X-Git-Url: http://git.osdn.net/view?a=blobdiff_plain;f=libavformat%2Frtpdec.c;h=a80463b98b4c91813fc9b752d62cbfaa772b7b08;hb=ec96a89c3e507cf0fb1f2b159b28a53f2bad9a74;hp=2c5e6c817650243d570eece591fddd14af8b14d2;hpb=c4ef6a3e4ba50b7e3746a46b51c2f8d16e8cba7b;p=android-x86%2Fexternal-ffmpeg.git diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c index 2c5e6c8176..a80463b98b 100644 --- a/libavformat/rtpdec.c +++ b/libavformat/rtpdec.c @@ -24,91 +24,107 @@ #include "libavutil/time.h" #include "libavcodec/get_bits.h" #include "avformat.h" -#include "mpegts.h" -#include "url.h" - #include "network.h" - +#include "srtp.h" +#include "url.h" #include "rtpdec.h" #include "rtpdec_formats.h" -//#define DEBUG +#define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */ -/* TODO: - add RTCP statistics reporting (should be optional). +static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = { + .enc_name = "X-MP3-draft-00", + .codec_type = AVMEDIA_TYPE_AUDIO, + .codec_id = AV_CODEC_ID_MP3ADU, +}; + +static RTPDynamicProtocolHandler speex_dynamic_handler = { + .enc_name = "speex", + .codec_type = AVMEDIA_TYPE_AUDIO, + .codec_id = AV_CODEC_ID_SPEEX, +}; - - add support for h263/mpeg4 packetized output : IDEA: send a - buffer to 'rtp_write_packet' contains all the packets for ONE - frame. Each packet should have a four byte header containing - the length in big endian format (same trick as - 'ffio_open_dyn_packet_buf') -*/ +static RTPDynamicProtocolHandler opus_dynamic_handler = { + .enc_name = "opus", + .codec_type = AVMEDIA_TYPE_AUDIO, + .codec_id = AV_CODEC_ID_OPUS, +}; -static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = { - .enc_name = "X-MP3-draft-00", - .codec_type = AVMEDIA_TYPE_AUDIO, - .codec_id = CODEC_ID_MP3ADU, +static RTPDynamicProtocolHandler t140_dynamic_handler = { /* RFC 4103 */ + .enc_name = "t140", + .codec_type = AVMEDIA_TYPE_DATA, + .codec_id = AV_CODEC_ID_TEXT, }; -/* statistics functions */ -static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL; +static RTPDynamicProtocolHandler *rtp_first_dynamic_payload_handler = NULL; void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) { - handler->next= RTPFirstDynamicPayloadHandler; - RTPFirstDynamicPayloadHandler= handler; + handler->next = rtp_first_dynamic_payload_handler; + rtp_first_dynamic_payload_handler = handler; } -void av_register_rtp_dynamic_payload_handlers(void) +void ff_register_rtp_dynamic_payload_handlers(void) { - ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler); - ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_ac3_dynamic_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_dv_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_h261_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_h263_rfc2190_dynamic_handler); ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_hevc_dynamic_handler); ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler); - ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler); - ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler); - ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler); - ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler); ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler); - ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler); - ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler); - ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler); - - ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler); + ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_mpeg_audio_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_mpeg_audio_robust_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_mpeg_video_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_mpegts_dynamic_handler); ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler); - + ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler); + ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler); ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler); ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler); ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler); ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler); - - ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler); - ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler); - ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler); - ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler); + ff_register_dynamic_payload_handler(&opus_dynamic_handler); + ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler); + ff_register_dynamic_payload_handler(&speex_dynamic_handler); + ff_register_dynamic_payload_handler(&t140_dynamic_handler); } RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name, - enum AVMediaType codec_type) + enum AVMediaType codec_type) { RTPDynamicProtocolHandler *handler; - for (handler = RTPFirstDynamicPayloadHandler; + for (handler = rtp_first_dynamic_payload_handler; handler; handler = handler->next) - if (!av_strcasecmp(name, handler->enc_name) && + if (handler->enc_name && + !av_strcasecmp(name, handler->enc_name) && codec_type == handler->codec_type) return handler; return NULL; } RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id, - enum AVMediaType codec_type) + enum AVMediaType codec_type) { RTPDynamicProtocolHandler *handler; - for (handler = RTPFirstDynamicPayloadHandler; + for (handler = rtp_first_dynamic_payload_handler; handler; handler = handler->next) if (handler->static_payload_id && handler->static_payload_id == id && codec_type == handler->codec_type) @@ -116,7 +132,8 @@ RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id, return NULL; } -static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len) +static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, + int len) { int payload_len; while (len >= 4) { @@ -125,17 +142,19 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l switch (buf[1]) { case RTCP_SR: if (payload_len < 20) { - av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n"); + av_log(NULL, AV_LOG_ERROR, + "Invalid length for RTCP SR packet\n"); return AVERROR_INVALIDDATA; } - s->last_rtcp_ntp_time = AV_RB64(buf + 8); + s->last_rtcp_reception_time = av_gettime_relative(); + s->last_rtcp_ntp_time = AV_RB64(buf + 8); s->last_rtcp_timestamp = AV_RB32(buf + 16); if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) { s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; if (!s->base_timestamp) s->base_timestamp = s->last_rtcp_timestamp; - s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp; + s->rtcp_ts_offset = (int32_t)(s->last_rtcp_timestamp - s->base_timestamp); } break; @@ -149,73 +168,70 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l return -1; } -#define RTP_SEQ_MOD (1<<16) +#define RTP_SEQ_MOD (1 << 16) -/** -* called on parse open packet -*/ -static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet. +static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) { memset(s, 0, sizeof(RTPStatistics)); - s->max_seq= base_sequence; - s->probation= 1; + s->max_seq = base_sequence; + s->probation = 1; } -/** -* called whenever there is a large jump in sequence numbers, or when they get out of probation... -*/ +/* + * Called whenever there is a large jump in sequence numbers, + * or when they get out of probation... + */ static void rtp_init_sequence(RTPStatistics *s, uint16_t seq) { - s->max_seq= seq; - s->cycles= 0; - s->base_seq= seq -1; - s->bad_seq= RTP_SEQ_MOD + 1; - s->received= 0; - s->expected_prior= 0; - s->received_prior= 0; - s->jitter= 0; - s->transit= 0; + s->max_seq = seq; + s->cycles = 0; + s->base_seq = seq - 1; + s->bad_seq = RTP_SEQ_MOD + 1; + s->received = 0; + s->expected_prior = 0; + s->received_prior = 0; + s->jitter = 0; + s->transit = 0; } -/** -* returns 1 if we should handle this packet. -*/ +/* Returns 1 if we should handle this packet. */ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) { - uint16_t udelta= seq - s->max_seq; - const int MAX_DROPOUT= 3000; - const int MAX_MISORDER = 100; + uint16_t udelta = seq - s->max_seq; + const int MAX_DROPOUT = 3000; + const int MAX_MISORDER = 100; const int MIN_SEQUENTIAL = 2; - /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */ - if(s->probation) - { - if(seq==s->max_seq + 1) { + /* source not valid until MIN_SEQUENTIAL packets with sequence + * seq. numbers have been received */ + if (s->probation) { + if (seq == s->max_seq + 1) { s->probation--; - s->max_seq= seq; - if(s->probation==0) { + s->max_seq = seq; + if (s->probation == 