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![Carl Eugen Hoyos](/assets/img/avatar_default.png)
Fixes two warnings: libavformat/rtpdec.c:155:20: warning: return discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] libavformat/rtpdec.c:168:20: warning: return discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers]
941 lines
29 KiB
C
941 lines
29 KiB
C
/*
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* RTP input format
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* Copyright (c) 2002 Fabrice Bellard
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/mathematics.h"
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#include "libavutil/avstring.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/time.h"
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#include "avformat.h"
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#include "network.h"
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#include "srtp.h"
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#include "url.h"
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#include "rtpdec.h"
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#include "rtpdec_formats.h"
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#define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */
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static RTPDynamicProtocolHandler l24_dynamic_handler = {
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.enc_name = "L24",
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.codec_type = AVMEDIA_TYPE_AUDIO,
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.codec_id = AV_CODEC_ID_PCM_S24BE,
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};
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static RTPDynamicProtocolHandler gsm_dynamic_handler = {
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.enc_name = "GSM",
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.codec_type = AVMEDIA_TYPE_AUDIO,
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.codec_id = AV_CODEC_ID_GSM,
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};
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static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
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.enc_name = "X-MP3-draft-00",
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.codec_type = AVMEDIA_TYPE_AUDIO,
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.codec_id = AV_CODEC_ID_MP3ADU,
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};
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static RTPDynamicProtocolHandler speex_dynamic_handler = {
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.enc_name = "speex",
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.codec_type = AVMEDIA_TYPE_AUDIO,
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.codec_id = AV_CODEC_ID_SPEEX,
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};
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static RTPDynamicProtocolHandler opus_dynamic_handler = {
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.enc_name = "opus",
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.codec_type = AVMEDIA_TYPE_AUDIO,
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.codec_id = AV_CODEC_ID_OPUS,
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};
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static RTPDynamicProtocolHandler t140_dynamic_handler = { /* RFC 4103 */
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.enc_name = "t140",
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.codec_type = AVMEDIA_TYPE_SUBTITLE,
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.codec_id = AV_CODEC_ID_TEXT,
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};
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extern RTPDynamicProtocolHandler ff_rdt_video_handler;
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extern RTPDynamicProtocolHandler ff_rdt_audio_handler;
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extern RTPDynamicProtocolHandler ff_rdt_live_video_handler;
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extern RTPDynamicProtocolHandler ff_rdt_live_audio_handler;
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static const RTPDynamicProtocolHandler *rtp_dynamic_protocol_handler_list[] = {
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/* rtp */
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&ff_ac3_dynamic_handler,
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&ff_amr_nb_dynamic_handler,
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&ff_amr_wb_dynamic_handler,
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&ff_dv_dynamic_handler,
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&ff_g726_16_dynamic_handler,
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&ff_g726_24_dynamic_handler,
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&ff_g726_32_dynamic_handler,
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&ff_g726_40_dynamic_handler,
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&ff_g726le_16_dynamic_handler,
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&ff_g726le_24_dynamic_handler,
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&ff_g726le_32_dynamic_handler,
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&ff_g726le_40_dynamic_handler,
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&ff_h261_dynamic_handler,
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&ff_h263_1998_dynamic_handler,
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&ff_h263_2000_dynamic_handler,
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&ff_h263_rfc2190_dynamic_handler,
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&ff_h264_dynamic_handler,
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&ff_hevc_dynamic_handler,
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&ff_ilbc_dynamic_handler,
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&ff_jpeg_dynamic_handler,
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&ff_mp4a_latm_dynamic_handler,
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&ff_mp4v_es_dynamic_handler,
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&ff_mpeg_audio_dynamic_handler,
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&ff_mpeg_audio_robust_dynamic_handler,
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&ff_mpeg_video_dynamic_handler,
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&ff_mpeg4_generic_dynamic_handler,
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&ff_mpegts_dynamic_handler,
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&ff_ms_rtp_asf_pfa_handler,
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&ff_ms_rtp_asf_pfv_handler,
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&ff_qcelp_dynamic_handler,
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&ff_qdm2_dynamic_handler,
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&ff_qt_rtp_aud_handler,
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&ff_qt_rtp_vid_handler,
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&ff_quicktime_rtp_aud_handler,
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&ff_quicktime_rtp_vid_handler,
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&ff_rfc4175_rtp_handler,
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&ff_svq3_dynamic_handler,
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&ff_theora_dynamic_handler,
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&ff_vc2hq_dynamic_handler,
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&ff_vorbis_dynamic_handler,
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&ff_vp8_dynamic_handler,
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&ff_vp9_dynamic_handler,
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&gsm_dynamic_handler,
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&l24_dynamic_handler,
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&opus_dynamic_handler,
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&realmedia_mp3_dynamic_handler,
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&speex_dynamic_handler,
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&t140_dynamic_handler,
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/* rdt */
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&ff_rdt_video_handler,
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&ff_rdt_audio_handler,
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&ff_rdt_live_video_handler,
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&ff_rdt_live_audio_handler,
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NULL,
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};
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const RTPDynamicProtocolHandler *ff_rtp_handler_iterate(void **opaque)
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{
