* Realmedia RTSP protocol (RDT) support.
* Copyright (c) 2007 Ronald S. Bultje
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
num = avio_rb16(&pb);
if (rule_nr < 0 || rule_nr >= num)
return -1;
- url_fskip(&pb, rule_nr * 2);
+ avio_skip(&pb, rule_nr * 2);
chunk_nr = avio_rb16(&pb);
- url_fskip(&pb, (num - 1 - rule_nr) * 2);
+ avio_skip(&pb, (num - 1 - rule_nr) * 2);
/* read MDPR chunks */
num = avio_rb16(&pb);
if (chunk_nr >= num)
return -1;
while (chunk_nr--)
- url_fskip(&pb, avio_rb32(&pb));
+ avio_skip(&pb, avio_rb32(&pb));
size = avio_rb32(&pb);
} else {
size = rdt->mlti_data_size;
static int
rdt_parse_packet (AVFormatContext *ctx, PayloadContext *rdt, AVStream *st,
AVPacket *pkt, uint32_t *timestamp,
- const uint8_t *buf, int len, int flags)
+ const uint8_t *buf, int len, uint16_t rtp_seq, int flags)
{
int seq = 1, res;
AVIOContext pb;
flags = (flags & RTP_FLAG_KEY) ? 2 : 0;
res = ff_rm_parse_packet (rdt->rmctx, &pb, st, rdt->rmst[st->index], len, pkt,
&seq, flags, *timestamp);
- pos = url_ftell(&pb);
+ pos = avio_tell(&pb);
if (res < 0)
return res;
if (res > 0) {
- if (st->codec->codec_id == CODEC_ID_AAC) {
+ if (st->codec->codec_id == AV_CODEC_ID_AAC) {
memcpy (rdt->buffer, buf + pos, len - pos);
rdt->rmctx->pb = avio_alloc_context (rdt->buffer, len - pos, 0,
NULL, NULL, NULL, NULL);
ff_rm_retrieve_cache (rdt->rmctx, rdt->rmctx->pb,
st, rdt->rmst[st->index], pkt);
if (rdt->audio_pkt_cnt == 0 &&
- st->codec->codec_id == CODEC_ID_AAC)
+ st->codec->codec_id == AV_CODEC_ID_AAC)
av_freep(&rdt->rmctx->pb);
}
pkt->stream_index = st->index;
timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
s->streams[s->prev_stream_id],
- pkt, ×tamp, NULL, 0, flags);
+ pkt, ×tamp, NULL, 0, 0, flags);
return rv;
}
rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
s->streams[s->prev_stream_id],
- pkt, ×tamp, buf, len, flags);
+ pkt, ×tamp, buf, len, 0, flags);
return rv;
}
for (n = 0; n < s->nb_streams; n++)
if (s->streams[n]->id == stream->id) {
- int count = s->streams[n]->index + 1;
+ int count = s->streams[n]->index + 1, err;
if (first == -1) first = n;
if (rdt->nb_rmst < count) {
- RMStream **rmst= av_realloc(rdt->rmst, count*sizeof(*rmst));
- if (!rmst)
- return AVERROR(ENOMEM);
- memset(rmst + rdt->nb_rmst, 0,
- (count - rdt->nb_rmst) * sizeof(*rmst));
- rdt->rmst = rmst;
+ if ((err = av_reallocp(&rdt->rmst,
+ count * sizeof(*rdt->rmst))) < 0) {
+ rdt->nb_rmst = 0;
+ return err;
+ }
+ memset(rdt->rmst + rdt->nb_rmst, 0,
+ (count - rdt->nb_rmst) * sizeof(*rdt->rmst));
rdt->nb_rmst = count;
}
rdt->rmst[s->streams[n]->index] = ff_rm_alloc_rmstream();
rdt_load_mdpr(rdt, s->streams[n], (n - first) * 2);
-
- if (s->streams[n]->codec->codec_id == CODEC_ID_AAC)
- s->streams[n]->codec->frame_size = 1; // FIXME
}
}
{
AVStream *st;
- if (!(st = av_new_stream(s, orig_st->id)))
+ if (!(st = avformat_new_stream(s, NULL)))
return NULL;
+ st->id = orig_st->id;
st->codec->codec_type = orig_st->codec->codec_type;
st->first_dts = orig_st->first_dts;
* is set and once for if it isn't. We only read the first because we
* don't care much (that's what the "odd" variable is for).
* Each rule contains a set of one or more statements, optionally
- * preceeded by a single condition. If there's a condition, the rule
+ * preceded by a single condition. If there's a condition, the rule
* starts with a '#'. Multiple conditions are merged between brackets,
* so there are never multiple conditions spread out over separate
* statements. Generally, these conditions are bitrate limits (min/max)
{
PayloadContext *rdt = av_mallocz(sizeof(PayloadContext));
- av_open_input_stream(&rdt->rmctx, NULL, "", &ff_rdt_demuxer, NULL);
+ int ret = avformat_open_input(&rdt->rmctx, "", &ff_rdt_demuxer, NULL);
+ if (ret < 0) {
+ av_free(rdt);
+ return NULL;
+ }
return rdt;
}
av_freep(&rdt->rmst[i]);
}
if (rdt->rmctx)
- av_close_input_stream(rdt->rmctx);
+ avformat_close_input(&rdt->rmctx);
av_freep(&rdt->mlti_data);
av_freep(&rdt->rmst);
av_free(rdt);
}
#define RDT_HANDLER(n, s, t) \
-static RTPDynamicProtocolHandler ff_rdt_ ## n ## _handler = { \
+static RTPDynamicProtocolHandler rdt_ ## n ## _handler = { \
.enc_name = s, \
.codec_type = t, \
- .codec_id = CODEC_ID_NONE, \
+ .codec_id = AV_CODEC_ID_NONE, \
.parse_sdp_a_line = rdt_parse_sdp_line, \
- .open = rdt_new_context, \
- .close = rdt_free_context, \
+ .alloc = rdt_new_context, \
+ .free = rdt_free_context, \
.parse_packet = rdt_parse_packet \
}
RDT_HANDLER(video, "x-pn-realvideo", AVMEDIA_TYPE_VIDEO);
RDT_HANDLER(audio, "x-pn-realaudio", AVMEDIA_TYPE_AUDIO);
-void av_register_rdt_dynamic_payload_handlers(void)
+void ff_register_rdt_dynamic_payload_handlers(void)
{
- ff_register_dynamic_payload_handler(&ff_rdt_video_handler);
- ff_register_dynamic_payload_handler(&ff_rdt_audio_handler);
- ff_register_dynamic_payload_handler(&ff_rdt_live_video_handler);
- ff_register_dynamic_payload_handler(&ff_rdt_live_audio_handler);
+ ff_register_dynamic_payload_handler(&rdt_video_handler);
+ ff_register_dynamic_payload_handler(&rdt_audio_handler);
+ ff_register_dynamic_payload_handler(&rdt_live_video_handler);
+ ff_register_dynamic_payload_handler(&rdt_live_audio_handler);
}