int main(int argc, char **argv)
{
struct sigaction sigact = { { 0 } };
int ret = 0;
config_filename = av_strdup("/etc/ffserver.conf");
av_lfg_init(&random_state, av_get_random_seed());
if ((ret = parse_ffconfig(config_filename)) < 0) {
fprintf(stderr, "Error reading configuration file '%s': %s\n",
config_filename, av_err2str(ret));
exit(1);
}
av_freep(&config_filename);
// set logfile
logfile = stdout;
build_file_streams();
/* signal init */
signal(SIGPIPE, SIG_IGN);
if (start_event_loop() < 0) {
// http_log("Could not start server\n");
printf("could not start server\n");
exit(1);
}
return 0;
}
int av_parse_color(uint8_t *rgba_color, const char *color_string, int slen,
void *log_ctx)
{
char *tail, color_string2[128];
const ColorEntry *entry;
int len, hex_offset = 0;
if (color_string[0] == '#') {
hex_offset = 1;
} else if (!strncmp(color_string, "0x", 2))
hex_offset = 2;
if (slen < 0)
slen = strlen(color_string);
av_strlcpy(color_string2, color_string + hex_offset,
FFMIN(slen-hex_offset+1, sizeof(color_string2)));
if ((tail = strchr(color_string2, ALPHA_SEP)))
*tail++ = 0;
len = strlen(color_string2);
rgba_color[3] = 255;
if (!strcasecmp(color_string2, "random") || !strcasecmp(color_string2, "bikeshed")) {
int rgba = av_get_random_seed();
rgba_color[0] = rgba >> 24;
rgba_color[1] = rgba >> 16;
rgba_color[2] = rgba >> 8;
rgba_color[3] = rgba;
} else if (hex_offset ||
int av_parse_color(uint8_t *rgba_color, const char *color_string, void *log_ctx)
{
if (!strcasecmp(color_string, "random") || !strcasecmp(color_string, "bikeshed")) {
int rgba = av_get_random_seed();
rgba_color[0] = rgba >> 24;
rgba_color[1] = rgba >> 16;
rgba_color[2] = rgba >> 8;
rgba_color[3] = rgba;
} else
int generate_playback_session_id(char * ssid, int size)
{
AVLFG rnd;
char *ex_id = "314F1B49-AA3C-4715-950D-";
unsigned int session_id = 0;
char session_name[SESSLEN] = "";
av_lfg_init(&rnd, av_get_random_seed());
session_id = av_lfg_get(&rnd);
snprintf(session_name, SESSLEN, "%s%012u", ex_id, session_id);
if (ssid != NULL) {
snprintf(ssid, size, "%s", session_name);
return 0;
}
return -1;
}
int generate_segment_session_id(char * ssid, int size)
{
AVLFG rnd;
char *ex_id = "E4499E08-D4C5-4DB0-A21C-";
unsigned int session_id = 0;
char session_name[SESSLEN] = "";
av_lfg_init(&rnd, av_get_random_seed());
session_id = av_lfg_get(&rnd);
snprintf(session_name, SESSLEN, "%s%012u", ex_id, session_id);
if (ssid != NULL) {
snprintf(ssid, size, "%s", session_name);
return 0;
}
return -1;
}
int main(void)
{
int i, j, retry;
uint32_t seeds[N];
for (retry=0; retry<3; retry++){
for (i=0; i<N; i++){
seeds[i] = av_get_random_seed();
for (j=0; j<i; j++)
if (seeds[j] == seeds[i])
goto retry;
}
printf("seeds OK\n");
return 0;
retry:;
}
printf("FAIL at %d with %X\n", j, seeds[j]);
return 1;
}
static int rtp_open( hnd_t *p_handle, obe_udp_opts_t *udp_opts )
{
obe_rtp_ctx *p_rtp = calloc( 1, sizeof(*p_rtp) );
if( !p_rtp )
{
fprintf( stderr, "[rtp] malloc failed" );
return -1;
}
if( udp_open( &p_rtp->udp_handle, udp_opts ) < 0 )
{
fprintf( stderr, "[rtp] Could not create udp output" );
return -1;
}
p_rtp->ssrc = av_get_random_seed();
*p_handle = p_rtp;
return 0;
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
DrawTextContext *s = ctx->priv;
int ret;
ff_draw_init(&s->dc, inlink->format, FF_DRAW_PROCESS_ALPHA);
ff_draw_color(&s->dc, &s->fontcolor, s->fontcolor.rgba);
ff_draw_color(&s->dc, &s->shadowcolor, s->shadowcolor.rgba);
ff_draw_color(&s->dc, &s->bordercolor, s->bordercolor.rgba);
ff_draw_color(&s->dc, &s->boxcolor, s->boxcolor.rgba);
s->var_values[VAR_w] = s->var_values[VAR_W] = s->var_values[VAR_MAIN_W] = inlink->w;
s->var_values[VAR_h] = s->var_values[VAR_H] = s->var_values[VAR_MAIN_H] = inlink->h;
s->var_values[VAR_SAR] = inlink->sample_aspect_ratio.num ? av_q2d(inlink->sample_aspect_ratio) : 1;
s->var_values[VAR_DAR] = (double)inlink->w / inlink->h * s->var_values[VAR_SAR];
s->var_values[VAR_HSUB] = 1 << s->dc.hsub_max;
