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@ -25,49 +25,46 @@ |
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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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*/ |
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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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.enc_name = "X-MP3-draft-00", |
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.codec_type = AVMEDIA_TYPE_AUDIO, |
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.codec_id = AV_CODEC_ID_MP3ADU, |
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.enc_name = "X-MP3-draft-00", |
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.codec_type = AVMEDIA_TYPE_AUDIO, |
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.codec_id = AV_CODEC_ID_MP3ADU, |
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}; |
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static RTPDynamicProtocolHandler speex_dynamic_handler = { |
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.enc_name = "speex", |
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.codec_type = AVMEDIA_TYPE_AUDIO, |
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.codec_id = AV_CODEC_ID_SPEEX, |
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.enc_name = "speex", |
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.codec_type = AVMEDIA_TYPE_AUDIO, |
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.codec_id = AV_CODEC_ID_SPEEX, |
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}; |
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static RTPDynamicProtocolHandler opus_dynamic_handler = { |
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.enc_name = "opus", |
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.codec_type = AVMEDIA_TYPE_AUDIO, |
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.codec_id = AV_CODEC_ID_OPUS, |
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.enc_name = "opus", |
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.codec_type = AVMEDIA_TYPE_AUDIO, |
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.codec_id = AV_CODEC_ID_OPUS, |
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}; |
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/* statistics functions */ |
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static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL; |
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static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler = NULL; |
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void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) |
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{ |
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handler->next= RTPFirstDynamicPayloadHandler; |
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RTPFirstDynamicPayloadHandler= handler; |
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handler->next = RTPFirstDynamicPayloadHandler; |
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RTPFirstDynamicPayloadHandler = handler; |
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} |
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void av_register_rtp_dynamic_payload_handlers(void) |
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@ -108,7 +105,7 @@ void av_register_rtp_dynamic_payload_handlers(void) |
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} |
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RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name, |
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enum AVMediaType codec_type) |
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enum AVMediaType codec_type) |
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{ |
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RTPDynamicProtocolHandler *handler; |
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for (handler = RTPFirstDynamicPayloadHandler; |
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@ -120,7 +117,7 @@ RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name, |
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} |
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RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id, |
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enum AVMediaType codec_type) |
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enum AVMediaType codec_type) |
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{ |
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RTPDynamicProtocolHandler *handler; |
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for (handler = RTPFirstDynamicPayloadHandler; |
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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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@ -140,11 +138,12 @@ 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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s->last_rtcp_ntp_time = AV_RB64(buf + 8); |
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s->last_rtcp_ntp_time = AV_RB64(buf + 8); |
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s->last_rtcp_timestamp = AV_RB32(buf + 16); |
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if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) { |
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s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; |
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@ -164,7 +163,7 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l |
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return -1; |
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} |
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#define RTP_SEQ_MOD (1<<16) |
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#define RTP_SEQ_MOD (1 << 16) |
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static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) |
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{ |
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@ -174,8 +173,9 @@ 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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*/ |
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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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s->max_seq = seq; |
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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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@ -211,7 +210,7 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) |
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} |
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} else { |
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s->probation = MIN_SEQUENTIAL - 1; |
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s->max_seq = seq; |
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s->max_seq = seq; |
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} |
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} else if (udelta < MAX_DROPOUT) { |
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// in order, with permissible gap
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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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@ -277,15 +277,15 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) |
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avio_wb32(pb, s->ssrc); // server SSRC
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// some placeholders we should really fill...
