| /* | |
| * RTP input format | |
| * Copyright (c) 2002 Fabrice Bellard | |
| * | |
| * This file is part of FFmpeg. | |
| * | |
| * FFmpeg is free software; you can redistribute it and/or | |
| * modify it under the terms of the GNU Lesser General Public | |
| * License as published by the Free Software Foundation; either | |
| * version 2.1 of the License, or (at your option) any later version. | |
| * | |
| * FFmpeg is distributed in the hope that it will be useful, | |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
| * Lesser General Public License for more details. | |
| * | |
| * You should have received a copy of the GNU Lesser General Public | |
| * License along with FFmpeg; if not, write to the Free Software | |
| * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
| */ | |
| #include "libavutil/mathematics.h" | |
| #include "libavutil/avstring.h" | |
| #include "libavutil/intreadwrite.h" | |
| #include "libavutil/time.h" | |
| #include "libavcodec/bytestream.h" | |
| #include "avformat.h" | |
| #include "network.h" | |
| #include "srtp.h" | |
| #include "url.h" | |
| #include "rtpdec.h" | |
| #include "rtpdec_formats.h" | |
| #include "internal.h" | |
| #define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */ | |
| static const RTPDynamicProtocolHandler l24_dynamic_handler = { | |
| .enc_name = "L24", | |
| .codec_type = AVMEDIA_TYPE_AUDIO, | |
| .codec_id = AV_CODEC_ID_PCM_S24BE, | |
| }; | |
| static const RTPDynamicProtocolHandler gsm_dynamic_handler = { | |
| .enc_name = "GSM", | |
| .codec_type = AVMEDIA_TYPE_AUDIO, | |
| .codec_id = AV_CODEC_ID_GSM, | |
| }; | |
| static const RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = { | |
| .enc_name = "X-MP3-draft-00", | |
| .codec_type = AVMEDIA_TYPE_AUDIO, | |
| .codec_id = AV_CODEC_ID_MP3ADU, | |
| }; | |
| static const RTPDynamicProtocolHandler speex_dynamic_handler = { | |
| .enc_name = "speex", | |
| .codec_type = AVMEDIA_TYPE_AUDIO, | |
| .codec_id = AV_CODEC_ID_SPEEX, | |
| }; | |
| static const RTPDynamicProtocolHandler opus_dynamic_handler = { | |
| .enc_name = "opus", | |
| .codec_type = AVMEDIA_TYPE_AUDIO, | |
| .codec_id = AV_CODEC_ID_OPUS, | |
| }; | |
| static const RTPDynamicProtocolHandler t140_dynamic_handler = { /* RFC 4103 */ | |
| .enc_name = "t140", | |
| .codec_type = AVMEDIA_TYPE_SUBTITLE, | |
| .codec_id = AV_CODEC_ID_TEXT, | |
| }; | |
| extern const RTPDynamicProtocolHandler ff_rdt_video_handler; | |
| extern const RTPDynamicProtocolHandler ff_rdt_audio_handler; | |
| extern const RTPDynamicProtocolHandler ff_rdt_live_video_handler; | |
| extern const RTPDynamicProtocolHandler ff_rdt_live_audio_handler; | |
| static const RTPDynamicProtocolHandler *const rtp_dynamic_protocol_handler_list[] = { | |
| /* rtp */ | |
| &ff_ac3_dynamic_handler, | |
| &ff_amr_nb_dynamic_handler, | |
| &ff_amr_wb_dynamic_handler, | |
| &ff_dv_dynamic_handler, | |
| &ff_g726_16_dynamic_handler, | |
| &ff_g726_24_dynamic_handler, | |
| &ff_g726_32_dynamic_handler, | |
| &ff_g726_40_dynamic_handler, | |
| &ff_g726le_16_dynamic_handler, | |
| &ff_g726le_24_dynamic_handler, | |
| &ff_g726le_32_dynamic_handler, | |
| &ff_g726le_40_dynamic_handler, | |
| &ff_h261_dynamic_handler, | |
| &ff_h263_1998_dynamic_handler, | |
| &ff_h263_2000_dynamic_handler, | |
| &ff_h263_rfc2190_dynamic_handler, | |
| &ff_h264_dynamic_handler, | |
| &ff_hevc_dynamic_handler, | |
| &ff_ilbc_dynamic_handler, | |
| &ff_jpeg_dynamic_handler, | |
| &ff_mp4a_latm_dynamic_handler, | |
| &ff_mp4v_es_dynamic_handler, | |
| &ff_mpeg_audio_dynamic_handler, | |
