| /* |
| * 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 "libavcodec/get_bits.h" |
| #include "avformat.h" |
| #include "mpegts.h" |
| #include "url.h" |
| |
| #include <unistd.h> |
| #include "network.h" |
| |
| #include "rtpdec.h" |
| #include "rtpdec_formats.h" |
| |
| //#define DEBUG |
| |
| /* TODO: - add RTCP statistics reporting (should be optional). |
| |
| - add support for h263/mpeg4 packetized output : IDEA: send a |
| buffer to 'rtp_write_packet' contains all the packets for ONE |
| frame. Each packet should have a four byte header containing |
| the length in big endian format (same trick as |
| 'ffio_open_dyn_packet_buf') |
| */ |
| |
| static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = { |
| .enc_name = "X-MP3-draft-00", |
| .codec_type = AVMEDIA_TYPE_AUDIO, |
| .codec_id = CODEC_ID_MP3ADU, |
| }; |
| |
| /* statistics functions */ |
| static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL; |
| |
| void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) |
| { |
| handler->next= RTPFirstDynamicPayloadHandler; |
| RTPFirstDynamicPayloadHandler= handler; |
| } |
| |
| void av_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_h263_1998_dynamic_handler); |
| ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler); |
| ff_register_dynamic_payload_handler(&ff_h264_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(&ff_realmedia_mp3_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_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); |
| } |
| |
| RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name, |
| enum AVMediaType codec_type) |
| { |
| RTPDynamicProtocolHandler *handler; |
| for (handler = RTPFirstDynamicPayloadHandler; |
| 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 = RTPFirstDynamicPayloadHandler; |
| 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_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) |
| |
| /** |
| * called on parse open packet |
| */ |
| static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet. |
| { |
| 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 antother 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; |
| } |
| |
| int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, 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; |
| uint32_t lost_interval; |
| uint32_t expected; |
| uint32_t fraction; |
| uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time? |
| |
| if (!s->rtp_ctx || (count < 1)) |
| 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 (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 + 1; |
| lost= expected - stats->received; |
| lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits... |
| expected_interval= expected - stats->expected_prior; |
| stats->expected_prior= expected; |
| received_interval= stats->received - stats->received_prior; |
| stats->received_prior= stats->received; |
| lost_interval= expected_interval - received_interval; |
| if (expected_interval==0 || lost_interval<=0) fraction= 0; |
| else fraction = (lost_interval<<8)/expected_interval; |
| |
| fraction= (fraction<<24) | lost; |
| |
| 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= ntp_time - s->last_rtcp_ntp_time; |
| |
| 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, (6 + 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); |
| // padding |
| for (len = (6 + len) % 4; len % 4; len++) { |
| avio_w8(pb, 0); |
| } |
| |
| avio_flush(pb); |
| len = avio_close_dyn_buf(pb, &buf); |
| if ((len > 0) && buf) { |
| int av_unused result; |
| av_dlog(s->ic, "sending %d bytes of RR\n", len); |
| result= ffurl_write(s->rtp_ctx, buf, len); |
| 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); |
| } |
| |
| |
| /** |
| * open a new RTP parse context for stream 'st'. 'st' can be NULL for |
| * MPEG2TS streams to indicate that they should be demuxed inside the |
| * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) |
| */ |
| RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, 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); // 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 { |
| switch(st->codec->codec_id) { |
| case CODEC_ID_MPEG1VIDEO: |
| case CODEC_ID_MPEG2VIDEO: |
| case CODEC_ID_MP2: |
| case CODEC_ID_MP3: |