s->var_values[VAR_VSUB] = 1 << s->dc.vsub_max;
s->var_values[VAR_X] = NAN;
s->var_values[VAR_Y] = NAN;
s->var_values[VAR_T] = NAN;
av_lfg_init(&s->prng, av_get_random_seed());
av_expr_free(s->x_pexpr);
av_expr_free(s->y_pexpr);
s->x_pexpr = s->y_pexpr = NULL;
if ((ret = av_expr_parse(&s->x_pexpr, s->x_expr, var_names,
NULL, NULL, fun2_names, fun2, 0, ctx)) < 0 ||
(ret = av_expr_parse(&s->y_pexpr, s->y_expr, var_names,
NULL, NULL, fun2_names, fun2, 0, ctx)) < 0 ||
(ret = av_expr_parse(&s->a_pexpr, s->a_expr, var_names,
NULL, NULL, fun2_names, fun2, 0, ctx)) < 0)
return AVERROR(EINVAL);
return 0;
}
static int rtp_write_header(AVFormatContext *s1)
{
RTPMuxContext *s = s1->priv_data;
int n, ret = AVERROR(EINVAL);
AVStream *st;
if (s1->nb_streams != 1) {
av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
return AVERROR(EINVAL);
}
st = s1->streams[0];
if (!is_supported(st->codec->codec_id)) {
av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));
return -1;
}
if (s->payload_type < 0) {
/* Re-validate non-dynamic payload types */
if (st->id < RTP_PT_PRIVATE)
st->id = ff_rtp_get_payload_type(s1, st->codec, -1);
s->payload_type = st->id;
} else {
/* private option takes priority */
st->id = s->payload_type;
}
s->base_timestamp = av_get_random_seed();
s->timestamp = s->base_timestamp;
s->cur_timestamp = 0;
if (!s->ssrc)
s->ssrc = av_get_random_seed();
s->first_packet = 1;
s->first_rtcp_ntp_time = ff_ntp_time();
if (s1->start_time_realtime != 0 && s1->start_time_realtime != AV_NOPTS_VALUE)
/* Round the NTP time to whole milliseconds. */
s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
NTP_OFFSET_US;
// Pick a random sequence start number, but in the lower end of the
// available range, so that any wraparound doesn't happen immediately.
// (Immediate wraparound would be an issue for SRTP.)
if (s->seq < 0) {
if (s1->flags & AVFMT_FLAG_BITEXACT) {
s->seq = 0;
} else
s->seq = av_get_random_seed() & 0x0fff;
} else
s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
if (s1->packet_size) {
if (s1->pb->max_packet_size)
s1->packet_size = FFMIN(s1->packet_size,
s1->pb->max_packet_size);
} else
s1->packet_size = s1->pb->max_packet_size;
if (s1->packet_size <= 12) {
av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
return AVERROR(EIO);
}
s->buf = av_malloc(s1->packet_size);
if (!s->buf) {
return AVERROR(ENOMEM);
}
s->max_payload_size = s1->packet_size - 12;
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
} else {
avpriv_set_pts_info(st, 32, 1, 90000);
}
s->buf_ptr = s->buf;
switch(st->codec->codec_id) {
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
s->buf_ptr = s->buf + 4;
avpriv_set_pts_info(st, 32, 1, 90000);
break;
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
break;
case AV_CODEC_ID_MPEG2TS:
n = s->max_payload_size / TS_PACKET_SIZE;
if (n < 1)
n = 1;
s->max_payload_size = n * TS_PACKET_SIZE;
break;
case AV_CODEC_ID_H261:
if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
av_log(s, AV_LOG_ERROR,
"Packetizing H261 is experimental and produces incorrect "
"packetization for cases where GOBs don't fit into packets "
"(even though most receivers may handle it just fine). "
"Please set -f_strict experimental in order to enable it.\n");
ret = AVERROR_EXPERIMENTAL;
goto fail;
}
break;
case AV_CODEC_ID_H264:
/* check for H.264 MP4 syntax */
if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
}
break;
case AV_CODEC_ID_HEVC:
/* Only check for the standardized hvcC version of extradata, keeping
* things simple and similar to the avcC/H264 case above, instead
* of trying to handle the pre-standardization versions (as in
* libavcodec/hevc.c). */
if (st->codec->extradata_size > 21 && st->codec->extradata[0] == 1) {
s->nal_length_size = (st->codec->extradata[21] & 0x03) + 1;
}
break;
case AV_CODEC_ID_VORBIS:
case AV_CODEC_ID_THEORA:
s->max_frames_per_packet = 15;
break;
case AV_CODEC_ID_ADPCM_G722:
/* Due to a historical error, the clock rate for G722 in RTP is