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// RFC 1889/p64
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extended_max = stats->cycles + stats->max_seq; |
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expected = extended_max - stats->base_seq + 1; |
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lost = expected - stats->received; |
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lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
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expected_interval = expected - stats->expected_prior; |
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extended_max = stats->cycles + stats->max_seq; |
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expected = extended_max - stats->base_seq + 1; |
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lost = expected - stats->received; |
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lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
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expected_interval = expected - stats->expected_prior; |
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stats->expected_prior = expected; |
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received_interval = stats->received - stats->received_prior; |
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received_interval = stats->received - stats->received_prior; |
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stats->received_prior = stats->received; |
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lost_interval = expected_interval - received_interval; |
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lost_interval = expected_interval - received_interval; |
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if (expected_interval == 0 || lost_interval <= 0) |
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fraction = 0; |
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else |
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@ -301,7 +301,7 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) |
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avio_wb32(pb, 0); /* last SR timestamp */ |
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avio_wb32(pb, 0); /* delay since last SR */ |
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} else { |
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uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
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uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
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uint32_t delay_since_last = ntp_time - s->last_rtcp_ntp_time; |
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avio_wb32(pb, middle_32_bits); /* last SR timestamp */ |
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@ -318,23 +318,22 @@ 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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if ((len > 0) && buf) { |
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int av_unused result; |
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av_dlog(s->ic, "sending %d bytes of RR\n", len); |
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result= ffurl_write(s->rtp_ctx, buf, len); |
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result = ffurl_write(s->rtp_ctx, buf, len); |
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av_dlog(s->ic, "result from ffurl_write: %d\n", result); |
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av_free(buf); |
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} |
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return 0; |
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} |
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void ff_rtp_send_punch_packets(URLContext* rtp_handle) |
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void ff_rtp_send_punch_packets(URLContext *rtp_handle) |
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{ |
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AVIOContext *pb; |
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uint8_t *buf; |
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@ -372,25 +371,26 @@ 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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s = av_mallocz(sizeof(RTPDemuxContext)); |
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if (!s) |
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return NULL; |
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s->payload_type = payload_type; |
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s->last_rtcp_ntp_time = AV_NOPTS_VALUE; |
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s->payload_type = payload_type; |
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s->last_rtcp_ntp_time = AV_NOPTS_VALUE; |
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s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
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s->ic = s1; |
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s->st = st; |
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s->queue_size = queue_size; |
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s->ic = s1; |
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s->st = st; |
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s->queue_size = queue_size; |
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rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
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if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) { |
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s->ts = ff_mpegts_parse_open(s->ic); |
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@ -399,7 +399,7 @@ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext |
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return NULL; |
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} |
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} else if (st) { |
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switch(st->codec->codec_id) { |
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switch (st->codec->codec_id) { |
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case AV_CODEC_ID_MPEG1VIDEO: |
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case AV_CODEC_ID_MPEG2VIDEO: |
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case AV_CODEC_ID_MP2: |
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@ -432,11 +432,12 @@ void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, |
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RTPDynamicProtocolHandler *handler) |
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{ |
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s->dynamic_protocol_context = ctx; |
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s->parse_packet = handler->parse_packet; |
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s->parse_packet = handler->parse_packet; |
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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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@ -477,15 +482,15 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, |
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int ext; |
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AVStream *st; |
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|
uint32_t timestamp; |
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int rv= 0; |
