static int mpegts_handle_packet(AVFormatContext *ctx, PayloadContext *data,
AVStream *st, AVPacket *pkt, uint32_t *timestamp,
const uint8_t *buf, int len, uint16_t seq,
int flags)
{
int ret;
// We don't want to use the RTP timestamps at all. If the mpegts demuxer
// doesn't set any pts/dts, the generic rtpdec code shouldn't try to
// fill it in either, since the mpegts and RTP timestamps are in totally
// different ranges.
*timestamp = RTP_NOTS_VALUE;
if (!data->ts)
return AVERROR(EINVAL);
if (!buf) {
if (data->read_buf_index >= data->read_buf_size)
return AVERROR(EAGAIN);
ret = ff_mpegts_parse_packet(data->ts, pkt, data->buf + data->read_buf_index,
data->read_buf_size - data->read_buf_index);
if (ret < 0)
return AVERROR(EAGAIN);
data->read_buf_index += ret;
if (data->read_buf_index < data->read_buf_size)
return 1;
else
return 0;
}
ret = ff_mpegts_parse_packet(data->ts, pkt, buf, len);
/* The only error that can be returned from ff_mpegts_parse_packet
* is "no more data to return from the provided buffer", so return
* AVERROR(EAGAIN) for all errors */
if (ret < 0)
return AVERROR(EAGAIN);
if (ret < len) {
data->read_buf_size = FFMIN(len - ret, sizeof(data->buf));
memcpy(data->buf, buf + ret, data->read_buf_size);
data->read_buf_index = 0;
return 1;
}
return 0;
}
static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
uint8_t **bufptr, int len)
{
uint8_t* buf = bufptr ? *bufptr : NULL;
int ret, flags = 0;
uint32_t timestamp;
int rv= 0;
if (!buf) {
/* If parsing of the previous packet actually returned 0 or an error,
* there's nothing more to be parsed from that packet, but we may have
* indicated that we can return the next enqueued packet. */
if (s->prev_ret <= 0)
return rtp_parse_queued_packet(s, pkt);
/* return the next packets, if any */
if(s->st && s->parse_packet) {
/* timestamp should be overwritten by parse_packet, if not,
* the packet is left with pts == AV_NOPTS_VALUE */
timestamp = RTP_NOTS_VALUE;
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 {
// TODO: Move to a dynamic packet handler (like above)
if (s->read_buf_index >= s->read_buf_size)
return AVERROR(EAGAIN);
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 AVERROR(EAGAIN);
s->read_buf_index += ret;
if (s->read_buf_index < s->read_buf_size)
return 1;
else
return 0;
}
}
if (len < 12)
return -1;
if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
return -1;
if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
return rtcp_parse_packet(s, buf, len);
}
if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
/* First packet, or no reordering */
return rtp_parse_packet_internal(s, pkt, buf, len);
} else {
uint16_t seq = AV_RB16(buf + 2);
int16_t diff = seq - s->seq;
if (diff < 0) {
/* Packet older than the previously emitted one, drop */
av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
"RTP: dropping old packet received too late\n");
return -1;
} else if (diff <= 1) {
/* Correct packet */
rv = rtp_parse_packet_internal(s, pkt, buf, len);
return rv;
} else {
/* Still missing some packet, enqueue this one. */
enqueue_packet(s, buf, len);
*bufptr = NULL;
/* Return the first enqueued packet if the queue is full,
* even if we're missing something */
if (s->queue_len >= s->queue_size)
return rtp_parse_queued_packet(s, pkt);
return -1;
}
}
}
static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
const uint8_t *buf, int len)
{
unsigned int ssrc, h;
int payload_type, seq, ret, flags = 0;
int ext;
AVStream *st;
uint32_t timestamp;
int rv= 0;
ext = buf[0] & 0x10;
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);
/* store the ssrc in the RTPDemuxContext */
s->ssrc = ssrc;
/* NOTE: we can handle only one payload type */
if (s->payload_type != payload_type)
return -1;
st = s->st;
// only do something with this if all the rtp checks pass...