* 8000, even if the sample rate is 16000. See RFC 3551. */
avpriv_set_pts_info(st, 32, 1, 8000);
break;
case AV_CODEC_ID_OPUS:
if (st->codec->channels > 2) {
av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
goto fail;
}
/* The opus RTP RFC says that all opus streams should use 48000 Hz
* as clock rate, since all opus sample rates can be expressed in
* this clock rate, and sample rate changes on the fly are supported. */
avpriv_set_pts_info(st, 32, 1, 48000);
break;
case AV_CODEC_ID_ILBC:
if (st->codec->block_align != 38 && st->codec->block_align != 50) {
av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
goto fail;
}
s->max_frames_per_packet = s->max_payload_size / st->codec->block_align;
break;
case AV_CODEC_ID_AMR_NB:
case AV_CODEC_ID_AMR_WB:
s->max_frames_per_packet = 50;
if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
n = 31;
else
n = 61;
/* max_header_toc_size + the largest AMR payload must fit */
if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
goto fail;
}
if (st->codec->channels != 1) {
av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
goto fail;
}
break;
case AV_CODEC_ID_AAC:
s->max_frames_per_packet = 50;
break;
default:
break;
}
return 0;
fail:
av_freep(&s->buf);
return ret;
}
static int rtsp_read_setup(AVFormatContext *s, char* host, char *controlurl)
{
RTSPState *rt = s->priv_data;
RTSPMessageHeader request = { 0 };
int ret = 0;
char url[1024];
RTSPStream *rtsp_st;
char responseheaders[1024];
int localport = -1;
int transportidx = 0;
int streamid = 0;
ret = rtsp_read_request(s, &request, "SETUP");
if (ret)
return ret;
rt->seq++;
if (!request.nb_transports) {
av_log(s, AV_LOG_ERROR, "No transport defined in SETUP\n");
return AVERROR_INVALIDDATA;
}
for (transportidx = 0; transportidx < request.nb_transports;
transportidx++) {
if (!request.transports[transportidx].mode_record ||
(request.transports[transportidx].lower_transport !=
RTSP_LOWER_TRANSPORT_UDP &&
request.transports[transportidx].lower_transport !=
RTSP_LOWER_TRANSPORT_TCP)) {
av_log(s, AV_LOG_ERROR, "mode=record/receive not set or transport"
" protocol not supported (yet)\n");
return AVERROR_INVALIDDATA;
}
}
if (request.nb_transports > 1)
av_log(s, AV_LOG_WARNING, "More than one transport not supported, "
"using first of all\n");
for (streamid = 0; streamid < rt->nb_rtsp_streams; streamid++) {
if (!strcmp(rt->rtsp_streams[streamid]->control_url,
controlurl))
break;
}
if (streamid == rt->nb_rtsp_streams) {
av_log(s, AV_LOG_ERROR, "Unable to find requested track\n");
return AVERROR_STREAM_NOT_FOUND;
}
rtsp_st = rt->rtsp_streams[streamid];
localport = rt->rtp_port_min;
if (request.transports[0].lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
rt->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
if ((ret = ff_rtsp_open_transport_ctx(s, rtsp_st))) {
rtsp_send_reply(s, RTSP_STATUS_TRANSPORT, NULL, request.seq);
return ret;
}
rtsp_st->interleaved_min = request.transports[0].interleaved_min;
rtsp_st->interleaved_max = request.transports[0].interleaved_max;
snprintf(responseheaders, sizeof(responseheaders), "Transport: "
"RTP/AVP/TCP;unicast;mode=receive;interleaved=%d-%d"
"\r\n", request.transports[0].interleaved_min,
request.transports[0].interleaved_max);
} else {
do {
ff_url_join(url, sizeof(url), "rtp", NULL, host, localport, NULL);
av_dlog(s, "Opening: %s", url);
ret = ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
&s->interrupt_callback, NULL);
if (ret)
localport += 2;
} while (ret || localport > rt->rtp_port_max);
if (localport > rt->rtp_port_max) {
rtsp_send_reply(s, RTSP_STATUS_TRANSPORT, NULL, request.seq);
return ret;
}
av_dlog(s, "Listening on: %d",
ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle));
if ((ret = ff_rtsp_open_transport_ctx(s, rtsp_st))) {
rtsp_send_reply(s, RTSP_STATUS_TRANSPORT, NULL, request.seq);
return ret;
}
localport = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
snprintf(responseheaders, sizeof(responseheaders), "Transport: "
"RTP/AVP/UDP;unicast;mode=receive;source=%s;"
