*/
/**
- * @file rdt.c
+ * @file libavformat/rdt.c
* @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/get_bits.h"
struct RDTDemuxContext {
- AVFormatContext *ic;
- AVStream *st;
+ AVFormatContext *ic; /**< the containing (RTSP) demux context */
+ /** Each RDT stream-set (represented by one RTSPStream) can contain
+ * multiple streams (of the same content, but with possibly different
+ * codecs/bitrates). Each such stream is represented by one AVStream
+ * 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 */
void *dynamic_protocol_context;
DynamicPayloadPacketHandlerProc parse_packet;
- uint32_t prev_set_id, prev_timestamp;
+ uint32_t prev_timestamp;
+ int prev_set_id, prev_stream_id;
};
RDTDemuxContext *
-ff_rdt_parse_open(AVFormatContext *ic, AVStream *st,
+ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx,
void *priv_data, RTPDynamicProtocolHandler *handler)
{
RDTDemuxContext *s = av_mallocz(sizeof(RDTDemuxContext));
return NULL;
s->ic = ic;
- s->st = st;
+ s->streams = &ic->streams[first_stream_of_set_idx];
+ 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->prev_set_id = -1;
+ s->prev_stream_id = -1;
s->prev_timestamp = -1;
s->parse_packet = handler->parse_packet;
s->dynamic_protocol_context = priv_data;
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];
uint8_t *mlti_data;
unsigned int mlti_data_size;
char buffer[RTP_MAX_PACKET_LENGTH + FF_INPUT_BUFFER_PADDING_SIZE];
+ int audio_pkt_cnt; /**< remaining audio packets in rmdec */
};
void
static int
rdt_load_mdpr (PayloadContext *rdt, AVStream *st, int rule_nr)
{
- ByteIOContext *pb;
+ ByteIOContext pb;
int size;
uint32_t tag;
*/
if (!rdt->mlti_data)
return -1;
- url_open_buf(&pb, rdt->mlti_data, rdt->mlti_data_size, URL_RDONLY);
- tag = get_le32(pb);
+ init_put_byte(&pb, rdt->mlti_data, rdt->mlti_data_size, 0,
+ NULL, NULL, NULL, NULL);
+ tag = get_le32(&pb);
if (tag == MKTAG('M', 'L', 'T', 'I')) {
int num, chunk_nr;
/* read index of MDPR chunk numbers */
- num = get_be16(pb);
+ num = get_be16(&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);
+ url_fskip(&pb, rule_nr * 2);
+ chunk_nr = get_be16(&pb);
+ url_fskip(&pb, (num - 1 - rule_nr) * 2);
/* read MDPR chunks */
- num = get_be16(pb);
+ num = get_be16(&pb);
if (chunk_nr >= num)
return -1;
while (chunk_nr--)
- url_fskip(pb, get_be32(pb));
- size = get_be32(pb);
+ url_fskip(&pb, get_be32(&pb));
+ size = get_be32(&pb);
} else {
size = rdt->mlti_data_size;
- url_fseek(pb, 0, SEEK_SET);
+ url_fseek(&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;
- url_close_buf(pb);
return 0;
}
int
ff_rdt_parse_header(const uint8_t *buf, int len,
- int *set_id, int *seq_no, int *stream_id,
- int *is_keyframe, uint32_t *timestamp)
+ int *pset_id, int *pseq_no, int *pstream_id,
+ int *pis_keyframe, uint32_t *ptimestamp)
{
- int consumed = 10;
+ GetBitContext gb;
+ int consumed = 0, set_id, seq_no, stream_id, is_keyframe,
+ len_included, need_reliable;
+ uint32_t timestamp;
/* skip status packets */
while (len >= 5 && buf[1] == 0xFF /* status packet */) {
len -= pkt_len;
consumed += pkt_len;
}
- if (len < 10)
+ if (len < 16)
return -1;
/**
* Layout of the header (in bits):
* [2] http://www.wireshark.org/docs/dfref/r/rdt.html and
* http://anonsvn.wireshark.org/viewvc/trunk/epan/dissectors/packet-rdt.c
*/
- if (set_id) *set_id = (buf[0]>>1) & 0x1f;
- if (seq_no) *seq_no = AV_RB16(buf+1);
- if (timestamp) *timestamp = AV_RB32(buf+4);
- if (stream_id) *stream_id = (buf[3]>>1) & 0x1f;
- if (is_keyframe) *is_keyframe = !(buf[3] & 0x1);
-
- return consumed;
+ init_get_bits(&gb, buf, len << 3);
+ len_included = get_bits1(&gb);
+ need_reliable = get_bits1(&gb);
+ set_id = get_bits(&gb, 5);
+ skip_bits(&gb, 1);
+ seq_no = get_bits(&gb, 16);
+ if (len_included)
+ skip_bits(&gb, 16);
+ skip_bits(&gb, 2);
+ stream_id = get_bits(&gb, 5);
+ is_keyframe = !get_bits1(&gb);
+ timestamp = get_bits_long(&gb, 32);
+ if (set_id == 0x1f)
+ set_id = get_bits(&gb, 16);
+ if (need_reliable)
+ skip_bits(&gb, 16);
+ if (stream_id == 0x1f)
