
+ /* ensure that we do not wrap before the end of file */
+ if (feed->feed_max_size && feed->feed_max_size < feed->feed_size)
+ feed->feed_max_size = feed->feed_size;
+
+ close(fd);
+ }
+ return 0;
+
+bail:
+ return -1;
+}
+
+/* compute the bandwidth used by each stream */
+static void compute_bandwidth(void)
+{
+ unsigned bandwidth;
+ int i;
+ FFServerStream *stream;
+
+ for(stream = config.first_stream; stream; stream = stream->next) {
+ bandwidth = 0;
+ for(i=0;i<stream->nb_streams;i++) {
+ AVStream *st = stream->streams[i];
+ switch(st->codec->codec_type) {
+ case AVMEDIA_TYPE_AUDIO:
+ case AVMEDIA_TYPE_VIDEO:
+ bandwidth += st->codec->bit_rate;
+ break;
+ default:
+ break;
+ }
+ }
+ stream->bandwidth = (bandwidth + 999) / 1000;
+ }
+}
+
+static void handle_child_exit(int sig)
+{
+ pid_t pid;
+ int status;
+ time_t uptime;
+
+ while ((pid = waitpid(-1, &status, WNOHANG)) > 0) {
+ FFServerStream *feed;
+
+ for (feed = config.first_feed; feed; feed = feed->next) {
+ if (feed->pid != pid)
+ continue;
+
+ uptime = time(0) - feed->pid_start;
+ feed->pid = 0;
+ fprintf(stderr,
+ "%s: Pid %"PRId64" exited with status %d after %"PRId64" "
+ "seconds\n",
+ feed->filename, (int64_t) pid, status, (int64_t)uptime);
+
+ if (uptime < 30)
+ /* Turn off any more restarts */
+ ffserver_free_child_args(&feed->child_argv);
+ }
+ }
+
+ need_to_start_children = 1;
+}
+
+static void opt_debug(void)
+{
+ config.debug = 1;
+ snprintf(config.logfilename, sizeof(config.logfilename), "-");
+}
+
+void show_help_default(const char *opt, const char *arg)
+{
+ printf("usage: ffserver [options]\n"
+ "Hyper fast multi format Audio/Video streaming server\n");
+ printf("\n");
+ show_help_options(options, "Main options:", 0, 0, 0);
+}
+
+static const OptionDef options[] = {
+#include "cmdutils_common_opts.h"
+ { "n", OPT_BOOL, {(void *)&no_launch }, "enable no-launch mode" },
+ { "d", 0, {(void*)opt_debug}, "enable debug mode" },
+ { "f", HAS_ARG | OPT_STRING, {(void*)&config.filename }, "use configfile instead of /etc/ffserver.conf", "configfile" },
+ { NULL },
+};
+
+int main(int argc, char **argv)
+{
+ struct sigaction sigact = { { 0 } };
+ int cfg_parsed;
+ int ret = EXIT_FAILURE;
+
+
+ config.filename = av_strdup("/etc/ffserver.conf");
+
+ parse_loglevel(argc, argv, options);
+ av_register_all();
+ avformat_network_init();
+
+ show_banner(argc, argv, options);
+
+ my_program_name = argv[0];
+
+ parse_options(NULL, argc, argv, options, NULL);
+
+ unsetenv("http_proxy"); /* Kill the http_proxy */
+
+ av_lfg_init(&random_state, av_get_random_seed());
+
+ sigact.sa_handler = handle_child_exit;
+ sigact.sa_flags = SA_NOCLDSTOP | SA_RESTART;
+ sigaction(SIGCHLD, &sigact, 0);
+
+ if ((cfg_parsed = ffserver_parse_ffconfig(config.filename, &config)) < 0) {
+ fprintf(stderr, "Error reading configuration file '%s': %s\n",
+ config.filename, av_err2str(cfg_parsed));
+ goto bail;
+ }
+
+ /* open log file if needed */
+ if (config.logfilename[0] != '\0') {
+ if (!strcmp(config.logfilename, "-"))
+ logfile = stdout;
+ else
+ logfile = fopen(config.logfilename, "a");
+ av_log_set_callback(http_av_log);
+ }
+
+ build_file_streams();
+
+ if (build_feed_streams() < 0) {
+ http_log("Could not setup feed streams\n");
+ goto bail;
+ }
+
+ compute_bandwidth();
+
+ /* signal init */
+ signal(SIGPIPE, SIG_IGN);
+
+ if (http_server() < 0) {
+ http_log("Could not start server\n");
+ goto bail;
+ }
+
+ ret=EXIT_SUCCESS;
+
+bail:
+ av_freep (&config.filename);
+ avformat_network_deinit();
+ return ret;
+}