* 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 rdt.c
+ * @file
* @brief Realmedia RTSP protocol (RDT) support
* @author Ronald S. Bultje <rbultje@ronald.bitfreak.net>
*/
#include "avformat.h"
#include "libavutil/avstring.h"
-#include "rtp_internal.h"
+#include "rtpdec.h"
#include "rdt.h"
#include "libavutil/base64.h"
#include "libavutil/md5.h"
#include "rm.h"
#include "internal.h"
-#include <libavcodec/bitstream.h>
+#include "avio_internal.h"
+#include "libavcodec/get_bits.h"
struct RDTDemuxContext {
AVFormatContext *ic; /**< the containing (RTSP) demux context */
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;
- s->parse_packet = handler->parse_packet;
+ s->parse_packet = handler ? handler->parse_packet : NULL;
s->dynamic_protocol_context = priv_data;
return s;
struct PayloadContext {
AVFormatContext *rmctx;
+ 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];
- int audio_pkt_cnt[MAX_STREAMS]; /**< remaining audio packets in rmdec */
+ int audio_pkt_cnt; /**< remaining audio packets in rmdec */
};
void
buf[8 + i] ^= xor_table[i];
av_md5_sum(zres, buf, 64);
- ff_data_to_hex(response, zres, 16);
- for (i=0;i<32;i++) response[i] = tolower(response[i]);
+ ff_data_to_hex(response, zres, 16, 1);
/* add tail */
strcpy (response + 32, "01d0a8e3");
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, size) < 0)
+ if (ff_rm_read_mdpr_codecdata(rdt->rmctx, &pb, st, rdt->rmst[st->index], size) < 0)
return -1;
return 0;
/**< return 0 on packet, no more left, 1 on packet, 1 on partial packet... */
static int
-rdt_parse_packet (PayloadContext *rdt, AVStream *st,
+rdt_parse_packet (AVFormatContext *ctx, PayloadContext *rdt, AVStream *st,
AVPacket *pkt, uint32_t *timestamp,
const uint8_t *buf, int len, 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);
- flags = (flags & PKT_FLAG_KEY) ? 2 : 0;
- res = ff_rm_parse_packet (rdt->rmctx, &pb, st, len, pkt,
- &seq, &flags, timestamp);
- pos = url_ftell(&pb);
+ 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 = avio_tell(&pb);
if (res < 0)
return res;
- rdt->audio_pkt_cnt[st->id] = res;
- if (rdt->audio_pkt_cnt[st->id] > 0 &&
- st->codec->codec_id == CODEC_ID_AAC) {
- memcpy (rdt->buffer, buf + pos, len - pos);
- rdt->rmctx->pb = av_alloc_put_byte (rdt->buffer, len - pos, 0,
- NULL, NULL, NULL, NULL);
+ if (res > 0) {
+ if (st->codec->codec_id == 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);
+ }
+ goto get_cache;
}
} else {
- ff_rm_retrieve_cache (rdt->rmctx, rdt->rmctx->pb, st, pkt);
- if (rdt->audio_pkt_cnt[st->id] == 0 &&
+get_cache:
+ rdt->audio_pkt_cnt =
+ 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)
av_freep(&rdt->rmctx->pb);
}
pkt->stream_index = st->index;
pkt->pts = *timestamp;
- return rdt->audio_pkt_cnt[st->id] > 0;
+ return rdt->audio_pkt_cnt > 0;
}
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;
if (!buf && s->prev_stream_id != -1) {
/* return the next packets, if any */
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->dynamic_protocol_context,
+ rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
s->streams[s->prev_stream_id],
pkt, ×tamp, NULL, 0, flags);
return rv;
if (is_keyframe &&
(set_id != s->prev_set_id || timestamp != s->prev_timestamp ||
stream_id != s->prev_stream_id)) {
- flags |= PKT_FLAG_KEY;
+ flags |= RTP_FLAG_KEY;
s->prev_set_id = set_id;
s->prev_timestamp = timestamp;
}
return -1;
}
- rv = s->parse_packet(s->dynamic_protocol_context,
+ rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
s->streams[s->prev_stream_id],
