/*
* RTP input format
- * Copyright (c) 2002 Fabrice Bellard.
+ * Copyright (c) 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
/* needed for gethostname() */
#define _XOPEN_SOURCE 600
-#include "libavcodec/bitstream.h"
+#include "libavcodec/get_bits.h"
#include "avformat.h"
#include "mpegts.h"
#include <unistd.h>
#include "network.h"
-#include "rtp_internal.h"
-#include "rtp_h264.h"
+#include "rtpdec.h"
+#include "rtpdec_amr.h"
+#include "rtpdec_asf.h"
+#include "rtpdec_h263.h"
+#include "rtpdec_h264.h"
+#include "rtpdec_vorbis.h"
//#define DEBUG
{
ff_register_dynamic_payload_handler(&mp4v_es_handler);
ff_register_dynamic_payload_handler(&mpeg4_generic_handler);
+ ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
+
+ ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
+ ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
}
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
if (buf[1] != 200)
return -1;
s->last_rtcp_ntp_time = AV_RB64(buf + 8);
- if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
- s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
s->last_rtcp_timestamp = AV_RB32(buf + 16);
return 0;
}
return 0;
}
+void rtp_send_punch_packets(URLContext* rtp_handle)
+{
+ ByteIOContext *pb;
+ uint8_t *buf;
+ int len;
+
+ /* Send a small RTP packet */
+ if (url_open_dyn_buf(&pb) < 0)
+ return;
+
+ put_byte(pb, (RTP_VERSION << 6));
+ put_byte(pb, 0); /* Payload type */
+ put_be16(pb, 0); /* Seq */
+ put_be32(pb, 0); /* Timestamp */
+ put_be32(pb, 0); /* SSRC */
+
+ put_flush_packet(pb);
+ len = url_close_dyn_buf(pb, &buf);
+ if ((len > 0) && buf)
+ url_write(rtp_handle, buf, len);
+ av_free(buf);
+
+ /* Send a minimal RTCP RR */
+ if (url_open_dyn_buf(&pb) < 0)
+ return;
+
+ put_byte(pb, (RTP_VERSION << 6));
+ put_byte(pb, 201); /* receiver report */
+ put_be16(pb, 1); /* length in words - 1 */
+ put_be32(pb, 0); /* our own SSRC */
+
+ put_flush_packet(pb);
+ len = url_close_dyn_buf(pb, &buf);
+ if ((len > 0) && buf)
+ url_write(rtp_handle, buf, len);
+ av_free(buf);
+}
+
+
/**
* open a new RTP parse context for stream 'st'. 'st' can be NULL for
* MPEG2TS streams to indicate that they should be demuxed inside the
return NULL;
s->payload_type = payload_type;
s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
- s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
s->ic = s1;
s->st = st;
s->rtp_payload_data = rtp_payload_data;
rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
- s->ts = mpegts_parse_open(s->ic);
+ s->ts = ff_mpegts_parse_open(s->ic);
if (s->ts == NULL) {
av_free(s);
return NULL;
case CODEC_ID_MP2:
case CODEC_ID_MP3:
case CODEC_ID_MPEG4:
+ case CODEC_ID_H263:
case CODEC_ID_H264:
st->need_parsing = AVSTREAM_PARSE_FULL;
break;
/* 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);
+ addend = av_rescale(s->last_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->dynamic_protocol_context,
+ rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
s->st, pkt, ×tamp, NULL, 0, flags);
finalize_packet(s, pkt, timestamp);
return rv;
// TODO: Move to a dynamic packet handler (like above)
if (s->read_buf_index >= s->read_buf_size)
return -1;
- ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
+ ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
s->read_buf_size - s->read_buf_index);
if (ret < 0)
return -1;
return -1;
}
payload_type = buf[1] & 0x7f;
+ if (buf[1] & 0x80)
+ flags |= RTP_FLAG_MARKER;
seq = AV_RB16(buf + 2);
timestamp = AV_RB32(buf + 4);
ssrc = AV_RB32(buf + 8);
if (!st) {
/* specific MPEG2TS demux support */
- ret = mpegts_parse_packet(s->ts, pkt, buf, len);
+ ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
if (ret < 0)
return -1;
if (ret < len) {
s->read_buf_index = 0;
return 1;
}
+ return 0;
} else if (s->parse_packet) {
- rv = s->parse_packet(s->dynamic_protocol_context,
+ 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) {
case CODEC_ID_MP2:
+ case CODEC_ID_MP3:
/* better than nothing: skip mpeg audio RTP header */
if (len <= 4)
return -1;
break;
}
- // now perform timestamp things....
- finalize_packet(s, pkt, timestamp);
+ pkt->stream_index = st->index;
}
+
+ // now perform timestamp things....
+ finalize_packet(s, pkt, timestamp);
+
return rv;
}
{
// TODO: fold this into the protocol specific data fields.
if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
- mpegts_parse_close(s->ts);
+ ff_mpegts_parse_close(s->ts);
}
av_free(s);
}