| &ff_mpeg_audio_robust_dynamic_handler, | |
| &ff_mpeg_video_dynamic_handler, | |
| &ff_mpeg4_generic_dynamic_handler, | |
| &ff_mpegts_dynamic_handler, | |
| &ff_ms_rtp_asf_pfa_handler, | |
| &ff_ms_rtp_asf_pfv_handler, | |
| &ff_qcelp_dynamic_handler, | |
| &ff_qdm2_dynamic_handler, | |
| &ff_qt_rtp_aud_handler, | |
| &ff_qt_rtp_vid_handler, | |
| &ff_quicktime_rtp_aud_handler, | |
| &ff_quicktime_rtp_vid_handler, | |
| &ff_rfc4175_rtp_handler, | |
| &ff_svq3_dynamic_handler, | |
| &ff_theora_dynamic_handler, | |
| &ff_vc2hq_dynamic_handler, | |
| &ff_vorbis_dynamic_handler, | |
| &ff_vp8_dynamic_handler, | |
| &ff_vp9_dynamic_handler, | |
| &gsm_dynamic_handler, | |
| &l24_dynamic_handler, | |
| &opus_dynamic_handler, | |
| &realmedia_mp3_dynamic_handler, | |
| &speex_dynamic_handler, | |
| &t140_dynamic_handler, | |
| /* rdt */ | |
| &ff_rdt_video_handler, | |
| &ff_rdt_audio_handler, | |
| &ff_rdt_live_video_handler, | |
| &ff_rdt_live_audio_handler, | |
| NULL, | |
| }; | |
| /** | |
| * Iterate over all registered rtp dynamic protocol handlers. | |
| * | |
| * @param opaque a pointer where libavformat will store the iteration state. | |
| * Must point to NULL to start the iteration. | |
| * | |
| * @return the next registered rtp dynamic protocol handler | |
| * or NULL when the iteration is finished | |
| */ | |
| static const RTPDynamicProtocolHandler *rtp_handler_iterate(void **opaque) | |
| { | |
| uintptr_t i = (uintptr_t)*opaque; | |
| const RTPDynamicProtocolHandler *r = rtp_dynamic_protocol_handler_list[i]; | |
| if (r) | |
| *opaque = (void*)(i + 1); | |
| return r; | |
| } | |
| const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name, | |
| enum AVMediaType codec_type) | |
| { | |
| void *i = 0; | |
| const RTPDynamicProtocolHandler *handler; | |
| while (handler = rtp_handler_iterate(&i)) { | |
| if (handler->enc_name && | |
| !av_strcasecmp(name, handler->enc_name) && | |
| codec_type == handler->codec_type) | |
| return handler; | |
| } | |
| return NULL; | |
| } | |
| const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id, | |
| enum AVMediaType codec_type) | |
| { | |
| void *i = 0; | |
| const RTPDynamicProtocolHandler *handler; | |
| while (handler = rtp_handler_iterate(&i)) { | |
| if (handler->static_payload_id && handler->static_payload_id == id && | |
| codec_type == handler->codec_type) | |
| return handler; | |
| } | |
| return NULL; | |
| } | |
| static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, | |
| int len) | |
| { | |
| int payload_len; | |
| while (len >= 4) { | |
| payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4); | |
| switch (buf[1]) { | |
| case RTCP_SR: | |
| if (payload_len < 20) { | |
| av_log(s->ic, AV_LOG_ERROR, "Invalid RTCP SR packet length\n"); | |
| return AVERROR_INVALIDDATA; | |
| } | |
| 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 = (int32_t)(s->last_rtcp_timestamp - s->base_timestamp); | |
| } | |
| break; | |
| case RTCP_BYE: | |
| return -RTCP_BYE; | |
| } | |
| buf += payload_len; | |
| len -= payload_len; | |
| } | |
| return -1; | |
| } | |
| #define RTP_SEQ_MOD (1 << 16) | |
| static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) | |
| { | |
| memset(s, 0, sizeof(RTPStatistics)); | |
| 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... | |
| */ | |
| 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; | |
| } | |
| /* 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; | |
| 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) { | |
| s->probation--; | |