"client_port=%d-%d;server_port=%d-%d\r\n",
host, request.transports[0].client_port_min,
request.transports[0].client_port_max, localport,
localport + 1);
}
/* Establish sessionid if not previously set */
/* Put this in a function? */
/* RFC 2326: session id must be at least 8 digits */
while (strlen(rt->session_id) < 8)
av_strlcatf(rt->session_id, 512, "%u", av_get_random_seed());
av_strlcatf(responseheaders, sizeof(responseheaders), "Session: %s\r\n",
rt->session_id);
/* Send Reply */
rtsp_send_reply(s, RTSP_STATUS_OK, responseheaders, request.seq);
rt->state = RTSP_STATE_PAUSED;
return 0;
}
static int rtp_write_header(AVFormatContext *s1)
{
RTPMuxContext *s = s1->priv_data;
int max_packet_size, n;
AVStream *st;
if (s1->nb_streams != 1)
return -1;
st = s1->streams[0];
if (!is_supported(st->codec->codec_id)) {
av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
return -1;
}
s->payload_type = ff_rtp_get_payload_type(st->codec);
if (s->payload_type < 0)
s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == AVMEDIA_TYPE_AUDIO);
s->base_timestamp = av_get_random_seed();
s->timestamp = s->base_timestamp;
s->cur_timestamp = 0;
s->ssrc = av_get_random_seed();
s->first_packet = 1;
s->first_rtcp_ntp_time = ff_ntp_time();
if (s1->start_time_realtime)
/* Round the NTP time to whole milliseconds. */
s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
NTP_OFFSET_US;
max_packet_size = url_fget_max_packet_size(s1->pb);
if (max_packet_size <= 12)
return AVERROR(EIO);
s->buf = av_malloc(max_packet_size);
if (s->buf == NULL) {
return AVERROR(ENOMEM);
}
s->max_payload_size = max_packet_size - 12;
s->max_frames_per_packet = 0;
if (s1->max_delay) {
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
if (st->codec->frame_size == 0) {
av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
} else {
s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN);
}
}
if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
/* FIXME: We should round down here... */
s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
}
}
av_set_pts_info(st, 32, 1, 90000);
switch(st->codec->codec_id) {
case CODEC_ID_MP2:
case CODEC_ID_MP3:
s->buf_ptr = s->buf + 4;
break;
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO:
break;
case CODEC_ID_MPEG2TS:
n = s->max_payload_size / TS_PACKET_SIZE;
if (n < 1)
n = 1;
s->max_payload_size = n * TS_PACKET_SIZE;
s->buf_ptr = s->buf;
break;
case CODEC_ID_H264:
/* check for H.264 MP4 syntax */
if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
}
break;
case CODEC_ID_AMR_NB:
case CODEC_ID_AMR_WB:
if (!s->max_frames_per_packet)
s->max_frames_per_packet = 12;
if (st->codec->codec_id == CODEC_ID_AMR_NB)
n = 31;
else
n = 61;
/* max_header_toc_size + the largest AMR payload must fit */
if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
return -1;
}
if (st->codec->channels != 1) {
av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
return -1;
}
case CODEC_ID_AAC:
s->num_frames = 0;
default:
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
av_set_pts_info(st, 32, 1, st->codec->sample_rate);
}
s->buf_ptr = s->buf;
break;
}
return 0;
}
/* Generate a digest reply, according to RFC 2617. */
static char *make_digest_auth(HTTPAuthState *state, const char *username,
const char *password, const char *uri,
const char *method)
{
DigestParams *digest = &state->digest_params;
int len;
uint32_t cnonce_buf[2];
char cnonce[17];
char nc[9];
int i;
char A1hash[33], A2hash[33], response[33];
struct AVMD5 *md5ctx;
uint8_t hash[16];
char *authstr;
digest->nc++;
snprintf(nc, sizeof(nc), "%08x", digest->nc);
/* Generate a client nonce. */
for (i = 0; i < 2; i++)
cnonce_buf[i] = av_get_random_seed();
ff_data_to_hex(cnonce, (const uint8_t*) cnonce_buf, sizeof(cnonce_buf), 1);
cnonce[2*sizeof(cnonce_buf)] = 0;
md5ctx = av_md5_alloc();
if (!md5ctx)
return NULL;
av_md5_init(md5ctx);
update_md5_strings(md5ctx, username, ":", state->realm, ":", password, NULL);
av_md5_final(md5ctx, hash);