+ stream_id = get_bits(&gb, 16);
+
+ if (pset_id) *pset_id = set_id;
+ if (pseq_no) *pseq_no = seq_no;
+ if (pstream_id) *pstream_id = stream_id;
+ if (pis_keyframe) *pis_keyframe = is_keyframe;
+ if (ptimestamp) *ptimestamp = timestamp;
+
+ return consumed + (get_bits_count(&gb) >> 3);
}
/**< 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;
- RMContext *rm = rdt->rmctx->priv_data;
+ ByteIOContext pb;
- if (rm->audio_pkt_cnt == 0) {
+ if (rdt->audio_pkt_cnt == 0) {
int pos;
- url_open_buf (&pb, buf, len, URL_RDONLY);
- flags = (flags & PKT_FLAG_KEY) ? 2 : 0;
- res = ff_rm_parse_packet (rdt->rmctx, pb, st, len, pkt,
+ init_put_byte(&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);
- url_close_buf (pb);
+ pos = url_ftell(&pb);
if (res < 0)
return res;
- if (rm->audio_pkt_cnt > 0 &&
- st->codec->codec_id == CODEC_ID_AAC) {
- memcpy (rdt->buffer, buf + pos, len - pos);
- url_open_buf (&rdt->rmctx->pb, rdt->buffer, len - pos, URL_RDONLY);
+ if (res > 0) {
+ if (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);
+ }
+ goto get_cache;
}
} else {
- ff_rm_retrieve_cache (rdt->rmctx, rdt->rmctx->pb, st, pkt);
- if (rm->audio_pkt_cnt == 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)
- url_close_buf (rdt->rmctx->pb);
+ av_freep(&rdt->rmctx->pb);
}
pkt->stream_index = st->index;
pkt->pts = *timestamp;
- return rm->audio_pkt_cnt > 0;
+ return rdt->audio_pkt_cnt > 0;
}
int
if (!s->parse_packet)
return -1;
- if (!buf) {
+ 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,
- s->st, pkt, ×tamp, NULL, 0, flags);
+ rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
+ s->streams[s->prev_stream_id],
+ pkt, ×tamp, NULL, 0, flags);
return rv;
}
rv = ff_rdt_parse_header(buf, len, &set_id, &seq_no, &stream_id, &is_keyframe, ×tamp);
if (rv < 0)
return rv;
- if (is_keyframe && (set_id != s->prev_set_id || timestamp != s->prev_timestamp)) {
- flags |= PKT_FLAG_KEY;
+ if (is_keyframe &&
+ (set_id != s->prev_set_id || timestamp != s->prev_timestamp ||
+ stream_id != s->prev_stream_id)) {
+ flags |= RTP_FLAG_KEY;
s->prev_set_id = set_id;
s->prev_timestamp = timestamp;
}
+ s->prev_stream_id = stream_id;
buf += rv;
len -= rv;
- rv = s->parse_packet(s->dynamic_protocol_context,
- s->st, pkt, ×tamp, buf, len, flags);
+ if (s->prev_stream_id >= s->n_streams) {
+ s->prev_stream_id = -1;
+ return -1;
+ }
+
+ rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
+ s->streams[s->prev_stream_id],
+ pkt, ×tamp, buf, len, flags);
return rv;
}
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->st, rule_nr * 2);
-}
-
static unsigned char *
rdt_parse_b64buf (unsigned int *target_len, const char *p)
{
}
static int
-rdt_parse_sdp_line (AVStream *stream, PayloadContext *rdt, const char *line)
+rdt_parse_sdp_line (AVFormatContext *s, int st_index,
+ PayloadContext *rdt, const char *line)
{
+ AVStream *stream = s->streams[st_index];
const char *p = line;
if (av_strstart(p, "OpaqueData:buffer;", &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 = st_index, first = -1;
+
+ for (n = 0; n < s->nb_streams; n++)
+ if (s->streams[n]->priv_data == stream->priv_data) {
+ if (first == -1) first = n;
+ 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
+ }
+ }
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 = av_new_stream(s, 0)))
+ return NULL;
+ 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;
+}
+
+static void
+real_parse_asm_rulebook(AVFormatContext *s, AVStream *orig_st,
+ const char *p)
+{
+ const char *end;
+ int n_rules, 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
+ * preceeded 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;) {
+ if (!(end = strchr(p, ';')))
+ break;
+ if (!odd && end != p) {
+ if (n_rules > 0)
+ st = add_dstream(s, orig_st);
+ else
+ st = orig_st;
+ 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)
{
static void
rdt_free_extradata (PayloadContext *rdt)
{
+ int i;
+
+ for (i = 0; i < MAX_STREAMS; 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);
av_freep(&rdt->mlti_data);
projects / frescor / ffmpeg.git / blobdiff