| /* |
| * 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 "avformat.h" |
| #include "network.h" |
| #include "srtp.h" |
| #include "url.h" |
| #include "rtpdec.h" |
| #include "rtpdec_formats.h" |
| |
| #define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */ |
| |
| static RTPDynamicProtocolHandler l24_dynamic_handler = { |
| .enc_name = "L24", |
| .codec_type = AVMEDIA_TYPE_AUDIO, |
| .codec_id = AV_CODEC_ID_PCM_S24BE, |
| }; |
| |
| static RTPDynamicProtocolHandler gsm_dynamic_handler = { |
| .enc_name = "GSM", |
| .codec_type = AVMEDIA_TYPE_AUDIO, |
| .codec_id = AV_CODEC_ID_GSM, |
| }; |
| |
| 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, |
| }; |
| |
| static RTPDynamicProtocolHandler opus_dynamic_handler = { |
| .enc_name = "opus", |
| .codec_type = AVMEDIA_TYPE_AUDIO, |
| .codec_id = AV_CODEC_ID_OPUS, |
| }; |
| |
| static RTPDynamicProtocolHandler t140_dynamic_handler = { /* RFC 4103 */ |
| .enc_name = "t140", |
| .codec_type = AVMEDIA_TYPE_SUBTITLE, |
| .codec_id = AV_CODEC_ID_TEXT, |
| }; |
| |
| extern RTPDynamicProtocolHandler ff_rdt_video_handler; |
| extern RTPDynamicProtocolHandler ff_rdt_audio_handler; |
| extern RTPDynamicProtocolHandler ff_rdt_live_video_handler; |
| extern RTPDynamicProtocolHandler ff_rdt_live_audio_handler; |
| |
| static const RTPDynamicProtocolHandler *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, |
| }; |
| |
| const RTPDynamicProtocolHandler *ff_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 = ff_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 = ff_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) |
| { |
| AVIOContext *pb; |
| uint8_t *buf; |
| int len; |
| |
| /* Send a small RTP packet */ |
| if (avio_open_dyn_buf(&pb) < 0) |
| return; |
| |
| avio_w8(pb, (RTP_VERSION << 6)); |
| avio_w8(pb, 0); /* Payload type */ |
| avio_wb16(pb, 0); /* Seq */ |
| avio_wb32(pb, 0); /* Timestamp */ |
| avio_wb32(pb, 0); /* SSRC */ |
| |
| len = avio_close_dyn_buf(pb, &buf); |
| if ((len > 0) && buf) |
| ffurl_write(rtp_handle, buf, len); |
| av_free(buf); |
| |
| /* Send a minimal RTCP RR */ |
| if (avio_open_dyn_buf(&pb) < 0) |
| return; |
| |
| avio_w8(pb, (RTP_VERSION << 6)); |
| avio_w8(pb, RTCP_RR); /* receiver report */ |
| avio_wb16(pb, 1); /* length in words - 1 */ |
| avio_wb32(pb, 0); /* our own SSRC */ |
| |
| len = avio_close_dyn_buf(pb, &buf); |
| if ((len > 0) && buf) |
| ffurl_write(rtp_handle, buf, len); |
| 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 |
| * MPEG-2 TS streams. |
| */ |
| 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->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; |
| 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; |
| } |
| |
| /** |
| * 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 && 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)) |
| av_log(s->ic, AV_LOG_WARNING, |
| "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); |
| 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_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; |
| } |