ff_data_to_hex(A1hash, hash, 16, 1);
A1hash[32] = 0;
if (!strcmp(digest->algorithm, "") || !strcmp(digest->algorithm, "MD5")) {
} else if (!strcmp(digest->algorithm, "MD5-sess")) {
av_md5_init(md5ctx);
update_md5_strings(md5ctx, A1hash, ":", digest->nonce, ":", cnonce, NULL);
av_md5_final(md5ctx, hash);
ff_data_to_hex(A1hash, hash, 16, 1);
A1hash[32] = 0;
} else {
/* Unsupported algorithm */
av_free(md5ctx);
return NULL;
}
av_md5_init(md5ctx);
update_md5_strings(md5ctx, method, ":", uri, NULL);
av_md5_final(md5ctx, hash);
ff_data_to_hex(A2hash, hash, 16, 1);
A2hash[32] = 0;
av_md5_init(md5ctx);
update_md5_strings(md5ctx, A1hash, ":", digest->nonce, NULL);
if (!strcmp(digest->qop, "auth") || !strcmp(digest->qop, "auth-int")) {
update_md5_strings(md5ctx, ":", nc, ":", cnonce, ":", digest->qop, NULL);
}
update_md5_strings(md5ctx, ":", A2hash, NULL);
av_md5_final(md5ctx, hash);
ff_data_to_hex(response, hash, 16, 1);
response[32] = 0;
av_free(md5ctx);
if (!strcmp(digest->qop, "") || !strcmp(digest->qop, "auth")) {
} else if (!strcmp(digest->qop, "auth-int")) {
/* qop=auth-int not supported */
return NULL;
} else {
/* Unsupported qop value. */
return NULL;
}
len = strlen(username) + strlen(state->realm) + strlen(digest->nonce) +
strlen(uri) + strlen(response) + strlen(digest->algorithm) +
strlen(digest->opaque) + strlen(digest->qop) + strlen(cnonce) +
strlen(nc) + 150;
authstr = av_malloc(len);
if (!authstr)
return NULL;
snprintf(authstr, len, "Authorization: Digest ");
/* TODO: Escape the quoted strings properly. */
av_strlcatf(authstr, len, "username=\"%s\"", username);
av_strlcatf(authstr, len, ", realm=\"%s\"", state->realm);
av_strlcatf(authstr, len, ", nonce=\"%s\"", digest->nonce);
av_strlcatf(authstr, len, ", uri=\"%s\"", uri);
av_strlcatf(authstr, len, ", response=\"%s\"", response);
if (digest->algorithm[0])
av_strlcatf(authstr, len, ", algorithm=%s", digest->algorithm);
if (digest->opaque[0])
av_strlcatf(authstr, len, ", opaque=\"%s\"", digest->opaque);
if (digest->qop[0]) {
av_strlcatf(authstr, len, ", qop=\"%s\"", digest->qop);
av_strlcatf(authstr, len, ", cnonce=\"%s\"", cnonce);
av_strlcatf(authstr, len, ", nc=%s", nc);
}
av_strlcatf(authstr, len, "\r\n");
return authstr;
}
uint32_t ff_random_get_seed(void)
{
return av_get_random_seed();
}
static int sap_write_header(AVFormatContext *s)
{
struct SAPState *sap = s->priv_data;
char host[1024], path[1024], url[1024], announce_addr[50] = "";
char *option_list;
int port = 9875, base_port = 5004, i, pos = 0, same_port = 0, ttl = 255;
AVFormatContext **contexts = NULL;
int ret = 0;
struct sockaddr_storage localaddr;
socklen_t addrlen = sizeof(localaddr);
int udp_fd;
if (!ff_network_init())
return AVERROR(EIO);
/* extract hostname and port */
av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &base_port,
path, sizeof(path), s->filename);
if (base_port < 0)
base_port = 5004;
/* search for options */
option_list = strrchr(path, '?');
if (option_list) {
char buf[50];
if (av_find_info_tag(buf, sizeof(buf), "announce_port", option_list)) {
port = strtol(buf, NULL, 10);
}
if (av_find_info_tag(buf, sizeof(buf), "same_port", option_list)) {
same_port = strtol(buf, NULL, 10);
}
if (av_find_info_tag(buf, sizeof(buf), "ttl", option_list)) {
ttl = strtol(buf, NULL, 10);
}
if (av_find_info_tag(buf, sizeof(buf), "announce_addr", option_list)) {
av_strlcpy(announce_addr, buf, sizeof(announce_addr));
}
}
if (!announce_addr[0]) {
struct addrinfo hints = { 0 }, *ai = NULL;
hints.ai_family = AF_UNSPEC;
if (getaddrinfo(host, NULL, &hints, &ai)) {
av_log(s, AV_LOG_ERROR, "Unable to resolve %s\n", host);
ret = AVERROR(EIO);
goto fail;
}
if (ai->ai_family == AF_INET) {
/* Also known as sap.mcast.net */
av_strlcpy(announce_addr, "224.2.127.254", sizeof(announce_addr));
#if HAVE_STRUCT_SOCKADDR_IN6
} else if (ai->ai_family == AF_INET6) {
/* With IPv6, you can use the same destination in many different
* multicast subnets, to choose how far you want it routed.