0) { rtp_init_sequence(s, seq); s->received++; return 1; } } else { - s->probation= MIN_SEQUENTIAL - 1; - s->max_seq = seq; + s->probation = MIN_SEQUENTIAL - 1; + s->max_seq = seq; } } else if (udelta < MAX_DROPOUT) { // in order, with permissible gap - if(seq < s->max_seq) { - //sequence number wrapped; count antother 64k cycles + if (seq < s->max_seq) { + // sequence number wrapped; count another 64k cycles s->cycles += RTP_SEQ_MOD; } - s->max_seq= seq; + s->max_seq = seq; } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) { // sequence made a large jump... - if(seq==s->bad_seq) { - // two sequential packets-- assume that the other side restarted without telling us; just resync. + if (seq == s->bad_seq) { + /* two sequential packets -- assume that the other side + * restarted without telling us; just resync. */ rtp_init_sequence(s, seq); } else { - s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1); + s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1); return 0; } } else { @@ -225,27 +241,45 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) return 1; } -int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) +static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, + uint32_t arrival_timestamp) +{ + // Most of this is pretty straight from RFC 3550 appendix A.8 + uint32_t transit = arrival_timestamp - sent_timestamp; + uint32_t prev_transit = s->transit; + int32_t d = transit - prev_transit; + // Doing the FFABS() call directly on the "transit - prev_transit" + // expression doesn't work, since it's an unsigned expression. Doing the + // transit calculation in unsigned is desired though, since it most + // probably will need to wrap around. + d = FFABS(d); + s->transit = transit; + if (!prev_transit) + return; + s->jitter += d - (int32_t) ((s->jitter + 8) >> 4); +} + +int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, + AVIOContext *avio, int count) { AVIOContext *pb; uint8_t *buf; int len; int rtcp_bytes; - RTPStatistics *stats= &s->statistics; + RTPStatistics *stats = &s->statistics; uint32_t lost; uint32_t extended_max; uint32_t expected_interval; uint32_t received_interval; - uint32_t lost_interval; + int32_t lost_interval; uint32_t expected; uint32_t fraction; - uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time? - if (!s->rtp_ctx || (count < 1)) + if ((!fd && !avio) || (count < 1)) return -1; /* TODO: I think this is way too often; RFC 1889 has algorithm for this */ - /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ + /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */ s->octet_count += count; rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / RTCP_TX_RATIO_DEN; @@ -254,7 +288,9 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) return -1; s->last_octet_count = s->octet_count; - if (avio_open_dyn_buf(&pb) < 0) + if (!fd) + pb = avio; + else if (avio_open_dyn_buf(&pb) < 0) return -1; // Receiver Report @@ -266,31 +302,33 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) avio_wb32(pb, s->ssrc); // server SSRC // some placeholders we should really fill... // RFC 1889/p64 - extended_max= stats->cycles + stats->max_seq; - expected= extended_max - stats->base_seq + 1; - lost= expected - stats->received; - lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits... - expected_interval= expected - stats->expected_prior; - stats->expected_prior= expected; - received_interval= stats->received - stats->received_prior; - stats->received_prior= stats->received; - lost_interval= expected_interval - received_interval; - if (expected_interval==0 || lost_interval<=0) fraction= 0; - else fraction = (lost_interval<<8)/expected_interval; - - fraction= (fraction<<24) | lost; + extended_max = stats->cycles + stats->max_seq; + expected = extended_max - stats->base_seq; + lost = expected - stats->received; + lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits... + expected_interval = expected - stats->expected_prior; + stats->expected_prior = expected; + received_interval = stats->received - stats->received_prior; + stats->received_prior = stats->received; + lost_interval = expected_interval - received_interval; + if (expected_interval == 0 || lost_interval <= 0) + fraction = 0; + else + fraction = (lost_interval << 