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uintptr_t i = (uintptr_t)*opaque;
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const RTPDynamicProtocolHandler *r = rtp_dynamic_protocol_handler_list[i];
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if (r)
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*opaque = (void*)(i + 1);
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return r;
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}
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const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
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enum AVMediaType codec_type)
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{
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void *i = 0;
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const RTPDynamicProtocolHandler *handler;
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while (handler = ff_rtp_handler_iterate(&i)) {
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if (handler->enc_name &&
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!av_strcasecmp(name, handler->enc_name) &&
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codec_type == handler->codec_type)
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return handler;
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}
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return NULL;
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}
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const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
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enum AVMediaType codec_type)
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{
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void *i = 0;
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const RTPDynamicProtocolHandler *handler;
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while (handler = ff_rtp_handler_iterate(&i)) {
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if (handler->static_payload_id && handler->static_payload_id == id &&
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codec_type == handler->codec_type)
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return handler;
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}
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return NULL;
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}
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static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
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int len)
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{
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int payload_len;
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while (len >= 4) {
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payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
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switch (buf[1]) {
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case RTCP_SR:
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if (payload_len < 20) {
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av_log(s->ic, AV_LOG_ERROR, "Invalid RTCP SR packet length\n");
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return AVERROR_INVALIDDATA;
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}
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s->last_rtcp_reception_time = av_gettime_relative();
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s->last_rtcp_ntp_time = AV_RB64(buf + 8);
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s->last_rtcp_timestamp = AV_RB32(buf + 16);
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if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
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s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
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if (!s->base_timestamp)
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s->base_timestamp = s->last_rtcp_timestamp;
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s->rtcp_ts_offset = (int32_t)(s->last_rtcp_timestamp - s->base_timestamp);
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}
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break;
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case RTCP_BYE:
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return -RTCP_BYE;
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}
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buf += payload_len;
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len -= payload_len;
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}
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return -1;
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}
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#define RTP_SEQ_MOD (1 << 16)
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static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
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{
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memset(s, 0, sizeof(RTPStatistics));
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s->max_seq = base_sequence;
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s->probation = 1;
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}
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/*
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* Called whenever there is a large jump in sequence numbers,
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* or when they get out of probation...
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*/
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static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
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{
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s->max_seq = seq;
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s->cycles = 0;
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s->base_seq = seq - 1;
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s->bad_seq = RTP_SEQ_MOD + 1;
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s->received = 0;
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s->expected_prior = 0;
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s->received_prior = 0;
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s->jitter = 0;
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s->transit = 0;
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}
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/* Returns 1 if we should handle this packet. */
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static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
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{
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uint16_t udelta = seq - s->max_seq;
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const int MAX_DROPOUT = 3000;
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const int MAX_MISORDER = 100;
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const int MIN_SEQUENTIAL = 2;
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/* source not valid until MIN_SEQUENTIAL packets with sequence
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* seq. numbers have been received */
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if (s->probation) {
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if (seq == s->max_seq + 1) {
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s->probation--;
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s->max_seq = seq;
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if (s->probation == 0) {
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rtp_init_sequence(s, seq);
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s->received++;
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return 1;
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}
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} else {
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s->probation = MIN_SEQUENTIAL - 1;
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s->max_seq = seq;
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}
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} else if (udelta < MAX_DROPOUT) {
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// in order, with permissible gap
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if (seq < s->max_seq) {
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// sequence number wrapped; count another 64k cycles
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s->cycles += RTP_SEQ_MOD;
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}
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s->max_seq = seq;
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} else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
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// sequence made a large jump...