| 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; | |
| } | |
| } else if (udelta < MAX_DROPOUT) { | |
| // in order, with permissible gap | |
| if (seq < s->max_seq) { | |
| // sequence number wrapped; count another 64k cycles | |
| s->cycles += RTP_SEQ_MOD; | |
| } | |
| 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. */ | |
| rtp_init_sequence(s, seq); | |
| } else { | |
| s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1); | |
| return 0; | |
| } | |
| } else { | |
| // duplicate or reordered packet... | |
| } | |
| s->received++; | |
| return 1; | |
| } | |
| 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; | |
| uint32_t lost; | |
| uint32_t extended_max; | |
| uint32_t expected_interval; | |
| uint32_t received_interval; | |
| int32_t lost_interval; | |
| uint32_t expected; | |
| uint32_t fraction; | |
| 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 */ | |
| s->octet_count += count; | |
| rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / | |
| RTCP_TX_RATIO_DEN; | |
| rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !? | |
| if (rtcp_bytes < 28) | |
| return -1; | |
| s->last_octet_count = s->octet_count; | |
| if (!fd) | |
| pb = avio; | |
| else if (avio_open_dyn_buf(&pb) < 0) | |
| return -1; | |
| // Receiver Report | |
| avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ | |
| avio_w8(pb, RTCP_RR); | |
| avio_wb16(pb, 7); /* 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 | |
| // some placeholders we should really fill... | |
| // RFC 1889/p64 | |
| 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 */ | |
| 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 = 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 */ | |
| } | |
| // CNAME | |
| avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ | |
| avio_w8(pb, RTCP_SDES); | |
| len = strlen(s->hostname); | |
| 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 = (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_log(s->ic, AV_LOG_TRACE, "sending %d bytes of RR\n", len); | |
| result = ffurl_write(fd, buf, len); | |
| av_log(s->ic, AV_LOG_TRACE, "result from ffurl_write: %d\n", result); | |
| av_free(buf); | |
| } | |
| return 0; | |
| } | |
| void ff_rtp_send_punch_packets(URLContext *rtp_handle) | |
| { | |
| uint8_t buf[RTP_MIN_PACKET_LENGTH], *ptr = buf; | |
| /* Send a small RTP packet */ | |
| bytestream_put_byte(&ptr, (RTP_VERSION << 6)); | |
| bytestream_put_byte(&ptr, 0); /* Payload type */ | |
| bytestream_put_be16(&ptr, 0); /* Seq */ | |
| bytestream_put_be32(&ptr, 0); /* Timestamp */ | |
| bytestream_put_be32(&ptr, 0); /* SSRC */ | |
| ffurl_write(rtp_handle, buf, ptr - buf); | |
| /* Send a minimal RTCP RR */ | |
| ptr = buf; | |
| bytestream_put_byte(&ptr, (RTP_VERSION << 6)); | |
| bytestream_put_byte(&ptr, RTCP_RR); /* receiver report */ | |
| bytestream_put_be16(&ptr, 1); /* length in words - 1 */ | |
| bytestream_put_be32(&ptr, 0); /* our own SSRC */ | |
| ffurl_write(rtp_handle, buf, ptr - 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; | |
| } | |
| static int opus_write_extradata(AVCodecParameters *codecpar) | |
| { | |
| uint8_t *bs; | |
| int ret; | |
| /* This function writes an extradata with a channel mapping family of 0. | |
| * This mapping family only supports mono and stereo layouts. And RFC7587 | |
| * specifies that the number of channels in the SDP must be 2. | |
| */ | |
| if (codecpar->ch_layout.nb_channels > 2) { | |
| return AVERROR_INVALIDDATA; | |
| } | |
| ret = ff_alloc_extradata(codecpar, 19); | |
| if (ret < 0) | |
| return ret; | |