8) / expected_interval; + + fraction = (fraction << 24) | lost; avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */ avio_wb32(pb, extended_max); /* max sequence received */ - avio_wb32(pb, stats->jitter>>4); /* jitter */ + avio_wb32(pb, stats->jitter >> 4); /* jitter */ - if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE) - { + if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) { avio_wb32(pb, 0); /* last SR timestamp */ avio_wb32(pb, 0); /* delay since last SR */ } else { - uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special? - uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time; + uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special? + uint32_t delay_since_last = av_rescale(av_gettime_relative() - s->last_rtcp_reception_time, + 65536, AV_TIME_BASE); avio_wb32(pb, middle_32_bits); /* last SR timestamp */ avio_wb32(pb, delay_since_last); /* delay since last SR */ @@ -300,29 +338,31 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ avio_w8(pb, RTCP_SDES); len = strlen(s->hostname); - avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */ + avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */ avio_wb32(pb, s->ssrc + 1); avio_w8(pb, 0x01); avio_w8(pb, len); avio_write(pb, s->hostname, len); + avio_w8(pb, 0); /* END */ // padding - for (len = (6 + len) % 4; len % 4; len++) { + for (len = (7 + len) % 4; len % 4; len++) avio_w8(pb, 0); - } avio_flush(pb); + if (!fd) + return 0; len = avio_close_dyn_buf(pb, &buf); if ((len > 0) && buf) { int av_unused result; av_dlog(s->ic, "sending %d bytes of RR\n", len); - result= ffurl_write(s->rtp_ctx, buf, len); + result = ffurl_write(fd, buf, len); av_dlog(s->ic, "result from ffurl_write: %d\n", result); av_free(buf); } return 0; } -void ff_rtp_send_punch_packets(URLContext* rtp_handle) +void ff_rtp_send_punch_packets(URLContext *rtp_handle) { AVIOContext *pb; uint8_t *buf; @@ -360,47 +400,122 @@ void ff_rtp_send_punch_packets(URLContext* rtp_handle) av_free(buf); } +static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing, + uint16_t *missing_mask) +{ + int i; + uint16_t next_seq = s->seq + 1; + RTPPacket *pkt = s->queue; + + if (!pkt || pkt->seq == next_seq) + return 0; + + *missing_mask = 0; + for (i = 1; i <= 16; i++) { + uint16_t missing_seq = next_seq + i; + while (pkt) { + int16_t diff = pkt->seq - missing_seq; + if (diff >= 0) + break; + pkt = pkt->next; + } + if (!pkt) + break; + if (pkt->seq == missing_seq) + continue; + *missing_mask |= 1 << (i - 1); + } + + *first_missing = next_seq; + return 1; +} + +int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, + AVIOContext *avio) +{ + int len, need_keyframe, missing_packets; + AVIOContext *pb; + uint8_t *buf; + int64_t now; + uint16_t first_missing = 0, missing_mask = 0; + + if (!fd && !avio) + return -1; + + need_keyframe = s->handler && s->handler->need_keyframe && + s->handler->need_keyframe(s->dynamic_protocol_context); + missing_packets = find_missing_packets(s, &first_missing, &missing_mask); + + if (!need_keyframe && !missing_packets) + return 0; + + /* Send new feedback if enough time has elapsed since the last + * feedback packet. */ + + now = av_gettime_relative(); + if (s->last_feedback_time && + (now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL) + return 0; + s->last_feedback_time = now; + + if (!fd) + pb = avio; + else if (avio_open_dyn_buf(&pb) < 0) + return -1; + + if (need_keyframe) { + avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */ + avio_w8(pb, RTCP_PSFB); + avio_wb16(pb, 2); /* length in words - 1 */ + // our own SSRC: we use the server's SSRC + 1 to avoid conflicts + avio_wb32(pb, s->ssrc + 1); + avio_wb32(pb, s->ssrc); // server SSRC + } + + if (missing_packets) { + avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */ + avio_w8(pb, RTCP_RTPFB); + avio_wb16(pb, 3); /* length in words - 1 */ + avio_wb32(pb, s->ssrc + 1); + avio_wb32(pb, s->ssrc); // server SSRC + + avio_wb16(pb, first_missing); + avio_wb16(pb, missing_mask); + } + + avio_flush(pb); + if (!fd) + return 0; + len = avio_close_dyn_buf(pb, &buf); + if (len > 0 && buf) { + ffurl_write(fd, buf, len); + av_free(buf); + } + return 0; +} /** * 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 - * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) + * MPEG2-TS