X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtpdec.c;h=41e6eb4cabfc9d11929fd50a681f7d2c885a427d;hb=b7327887ea260d26e4fe98d34149cd57168f2ba3;hp=0b316e3c4864d11e739c771574d47db3e4d26deb;hpb=9b3788efc341d99dea6107a3683a2d73a1b1effe;p=ffmpeg diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c index 0b316e3c486..41e6eb4cabf 100644 --- a/libavformat/rtpdec.c +++ b/libavformat/rtpdec.c @@ -2,40 +2,35 @@ * RTP input format * Copyright (c) 2002 Fabrice Bellard * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -/* needed for gethostname() */ -#define _XOPEN_SOURCE 600 - +#include "libavutil/mathematics.h" +#include "libavutil/avstring.h" #include "libavcodec/get_bits.h" #include "avformat.h" #include "mpegts.h" +#include "url.h" #include #include "network.h" #include "rtpdec.h" -#include "rtpdec_amr.h" -#include "rtpdec_asf.h" -#include "rtpdec_h263.h" -#include "rtpdec_h264.h" -#include "rtpdec_mpeg4.h" -#include "rtpdec_xiph.h" +#include "rtpdec_formats.h" //#define DEBUG @@ -45,11 +40,17 @@ 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 - 'url_open_dyn_packet_buf') + '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 */ -RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL; +static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL; void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) { @@ -65,23 +66,86 @@ void av_register_rtp_dynamic_payload_handlers(void) 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_h263_rfc2190_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) { - if (buf[1] != 200) - return -1; - s->last_rtcp_ntp_time = AV_RB64(buf + 8); - if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) - s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; - s->last_rtcp_timestamp = AV_RB32(buf + 16); - return 0; + 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) @@ -160,25 +224,9 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) return 1; } -#if 0 -/** -* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the -* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values -* never change. I left this in in case someone else can see a way. (rdm) -*/ -static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp) -{ - uint32_t transit= arrival_timestamp - sent_timestamp; - int d; - s->transit= transit; - d= FFABS(transit - s->transit); - s->jitter += d - ((s->jitter + 8)>>4); -} -#endif - -int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) +int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) { - ByteIOContext *pb; + AVIOContext *pb; uint8_t *buf; int len; int rtcp_bytes; @@ -205,15 +253,16 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) return -1; s->last_octet_count = s->octet_count; - if (url_open_dyn_buf(&pb) < 0) + if (avio_open_dyn_buf(&pb) < 0) return -1; // Receiver Report - put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ - put_byte(pb, 201); - put_be16(pb, 7); /* length in words - 1 */ - put_be32(pb, s->ssrc); // our own SSRC - put_be32(pb, s->ssrc); // XXX: should be the server's here! + 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; @@ -230,83 +279,83 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) fraction= (fraction<<24) | lost; - put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */ - put_be32(pb, extended_max); /* max sequence received */ - put_be32(pb, stats->jitter>>4); /* jitter */ + 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) { - put_be32(pb, 0); /* last SR timestamp */ - put_be32(pb, 0); /* delay since last SR */ + 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; - put_be32(pb, middle_32_bits); /* last SR timestamp */ - put_be32(pb, delay_since_last); /* delay since last SR */ + avio_wb32(pb, middle_32_bits); /* last SR timestamp */ + avio_wb32(pb, delay_since_last); /* delay since last SR */ } // CNAME - put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ - put_byte(pb, 202); + avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ + avio_w8(pb, RTCP_SDES); len = strlen(s->hostname); - put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */ - put_be32(pb, s->ssrc); - put_byte(pb, 0x01); - put_byte(pb, len); - put_buffer(pb, s->hostname, len); + 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++) { - put_byte(pb, 0); + avio_w8(pb, 0); } - put_flush_packet(pb); - len = url_close_dyn_buf(pb, &buf); + avio_flush(pb); + len = avio_close_dyn_buf(pb, &buf); if ((len > 0) && buf) { - int result; - dprintf(s->ic, "sending %d bytes of RR\n", len); - result= url_write(s->rtp_ctx, buf, len); - dprintf(s->ic, "result from url_write: %d\n", result); + 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 rtp_send_punch_packets(URLContext* rtp_handle) +void ff_rtp_send_punch_packets(URLContext* rtp_handle) { - ByteIOContext *pb; + AVIOContext *pb; uint8_t *buf; int len; /* Send a small RTP packet */ - if (url_open_dyn_buf(&pb) < 0) + if (avio_open_dyn_buf(&pb) < 0) return; - put_byte(pb, (RTP_VERSION << 6)); - put_byte(pb, 0); /* Payload type */ - put_be16(pb, 0); /* Seq */ - put_be32(pb, 0); /* Timestamp */ - put_be32(pb, 0); /* SSRC */ + 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 */ - put_flush_packet(pb); - len = url_close_dyn_buf(pb, &buf); + avio_flush(pb); + len = avio_close_dyn_buf(pb, &buf); if ((len > 0) && buf) - url_write(rtp_handle, buf, len); + ffurl_write(rtp_handle, buf, len); av_free(buf); /* Send a minimal RTCP RR */ - if (url_open_dyn_buf(&pb) < 0) + if (avio_open_dyn_buf(&pb) < 0) return; - put_byte(pb, (RTP_VERSION << 6)); - put_byte(pb, 201); /* receiver report */ - put_be16(pb, 1); /* length in words - 1 */ - put_be32(pb, 0); /* our own SSRC */ + 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 */ - put_flush_packet(pb); - len = url_close_dyn_buf(pb, &buf); + avio_flush(pb); + len = avio_close_dyn_buf(pb, &buf); if ((len > 0) && buf) - url_write(rtp_handle, buf, len); + ffurl_write(rtp_handle, buf, len); av_free(buf); } @@ -315,9 +364,8 @@ void rtp_send_punch_packets(URLContext* rtp_handle) * 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) - * TODO: change this to not take rtp_payload data, and use the new dynamic payload system. */ -RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data) +RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size) { RTPDemuxContext *s; @@ -329,7 +377,7 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r s->first_rtcp_ntp_time = AV_NOPTS_VALUE; s->ic = s1; s->st = st; - s->rtp_payload_data = rtp_payload_data; + 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); @@ -337,8 +385,7 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r av_free(s); return NULL; } - } else { - av_set_pts_info(st, 32, 1, 90000); + } else if (st) { switch(st->codec->codec_id) { case CODEC_ID_MPEG1VIDEO: case CODEC_ID_MPEG2VIDEO: @@ -349,10 +396,16 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r case CODEC_ID_H264: st->need_parsing = AVSTREAM_PARSE_FULL; break; + case CODEC_ID_VORBIS: + st->need_parsing = AVSTREAM_PARSE_HEADERS; + 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: - if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { - av_set_pts_info(st, 32, 1, st->codec->sample_rate); - } break; } } @@ -363,71 +416,24 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r } void -rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, - RTPDynamicProtocolHandler *handler) +ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, + RTPDynamicProtocolHandler *handler) { s->dynamic_protocol_context = ctx; s->parse_packet = handler->parse_packet; } -static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf) -{ - int au_headers_length, au_header_size, i; - GetBitContext getbitcontext; - RTPPayloadData *infos; - - infos = s->rtp_payload_data; - - if (infos == NULL) - return -1; - - /* decode the first 2 bytes where the AUHeader sections are stored - length in bits */ - au_headers_length = AV_RB16(buf); - - if (au_headers_length > RTP_MAX_PACKET_LENGTH) - return -1; - - infos->au_headers_length_bytes = (au_headers_length + 7) / 8; - - /* skip AU headers length section (2 bytes) */ - buf += 2; - - init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8); - - /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */ - au_header_size = infos->sizelength + infos->indexlength; - if (au_header_size <= 0 || (au_headers_length % au_header_size != 0)) - return -1; - - infos->nb_au_headers = au_headers_length / au_header_size; - if (!infos->au_headers || infos->au_headers_allocated < infos->nb_au_headers) { - av_free(infos->au_headers); - infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers); - infos->au_headers_allocated = infos->nb_au_headers; - } - - /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving) - In my test, the FAAD decoder does not behave correctly when sending each AU one by one - but does when sending the whole as one big packet... */ - infos->au_headers[0].size = 0; - infos->au_headers[0].index = 0; - for (i = 0; i < infos->nb_au_headers; ++i) { - infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength); - infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength); - } - - infos->nb_au_headers = 1; - - return 0; -} - /** * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc. */ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) { - if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { + 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; @@ -435,61 +441,33 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam 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 + addend + delta_timestamp; + 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; } -/** - * Parse an RTP or RTCP packet directly sent as a buffer. - * @param s RTP parse context. - * @param pkt returned packet - * @param buf 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 rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, - const uint8_t *buf, int len) +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; - if (!buf) { - /* return the next packets, if any */ - if(s->st && s->parse_packet) { - timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned.... - 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 -1; - 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 -1; - 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] >= 200 && buf[1] <= 204) { - rtcp_parse_packet(s, buf, len); - return -1; - } + ext = buf[0] & 0x10; payload_type = buf[1] & 0x7f; if (buf[1] & 0x80) flags |= RTP_FLAG_MARKER; @@ -512,15 +490,39 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, 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 -1; + return AVERROR(EAGAIN); if (ret < len) { s->read_buf_size = len - ret; memcpy(s->buf, buf + ret, s->read_buf_size); @@ -563,29 +565,6 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, av_new_packet(pkt, len); memcpy(pkt->data, buf, len); break; - // moved from below, verbatim. this is because this section handles packets, and the lower switch handles - // timestamps. - // TODO: Put this into a dynamic packet handler... - case CODEC_ID_AAC: - if (rtp_parse_mp4_au(s, buf)) - return -1; - { - RTPPayloadData *infos = s->rtp_payload_data; - if (infos == NULL) - return -1; - buf += infos->au_headers_length_bytes + 2; - len -= infos->au_headers_length_bytes + 2; - - /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define - one au_header */ - av_new_packet(pkt, infos->au_headers[0].size); - memcpy(pkt->data, buf, infos->au_headers[0].size); - buf += infos->au_headers[0].size; - len -= infos->au_headers[0].size; - } - s->read_buf_size = len; - rv= 0; - break; default: av_new_packet(pkt, len); memcpy(pkt->data, buf, len); @@ -601,13 +580,214 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, return rv; } -void rtp_parse_close(RTPDemuxContext *s) +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) { - // TODO: fold this into the protocol specific data fields. - av_free(s->rtp_payload_data->mode); - av_free(s->rtp_payload_data->au_headers); + 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 (RTP_PT_IS_RTCP(buf[1])) { + 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; +}