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if (seq == s->bad_seq) {
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/* two sequential packets -- assume that the other side
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* restarted without telling us; just resync. */
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rtp_init_sequence(s, seq);
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} else {
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s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
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return 0;
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}
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} else {
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// duplicate or reordered packet...
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}
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s->received++;
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return 1;
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}
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static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp,
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uint32_t arrival_timestamp)
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{
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// Most of this is pretty straight from RFC 3550 appendix A.8
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uint32_t transit = arrival_timestamp - sent_timestamp;
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uint32_t prev_transit = s->transit;
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int32_t d = transit - prev_transit;
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// Doing the FFABS() call directly on the "transit - prev_transit"
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// expression doesn't work, since it's an unsigned expression. Doing the
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// transit calculation in unsigned is desired though, since it most
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// probably will need to wrap around.
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d = FFABS(d);
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s->transit = transit;
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if (!prev_transit)
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return;
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s->jitter += d - (int32_t) ((s->jitter + 8) >> 4);
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}
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int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
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AVIOContext *avio, int count)
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{
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AVIOContext *pb;
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uint8_t *buf;
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int len;
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int rtcp_bytes;
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RTPStatistics *stats = &s->statistics;
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uint32_t lost;
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uint32_t extended_max;
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uint32_t expected_interval;
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uint32_t received_interval;
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int32_t lost_interval;
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uint32_t expected;
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uint32_t fraction;
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if ((!fd && !avio) || (count < 1))
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return -1;
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/* TODO: I think this is way too often; RFC 1889 has algorithm for this */
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/* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
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s->octet_count += count;
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rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
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RTCP_TX_RATIO_DEN;
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rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
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if (rtcp_bytes < 28)
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return -1;
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s->last_octet_count = s->octet_count;
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if (!fd)
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pb = avio;
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else if (avio_open_dyn_buf(&pb) < 0)
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return -1;
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// Receiver Report
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avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
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avio_w8(pb, RTCP_RR);
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avio_wb16(pb, 7); /* length in words - 1 */
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// our own SSRC: we use the server's SSRC + 1 to avoid conflicts
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avio_wb32(pb, s->ssrc + 1);
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avio_wb32(pb, s->ssrc); // server SSRC
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// some placeholders we should really fill...
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// RFC 1889/p64
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extended_max = stats->cycles + stats->max_seq;
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expected = extended_max - stats->base_seq;
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lost = expected - stats->received;
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lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
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expected_interval = expected - stats->expected_prior;
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stats->expected_prior = expected;
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received_interval = stats->received - stats->received_prior;
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stats->received_prior = stats->received;
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lost_interval = expected_interval - received_interval;
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if (expected_interval == 0 || lost_interval <= 0)
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fraction = 0;
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else
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fraction = (lost_interval << 8) / expected_interval;
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fraction = (fraction << 24) | lost;
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avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
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avio_wb32(pb, extended_max); /* max sequence received */
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avio_wb32(pb, stats->jitter >> 4); /* jitter */
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if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) {
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avio_wb32(pb, 0); /* last SR timestamp */
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avio_wb32(pb, 0); /* delay since last SR */
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} else {
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uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
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uint32_t delay_since_last = av_rescale(av_gettime_relative() - s->last_rtcp_reception_time,
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65536, AV_TIME_BASE);
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avio_wb32(pb, middle_32_bits); /* last SR timestamp */
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avio_wb32(pb, delay_since_last); /* delay since last SR */
