*/
/**
- * @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"
-typedef struct rdt_data {
+struct RDTDemuxContext {
+ 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_timestamp;
+ int prev_set_id, prev_stream_id;
+};
+
+RDTDemuxContext *
+ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx,
+ void *priv_data, RTPDynamicProtocolHandler *handler)
+{
+ RDTDemuxContext *s = av_mallocz(sizeof(RDTDemuxContext));
+ if (!s)
+ return NULL;
+
+ s->ic = ic;
+ 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;
+
+ return s;
+}
+
+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;
-} rdt_data;
+ char buffer[RTP_MAX_PACKET_LENGTH + FF_INPUT_BUFFER_PADDING_SIZE];
+ int audio_pkt_cnt; /**< remaining audio packets in rmdec */
+};
void
ff_rdt_calc_response_and_checksum(char response[41], char chksum[9],
}
static int
-rdt_load_mdpr (rdt_data *rdt, AVStream *st, int rule_nr)
+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);
}
- rdt->rmctx->pb = pb;
- if (ff_rm_read_mdpr_codecdata(rdt->rmctx, 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;
}
+/**
+ * Actual data handling.
+ */
+
+int
+ff_rdt_parse_header(const uint8_t *buf, int len,
+ int *pset_id, int *pseq_no, int *pstream_id,
+ int *pis_keyframe, uint32_t *ptimestamp)
+{
+ 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 */) {
+ int pkt_len;
+
+ if (!(buf[0] & 0x80))
+ return -1; /* not followed by a data packet */
+
+ pkt_len = AV_RB16(buf+3);
+ buf += pkt_len;
+ len -= pkt_len;
+ consumed += pkt_len;
+ }
+ if (len < 16)
+ return -1;
+ /**
+ * Layout of the header (in bits):
+ * 1: len_included
+ * Flag indicating whether this header includes a length field;
+ * this can be used to concatenate multiple RDT packets in a
+ * single UDP/TCP data frame and is used to precede RDT data
+ * by stream status packets
+ * 1: need_reliable
+ * Flag indicating whether this header includes a "reliable
+ * sequence number"; these are apparently sequence numbers of
+ * data packets alone. For data packets, this flag is always
+ * set, according to the Real documentation [1]
+ * 5: set_id
+ * ID of a set of streams of identical content, possibly with
+ * different codecs or bitrates
+ * 1: is_reliable
+ * Flag set for certain streams deemed less tolerable for packet
+ * loss
+ * 16: seq_no
+ * Packet sequence number; if >=0xFF00, this is a non-data packet
+ * containing stream status info, the second byte indicates the
+ * type of status packet (see wireshark docs / source code [2])
+ * if (len_included) {
+ * 16: packet_len
+ * } else {
+ * packet_len = remainder of UDP/TCP frame
+ * }
+ * 1: is_back_to_back
+ * Back-to-Back flag; used for timing, set for one in every 10
+ * packets, according to the Real documentation [1]
+ * 1: is_slow_data
+ * Slow-data flag; currently unused, according to Real docs [1]
+ * 5: stream_id
+ * ID of the stream within this particular set of streams
+ * 1: is_no_keyframe
+ * Non-keyframe flag (unset if packet belongs to a keyframe)
+ * 32: timestamp (PTS)
+ * if (set_id == 0x1F) {
+ * 16: set_id (extended set-of-streams ID; see set_id)
+ * }
+ * if (need_reliable) {
+ * 16: reliable_seq_no
+ * Reliable sequence number (see need_reliable)
+ * }
+ * if (stream_id == 0x3F) {
+ * 16: stream_id (extended stream ID; see stream_id)
+ * }
+ * [1] https://protocol.helixcommunity.org/files/2005/devdocs/RDT_Feature_Level_20.txt
+ * [2] http://www.wireshark.org/docs/dfref/r/rdt.html and
+ * http://anonsvn.wireshark.org/viewvc/trunk/epan/dissectors/packet-rdt.c
+ */
+ 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 (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;
+
+ if (rdt->audio_pkt_cnt == 0) {
+ int pos;
+
+ 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);
+ if (res < 0)
+ return res;
+ 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 {
+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 > 0;
+}
+
+int
+ff_rdt_parse_packet(RDTDemuxContext *s, AVPacket *pkt,
+ const uint8_t *buf, int len)
+{
+ int seq_no, flags = 0, stream_id, set_id, is_keyframe;
+ uint32_t timestamp;
+ int rv= 0;
+
+ if (!s->parse_packet)
+ return -1;
+
+ 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->ic, s->dynamic_protocol_context,
+ s->streams[s->prev_stream_id],
+ pkt, ×tamp, NULL, 0, flags);
+ return rv;
+ }
+
+ if (len < 12)
+ return -1;
+ 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 ||
+ 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;
+
+ 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;
+}
+
+void
+ff_rdt_subscribe_rule (char *cmd, int size,
+ int stream_nr, int rule_nr)
+{
+ av_strlcatf(cmd, size, "stream=%d;rule=%d,stream=%d;rule=%d",
+ stream_nr, rule_nr * 2, stream_nr, rule_nr * 2 + 1);
+}
+
static unsigned char *
rdt_parse_b64buf (unsigned int *target_len, const char *p)
{
}
static int
-rdt_parse_sdp_line (AVStream *stream, void *d, const char *line)
+rdt_parse_sdp_line (AVFormatContext *s, int st_index,
+ PayloadContext *rdt, const char *line)
{
- rdt_data *rdt = d;
+ 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);
- rdt_load_mdpr(rdt, stream, 0);
} 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 *
+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)
{
- rdt_data *rdt = av_mallocz(sizeof(rdt_data));
+ PayloadContext *rdt = av_mallocz(sizeof(PayloadContext));
av_open_input_stream(&rdt->rmctx, NULL, "", &rdt_demuxer, NULL);
}
static void
-rdt_free_extradata (void *d)
+rdt_free_extradata (PayloadContext *rdt)
{
- rdt_data *rdt = d;
+ 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);
CODEC_ID_NONE, \
rdt_parse_sdp_line, \
rdt_new_extradata, \
- rdt_free_extradata \
+ rdt_free_extradata, \
+ rdt_parse_packet \
};
RDT_HANDLER(live_video, "x-pn-multirate-realvideo-live", CODEC_TYPE_VIDEO);
projects / frescor / ffmpeg.git / blobdiff