| case CODEC_ID_MPEG4: |
| case CODEC_ID_H263: |
| case CODEC_ID_H264: |
| st->need_parsing = AVSTREAM_PARSE_FULL; |
| break; |
| case CODEC_ID_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 |
| s->rtp_ctx = rtpc; |
| gethostname(s->hostname, sizeof(s->hostname)); |
| return s; |
| } |
| |
| void |
| ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, |
| RTPDynamicProtocolHandler *handler) |
| { |
| s->dynamic_protocol_context = ctx; |
| s->parse_packet = handler->parse_packet; |
| } |
| |
| /** |
| * 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, h; |
| int payload_type, seq, ret, flags = 0; |
| int ext; |
| AVStream *st; |
| uint32_t timestamp; |
| int rv= 0; |
| |
| 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; |
| |
| /* 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 (!st) { |
| /* specific MPEG2TS demux support */ |
| ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len); |
| /* The only error that can be returned from ff_mpegts_parse_packet |
| * is "no more data to return from the provided buffer", so return |
| * AVERROR(EAGAIN) for all errors */ |
| if (ret < 0) |
| return AVERROR(EAGAIN); |
| if (ret < len) { |
| s->read_buf_size = len - ret; |
| memcpy(s->buf, buf + ret, s->read_buf_size); |
| s->read_buf_index = 0; |
| return 1; |
| } |
| return 0; |
| } else if (s->parse_packet) { |
| rv = s->parse_packet(s->ic, s->dynamic_protocol_context, |
| s->st, pkt, ×tamp, buf, len, flags); |
| } else { |
| // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise. |
| switch(st->codec->codec_id) { |
| case CODEC_ID_MP2: |
| case CODEC_ID_MP3: |
| /* better than nothing: skip mpeg audio RTP header */ |
| if (len <= 4) |
| return -1; |
| h = AV_RB32(buf); |
| len -= 4; |
| buf += 4; |
| av_new_packet(pkt, len); |
| memcpy(pkt->data, buf, len); |
| break; |
| case CODEC_ID_MPEG1VIDEO: |
| case CODEC_ID_MPEG2VIDEO: |
| /* better than nothing: skip mpeg video RTP header */ |
| if (len <= 4) |
| return -1; |
| h = AV_RB32(buf); |
| buf += 4; |
| len -= 4; |
| if (h & (1 << 26)) { |
| /* mpeg2 */ |
| if (len <= 4) |
| return -1; |
| buf += 4; |
| len -= 4; |
| } |
| av_new_packet(pkt, len); |
| memcpy(pkt->data, buf, len); |
| break; |
| default: |
| av_new_packet(pkt, len); |
| memcpy(pkt->data, buf, len); |
| break; |
| } |
| |
| pkt->stream_index = st->index; |
| } |
| |
| // 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, *prev = NULL, *packet; |
| |
| /* Find the correct place in the queue to insert the packet */ |
| while (cur) { |
| int16_t diff = seq - cur->seq; |
| if (diff < 0) |
| break; |
| prev = cur; |
| cur = cur->next; |
| } |
| |
| packet = av_mallocz(sizeof(*packet)); |
| if (!packet) |
| return; |
| packet->recvtime = av_gettime(); |
| packet->seq = seq; |
| packet->len = len; |
| packet->buf = buf; |
| packet->next = cur; |
| if (prev) |
| prev->next = packet; |
| else |
| s->queue = packet; |
| 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 ret, 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->st && s->parse_packet) { |
| /* timestamp should be overwritten by parse_packet, if not, |
| * the packet is left with pts == AV_NOPTS_VALUE */ |
| timestamp = RTP_NOTS_VALUE; |
| rv= s->parse_packet(s->ic, s->dynamic_protocol_context, |
| s->st, pkt, ×tamp, NULL, 0, flags); |
| 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; |
| } |
| } |
| |
| if (len < 12) |
| return -1; |
| |
| if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) |
| return -1; |
| if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) { |
| return rtcp_parse_packet(s, buf, len); |
| } |
| |
| 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 = 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); |
| if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) { |
| ff_mpegts_parse_close(s->ts); |
| } |
| 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(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP."); |
| return AVERROR(ENOMEM); |
| } |
| |
| // remove protocol identifier |
| while (*p && *p == ' ') p++; // strip spaces |
| while (*p && *p != ' ') p++; // eat protocol identifier |
| while (*p && *p == ' ') p++; // strip trailing spaces |
| |
| while (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; |
| } |
