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@ -25,22 +25,19 @@ |
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#include "libavcodec/get_bits.h" |
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#include "avformat.h" |
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#include "mpegts.h" |
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#include "url.h" |
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#include "network.h" |
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#include "url.h" |
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#include "rtpdec.h" |
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#include "rtpdec_formats.h" |
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//#define DEBUG
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/* TODO: - add RTCP statistics reporting (should be optional).
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- add support for h263/mpeg4 packetized output : IDEA: send a |
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buffer to 'rtp_write_packet' contains all the packets for ONE |
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frame. Each packet should have a four byte header containing |
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the length in big endian format (same trick as |
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'ffio_open_dyn_packet_buf') |
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/* TODO:
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* - add RTCP statistics reporting (should be optional). |
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* |
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* - add support for H.263/MPEG-4 packetized output: IDEA: send a |
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* buffer to 'rtp_write_packet' contains all the packets for ONE |
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* frame. Each packet should have a four byte header containing |
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* the length in big-endian format (same trick as |
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* 'ffio_open_dyn_packet_buf'). |
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*/ |
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static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = { |
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@ -131,7 +128,8 @@ RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id, |
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return NULL; |
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} |
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static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len) |
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static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, |
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int len) |
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{ |
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int payload_len; |
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while (len >= 4) { |
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@ -140,7 +138,8 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l |
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switch (buf[1]) { |
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case RTCP_SR: |
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if (payload_len < 20) { |
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av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n"); |
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av_log(NULL, AV_LOG_ERROR, |
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"Invalid length for RTCP SR packet\n"); |
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return AVERROR_INVALIDDATA; |
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} |
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@ -174,7 +173,8 @@ static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) |
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} |
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/*
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* called whenever there is a large jump in sequence numbers, or when they get out of probation... |
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* Called whenever there is a large jump in sequence numbers, |
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* or when they get out of probation... |
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*/ |
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static void rtp_init_sequence(RTPStatistics *s, uint16_t seq) |
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{ |
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@ -189,9 +189,7 @@ static void rtp_init_sequence(RTPStatistics *s, uint16_t seq) |
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s->transit = 0; |
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} |
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/*
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* returns 1 if we should handle this packet. |
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*/ |
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/* Returns 1 if we should handle this packet. */ |
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static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) |
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{ |
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uint16_t udelta = seq - s->max_seq; |
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@ -199,7 +197,8 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) |
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const int MAX_MISORDER = 100; |
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const int MIN_SEQUENTIAL = 2; |
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/* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */ |
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/* source not valid until MIN_SEQUENTIAL packets with sequence
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* seq. numbers have been received */ |
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if (s->probation) { |
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if (seq == s->max_seq + 1) { |
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s->probation--; |
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@ -223,7 +222,8 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) |
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} else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) { |
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// sequence made a large jump...
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if (seq == s->bad_seq) { |
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// two sequential packets-- assume that the other side restarted without telling us; just resync.
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/* two sequential packets -- assume that the other side
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* restarted without telling us; just resync. */ |
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rtp_init_sequence(s, seq); |
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} else { |
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s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1); |
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@ -256,7 +256,7 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) |
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return -1; |
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/* TODO: I think this is way too often; RFC 1889 has algorithm for this */ |
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/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ |
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/* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */ |
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s->octet_count += count; |
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rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
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RTCP_TX_RATIO_DEN; |
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@ -318,9 +318,8 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) |
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avio_w8(pb, len); |
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avio_write(pb, s->hostname, len); |
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// padding
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for (len = (6 + len) % 4; len % 4; len++) { |
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for (len = (6 + len) % 4; len % 4; len++) |
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avio_w8(pb, 0); |
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} |
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avio_flush(pb); |
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len = avio_close_dyn_buf(pb, &buf); |
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@ -372,13 +371,14 @@ void ff_rtp_send_punch_packets(URLContext* rtp_handle) |
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av_free(buf); |
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} |
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/**
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* open a new RTP parse context for stream 'st'. 'st' can be NULL for |
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* MPEG2TS streams to indicate that they should be demuxed inside the |
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* MPEG2-TS streams to indicate that they should be demuxed inside the |
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* rtp demux (otherwise AV_CODEC_ID_MPEG2TS packets are returned) |
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*/ |
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RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size) |
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RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, |
