X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtpdec.c;h=d6421d0a219baff8e91f6e8bb4c6c26092d8d0fc;hb=4d55e9de27894ddfb337b28cd7aa6d14a12666d2;hp=348b796d837f9b2ff6228f8b0b6f897c5ed43b63;hpb=42805eda554a7fc44341282771531e7837ac72b7;p=ffmpeg diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c index 348b796d837..d6421d0a219 100644 --- a/libavformat/rtpdec.c +++ b/libavformat/rtpdec.c @@ -24,12 +24,14 @@ #include "libavutil/time.h" #include "libavcodec/get_bits.h" #include "avformat.h" -#include "mpegts.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, @@ -48,7 +50,6 @@ static RTPDynamicProtocolHandler opus_dynamic_handler = { .codec_id = AV_CODEC_ID_OPUS, }; -/* statistics functions */ static RTPDynamicProtocolHandler *rtp_first_dynamic_payload_handler = NULL; void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) @@ -57,41 +58,43 @@ void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *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_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_h265_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_jpeg_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_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(&realmedia_mp3_dynamic_handler); - ff_register_dynamic_payload_handler(&speex_dynamic_handler); - ff_register_dynamic_payload_handler(&opus_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_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); } RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name, @@ -133,6 +136,7 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, 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) { @@ -226,6 +230,24 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) 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) { @@ -238,10 +260,9 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, 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 ((!fd && !avio) || (count < 1)) return -1; @@ -271,7 +292,7 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, // some placeholders we should really fill... // RFC 1889/p64 extended_max = stats->cycles + stats->max_seq; - expected = extended_max - stats->base_seq + 1; + 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; @@ -295,7 +316,8 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, 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 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 */ @@ -305,13 +327,14 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, 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); @@ -366,10 +389,103 @@ 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(); + 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 to indicate that they should be demuxed inside the - * rtp demux (otherwise AV_CODEC_ID_MPEG2TS packets are returned) + * MPEG2-TS streams. */ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, int queue_size) @@ -385,27 +501,9 @@ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, 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) { + rtp_init_statistics(&s->statistics, 0); + if (st) { switch (st->codec->codec_id) { - case AV_CODEC_ID_MPEG1VIDEO: - case AV_CODEC_ID_MPEG2VIDEO: - case AV_CODEC_ID_MP2: - case AV_CODEC_ID_MP3: - case AV_CODEC_ID_MPEG4: - case AV_CODEC_ID_H263: - case AV_CODEC_ID_H264: - st->need_parsing = AVSTREAM_PARSE_FULL; - break; - case AV_CODEC_ID_VORBIS: - st->need_parsing = AVSTREAM_PARSE_HEADERS; - break; case AV_CODEC_ID_ADPCM_G722: /* According to RFC 3551, the stream clock rate is 8000 * even if the sample rate is 16000. */ @@ -428,6 +526,13 @@ void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *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. @@ -470,13 +575,14 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam 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; + csrc = buf[0] & 0x0f; ext = buf[0] & 0x10; payload_type = buf[1] & 0x7f; if (buf[1] & 0x80) @@ -510,6 +616,11 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, 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) @@ -525,65 +636,17 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, buf += ext; } - if (!st) { - /* specific MPEG2-TS 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 = FFMIN(len - ret, sizeof(s->buf)); - memcpy(s->buf, buf + ret, s->read_buf_size); - s->read_buf_index = 0; - return 1; - } - return 0; - } else if (s->handler && s->handler->parse_packet) { + 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 { - /* At this point, the RTP header has been stripped; - * This is ASSUMING that there is only 1 CSRC, which isn't wise. */ - switch (st->codec->codec_id) { - case AV_CODEC_ID_MP2: - case AV_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 AV_CODEC_ID_MPEG1VIDEO: - case AV_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)) { - /* MPEG-2 */ - 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; - } - + } 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.... @@ -608,15 +671,14 @@ 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; + 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)); @@ -626,11 +688,8 @@ static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len) packet->seq = seq; packet->len = len; packet->buf = buf; - packet->next = cur; - if (prev) - prev->next = packet; - else - s->queue = packet; + packet->next = *cur; + *cur = packet; s->queue_len++; } @@ -670,7 +729,7 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len) { uint8_t *buf = bufptr ? *bufptr : NULL; - int ret, flags = 0; + int flags = 0; uint32_t timestamp; int rv = 0; @@ -681,7 +740,7 @@ 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->handler && s->handler->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; @@ -690,19 +749,6 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt, 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; } } @@ -715,6 +761,16 @@ 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(); + 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); @@ -755,7 +811,10 @@ 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); @@ -765,14 +824,14 @@ 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)) { @@ -797,7 +856,7 @@ int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p, 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; @@ -809,11 +868,15 @@ int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p, 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; - pkt->destruct = av_destruct_packet; - *dyn_buf = NULL; + *dyn_buf = NULL; + if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) { + av_freep(&pkt->data); + return ret; + } return pkt->size; }