| bs = (uint8_t *)codecpar->extradata; | |
| /* Opus magic */ | |
| bytestream_put_buffer(&bs, "OpusHead", 8); | |
| /* Version */ | |
| bytestream_put_byte (&bs, 0x1); | |
| /* Channel count */ | |
| bytestream_put_byte (&bs, codecpar->ch_layout.nb_channels); | |
| /* Pre skip */ | |
| bytestream_put_le16 (&bs, 0); | |
| /* Input sample rate */ | |
| bytestream_put_le32 (&bs, 48000); | |
| /* Output gain */ | |
| bytestream_put_le16 (&bs, 0x0); | |
| /* Mapping family */ | |
| bytestream_put_byte (&bs, 0x0); | |
| return 0; | |
| } | |
| /** | |
| * open a new RTP parse context for stream 'st'. 'st' can be NULL for | |
| * MPEG-2 TS streams. | |
| */ | |
| RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, | |
| int payload_type, int queue_size) | |
| { | |
| RTPDemuxContext *s; | |
| int ret; | |
| s = av_mallocz(sizeof(RTPDemuxContext)); | |
| if (!s) | |
| return NULL; | |
| 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; | |
| av_log(s->ic, AV_LOG_VERBOSE, "setting jitter buffer size to %d\n", | |
| s->queue_size); | |
| rtp_init_statistics(&s->statistics, 0); | |
| if (st) { | |
| switch (st->codecpar->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->codecpar->sample_rate == 8000) | |
| st->codecpar->sample_rate = 16000; | |
| break; | |
| case AV_CODEC_ID_OPUS: | |
| ret = opus_write_extradata(st->codecpar); | |
| if (ret < 0) { | |
| av_log(s1, AV_LOG_ERROR, | |
| "Error creating opus extradata: %s\n", | |
| av_err2str(ret)); | |
| av_free(s); | |
| return NULL; | |
| } | |
| break; | |
| default: | |
| break; | |
| } | |
| } | |
| // needed to send back RTCP RR in RTSP sessions | |
| gethostname(s->hostname, sizeof(s->hostname)); | |
| return s; | |
| } | |
| void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, | |
| const RTPDynamicProtocolHandler *handler) | |
| { | |
| s->dynamic_protocol_context = ctx; | |
| 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; | |
| } | |
| static int rtp_set_prft(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) { | |
| int64_t rtcp_time, delta_timestamp, delta_time; | |
| AVProducerReferenceTime *prft = | |
| (AVProducerReferenceTime *) av_packet_new_side_data( | |
| pkt, AV_PKT_DATA_PRFT, sizeof(AVProducerReferenceTime)); | |
| if (!prft) | |
| return AVERROR(ENOMEM); | |
| rtcp_time = ff_parse_ntp_time(s->last_rtcp_ntp_time) - NTP_OFFSET_US; | |
| delta_timestamp = (int64_t)timestamp - (int64_t)s->last_rtcp_timestamp; | |
| delta_time = av_rescale_q(delta_timestamp, s->st->time_base, AV_TIME_BASE_Q); | |
| prft->wallclock = rtcp_time + delta_time; | |
| prft->flags = 24; | |
| return 0; | |
| } | |
| /** | |
| * 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) | |
| { | |
| if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE) | |
| return; /* Timestamp already set by depacketizer */ | |
| if (timestamp == RTP_NOTS_VALUE) | |
| return; | |
| if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { | |
| if (rtp_set_prft(s, pkt, timestamp) < 0) { | |
| av_log(s->ic, AV_LOG_WARNING, "rtpdec: failed to set prft"); | |
| } | |
| } | |
| if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) { | |
| int64_t addend; | |
| int delta_timestamp; | |
| /* 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); | |
| pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend + | |
| delta_timestamp; | |
| return; | |
| } | |
| 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 */ | |
| 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; | |
| } | |
| static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, | |