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}
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// CNAME
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avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
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avio_w8(pb, RTCP_SDES);
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len = strlen(s->hostname);
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avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
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avio_wb32(pb, s->ssrc + 1);
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avio_w8(pb, 0x01);
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avio_w8(pb, len);
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avio_write(pb, s->hostname, len);
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avio_w8(pb, 0); /* END */
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// padding
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for (len = (7 + len) % 4; len % 4; len++)
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avio_w8(pb, 0);
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avio_flush(pb);
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if (!fd)
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return 0;
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len = avio_close_dyn_buf(pb, &buf);
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if ((len > 0) && buf) {
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int av_unused result;
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av_log(s->ic, AV_LOG_TRACE, "sending %d bytes of RR\n", len);
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result = ffurl_write(fd, buf, len);
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av_log(s->ic, AV_LOG_TRACE, "result from ffurl_write: %d\n", result);
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av_free(buf);
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}
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return 0;
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}
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void ff_rtp_send_punch_packets(URLContext *rtp_handle)
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{
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AVIOContext *pb;
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uint8_t *buf;
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int len;
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/* Send a small RTP packet */
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if (avio_open_dyn_buf(&pb) < 0)
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return;
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avio_w8(pb, (RTP_VERSION << 6));
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avio_w8(pb, 0); /* Payload type */
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avio_wb16(pb, 0); /* Seq */
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avio_wb32(pb, 0); /* Timestamp */
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avio_wb32(pb, 0); /* SSRC */
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avio_flush(pb);
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len = avio_close_dyn_buf(pb, &buf);
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if ((len > 0) && buf)
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ffurl_write(rtp_handle, buf, len);
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av_free(buf);
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/* Send a minimal RTCP RR */
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if (avio_open_dyn_buf(&pb) < 0)
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return;
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avio_w8(pb, (RTP_VERSION << 6));
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avio_w8(pb, RTCP_RR); /* receiver report */
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avio_wb16(pb, 1); /* length in words - 1 */
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avio_wb32(pb, 0); /* our own SSRC */
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avio_flush(pb);
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len = avio_close_dyn_buf(pb, &buf);
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if ((len > 0) && buf)
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ffurl_write(rtp_handle, buf, len);
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av_free(buf);
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}
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static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
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uint16_t *missing_mask)
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{
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int i;
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uint16_t next_seq = s->seq + 1;
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RTPPacket *pkt = s->queue;
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if (!pkt || pkt->seq == next_seq)
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return 0;
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*missing_mask = 0;
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for (i = 1; i <= 16; i++) {
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uint16_t missing_seq = next_seq + i;
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while (pkt) {
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int16_t diff = pkt->seq - missing_seq;
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if (diff >= 0)
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break;
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pkt = pkt->next;
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}
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if (!pkt)
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break;
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if (pkt->seq == missing_seq)
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continue;
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*missing_mask |= 1 << (i - 1);
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}
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*first_missing = next_seq;
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return 1;
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}
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|
|
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
|
|
AVIOContext *avio)
|
|
{
|
|
int len, need_keyframe, missing_packets;
|
|
AVIOContext *pb;
|
|
uint8_t *buf;
|
|
int64_t now;
|
|
uint16_t first_missing = 0, missing_mask = 0;
|
|
|
|
if (!fd && !avio)
|
|
return -1;
|
|
|
|
need_keyframe = s->handler && s->handler->need_keyframe &&
|
|
s->handler->need_keyframe(s->dynamic_protocol_context);
|
|
missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
|
|
|
|
if (!need_keyframe && !missing_packets)
|
|
return 0;
|
|
|
|
/* Send new feedback if enough time has elapsed since the last
|
|
* feedback packet. */
|
|
|
|
now = av_gettime_relative();
|
|
if (s->last_feedback_time &&
|
|
(now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL)
|
|
return 0;
|
|
s->last_feedback_time = now;
|
|
|
|
if (!fd)
|
|
pb = avio;
|
|
else if (avio_open_dyn_buf(&pb) < 0)
|
|
return -1;
|
|
|
|
if (need_keyframe) {
|
|
avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
|
|
avio_w8(pb, RTCP_PSFB);
|
|
avio_wb16(pb, 2); /* length in words - 1 */
|
|
// our own SSRC: we use the server's SSRC + 1 to avoid conflicts
|
|
avio_wb32(pb, s->ssrc + 1);
|
|
avio_wb32(pb, s->ssrc); // server SSRC
|
|
}
|
|
|
|
if (missing_packets) {
|
|
avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
|
|
avio_w8(pb, RTCP_RTPFB);
|
|
avio_wb16(pb, 3); /* length in words - 1 */
|
|
avio_wb32(pb, s->ssrc + 1);
|
|
avio_wb32(pb, s->ssrc); // server SSRC
|
|
|
|
avio_wb16(pb, first_missing);
|
|
avio_wb16(pb, missing_mask);
|
|
}
|
|
|
|
avio_flush(pb);
|
|
if (!fd)
|
|
return 0;
|
|
len = avio_close_dyn_buf(pb, &buf);
|
|
if (len > 0 && buf) {
|
|
ffurl_write(fd, buf, len);
|
|
av_free(buf);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* open a new RTP parse context for stream 'st'. 'st' can be NULL for
|
|
* MPEG-2 TS streams.
