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; }