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int rv = 0; |
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ext = buf[0] & 0x10; |
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ext = buf[0] & 0x10; |
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payload_type = buf[1] & 0x7f; |
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if (buf[1] & 0x80) |
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flags |= RTP_FLAG_MARKER; |
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seq = AV_RB16(buf + 2); |
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seq = AV_RB16(buf + 2); |
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timestamp = AV_RB32(buf + 4); |
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|
ssrc = AV_RB32(buf + 8); |
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|
ssrc = AV_RB32(buf + 8); |
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|
/* store the ssrc in the RTPDemuxContext */ |
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|
s->ssrc = ssrc; |
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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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|
@ -509,8 +514,8 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, |
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|
} |
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|
s->seq = seq; |
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|
len -= 12; |
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|
buf += 12; |
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|
len -= 12; |
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|
buf += 12; |
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|
|
/* RFC 3550 Section 5.3.1 RTP Header Extension handling */ |
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|
if (ext) { |
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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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|
|
@ -546,14 +551,15 @@ 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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|
|
|
switch(st->codec->codec_id) { |
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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) { |
|
|
|
|
case AV_CODEC_ID_MP2: |
|
|
|
|
case AV_CODEC_ID_MP3: |
|
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|
|
/* better than nothing: skip mpeg audio RTP header */ |
|
|
|
|
/* better than nothing: skip MPEG audio RTP header */ |
|
|
|
|
if (len <= 4) |
|
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|
|
return -1; |
|
|
|
|
h = AV_RB32(buf); |
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|
|
h = AV_RB32(buf); |
|
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|
|
len -= 4; |
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|
|
|
buf += 4; |
|
|
|
|
if (av_new_packet(pkt, len) < 0) |
|
|
|
@ -562,14 +568,14 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, |
|
|
|
|
break; |
|
|
|
|
case AV_CODEC_ID_MPEG1VIDEO: |
|
|
|
|
case AV_CODEC_ID_MPEG2VIDEO: |
|
|
|
|
/* better than nothing: skip mpeg video RTP header */ |
|
|
|
|
/* better than nothing: skip MPEG video RTP header */ |
|
|
|
|
if (len <= 4) |
|
|
|
|
return -1; |
|
|
|
|
h = AV_RB32(buf); |
|
|
|
|
h = AV_RB32(buf); |
|
|
|
|
buf += 4; |
|
|
|
|
len -= 4; |
|
|
|
|
if (h & (1 << 26)) { |
|
|
|
|
/* mpeg2 */ |
|
|
|
|
/* MPEG-2 */ |
|
|
|
|
if (len <= 4) |
|
|
|
|
return -1; |
|
|
|
|
buf += 4; |
|
|
|
@ -610,7 +616,7 @@ void ff_rtp_reset_packet_queue(RTPDemuxContext *s) |
|
|
|
|
|
|
|
|
|
static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len) |
|
|
|
|
{ |
|
|
|
|
uint16_t seq = AV_RB16(buf + 2); |
|
|
|
|
uint16_t seq = AV_RB16(buf + 2); |
|
|
|
|
RTPPacket *cur = s->queue, *prev = NULL, *packet; |
|
|
|
|
|
|
|
|
|
/* Find the correct place in the queue to insert the packet */ |
|
|
|
@ -619,17 +625,17 @@ static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len) |
|
|
|
|
if (diff < 0) |
|
|
|
|
break; |
|
|
|
|
prev = cur; |
|
|
|
|
cur = cur->next; |
|
|
|
|
cur = cur->next; |
|
|
|
|
} |
|
|
|
|
|
|
|
|
|
packet = av_mallocz(sizeof(*packet)); |
|
|
|
|
if (!packet) |
|
|
|
|
return; |
|
|
|
|
packet->recvtime = av_gettime(); |
|
|
|
|
packet->seq = seq; |
|
|
|
|
packet->len = len; |
|
|
|
|
packet->buf = buf; |
|
|
|
|
packet->next = cur; |
|
|
|
|
packet->seq = seq; |
|
|
|
|
packet->len = len; |
|
|
|
|
packet->buf = buf; |
|
|
|
|
packet->next = cur; |
|
|
|
|
if (prev) |
|
|
|
|
prev->next = packet; |
|
|
|
|
else |
|
|
|
@ -660,7 +666,7 @@ static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt) |
|
|
|
|
"RTP: missed %d packets\n", s->queue->seq - s->seq - 1); |
|
|
|
|
|
|
|
|
|
/* Parse the first packet in the queue, and dequeue it */ |
|
|
|
|
rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len); |
|
|
|
|
rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len); |
|
|
|
|
next = s->queue->next; |
|
|
|
|
av_free(s->queue->buf); |
|
|
|
|
av_free(s->queue); |
|
|
|
@ -672,10 +678,10 @@ static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt) |
|
|
|
|
static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt, |
|
|
|
|
uint8_t **bufptr, int len) |
|
|
|
|
{ |
|
|
|
|
uint8_t* buf = bufptr ? *bufptr : NULL; |
|
|
|
|
uint8_t *buf = bufptr ? *bufptr : NULL; |
|
|
|
|
int ret, flags = 0; |
|
|
|
|
uint32_t timestamp; |
|
|
|
|
int rv= 0; |
|
|
|
|
int rv = 0; |
|
|
|
|
|
|
|
|
|
if (!buf) { |
|
|
|
|
/* If parsing of the previous packet actually returned 0 or an error,
|
|
|
|
@ -684,12 +690,12 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt, |
|
|
|
|
if (s->prev_ret <= 0) |
|
|
|
|
return rtp_parse_queued_packet(s, pkt); |
|
|
|
|
/* return the next packets, if any */ |
|
|
|
|
if(s->st && s->parse_packet) { |
|
|
|
|
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); |
|
|
|
|
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 { |
|
|
|
@ -697,7 +703,7 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt, |
|
|
|
|
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); |
|
|
|
|
s->read_buf_size - s->read_buf_index); |
|
|
|
|
if (ret < 0) |
|
|
|
|
return AVERROR(EAGAIN); |
|
|
|
|
s->read_buf_index += ret; |
|
|
|
@ -789,14 +795,16 @@ int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p, |
|
|
|
|
} |
|
|
|
|
|
|
|
|
|
// remove protocol identifier
|
|
|
|
|
while (*p && *p == ' ') p++; // strip spaces
|
|
|
|
|
while (*p && *p != ' ') p++; // eat protocol identifier
|
|
|
|
|
while (*p && *p == ' ') p++; // strip trailing spaces
|
|
|
|
|
while (*p && *p == ' ') |
|
|
|
|
p++; // strip spaces
|
|
|
|
|
while (*p && *p != ' ') |
|
|
|
|
p++; // eat protocol identifier
|
|
|
|
|
while (*p && *p == ' ') |
|
|
|
|
p++; // strip trailing spaces
|
|
|
|
|
|
|
|
|
|
while (ff_rtsp_next_attr_and_value(&p, |
|
|
|
|
attr, sizeof(attr), |
|
|
|
|
value, value_size)) { |
|
|
|
|
|
|
|
|
|
res = parse_fmtp(stream, data, attr, value); |
|
|
|
|
if (res < 0 && res != AVERROR_PATCHWELCOME) { |
|
|
|
|
av_free(value); |
|
|
|
@ -811,9 +819,9 @@ int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx) |
|
|
|
|
{ |
|
|
|
|
av_init_packet(pkt); |
|
|
|
|
|
|
|
|
|
pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data); |
|
|
|
|
pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data); |
|
|
|
|
pkt->stream_index = stream_idx; |
|
|
|
|
pkt->destruct = av_destruct_packet; |
|
|
|
|
*dyn_buf = NULL; |
|
|
|
|
*dyn_buf = NULL; |
|
|
|
|
return pkt->size; |
|
|
|
|
} |
|
|
|
|