static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
RTSPStream *rtsp_st, AVStream *st)
{
- AVCodecContext *codec = st ? st->codec : NULL;
+ AVCodecParameters *par = st ? st->codecpar : NULL;
if (!handler)
return;
- if (codec)
- codec->codec_id = handler->codec_id;
+ if (par)
+ par->codec_id = handler->codec_id;
rtsp_st->dynamic_handler = handler;
if (st)
st->need_parsing = handler->need_parsing;
AVStream *st, RTSPStream *rtsp_st,
int payload_type, const char *p)
{
- AVCodecContext *codec = st->codec;
+ AVCodecParameters *par = st->codecpar;
char buf[256];
int i;
AVCodec *c;
if (payload_type < RTP_PT_PRIVATE) {
/* We are in a standard case
* (from http://www.iana.org/assignments/rtp-parameters). */
- codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
+ par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
}
- if (codec->codec_id == AV_CODEC_ID_NONE) {
+ if (par->codec_id == AV_CODEC_ID_NONE) {
RTPDynamicProtocolHandler *handler =
- ff_rtp_handler_find_by_name(buf, codec->codec_type);
+ ff_rtp_handler_find_by_name(buf, par->codec_type);
init_rtp_handler(handler, rtsp_st, st);
/* If no dynamic handler was found, check with the list of standard
* allocated types, if such a stream for some reason happens to
* use a private payload type. This isn't handled in rtpdec.c, since
* the format name from the rtpmap line never is passed into rtpdec. */
if (!rtsp_st->dynamic_handler)
- codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
+ par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
}
- c = avcodec_find_decoder(codec->codec_id);
+ c = avcodec_find_decoder(par->codec_id);
if (c && c->name)
c_name = c->name;
else
get_word_sep(buf, sizeof(buf), "/", &p);
i = atoi(buf);
- switch (codec->codec_type) {
+ switch (par->codec_type) {
case AVMEDIA_TYPE_AUDIO:
av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
- codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
- codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
+ par->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
+ par->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
if (i > 0) {
- codec->sample_rate = i;
- avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
+ par->sample_rate = i;
+ avpriv_set_pts_info(st, 32, 1, par->sample_rate);
get_word_sep(buf, sizeof(buf), "/", &p);
i = atoi(buf);
if (i > 0)
- codec->channels = i;
+ par->channels = i;
}
av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
- codec->sample_rate);
+ par->sample_rate);
av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
- codec->channels);
+ par->channels);
break;
case AVMEDIA_TYPE_VIDEO:
av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
return;
st->id = rt->nb_rtsp_streams - 1;
rtsp_st->stream_index = st->index;
- st->codec->codec_type = codec_type;
+ st->codecpar->codec_type = codec_type;
if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
RTPDynamicProtocolHandler *handler;
/* if standard payload type, we can find the codec right now */
- ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
- if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
- st->codec->sample_rate > 0)
- avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
+ ff_rtp_get_codec_info(st->codecpar, rtsp_st->sdp_payload_type);
+ if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
+ st->codecpar->sample_rate > 0)
+ avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
/* Even static payload types may need a custom depacketizer */
handler = ff_rtp_handler_find_by_id(
- rtsp_st->sdp_payload_type, st->codec->codec_type);
+ rtsp_st->sdp_payload_type, st->codecpar->codec_type);
init_rtp_handler(handler, rtsp_st, st);
finalize_rtp_handler_init(s, rtsp_st, st);
}
} else if (av_strstart(p, "SampleRate:integer;", &p) &&
s->nb_streams > 0) {
st = s->streams[s->nb_streams - 1];
- st->codec->sample_rate = atoi(p);
+ st->codecpar->sample_rate = atoi(p);
} else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
// RFC 4568
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
/* we will use two ports per rtp stream (rtp and rtcp) */
j += 2;
err = ffurl_open_whitelist(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
- &s->interrupt_callback, &opts, s->protocol_whitelist);
+ &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist);
av_dict_free(&opts);
* will return an error. Therefore, we skip those streams. */
if (rt->server_type == RTSP_SERVER_WMS &&
(rtsp_st->stream_index < 0 ||
- s->streams[rtsp_st->stream_index]->codec->codec_type ==
+ s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
AVMEDIA_TYPE_DATA))
continue;
snprintf(transport, sizeof(transport) - 1,
ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
port, "%s", optbuf);
if (ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
- &s->interrupt_callback, NULL, s->protocol_whitelist) < 0) {
+ &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
host, port,
"?timeout=%d", rt->stimeout);
if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
- &s->interrupt_callback, NULL, s->protocol_whitelist)) < 0) {
+ &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist)) < 0) {
err = ret;
goto fail;
}
append_source_addrs(url, sizeof(url), "block",
rtsp_st->nb_exclude_source_addrs,
rtsp_st->exclude_source_addrs);
- err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
- &s->interrupt_callback, &opts, s->protocol_whitelist);
+ err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
+ &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist);
av_dict_free(&opts);
int ret, port;
URLContext* in = NULL;
int payload_type;
- AVCodecContext codec = { 0 };
+ AVCodecParameters *par = NULL;
struct sockaddr_storage addr;
AVIOContext pb;
socklen_t addrlen = sizeof(addr);
return AVERROR(EIO);
ret = ffurl_open_whitelist(&in, s->filename, AVIO_FLAG_READ,
- &s->interrupt_callback, NULL, s->protocol_whitelist);
+ &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist);
if (ret)
goto fail;
ffurl_close(in);
in = NULL;
- if (ff_rtp_get_codec_info(&codec, payload_type)) {
+ par = avcodec_parameters_alloc();
+ if (!par) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+
+ if (ff_rtp_get_codec_info(par, payload_type)) {
av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
"without an SDP file describing it\n",
payload_type);
goto fail;
}
- if (codec.codec_type != AVMEDIA_TYPE_DATA) {
+ if (par->codec_type != AVMEDIA_TYPE_DATA) {
av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
"properly you need an SDP file "
"describing it\n");
snprintf(sdp, sizeof(sdp),
"v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
addr.ss_family == AF_INET ? 4 : 6, host,
- codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
- codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
+ par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
+ par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
port, payload_type);
av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
+ avcodec_parameters_free(&par);
ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
s->pb = &pb;
return ret;
fail:
+ avcodec_parameters_free(&par);
if (in)
ffurl_close(in);
ff_network_close();
projects / ffmpeg / blobdiff