pkt, ×tamp, buf, len, flags);
stream_nr, rule_nr * 2, stream_nr, rule_nr * 2 + 1);
}
-void
-ff_rdt_subscribe_rule2 (RDTDemuxContext *s, char *cmd, int size,
- int stream_nr, int rule_nr)
-{
- PayloadContext *rdt = s->dynamic_protocol_context;
-
- rdt_load_mdpr(rdt, s->streams[0], rule_nr * 2);
-}
-
static unsigned char *
rdt_parse_b64buf (unsigned int *target_len, const char *p)
{
rdt->mlti_data = rdt_parse_b64buf(&rdt->mlti_data_size, p);
} else if (av_strstart(p, "StartTime:integer;", &p))
stream->first_dts = atoi(p);
+ else if (av_strstart(p, "ASMRuleBook:string;", &p)) {
+ int n, first = -1;
+
+ for (n = 0; n < s->nb_streams; n++)
+ 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);
+ }
+ }
return 0;
}
+static void
+real_parse_asm_rule(AVStream *st, const char *p, const char *end)
+{
+ do {
+ /* can be either averagebandwidth= or AverageBandwidth= */
+ if (sscanf(p, " %*1[Aa]verage%*1[Bb]andwidth=%d", &st->codec->bit_rate) == 1)
+ break;
+ if (!(p = strchr(p, ',')) || p > end)
+ p = end;
+ p++;
+ } while (p < end);
+}
+
+static AVStream *
+add_dstream(AVFormatContext *s, AVStream *orig_st)
+{
+ AVStream *st;
+
+ 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;
+
+ return st;
+}
+
+static void
+real_parse_asm_rulebook(AVFormatContext *s, AVStream *orig_st,
+ const char *p)
+{
+ const char *end;
+ int n_rules = 0, odd = 0;
+ AVStream *st;
+
+ /**
+ * The ASMRuleBook contains a list of comma-separated strings per rule,
+ * and each rule is separated by a ;. The last one also has a ; at the
+ * end so we can use it as delimiter.
+ * Every rule occurs twice, once for when the RTSP packet header marker
+ * 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
+ * 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++;
+ while (1) {
+ if (!(end = strchr(p, ';')))
+ break;
+ if (!odd && end != p) {
+ if (n_rules > 0)
+ st = add_dstream(s, orig_st);
+ else
+ st = orig_st;
+ if (!st)
+ break;
+ real_parse_asm_rule(st, p, end);
+ n_rules++;
+ }
+ p = end + 1;
+ odd ^= 1;
+ }
+}
+
+void
+ff_real_parse_sdp_a_line (AVFormatContext *s, int stream_index,
+ const char *line)
+{
+ const char *p = line;
+
+ if (av_strstart(p, "ASMRuleBook:string;", &p))
+ real_parse_asm_rulebook(s, s->streams[stream_index], p);
+}
+
static PayloadContext *
-rdt_new_extradata (void)
+rdt_new_context (void)
{
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;
}
static void
-rdt_free_extradata (PayloadContext *rdt)
+rdt_free_context (PayloadContext *rdt)
{
+ int 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);
}
#define RDT_HANDLER(n, s, t) \
static RTPDynamicProtocolHandler ff_rdt_ ## n ## _handler = { \
- s, \
- t, \
- CODEC_ID_NONE, \
- rdt_parse_sdp_line, \
- rdt_new_extradata, \
- rdt_free_extradata, \
- rdt_parse_packet \
-};
+ .enc_name = s, \
+ .codec_type = t, \
+ .codec_id = CODEC_ID_NONE, \
+ .parse_sdp_a_line = rdt_parse_sdp_line, \
+ .alloc = rdt_new_context, \
+ .free = rdt_free_context, \
+ .parse_packet = rdt_parse_packet \
+}
-RDT_HANDLER(live_video, "x-pn-multirate-realvideo-live", CODEC_TYPE_VIDEO);
-RDT_HANDLER(live_audio, "x-pn-multirate-realaudio-live", CODEC_TYPE_AUDIO);
-RDT_HANDLER(video, "x-pn-realvideo", CODEC_TYPE_VIDEO);
-RDT_HANDLER(audio, "x-pn-realaudio", CODEC_TYPE_AUDIO);
+RDT_HANDLER(live_video, "x-pn-multirate-realvideo-live", AVMEDIA_TYPE_VIDEO);
+RDT_HANDLER(live_audio, "x-pn-multirate-realaudio-live", AVMEDIA_TYPE_AUDIO);
+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)
{
projects / ffmpeg / blobdiff