/*
* RTP input format
- * Copyright (c) 2002 Fabrice Bellard.
+ * Copyright (c) 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+
+/* needed for gethostname() */
+#define _XOPEN_SOURCE 600
+
+#include "libavcodec/bitstream.h"
#include "avformat.h"
#include "mpegts.h"
-#include "bitstream.h"
#include <unistd.h>
#include "network.h"
-#include "rtp_internal.h"
+#include "rtpdec.h"
#include "rtp_h264.h"
//#define DEBUG
static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
-static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
+void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
{
handler->next= RTPFirstDynamicPayloadHandler;
RTPFirstDynamicPayloadHandler= handler;
void av_register_rtp_dynamic_payload_handlers(void)
{
- register_dynamic_payload_handler(&mp4v_es_handler);
- register_dynamic_payload_handler(&mpeg4_generic_handler);
- register_dynamic_payload_handler(&ff_h264_dynamic_handler);
+ ff_register_dynamic_payload_handler(&mp4v_es_handler);
+ ff_register_dynamic_payload_handler(&mpeg4_generic_handler);
+ ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
}
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
len = url_close_dyn_buf(pb, &buf);
if ((len > 0) && buf) {
int result;
-#if defined(DEBUG)
- printf("sending %d bytes of RR\n", len);
-#endif
+ dprintf(s->ic, "sending %d bytes of RR\n", len);
result= url_write(s->rtp_ctx, buf, len);
-#if defined(DEBUG)
- printf("result from url_write: %d\n", result);
-#endif
+ dprintf(s->ic, "result from url_write: %d\n", result);
av_free(buf);
}
return 0;
* rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
* TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
*/
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
+RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data)
{
RTPDemuxContext *s;
return NULL;
}
} else {
+ av_set_pts_info(st, 32, 1, 90000);
switch(st->codec->codec_id) {
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO:
st->need_parsing = AVSTREAM_PARSE_FULL;
break;
default:
+ if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+ av_set_pts_info(st, 32, 1, st->codec->sample_rate);
+ }
break;
}
}
return s;
}
+void
+rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
+ RTPDynamicProtocolHandler *handler)
+{
+ s->dynamic_protocol_context = ctx;
+ s->parse_packet = handler->parse_packet;
+}
+
static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
{
int au_headers_length, au_header_size, i;
GetBitContext getbitcontext;
- rtp_payload_data_t *infos;
+ RTPPayloadData *infos;
infos = s->rtp_payload_data;
if (infos == NULL)
return -1;
- /* decode the first 2 bytes where are stored the AUHeader sections
+ /* decode the first 2 bytes where the AUHeader sections are stored
length in bits */
au_headers_length = AV_RB16(buf);
*/
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
{
- switch(s->st->codec->codec_id) {
- case CODEC_ID_MP2:
- case CODEC_ID_MPEG1VIDEO:
- case CODEC_ID_MPEG2VIDEO:
- if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
- int64_t addend;
-
- int delta_timestamp;
- /* XXX: is it really necessary to unify the timestamp base ? */
- /* compute pts from timestamp with received ntp_time */
- delta_timestamp = timestamp - s->last_rtcp_timestamp;
- /* convert to 90 kHz without overflow */
- addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
- addend = (addend * 5625) >> 14;
- pkt->pts = addend + delta_timestamp;
- }
- break;
- case CODEC_ID_AAC:
- case CODEC_ID_H264:
- case CODEC_ID_MPEG4:
- pkt->pts = timestamp;
- break;
- default:
- /* no timestamp info yet */
- break;
+ if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
+ int64_t addend;
+ int delta_timestamp;
+
+ /* compute pts from timestamp with received ntp_time */
+ delta_timestamp = timestamp - s->last_rtcp_timestamp;
+ /* convert to the PTS timebase */
+ addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);
+ pkt->pts = addend + delta_timestamp;
}
pkt->stream_index = s->st->index;
}
/* return the next packets, if any */
if(s->st && s->parse_packet) {
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, pkt, ×tamp, NULL, 0, flags);
+ rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
+ s->st, pkt, ×tamp, NULL, 0, flags);
finalize_packet(s, pkt, timestamp);
return rv;
} else {
return 1;
}
} else if (s->parse_packet) {
- rv = s->parse_packet(s, pkt, ×tamp, buf, len, flags);
+ rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
+ s->st, pkt, ×tamp, buf, len, flags);
} else {
// at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
switch(st->codec->codec_id) {
if (rtp_parse_mp4_au(s, buf))
return -1;
{
- rtp_payload_data_t *infos = s->rtp_payload_data;
+ RTPPayloadData *infos = s->rtp_payload_data;
if (infos == NULL)
return -1;
buf += infos->au_headers_length_bytes + 2;