* This one is intended to be routed globally. */
av_strlcpy(announce_addr, "ff0e::2:7ffe", sizeof(announce_addr));
#endif
} else {
freeaddrinfo(ai);
av_log(s, AV_LOG_ERROR, "Host %s resolved to unsupported "
"address family\n", host);
ret = AVERROR(EIO);
goto fail;
}
freeaddrinfo(ai);
}
contexts = av_mallocz(sizeof(AVFormatContext*) * s->nb_streams);
if (!contexts) {
ret = AVERROR(ENOMEM);
goto fail;
}
s->start_time_realtime = av_gettime();
for (i = 0; i < s->nb_streams; i++) {
URLContext *fd;
ff_url_join(url, sizeof(url), "rtp", NULL, host, base_port,
"?ttl=%d", ttl);
if (!same_port)
base_port += 2;
ret = ffurl_open(&fd, url, AVIO_FLAG_WRITE, &s->interrupt_callback, NULL);
if (ret) {
ret = AVERROR(EIO);
goto fail;
}
s->streams[i]->priv_data = contexts[i] =
ff_rtp_chain_mux_open(s, s->streams[i], fd, 0);
av_strlcpy(contexts[i]->filename, url, sizeof(contexts[i]->filename));
}
ff_url_join(url, sizeof(url), "udp", NULL, announce_addr, port,
"?ttl=%d&connect=1", ttl);
ret = ffurl_open(&sap->ann_fd, url, AVIO_FLAG_WRITE,
&s->interrupt_callback, NULL);
if (ret) {
ret = AVERROR(EIO);
goto fail;
}
udp_fd = ffurl_get_file_handle(sap->ann_fd);
if (getsockname(udp_fd, (struct sockaddr*) &localaddr, &addrlen)) {
ret = AVERROR(EIO);
goto fail;
}
if (localaddr.ss_family != AF_INET
#if HAVE_STRUCT_SOCKADDR_IN6
&& localaddr.ss_family != AF_INET6
#endif
) {
av_log(s, AV_LOG_ERROR, "Unsupported protocol family\n");
ret = AVERROR(EIO);
goto fail;
}
sap->ann_size = 8192;
sap->ann = av_mallocz(sap->ann_size);
if (!sap->ann) {
ret = AVERROR(EIO);
goto fail;
}
sap->ann[pos] = (1 << 5);
#if HAVE_STRUCT_SOCKADDR_IN6
if (localaddr.ss_family == AF_INET6)
sap->ann[pos] |= 0x10;
#endif
pos++;
sap->ann[pos++] = 0; /* Authentication length */
AV_WB16(&sap->ann[pos], av_get_random_seed());
pos += 2;
if (localaddr.ss_family == AF_INET) {
memcpy(&sap->ann[pos], &((struct sockaddr_in*)&localaddr)->sin_addr,
sizeof(struct in_addr));
pos += sizeof(struct in_addr);
#if HAVE_STRUCT_SOCKADDR_IN6
} else {
memcpy(&sap->ann[pos], &((struct sockaddr_in6*)&localaddr)->sin6_addr,
sizeof(struct in6_addr));
pos += sizeof(struct in6_addr);
#endif
}
av_strlcpy(&sap->ann[pos], "application/sdp", sap->ann_size - pos);
pos += strlen(&sap->ann[pos]) + 1;
if (av_sdp_create(contexts, s->nb_streams, &sap->ann[pos],
sap->ann_size - pos)) {
ret = AVERROR_INVALIDDATA;
goto fail;
}
av_freep(&contexts);
av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", &sap->ann[pos]);
pos += strlen(&sap->ann[pos]);
sap->ann_size = pos;
if (sap->ann_size > sap->ann_fd->max_packet_size) {
av_log(s, AV_LOG_ERROR, "Announcement too large to send in one "
"packet\n");
goto fail;
}
return 0;
fail:
av_free(contexts);
sap_write_close(s);
return ret;
}
static int rtp_write_header(AVFormatContext *s1)
{
RTPMuxContext *s = s1->priv_data;
int max_packet_size, n;
AVStream *st;
if (s1->nb_streams != 1)
return -1;
st = s1->streams[0];
if (!is_supported(st->codec->codec_id)) {
av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
return -1;
}
if (s->payload_type < 0)
s->payload_type = ff_rtp_get_payload_type(s1, st->codec);
s->base_timestamp = av_get_random_seed();
s->timestamp = s->base_timestamp;
s->cur_timestamp = 0;
s->ssrc = av_get_random_seed();
s->first_packet = 1;
s->first_rtcp_ntp_time = ff_ntp_time();
if (s1->start_time_realtime)
/* Round the NTP time to whole milliseconds. */
s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
NTP_OFFSET_US;
max_packet_size = s1->pb->max_packet_size;
if (max_packet_size <= 12)
return AVERROR(EIO);
s->buf = av_malloc(max_packet_size);