| const uint8_t *buf, int len) | |
| { | |
| unsigned int ssrc; | |
| int payload_type, seq, flags = 0; | |
| int ext, csrc; | |
| AVStream *st; | |
| uint32_t timestamp; | |
| int rv = 0; | |
| 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); | |
| timestamp = AV_RB32(buf + 4); | |
| ssrc = AV_RB32(buf + 8); | |
| /* store the ssrc in the RTPDemuxContext */ | |
| s->ssrc = ssrc; | |
| /* NOTE: we can handle only one payload type */ | |
| if (s->payload_type != payload_type) | |
| return -1; | |
| st = s->st; | |
| // only do something with this if all the rtp checks pass... | |
| if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) { | |
| av_log(s->ic, AV_LOG_ERROR, | |
| "RTP: PT=%02x: bad cseq %04x expected=%04x\n", | |
| payload_type, seq, ((s->seq + 1) & 0xffff)); | |
| return -1; | |
| } | |
| if (buf[0] & 0x20) { | |
| int padding = buf[len - 1]; | |
| if (len >= 12 + padding) | |
| len -= padding; | |
| } | |
| s->seq = seq; | |
| len -= 12; | |
| buf += 12; | |
| len -= 4 * csrc; | |
| buf += 4 * csrc; | |
| if (len < 0) | |
| return AVERROR_INVALIDDATA; | |
| /* RFC 3550 Section 5.3.1 RTP Header Extension handling */ | |
| if (ext) { | |
| if (len < 4) | |
| return -1; | |
| /* calculate the header extension length (stored as number | |
| * of 32-bit words) */ | |
| ext = (AV_RB16(buf + 2) + 1) << 2; | |
| if (len < ext) | |
| return -1; | |
| // skip past RTP header extension | |
| len -= ext; | |
| buf += ext; | |
| } | |
| if (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.... | |
| finalize_packet(s, pkt, timestamp); | |
| return rv; | |
| } | |
| void ff_rtp_reset_packet_queue(RTPDemuxContext *s) | |
| { | |
| while (s->queue) { | |
| RTPPacket *next = s->queue->next; | |
| av_freep(&s->queue->buf); | |
| av_freep(&s->queue); | |
| s->queue = next; | |
| } | |
| s->seq = 0; | |
| s->queue_len = 0; | |
| s->prev_ret = 0; | |
| } | |
| static int enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len) | |
| { | |
| 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; | |
| if (diff < 0) | |
| break; | |
| cur = &(*cur)->next; | |
| } | |
| packet = av_mallocz(sizeof(*packet)); | |
| if (!packet) | |
| return AVERROR(ENOMEM); | |
| packet->recvtime = av_gettime_relative(); | |
| packet->seq = seq; | |
| packet->len = len; | |
| packet->buf = buf; | |
| packet->next = *cur; | |
| *cur = packet; | |
| s->queue_len++; | |
| return 0; | |
| } | |
| static int has_next_packet(RTPDemuxContext *s) | |
| { | |
| return s->queue && s->queue->seq == (uint16_t) (s->seq + 1); | |
| } | |
| int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s) | |
| { | |
| return s->queue ? s->queue->recvtime : 0; | |
| } | |
| static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt) | |
| { | |
| int rv; | |
| RTPPacket *next; | |
| if (s->queue_len <= 0) | |
| return -1; | |
| if (!has_next_packet(s)) { | |
| int pkt_missed = s->queue->seq - s->seq - 1; | |
| if (pkt_missed < 0) | |
| pkt_missed += UINT16_MAX; | |
| av_log(s->ic, AV_LOG_WARNING, | |
| "RTP: missed %d packets\n", pkt_missed); | |
| } | |
| /* Parse the first packet in the queue, and dequeue it */ | |
| rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len); | |
| next = s->queue->next; | |
| av_freep(&s->queue->buf); | |
| av_freep(&s->queue); | |
| s->queue = next; | |
| s->queue_len--; | |
| return rv; | |
| } | |
| static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt, | |
| uint8_t **bufptr, int len) | |
| { | |
| uint8_t *buf = bufptr ? *bufptr : NULL; | |
| int flags = 0; | |
| uint32_t timestamp; | |