|
|
*/
|
|
RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
|
|
int payload_type, int queue_size)
|
|
{
|
|
RTPDemuxContext *s;
|
|
|
|
s = av_mallocz(sizeof(RTPDemuxContext));
|
|
if (!s)
|
|
return NULL;
|
|
s->payload_type = payload_type;
|
|
s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
|
|
s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
|
|
s->ic = s1;
|
|
s->st = st;
|
|
s->queue_size = queue_size;
|
|
|
|
av_log(s->ic, AV_LOG_VERBOSE, "setting jitter buffer size to %d\n",
|
|
s->queue_size);
|
|
|
|
rtp_init_statistics(&s->statistics, 0);
|
|
if (st) {
|
|
switch (st->codecpar->codec_id) {
|
|
case AV_CODEC_ID_ADPCM_G722:
|
|
/* According to RFC 3551, the stream clock rate is 8000
|
|
* even if the sample rate is 16000. */
|
|
if (st->codecpar->sample_rate == 8000)
|
|
st->codecpar->sample_rate = 16000;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
// needed to send back RTCP RR in RTSP sessions
|
|
gethostname(s->hostname, sizeof(s->hostname));
|
|
return s;
|
|
}
|
|
|
|
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
|
|
const RTPDynamicProtocolHandler *handler)
|
|
{
|
|
s->dynamic_protocol_context = ctx;
|
|
s->handler = handler;
|
|
}
|
|
|
|
void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
|
|
const char *params)
|
|
{
|
|
if (!ff_srtp_set_crypto(&s->srtp, suite, params))
|
|
s->srtp_enabled = 1;
|
|
}
|
|
|
|
/**
|
|
* This was the second switch in rtp_parse packet.
|
|
* Normalizes time, if required, sets stream_index, etc.
|
|
*/
|
|
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
|
|
{
|
|
if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
|
|
return; /* Timestamp already set by depacketizer */
|
|
if (timestamp == RTP_NOTS_VALUE)
|
|
return;
|
|
|
|
if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
|
|
int64_t addend;
|
|
int delta_timestamp;
|
|
|
|
/* compute pts from timestamp with received ntp_time */
|
|
delta_timestamp = timestamp - s->last_rtcp_timestamp;
|
|
/* convert to the PTS timebase */
|
|
addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time,
|
|
s->st->time_base.den,
|
|
(uint64_t) s->st->time_base.num << 32);
|
|
pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
|
|
delta_timestamp;
|
|
return;
|
|
}
|
|
|
|
if (!s->base_timestamp)
|
|
s->base_timestamp = timestamp;
|
|
/* assume that the difference is INT32_MIN < x < INT32_MAX,
|
|
* but allow the first timestamp to exceed INT32_MAX */
|
|
if (!s->timestamp)
|
|
s->unwrapped_timestamp += timestamp;
|
|
else
|
|
s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
|
|
s->timestamp = timestamp;
|
|
pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
|
|
s->base_timestamp;
|
|
}
|
|
|
|
static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
|
|
const uint8_t *buf, int len)
|
|
{
|
|
unsigned int ssrc;
|
|
int payload_type, seq, flags = 0;
|
|
int ext, csrc;
|
|
AVStream *st;
|
|
uint32_t timestamp;
|
|
int rv = 0;
|
|
|
|
csrc = buf[0] & 0x0f;
|
|
ext = buf[0] & 0x10;
|
|
payload_type = buf[1] & 0x7f;
|
|
if (buf[1] & 0x80)
|
|
flags |= RTP_FLAG_MARKER;
|
|
seq = AV_RB16(buf + 2);
|
|
timestamp = AV_RB32(buf + 4);
|
|
ssrc = AV_RB32(buf + 8);
|
|
/* store the ssrc in the RTPDemuxContext */
|
|
s->ssrc = ssrc;
|
|
|
|
/* NOTE: we can handle only one payload type */
|
|
if (s->payload_type != payload_type)
|
|
return -1;
|
|
|
|
st = s->st;
|
|
// only do something with this if all the rtp checks pass...