streams. */ -RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size) +RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, + int payload_type, int queue_size) { RTPDemuxContext *s; s = av_mallocz(sizeof(RTPDemuxContext)); if (!s) return NULL; - s->payload_type = payload_type; - s->last_rtcp_ntp_time = AV_NOPTS_VALUE; + 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->queue_size = queue_size; - 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 = ff_mpegts_parse_open(s->ic); - if (s->ts == NULL) { - av_free(s); - return NULL; - } - } else if (st) { - switch(st->codec->codec_id) { - case CODEC_ID_MPEG1VIDEO: - case CODEC_ID_MPEG2VIDEO: - 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; - case CODEC_ID_VORBIS: - st->need_parsing = AVSTREAM_PARSE_HEADERS; - break; - case CODEC_ID_ADPCM_G722: + s->ic = s1; + s->st = st; + s->queue_size = queue_size; + rtp_init_statistics(&s->statistics, 0); + if (st) { + switch (st->codec->codec_id) { + case AV_CODEC_ID_ADPCM_G722: /* According to RFC 3551, the stream clock rate is 8000 * even if the sample rate is 16000. */ if (st->codec->sample_rate == 8000) @@ -411,21 +526,27 @@ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext } } // needed to send back RTCP RR in RTSP sessions - s->rtp_ctx = rtpc; gethostname(s->hostname, sizeof(s->hostname)); return s; } -void -ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, - RTPDynamicProtocolHandler *handler) +void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, + RTPDynamicProtocolHandler *handler) { s->dynamic_protocol_context = ctx; - s->parse_packet = handler->parse_packet; + s->handler = handler; +} + +void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite, + const char *params) +{ + if (!ff_srtp_set_crypto(&s->srtp, suite, params)) + s->srtp_enabled = 1; } /** - * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc. + * This was the second switch in rtp_parse packet. + * Normalizes time, if required, sets stream_index, etc. */ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) { @@ -441,7 +562,9 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam /* 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->first_rtcp_ntp_time, + s->st->time_base.den, + (uint64_t) s->st->time_base.num << 32); pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend + delta_timestamp; return; @@ -449,32 +572,35 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam if (!s->base_timestamp) s->base_timestamp = timestamp; - /* assume that the difference is INT32_MIN < x < INT32_MAX, but allow the first timestamp to exceed INT32_MAX */ + /* assume that the difference is INT32_MIN < x < INT32_MAX, + * but allow the first timestamp to exceed INT32_MAX */ if (!s->timestamp) s->unwrapped_timestamp += timestamp; else s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp); s->timestamp = timestamp; - pkt->pts = s->unwrapped_timestamp + s->range_start_offset - s->base_timestamp; + pkt->pts = s->unwrapped_timestamp + s->range_start_offset - + s->base_timestamp; } 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; + unsigned int ssrc; + int payload_type, seq, flags = 0; + int ext, csrc; AVStream *st; uint32_t timestamp; - int rv= 0; + int rv = 0; - ext = buf[0] & 0x10; + csrc = buf[0] & 0x0f; + ext = buf[0] & 0x10; payload_type = buf[1] & 0x7f; if (buf[1] & 0x80) flags |= RTP_FLAG_MARKER; - seq = AV_RB16(buf + 2); + seq = AV_RB16(buf + 2); timestamp = AV_RB32(buf + 4); - ssrc = AV_RB32(buf + 8); + ssrc = AV_RB32(buf + 8); /* store the ssrc in the RTPDemuxContext */ s->ssrc = ssrc; @@ -484,9 +610,9 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, 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", + 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; } @@ -498,8 +624,13 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, } s->seq = seq; - len -= 12; - buf += 12; + len -= 12; + buf += 12; + + len -= 4 * csrc; + buf += 4 * csrc; + if (len < 0) + return AVERROR_INVALIDDATA; /* RFC 3550 Section 5.3.1 RTP Header Extension handling */ if (ext) { @@ -516,63 +647,17 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, 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; - } - + if (s->handler && s->handler->parse_packet) { + rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context, + s->st, pkt, ×tamp, buf, len, seq, + flags); + } else if (st) { + if ((rv = av_new_packet(pkt, len)) < 0) + return rv; + memcpy(pkt->data, buf, len); pkt->stream_index = st->index; + } else { + return AVERROR(EINVAL); } // now perform timestamp things.... @@ -596,30 +681,26 @@ 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); - RTPPacket *cur = s->queue, *prev = NULL, *packet; + uint16_t seq = AV_RB16(buf + 2); + RTPPacket **cur = &s->queue, *packet; /* Find the correct place in the queue to insert the packet */ - while (cur) { - int16_t diff = seq - cur->seq; + while (*cur) { + int16_t diff = seq - (*cur)->seq; 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; - if (prev) - prev->next = packet; - else - s->queue = packet; + packet->recvtime = av_gettime_relative(); + packet->seq = seq; + packet->len = len; + packet->buf = buf; + packet->next = *cur; + *cur = packet; s->queue_len++; } @@ -646,7 +727,7 @@ 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); @@ -656,12 +737,12 @@ 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 **bufptr, int len) { - uint8_t* buf = bufptr ? *bufptr : NULL; - int ret, flags = 0; + uint8_t *buf = bufptr ? *bufptr : NULL; + int 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, @@ -670,27 +751,15 @@ 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->handler && s->handler->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->handler->parse_packet(s->ic, s->dynamic_protocol_context, + s->st, pkt, ×tamp, NULL, 0, 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; } } @@ -703,6 +772,16 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt, return rtcp_parse_packet(s, buf, len); } + if (s->st) { + int64_t received = av_gettime_relative(); + uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q, + s->st->time_base); + timestamp = AV_RB32(buf + 4); + // Calculate the jitter immediately, before queueing the packet + // into the reordering queue. + rtcp_update_jitter(&s->statistics, timestamp, arrival_ts); + } + if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) { /* First packet, or no reordering */ return rtp_parse_packet_internal(s, pkt, buf, len); @@ -743,7 +822,10 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt, int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len) { - int rv = rtp_parse_one_packet(s, pkt, bufptr, len); + int rv; + if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0) + return -1; + rv = rtp_parse_one_packet(s, pkt, bufptr, len); s->prev_ret = rv; while (rv == AVERROR(EAGAIN) && has_next_packet(s)) rv = rtp_parse_queued_packet(s, pkt); @@ -753,16 +835,16 @@ int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, void ff_rtp_parse_close(RTPDemuxContext *s) { ff_rtp_reset_packet_queue(s); - if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) { - ff_mpegts_parse_close(s->ts); - } + ff_srtp_free(&s->srtp); av_free(s); } -int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p, - int (*parse_fmtp)(AVStream *stream, +int ff_parse_fmtp(AVFormatContext *s, + AVStream *stream, PayloadContext *data, const char *p, + int (*parse_fmtp)(AVFormatContext *s, + AVStream *stream, PayloadContext *data, - char *attr, char *value)) + const char *attr, const char *value)) { char attr[256]; char *value; @@ -770,20 +852,22 @@ int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p, int value_size = strlen(p) + 1; if (!(value = av_malloc(value_size))) { - av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP."); + av_log(NULL, AV_LOG_ERROR, "Failed to allocate data for FMTP."); return AVERROR(ENOMEM); } // 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); + res = parse_fmtp(s, stream, data, attr, value); if (res < 0 && res != AVERROR_PATCHWELCOME) { av_free(value); return res; @@ -792,3 +876,18 @@ int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p, av_free(value); return 0; } + +int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx) +{ + int ret; + av_init_packet(pkt); + + pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data); + pkt->stream_index = stream_idx; + *dyn_buf = NULL; + if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) { + av_freep(&pkt->data); + return ret; + } + return pkt->size; +}