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URLContext *rtpc, int payload_type, |
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int queue_size) |
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{ |
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RTPDemuxContext *s; |
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@ -436,7 +436,8 @@ void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, |
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} |
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/**
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* This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc. |
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* This was the second switch in rtp_parse packet. |
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* Normalizes time, if required, sets stream_index, etc. |
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*/ |
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static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) |
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{ |
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@ -452,7 +453,9 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam |
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/* compute pts from timestamp with received ntp_time */ |
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delta_timestamp = timestamp - s->last_rtcp_timestamp; |
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/* convert to the PTS timebase */ |
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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); |
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addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, |
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s->st->time_base.den, |
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(uint64_t) s->st->time_base.num << 32); |
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pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend + |
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delta_timestamp; |
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return; |
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@ -460,13 +463,15 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam |
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if (!s->base_timestamp) |
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s->base_timestamp = timestamp; |
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/* assume that the difference is INT32_MIN < x < INT32_MAX, but allow the first timestamp to exceed INT32_MAX */ |
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/* assume that the difference is INT32_MIN < x < INT32_MAX,
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* but allow the first timestamp to exceed INT32_MAX */ |
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if (!s->timestamp) |
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s->unwrapped_timestamp += timestamp; |
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else |
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s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp); |
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s->timestamp = timestamp; |
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pkt->pts = s->unwrapped_timestamp + s->range_start_offset - s->base_timestamp; |
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pkt->pts = s->unwrapped_timestamp + s->range_start_offset - |
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s->base_timestamp; |
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} |
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static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, |
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@ -495,9 +500,9 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, |
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st = s->st; |
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// only do something with this if all the rtp checks pass...
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if(!rtp_valid_packet_in_sequence(&s->statistics, seq)) |
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{ |
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av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", |
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if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) { |
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av_log(st ? st->codec : NULL, AV_LOG_ERROR, |
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"RTP: PT=%02x: bad cseq %04x expected=%04x\n", |
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payload_type, seq, ((s->seq + 1) & 0xffff)); |
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return -1; |
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} |
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@ -528,7 +533,7 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, |
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} |
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if (!st) { |
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/* specific MPEG2TS demux support */ |
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/* specific MPEG2-TS demux support */ |
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ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len); |
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/* The only error that can be returned from ff_mpegts_parse_packet
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* is "no more data to return from the provided buffer", so return |
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@ -546,11 +551,12 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, |
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rv = s->parse_packet(s->ic, s->dynamic_protocol_context, |
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s->st, pkt, ×tamp, buf, len, flags); |
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} else { |
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// at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
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/* At this point, the RTP header has been stripped;
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* This is ASSUMING that there is only 1 CSRC, which isn't wise. */ |
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switch (st->codec->codec_id) { |
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case AV_CODEC_ID_MP2: |
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case AV_CODEC_ID_MP3: |
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/* better than nothing: skip mpeg audio RTP header */ |
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/* better than nothing: skip MPEG audio RTP header */ |
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if (len <= 4) |
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return -1; |
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h = AV_RB32(buf); |
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@ -562,14 +568,14 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, |
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break; |
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case AV_CODEC_ID_MPEG1VIDEO: |
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case AV_CODEC_ID_MPEG2VIDEO: |
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/* better than nothing: skip mpeg video RTP header */ |
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/* better than nothing: skip MPEG video RTP header */ |
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if (len <= 4) |
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return -1; |
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h = AV_RB32(buf); |
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buf += 4; |
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len -= 4; |
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if (h & (1 << 26)) { |
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/* mpeg2 */ |
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/* MPEG-2 */ |
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if (len <= 4) |
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return -1; |
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buf += 4; |
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@ -789,14 +795,16 @@ int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p, |
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} |
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// remove protocol identifier
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while (*p && *p == ' ') p++; // strip spaces
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while (*p && *p != ' ') p++; // eat protocol identifier
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while (*p && *p == ' ') p++; // strip trailing spaces
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while (*p && *p == ' ') |
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p++; // strip spaces
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while (*p && *p != ' ') |
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p++; // eat protocol identifier
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while (*p && *p == ' ') |
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p++; // strip trailing spaces
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while (ff_rtsp_next_attr_and_value(&p, |
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attr, sizeof(attr), |
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value, value_size)) { |
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res = parse_fmtp(stream, data, attr, value); |
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if (res < 0 && res != AVERROR_PATCHWELCOME) { |
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av_free(value); |
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Michael Niedermayer
13 years ago