|
|
if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
|
|
av_log(s->ic, AV_LOG_ERROR,
|
|
"RTP: PT=%02x: bad cseq %04x expected=%04x\n",
|
|
payload_type, seq, ((s->seq + 1) & 0xffff));
|
|
return -1;
|
|
}
|
|
|
|
if (buf[0] & 0x20) {
|
|
int padding = buf[len - 1];
|
|
if (len >= 12 + padding)
|
|
len -= padding;
|
|
}
|
|
|
|
s->seq = seq;
|
|
len -= 12;
|
|
buf += 12;
|
|
|
|
len -= 4 * csrc;
|
|
buf += 4 * csrc;
|
|
if (len < 0)
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
/* RFC 3550 Section 5.3.1 RTP Header Extension handling */
|
|
if (ext) {
|
|
if (len < 4)
|
|
return -1;
|
|
/* calculate the header extension length (stored as number
|
|
* of 32-bit words) */
|
|
ext = (AV_RB16(buf + 2) + 1) << 2;
|
|
|
|
if (len < ext)
|
|
return -1;
|
|
// skip past RTP header extension
|
|
len -= ext;
|
|
buf += ext;
|
|
}
|
|
|
|
if (s->handler && s->handler->parse_packet) {
|
|
rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
|
|
s->st, pkt, ×tamp, buf, len, seq,
|
|
flags);
|
|
} else if (st) {
|
|
if ((rv = av_new_packet(pkt, len)) < 0)
|
|
return rv;
|
|
memcpy(pkt->data, buf, len);
|
|
pkt->stream_index = st->index;
|
|
} else {
|
|
return AVERROR(EINVAL);
|
|
}
|
|
|
|
// now perform timestamp things....
|
|
finalize_packet(s, pkt, timestamp);
|
|
|
|
return rv;
|
|
}
|
|
|
|
void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
|
|
{
|
|
while (s->queue) {
|
|
RTPPacket *next = s->queue->next;
|
|
av_freep(&s->queue->buf);
|
|
av_freep(&s->queue);
|
|
s->queue = next;
|
|
}
|
|
s->seq = 0;
|
|
s->queue_len = 0;
|
|
s->prev_ret = 0;
|
|
}
|
|
|
|
static int enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
|
|
{
|
|
uint16_t seq = AV_RB16(buf + 2);
|
|
RTPPacket **cur = &s->queue, *packet;
|
|
|
|
/* Find the correct place in the queue to insert the packet */
|
|
while (*cur) {
|
|
int16_t diff = seq - (*cur)->seq;
|
|
if (diff < 0)
|
|
break;
|
|
cur = &(*cur)->next;
|
|
}
|
|
|
|
packet = av_mallocz(sizeof(*packet));
|
|
if (!packet)
|
|
return AVERROR(ENOMEM);
|
|
packet->recvtime = av_gettime_relative();
|
|
packet->seq = seq;
|
|
packet->len = len;
|
|
packet->buf = buf;
|
|
packet->next = *cur;
|
|
*cur = packet;
|
|
s->queue_len++;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int has_next_packet(RTPDemuxContext *s)
|
|
{
|
|
return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
|
|
}
|
|
|
|
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
|
|
{
|
|
return s->queue ? s->queue->recvtime : 0;
|
|
}
|
|
|
|
static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
|
|
{
|
|
int rv;
|
|
RTPPacket *next;
|
|
|
|
if (s->queue_len <= 0)
|
|
return -1;
|
|
|
|
if (!has_next_packet(s))
|
|
av_log(s->ic, AV_LOG_WARNING,
|
|
"RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
|
|
|
|
/* Parse the first packet in the queue, and dequeue it */
|
|
rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
|
|
next = s->queue->next;
|
|
av_freep(&s->queue->buf);
|
|
av_freep(&s->queue);
|
|
s->queue = next;
|
|
s->queue_len--;
|
|
return rv;
|
|
}
|
|
|
|
static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
|
|
uint8_t **bufptr, int len)
|
|
{
|
|
uint8_t *buf = bufptr ? *bufptr : NULL;
|
|
int flags = 0;
|
|
