* 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
*/
/**
- * @file libavformat/rdt.c
+ * @file
* @brief Realmedia RTSP protocol (RDT) support
* @author Ronald S. Bultje <rbultje@ronald.bitfreak.net>
*/
#include "libavutil/md5.h"
#include "rm.h"
#include "internal.h"
+#include "avio_internal.h"
#include "libavcodec/get_bits.h"
struct RDTDemuxContext {
do {
s->n_streams++;
} while (first_stream_of_set_idx + s->n_streams < ic->nb_streams &&
- s->streams[s->n_streams]->priv_data == s->streams[0]->priv_data);
+ s->streams[s->n_streams]->id == s->streams[0]->id);
s->prev_set_id = -1;
s->prev_stream_id = -1;
s->prev_timestamp = -1;
void
ff_rdt_parse_close(RDTDemuxContext *s)
{
- int i;
-
- for (i = 1; i < s->n_streams; i++)
- s->streams[i]->priv_data = NULL;
-
av_free(s);
}
struct PayloadContext {
AVFormatContext *rmctx;
- RMStream *rmst[MAX_STREAMS];
+ int nb_rmst;
+ RMStream **rmst;
uint8_t *mlti_data;
unsigned int mlti_data_size;
char buffer[RTP_MAX_PACKET_LENGTH + FF_INPUT_BUFFER_PADDING_SIZE];
static int
rdt_load_mdpr (PayloadContext *rdt, AVStream *st, int rule_nr)
{
- ByteIOContext pb;
+ AVIOContext pb;
int size;
uint32_t tag;
/**
* Layout of the MLTI chunk:
- * 4:MLTI
- * 2:<number of streams>
+ * 4: MLTI
+ * 2: number of streams
* Then for each stream ([number_of_streams] times):
- * 2:<mdpr index>
- * 2:<number of mdpr chunks>
+ * 2: mdpr index
+ * 2: number of mdpr chunks
* Then for each mdpr chunk ([number_of_mdpr_chunks] times):
- * 4:<size>
- * [size]:<data>
+ * 4: size
+ * [size]: data
* we skip MDPR chunks until we reach the one of the stream
* we're interested in, and forward that ([size]+[data]) to
* the RM demuxer to parse the stream-specific header data.
*/
if (!rdt->mlti_data)
return -1;
- init_put_byte(&pb, rdt->mlti_data, rdt->mlti_data_size, 0,
+ ffio_init_context(&pb, rdt->mlti_data, rdt->mlti_data_size, 0,
NULL, NULL, NULL, NULL);
- tag = get_le32(&pb);
+ tag = avio_rl32(&pb);
if (tag == MKTAG('M', 'L', 'T', 'I')) {
int num, chunk_nr;
/* read index of MDPR chunk numbers */
- num = get_be16(&pb);
+ num = avio_rb16(&pb);
if (rule_nr < 0 || rule_nr >= num)
return -1;
- url_fskip(&pb, rule_nr * 2);
- chunk_nr = get_be16(&pb);
- url_fskip(&pb, (num - 1 - rule_nr) * 2);
+ avio_skip(&pb, rule_nr * 2);
+ chunk_nr = avio_rb16(&pb);
+ avio_skip(&pb, (num - 1 - rule_nr) * 2);
/* read MDPR chunks */
- num = get_be16(&pb);
+ num = avio_rb16(&pb);
if (chunk_nr >= num)
return -1;
while (chunk_nr--)
- url_fskip(&pb, get_be32(&pb));
- size = get_be32(&pb);
+ avio_skip(&pb, avio_rb32(&pb));
+ size = avio_rb32(&pb);
} else {
size = rdt->mlti_data_size;
- url_fseek(&pb, 0, SEEK_SET);
+ avio_seek(&pb, 0, SEEK_SET);
}
if (ff_rm_read_mdpr_codecdata(rdt->rmctx, &pb, st, rdt->rmst[st->index], size) < 0)
return -1;
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;
- ByteIOContext pb;
+ AVIOContext pb;
if (rdt->audio_pkt_cnt == 0) {
int pos;
- init_put_byte(&pb, buf, len, 0, NULL, NULL, NULL, NULL);
+ ffio_init_context(&pb, buf, len, 0, NULL, NULL, NULL, NULL);
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 = av_alloc_put_byte (rdt->buffer, len - pos, 0,
+ rdt->rmctx->pb = avio_alloc_context (rdt->buffer, len - pos, 0,
NULL, NULL, NULL, NULL);
}
goto get_cache;
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;
int
ff_rdt_parse_packet(RDTDemuxContext *s, AVPacket *pkt,
- const uint8_t *buf, int len)
+ uint8_t **bufptr, int len)
{
+ uint8_t *buf = bufptr ? *bufptr : NULL;
int seq_no, flags = 0, stream_id, set_id, is_keyframe;
uint32_t timestamp;
int rv= 0;
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;
}
int n, first = -1;
for (n = 0; n < s->nb_streams; n++)
- if (s->streams[n]->priv_data == stream->priv_data) {
+ if (s->streams[n]->id == stream->id) {
+ int count = s->streams[n]->index + 1;
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;
+ 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, 0)))
+ if (!(st = avformat_new_stream(s, NULL)))
return NULL;
+ st->id = orig_st->id;
st->codec->codec_type = orig_st->codec->codec_type;
- st->priv_data = orig_st->priv_data;
st->first_dts = orig_st->first_dts;
return st;
const char *p)
{
const char *end;
- int n_rules, odd = 0;
+ int n_rules = 0, odd = 0;
AVStream *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)
* for multi-bitrate streams.
*/
if (*p == '\"') p++;
- for (n_rules = 0; s->nb_streams < MAX_STREAMS;) {
+ while (1) {
if (!(end = strchr(p, ';')))
break;
if (!odd && end != p) {
st = add_dstream(s, orig_st);
else
st = orig_st;
+ if (!st)
+ break;
real_parse_asm_rule(st, p, end);
n_rules++;
}
{
PayloadContext *rdt = av_mallocz(sizeof(PayloadContext));
- av_open_input_stream(&rdt->rmctx, NULL, "", &rdt_demuxer, NULL);
+ int ret = avformat_open_input(&rdt->rmctx, "", &ff_rdt_demuxer, NULL);
+ if (ret < 0) {
+ av_free(rdt);
+ return NULL;
+ }
return rdt;
}
{
int i;
- for (i = 0; i < MAX_STREAMS; i++)
+ for (i = 0; i < rdt->nb_rmst; i++)
if (rdt->rmst[i]) {
ff_rm_free_rmstream(rdt->rmst[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);
}
static RTPDynamicProtocolHandler ff_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(live_video, "x-pn-multirate-realvideo-live", AVMEDIA_TYPE_VIDEO);
RDT_HANDLER(live_audio, "x-pn-multirate-realaudio-live", AVMEDIA_TYPE_AUDIO);
projects / ffmpeg / blobdiff