if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
{
av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
payload_type, seq, ((s->seq + 1) & 0xffff));
return -1;
}
if (buf[0] & 0x20) {
int padding = buf[len - 1];
if (len >= 12 + padding)
len -= padding;
}
s->seq = seq;
len -= 12;
buf += 12;
/* RFC 3550 Section 5.3.1 RTP Header Extension handling */
if (ext) {
if (len < 4)
return -1;
/* calculate the header extension length (stored as number
* of 32-bit words) */
ext = (AV_RB16(buf + 2) + 1) << 2;
if (len < ext)
return -1;
// skip past RTP header extension
len -= ext;
buf += ext;
}
if (!st) {
/* specific MPEG2TS demux support */
ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
/* The only error that can be returned from ff_mpegts_parse_packet
* is "no more data to return from the provided buffer", so return
* AVERROR(EAGAIN) for all errors */
if (ret < 0)
return AVERROR(EAGAIN);
if (ret < len) {
s->read_buf_size = len - ret;
memcpy(s->buf, buf + ret, s->read_buf_size);
s->read_buf_index = 0;
return 1;
}
return 0;
} else if (s->parse_packet) {
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;
h = AV_RB32(buf);
len -= 4;
buf += 4;
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO:
/* better than nothing: skip mpeg video RTP header */
if (len <= 4)
return -1;
h = AV_RB32(buf);
buf += 4;
len -= 4;
if (h & (1 << 26)) {
/* mpeg2 */
if (len <= 4)
return -1;
buf += 4;
len -= 4;
}
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
default:
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
}
pkt->stream_index = st->index;
}
// now perform timestamp things....
finalize_packet(s, pkt, timestamp);
return rv;
}
/**
* Parse an RTP or RTCP packet directly sent as a buffer.
* @param s RTP parse context.
* @param pkt returned packet
* @param buf input buffer or NULL to read the next packets
* @param len buffer len
* @return 0 if a packet is returned, 1 if a packet is returned and more can follow
* (use buf as NULL to read the next). -1 if no packet (error or no more packet).
*/
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
const uint8_t *buf, int len)
{
unsigned int ssrc, h;
int payload_type, seq, ret, flags = 0;
AVStream *st;
uint32_t timestamp;
int rv= 0;
if (!buf) {
/* 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->ic, s->dynamic_protocol_context,
s->st, pkt, ×tamp, NULL, 0, flags);
finalize_packet(s, pkt, timestamp);
return rv;
} else {
// TODO: Move to a dynamic packet handler (like above)
if (s->read_buf_index >= s->read_buf_size)
return -1;
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;
s->read_buf_index += ret;
if (s->read_buf_index < s->read_buf_size)
return 1;
else
return 0;
}
}
if (len < 12)
return -1;
if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
return -1;
if (buf[1] >= 200 && buf[1] <= 204) {
rtcp_parse_packet(s, buf, len);
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);
/* store the ssrc in the RTPDemuxContext */
s->ssrc = ssrc;
/* NOTE: we can handle only one payload type */
if (s->payload_type != payload_type)
return -1;
st = s->st;
// only do something with this if all the rtp checks pass...
if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
{
av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
payload_type, seq, ((s->seq + 1) & 0xffff));
return -1;
}
s->seq = seq;
len -= 12;
buf += 12;
if (!st) {
/* specific MPEG2TS demux support */
ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
if (ret < 0)
return -1;
if (ret < len) {
s->read_buf_size = len - ret;
memcpy(s->buf, buf + ret, s->read_buf_size);
s->read_buf_index = 0;
return 1;
}
return 0;
} else if (s->parse_packet) {
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;
h = AV_RB32(buf);
len -= 4;
buf += 4;
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO:
/* better than nothing: skip mpeg video RTP header */
if (len <= 4)
return -1;
h = AV_RB32(buf);
buf += 4;
len -= 4;
if (h & (1 << 26)) {
/* mpeg2 */
if (len <= 4)
return -1;
buf += 4;
len -= 4;
}
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
// moved from below, verbatim. this is because this section handles packets, and the lower switch handles
// timestamps.
// TODO: Put this into a dynamic packet handler...
case CODEC_ID_AAC:
if (rtp_parse_mp4_au(s, buf))
return -1;
{
RTPPayloadData *infos = s->rtp_payload_data;
if (infos == NULL)
return -1;
buf += infos->au_headers_length_bytes + 2;
len -= infos->au_headers_length_bytes + 2;
/* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
one au_header */
av_new_packet(pkt, infos->au_headers[0].size);
memcpy(pkt->data, buf, infos->au_headers[0].size);
buf += infos->au_headers[0].size;
len -= infos->au_headers[0].size;
}
s->read_buf_size = len;
rv= 0;
break;
default:
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
}
pkt->stream_index = st->index;
}
// now perform timestamp things....
finalize_packet(s, pkt, timestamp);
return rv;
}
