* 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
*/
* in the AVFormatContext, and this variable points to the offset in
* that array such that the first is the first stream of this set. */
AVStream **streams;
- int n_streams; /**< streams with identifical content in this set */
+ int n_streams; /**< streams with identical content in this set */
void *dynamic_protocol_context;
DynamicPayloadPacketHandlerProc parse_packet;
uint32_t prev_timestamp;
RMStream **rmst;
uint8_t *mlti_data;
unsigned int mlti_data_size;
- char buffer[RTP_MAX_PACKET_LENGTH + FF_INPUT_BUFFER_PADDING_SIZE];
+ char buffer[RTP_MAX_PACKET_LENGTH + AV_INPUT_BUFFER_PADDING_SIZE];
int audio_pkt_cnt; /**< remaining audio packets in rmdec */
};
unsigned char zres[16],
buf[64] = { 0xa1, 0xe9, 0x14, 0x9d, 0x0e, 0x6b, 0x3b, 0x59 };
#define XOR_TABLE_SIZE 37
- const unsigned char xor_table[XOR_TABLE_SIZE] = {
+ static const unsigned char xor_table[XOR_TABLE_SIZE] = {
0x05, 0x18, 0x74, 0xd0, 0x0d, 0x09, 0x02, 0x53,
0xc0, 0x01, 0x05, 0x05, 0x67, 0x03, 0x19, 0x70,
0x08, 0x27, 0x66, 0x10, 0x10, 0x72, 0x08, 0x09,
rdt_load_mdpr (PayloadContext *rdt, AVStream *st, int rule_nr)
{
AVIOContext pb;
- int size;
+ unsigned int size;
uint32_t tag;
/**
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;
if (rdt->audio_pkt_cnt == 0) {
- int pos;
+ int pos, rmflags;
ffio_init_context(&pb, buf, len, 0, NULL, NULL, NULL, NULL);
- flags = (flags & RTP_FLAG_KEY) ? 2 : 0;
+ rmflags = (flags & RTP_FLAG_KEY) ? 2 : 0;
res = ff_rm_parse_packet (rdt->rmctx, &pb, st, rdt->rmst[st->index], len, pkt,
- &seq, flags, *timestamp);
+ &seq, rmflags, *timestamp);
pos = avio_tell(&pb);
if (res < 0)
return res;
if (res > 0) {
- if (st->codec->codec_id == CODEC_ID_AAC) {
+ if (st->codecpar->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->codecpar->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;
}
len -= 2; /* skip embracing " at start/end */
}
*target_len = len * 3 / 4;
- target = av_mallocz(*target_len + FF_INPUT_BUFFER_PADDING_SIZE);
+ target = av_mallocz(*target_len + AV_INPUT_BUFFER_PADDING_SIZE);
+ if (!target)
+ return NULL;
av_base64_decode(target, p, *target_len);
return target;
}
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();
+ if (!rdt->rmst[s->streams[n]->index])
+ return AVERROR(ENOMEM);
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
}
}
{
do {
/* can be either averagebandwidth= or AverageBandwidth= */
- if (sscanf(p, " %*1[Aa]verage%*1[Bb]andwidth=%d", &st->codec->bit_rate) == 1)
+ if (sscanf(p, " %*1[Aa]verage%*1[Bb]andwidth=%d", &st->codecpar->bit_rate) == 1)
break;
if (!(p = strchr(p, ',')) || p > end)
p = end;
{
AVStream *st;
- if (!(st = av_new_stream(s, orig_st->id)))
+ if (!(st = avformat_new_stream(s, NULL)))
return NULL;
- st->codec->codec_type = orig_st->codec->codec_type;
+ st->id = orig_st->id;
+ st->codecpar->codec_type = orig_st->codecpar->codec_type;
st->first_dts = orig_st->first_dts;
return st;
* 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)
real_parse_asm_rulebook(s, s->streams[stream_index], p);
}
-static PayloadContext *
-rdt_new_context (void)
+static av_cold int rdt_init(AVFormatContext *s, int st_index, PayloadContext *rdt)
{
- PayloadContext *rdt = av_mallocz(sizeof(PayloadContext));
-
- av_open_input_stream(&rdt->rmctx, NULL, "", &ff_rdt_demuxer, NULL);
-
- return rdt;
+ return avformat_open_input(&rdt->rmctx, "", &ff_rdt_demuxer, NULL);
}
static void
-rdt_free_context (PayloadContext *rdt)
+rdt_close_context (PayloadContext *rdt)
{
int i;
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, \
+ .priv_data_size = sizeof(PayloadContext), \
+ .init = rdt_init, \
.parse_sdp_a_line = rdt_parse_sdp_line, \
- .open = rdt_new_context, \
- .close = rdt_free_context, \
+ .close = rdt_close_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);
}
projects / ffmpeg / blobdiff