if (s->buf == NULL) {
return AVERROR(ENOMEM);
}
s->max_payload_size = max_packet_size - 12;
s->max_frames_per_packet = 0;
if (s1->max_delay) {
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
if (st->codec->frame_size == 0) {
av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
} else {
s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN);
}
}
if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
/* FIXME: We should round down here... */
s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
}
}
avpriv_set_pts_info(st, 32, 1, 90000);
switch(st->codec->codec_id) {
case CODEC_ID_MP2:
case CODEC_ID_MP3:
s->buf_ptr = s->buf + 4;
break;
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO:
break;
case CODEC_ID_MPEG2TS:
n = s->max_payload_size / TS_PACKET_SIZE;
if (n < 1)
n = 1;
s->max_payload_size = n * TS_PACKET_SIZE;
s->buf_ptr = s->buf;
break;
case CODEC_ID_H264:
/* check for H.264 MP4 syntax */
if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
}
break;
case CODEC_ID_VORBIS:
case CODEC_ID_THEORA:
if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
s->num_frames = 0;
goto defaultcase;
case CODEC_ID_VP8:
av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
"incompatible with the latest spec drafts.\n");
break;
case CODEC_ID_ADPCM_G722:
/* Due to a historical error, the clock rate for G722 in RTP is
* 8000, even if the sample rate is 16000. See RFC 3551. */
avpriv_set_pts_info(st, 32, 1, 8000);
break;
case CODEC_ID_AMR_NB:
case CODEC_ID_AMR_WB:
if (!s->max_frames_per_packet)
s->max_frames_per_packet = 12;
if (st->codec->codec_id == CODEC_ID_AMR_NB)
n = 31;
else
n = 61;
/* max_header_toc_size + the largest AMR payload must fit */
if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
return -1;
}
if (st->codec->channels != 1) {
av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
return -1;
}
case CODEC_ID_AAC:
s->num_frames = 0;
default:
defaultcase:
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
}
s->buf_ptr = s->buf;
break;
}
return 0;
}
$NetBSD$
--- ffserver.c.orig 2013-01-23 16:52:39.000000000 +0000
+++ ffserver.c
@@ -4700,7 +4700,10 @@ int main(int argc, char **argv)
av_lfg_init(&random_state, av_get_random_seed());
sigact.sa_handler = handle_child_exit;
- sigact.sa_flags = SA_NOCLDSTOP | SA_RESTART;
+ sigact.sa_flags = SA_NOCLDSTOP;
+#ifdef SA_RESTART
+ sigact.sa_flags |= SA_RESTART;
+#endif
sigaction(SIGCHLD, &sigact, 0);
if (parse_ffconfig(config_filename) < 0) {
static int rtp_write_header(AVFormatContext *s1)
{
RTPMuxContext *s = s1->priv_data;
int n;
AVStream *st;
if (s1->nb_streams != 1) {
av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
return AVERROR(EINVAL);
}
st = s1->streams[0];
if (!is_supported(st->codec->codec_id)) {
av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));
return -1;
}
if (s->payload_type < 0) {
/* Re-validate non-dynamic payload types */
if (st->id < RTP_PT_PRIVATE)
st->id = ff_rtp_get_payload_type(s1, st->codec, -1);
s->payload_type = st->id;
} else {
/* private option takes priority */
st->id = s->payload_type;
}
s->base_timestamp = av_get_random_seed();
s->timestamp = s->base_timestamp;
s->cur_timestamp = 0;
if (!s->ssrc)
s->ssrc = av_get_random_seed();
s->first_packet = 1;
s->first_rtcp_ntp_time = ff_ntp_time();
if (s1->start_time_realtime)
/* Round the NTP time to whole milliseconds. */
s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
NTP_OFFSET_US;
// Pick a random sequence start number, but in the lower end of the
// available range, so that any wraparound doesn't happen immediately.
