提交 4fce284c 编写于 作者: R Ronald S. Bultje

Implement RDT-specific data parsing routines. After these changes, simple

playback of RTSP/RDT streams should work. See discussion in "Realmedia patch"
thread on ML.

Originally committed as revision 15237 to svn://svn.ffmpeg.org/ffmpeg/trunk
上级 99b2ac07
......@@ -38,6 +38,8 @@ typedef struct rdt_data {
AVFormatContext *rmctx;
uint8_t *mlti_data;
unsigned int mlti_data_size;
uint32_t prev_sn, prev_ts;
char buffer[RTP_MAX_PACKET_LENGTH + FF_INPUT_BUFFER_PADDING_SIZE];
} rdt_data;
void
......@@ -134,6 +136,103 @@ rdt_load_mdpr (rdt_data *rdt, AVStream *st, int rule_nr)
return 0;
}
/**
* Actual data handling.
*/
static int rdt_parse_header(struct RTPDemuxContext *s, const uint8_t *buf,
int len, int *seq, uint32_t *timestamp, int *flags)
{
rdt_data *rdt = s->dynamic_protocol_context;
int consumed = 0, sn;
if (buf[0] < 0x40 || buf[0] > 0x42) {
buf += 9;
len -= 9;
consumed += 9;
}
sn = (buf[0]>>1) & 0x1f;
*seq = AV_RB16(buf+1);
*timestamp = AV_RB32(buf+4);
if (!(buf[3] & 1) && (sn != rdt->prev_sn || *timestamp != rdt->prev_ts)) {
*flags |= PKT_FLAG_KEY;
rdt->prev_sn = sn;
rdt->prev_ts = *timestamp;
}
return consumed + 10;
}
/**< return 0 on packet, no more left, 1 on packet, 1 on partial packet... */
static int
rdt_parse_packet (RTPDemuxContext *s, AVPacket *pkt, uint32_t *timestamp,
const uint8_t *buf, int len, int flags)
{
rdt_data *rdt = s->dynamic_protocol_context;
int seq = 1, res;
ByteIOContext *pb = rdt->rmctx->pb;
RMContext *rm = rdt->rmctx->priv_data;
AVStream *st = s->st;
if (rm->audio_pkt_cnt == 0) {
int pos;
url_open_buf (&pb, buf, len, URL_RDONLY);
flags = (flags & PKT_FLAG_KEY) ? 2 : 0;
rdt->rmctx->pb = pb;
res = ff_rm_parse_packet (rdt->rmctx, st, len, pkt,
&seq, &flags, timestamp);
pos = url_ftell(pb);
url_close_buf (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 (&pb, rdt->buffer, len - pos, URL_RDONLY);
rdt->rmctx->pb = pb;
}
} else {
ff_rm_retrieve_cache (rdt->rmctx, st, pkt);
if (rm->audio_pkt_cnt == 0 &&
st->codec->codec_id == CODEC_ID_AAC)
url_close_buf (pb);
}
pkt->stream_index = st->index;
pkt->pts = *timestamp;
return rm->audio_pkt_cnt > 0;
}
int
ff_rdt_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
const uint8_t *buf, int len)
{
int seq, flags = 0;
uint32_t timestamp;
int rv= 0;
if (!buf) {
/* 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= rdt_parse_packet(s, pkt, &timestamp, NULL, 0, flags);
return rv;
}
if (len < 12)
return -1;
rv = rdt_parse_header(s, buf, len, &seq, &timestamp, &flags);
if (rv < 0)
return rv;
buf += rv;
len -= rv;
s->seq = seq;
rv = rdt_parse_packet(s, pkt, &timestamp, buf, len, flags);
return rv;
}
void
ff_rdt_subscribe_rule (RTPDemuxContext *s, char *cmd, int size,
int stream_nr, int rule_nr)
......@@ -181,6 +280,8 @@ rdt_new_extradata (void)
rdt_data *rdt = av_mallocz(sizeof(rdt_data));
av_open_input_stream(&rdt->rmctx, NULL, "", &rdt_demuxer, NULL);
rdt->prev_ts = -1;
rdt->prev_sn = -1;
return rdt;
}
......
......@@ -54,4 +54,11 @@ void av_register_rdt_dynamic_payload_handlers(void);
void ff_rdt_subscribe_rule(RTPDemuxContext *s, char *cmd, int size,
int stream_nr, int rule_nr);
/**
* Parse RDT-style packet data (header + media data).
* Usage similar to rtp_parse_packet().
*/
int ff_rdt_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
const uint8_t *buf, int len);
#endif /* AVFORMAT_RDT_H */
......@@ -1326,7 +1326,10 @@ static int rtsp_read_packet(AVFormatContext *s,
/* get next frames from the same RTP packet */
if (rt->cur_rtp) {
ret = rtp_parse_packet(rt->cur_rtp, pkt, NULL, 0);
if (rt->server_type == RTSP_SERVER_RDT)
ret = ff_rdt_parse_packet(rt->cur_rtp, pkt, NULL, 0);
else
ret = rtp_parse_packet(rt->cur_rtp, pkt, NULL, 0);
if (ret == 0) {
rt->cur_rtp = NULL;
return 0;
......@@ -1353,7 +1356,10 @@ static int rtsp_read_packet(AVFormatContext *s,
}
if (len < 0)
return len;
ret = rtp_parse_packet(rtsp_st->rtp_ctx, pkt, buf, len);
if (rt->server_type == RTSP_SERVER_RDT)
ret = ff_rdt_parse_packet(rtsp_st->rtp_ctx, pkt, buf, len);
else
ret = rtp_parse_packet(rtsp_st->rtp_ctx, pkt, buf, len);
if (ret < 0)
goto redo;
if (ret == 1) {
......
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