diff options
Diffstat (limited to 'ffmpeg1/libavformat/rtpdec.c')
| -rw-r--r-- | ffmpeg1/libavformat/rtpdec.c | 878 |
1 files changed, 878 insertions, 0 deletions
diff --git a/ffmpeg1/libavformat/rtpdec.c b/ffmpeg1/libavformat/rtpdec.c new file mode 100644 index 0000000..b512b89 --- /dev/null +++ b/ffmpeg1/libavformat/rtpdec.c @@ -0,0 +1,878 @@ +/* + * 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/time.h" +#include "libavcodec/get_bits.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 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 *rtp_first_dynamic_payload_handler = NULL; + +void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) +{ + handler->next = rtp_first_dynamic_payload_handler; + rtp_first_dynamic_payload_handler = handler; +} + +void av_register_rtp_dynamic_payload_handlers(void) +{ + 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_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_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_ilbc_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_mp4v_es_dynamic_handler); + ff_register_dynamic_payload_handler(&ff_mpeg_audio_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_pfv_handler); + ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_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_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); +} + +RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name, + enum AVMediaType codec_type) +{ + RTPDynamicProtocolHandler *handler; + for (handler = rtp_first_dynamic_payload_handler; + handler; handler = handler->next) + if (!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) +{ + RTPDynamicProtocolHandler *handler; + 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) + 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(NULL, AV_LOG_ERROR, + "Invalid length for RTCP SR packet\n"); + return AVERROR_INVALIDDATA; + } + + s->last_rtcp_reception_time = av_gettime(); + 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; + } + + 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() - 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_dlog(s->ic, "sending %d bytes of RR\n", 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) +{ + 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 */ + + avio_flush(pb); + 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 */ + + avio_flush(pb); + 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(); + 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 + * MPEG2-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; + 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) + st->codec->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, + 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(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; + } + + 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_free(s->queue->buf); + av_free(s->queue); + s->queue = next; + } + s->seq = 0; + s->queue_len = 0; + s->prev_ret = 0; +} + +static void 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; + packet->recvtime = av_gettime(); + packet->seq = seq; + packet->len = len; + packet->buf = buf; + packet->next = *cur; + *cur = packet; + s->queue_len++; +} + +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->st ? s->st->codec : NULL, 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_free(s->queue->buf); + av_free(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(); + 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->st ? s->st->codec : NULL, 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. */ + enqueue_packet(s, buf, len); + *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) + 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 == AVERROR(EAGAIN) && 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(AVStream *stream, PayloadContext *data, const char *p, + int (*parse_fmtp)(AVStream *stream, + PayloadContext *data, + char *attr, char *value)) +{ + char attr[256]; + char *value; + int res; + int value_size = strlen(p) + 1; + + if (!(value = av_malloc(value_size))) { + av_log(NULL, 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(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; +} |