| int rv = 0; | |
| if (!buf) { | |
| /* If parsing of the previous packet actually returned 0 or an error, | |
| * there's nothing more to be parsed from that packet, but we may have | |
| * indicated that we can return the next enqueued packet. */ | |
| if (s->prev_ret <= 0) | |
| return rtp_parse_queued_packet(s, pkt); | |
| /* return the next packets, if any */ | |
| if (s->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->handler->parse_packet(s->ic, s->dynamic_protocol_context, | |
| s->st, pkt, ×tamp, NULL, 0, 0, | |
| flags); | |
| finalize_packet(s, pkt, timestamp); | |
| return rv; | |
| } | |
| } | |
| if (len < 12) | |
| return -1; | |
| if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) | |
| return -1; | |
| if (RTP_PT_IS_RTCP(buf[1])) { | |
| 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); | |
| } else { | |
| uint16_t seq = AV_RB16(buf + 2); | |
| int16_t diff = seq - s->seq; | |
| if (diff < 0) { | |
| /* Packet older than the previously emitted one, drop */ | |
| av_log(s->ic, AV_LOG_WARNING, | |
| "RTP: dropping old packet received too late\n"); | |
| return -1; | |
| } else if (diff <= 1) { | |
| /* Correct packet */ | |
| rv = rtp_parse_packet_internal(s, pkt, buf, len); | |
| return rv; | |
| } else { | |
| /* Still missing some packet, enqueue this one. */ | |
| rv = enqueue_packet(s, buf, len); | |
| if (rv < 0) | |
| return rv; | |
| *bufptr = NULL; | |
| /* Return the first enqueued packet if the queue is full, | |
| * even if we're missing something */ | |
| if (s->queue_len >= s->queue_size) { | |
| av_log(s->ic, AV_LOG_WARNING, "jitter buffer full\n"); | |
| return rtp_parse_queued_packet(s, pkt); | |
| } | |
| return -1; | |
| } | |
| } | |
| } | |
| /** | |
| * Parse an RTP or RTCP packet directly sent as a buffer. | |
| * @param s RTP parse context. | |
| * @param pkt returned packet | |
| * @param bufptr pointer to the input buffer or NULL to read the next packets | |
| * @param len buffer len | |
| * @return 0 if a packet is returned, 1 if a packet is returned and more can follow | |
| * (use buf as NULL to read the next). -1 if no packet (error or no more packet). | |
| */ | |
| int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, | |
| uint8_t **bufptr, int 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 < 0 && has_next_packet(s)) | |
| rv = rtp_parse_queued_packet(s, pkt); | |
| return rv ? rv : has_next_packet(s); | |
| } | |
| void ff_rtp_parse_close(RTPDemuxContext *s) | |
| { | |
| ff_rtp_reset_packet_queue(s); | |
| ff_srtp_free(&s->srtp); | |
| av_free(s); | |
| } | |
| int ff_parse_fmtp(AVFormatContext *s, | |
| AVStream *stream, PayloadContext *data, const char *p, | |
| int (*parse_fmtp)(AVFormatContext *s, | |
| AVStream *stream, | |
| PayloadContext *data, | |
| const char *attr, const char *value)) | |
| { | |
| char attr[256]; | |
| char *value; | |
| int res; | |
| int value_size = strlen(p) + 1; | |
| if (!(value = av_malloc(value_size))) { | |
| av_log(s, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n"); | |
| 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 (ff_rtsp_next_attr_and_value(&p, | |
| attr, sizeof(attr), | |
| value, value_size)) { | |
| res = parse_fmtp(s, stream, data, attr, value); | |
| if (res < 0 && res != AVERROR_PATCHWELCOME) { | |
| av_free(value); | |
| return res; | |
| } | |
| } | |
| av_free(value); | |
| return 0; | |
| } | |
| int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx) | |
| { | |
| int ret; | |
| av_packet_unref(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; | |
| } |