// (Immediate wraparound would be an issue for SRTP.)
if (s->seq < 0) {
if (st->codec->flags & CODEC_FLAG_BITEXACT) {
s->seq = 0;
} else
s->seq = av_get_random_seed() & 0x0fff;
} else
s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
if (s1->packet_size) {
if (s1->pb->max_packet_size)
s1->packet_size = FFMIN(s1->packet_size,
s1->pb->max_packet_size);
} else
s1->packet_size = s1->pb->max_packet_size;
if (s1->packet_size <= 12) {
av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
return AVERROR(EIO);
}
s->buf = av_malloc(s1->packet_size);
if (s->buf == NULL) {
return AVERROR(ENOMEM);
}
s->max_payload_size = s1->packet_size - 12;
s->max_frames_per_packet = 0;
if (s1->max_delay > 0) {
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
int frame_size = av_get_audio_frame_duration(st->codec, 0);
if (!frame_size)
frame_size = st->codec->frame_size;
if (frame_size == 0) {
av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
} else {
AVRational avr1 ={ frame_size, st->codec->sample_rate };
s->max_frames_per_packet = av_rescale_q_rnd(s1->max_delay,AV_TIME_BASE_Q,avr1, AV_ROUND_DOWN);
}
}
if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
/* FIXME: We should round down here... */
AVRational avr2 = {1, 1000000};
s->max_frames_per_packet = av_rescale_q(s1->max_delay, avr2, st->codec->time_base);
}
}
avpriv_set_pts_info(st, 32, 1, 90000);
switch(st->codec->codec_id) {
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
s->buf_ptr = s->buf + 4;
break;
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
break;
case AV_CODEC_ID_MPEG2TS:
n = s->max_payload_size / TS_PACKET_SIZE;
if (n < 1)
n = 1;
s->max_payload_size = n * TS_PACKET_SIZE;
s->buf_ptr = s->buf;
break;
case AV_CODEC_ID_H264:
/* check for H.264 MP4 syntax */
if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
}
break;
case AV_CODEC_ID_VORBIS:
case AV_CODEC_ID_THEORA:
if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
s->num_frames = 0;
goto defaultcase;
case AV_CODEC_ID_ADPCM_G722:
/* Due to a historical error, the clock rate for G722 in RTP is
* 8000, even if the sample rate is 16000. See RFC 3551. */
avpriv_set_pts_info(st, 32, 1, 8000);
break;
case AV_CODEC_ID_OPUS:
if (st->codec->channels > 2) {
av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
goto fail;
}
/* The opus RTP RFC says that all opus streams should use 48000 Hz
* as clock rate, since all opus sample rates can be expressed in
* this clock rate, and sample rate changes on the fly are supported. */
avpriv_set_pts_info(st, 32, 1, 48000);
break;
case AV_CODEC_ID_ILBC:
if (st->codec->block_align != 38 && st->codec->block_align != 50) {
av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
goto fail;
}
if (!s->max_frames_per_packet)
s->max_frames_per_packet = 1;
s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
s->max_payload_size / st->codec->block_align);
goto defaultcase;
case AV_CODEC_ID_AMR_NB:
case AV_CODEC_ID_AMR_WB:
if (!s->max_frames_per_packet)
s->max_frames_per_packet = 12;
if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
n = 31;
else
n = 61;
/* max_header_toc_size + the largest AMR payload must fit */
if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
goto fail;
}
if (st->codec->channels != 1) {
av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
goto fail;
}
case AV_CODEC_ID_AAC:
s->num_frames = 0;
default:
defaultcase:
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
}
s->buf_ptr = s->buf;
break;
}
return 0;
fail:
av_freep(&s->buf);
return AVERROR(EINVAL);
}