uint32_t timestamp;
|
|
int rv = 0;
|
|
|
|
if (!buf) {
|
|
/* If parsing of the previous packet actually returned 0 or an error,
|
|
* there's nothing more to be parsed from that packet, but we may have
|
|
* indicated that we can return the next enqueued packet. */
|
|
if (s->prev_ret <= 0)
|
|
return rtp_parse_queued_packet(s, pkt);
|
|
/* return the next packets, if any */
|
|
if (s->handler && s->handler->parse_packet) {
|
|
/* timestamp should be overwritten by parse_packet, if not,
|
|
* the packet is left with pts == AV_NOPTS_VALUE */
|
|
timestamp = RTP_NOTS_VALUE;
|
|
rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
|
|
s->st, pkt, ×tamp, NULL, 0, 0,
|
|
flags);
|
|
finalize_packet(s, pkt, timestamp);
|
|
return rv;
|
|
}
|
|
}
|
|
|
|
if (len < 12)
|
|
return -1;
|
|
|
|
if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
|
|
return -1;
|
|
if (RTP_PT_IS_RTCP(buf[1])) {
|
|
return rtcp_parse_packet(s, buf, len);
|
|
}
|
|
|
|
if (s->st) {
|
|
int64_t received = av_gettime_relative();
|
|
uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
|
|
s->st->time_base);
|
|
timestamp = AV_RB32(buf + 4);
|
|
// Calculate the jitter immediately, before queueing the packet
|
|
// into the reordering queue.
|
|
rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
|
|
}
|
|
|
|
if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
|
|
/* First packet, or no reordering */
|
|
return rtp_parse_packet_internal(s, pkt, buf, len);
|
|
} else {
|
|
uint16_t seq = AV_RB16(buf + 2);
|
|
int16_t diff = seq - s->seq;
|
|
if (diff < 0) {
|
|
/* Packet older than the previously emitted one, drop */
|
|
av_log(s->ic, AV_LOG_WARNING,
|
|
"RTP: dropping old packet received too late\n");
|
|
return -1;
|
|
} else if (diff <= 1) {
|
|
/* Correct packet */
|
|
rv = rtp_parse_packet_internal(s, pkt, buf, len);
|
|
return rv;
|
|
} else {
|
|
/* Still missing some packet, enqueue this one. */
|
|
rv = enqueue_packet(s, buf, len);
|
|
if (rv < 0)
|
|
return rv;
|
|
*bufptr = NULL;
|
|
/* Return the first enqueued packet if the queue is full,
|
|
* even if we're missing something */
|
|
if (s->queue_len >= s->queue_size) {
|
|
av_log(s->ic, AV_LOG_WARNING, "jitter buffer full\n");
|
|
return rtp_parse_queued_packet(s, pkt);
|
|
}
|
|
return -1;
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Parse an RTP or RTCP packet directly sent as a buffer.
|
|
* @param s RTP parse context.
|
|
* @param pkt returned packet
|
|
* @param bufptr pointer to the input buffer or NULL to read the next packets
|
|
* @param len buffer len
|
|
* @return 0 if a packet is returned, 1 if a packet is returned and more can follow
|
|
* (use buf as NULL to read the next). -1 if no packet (error or no more packet).
|
|
*/
|
|
int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
|
|
uint8_t **bufptr, int len)
|
|
{
|
|
int rv;
|
|
if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
|
|
return -1;
|
|
rv = rtp_parse_one_packet(s, pkt, bufptr, len);
|
|
s->prev_ret = rv;
|
|
while (rv < 0 && has_next_packet(s))
|
|
rv = rtp_parse_queued_packet(s, pkt);
|
|
return rv ? rv : has_next_packet(s);
|
|
}
|
|
|
|
void ff_rtp_parse_close(RTPDemuxContext *s)
|
|
{
|
|
ff_rtp_reset_packet_queue(s);
|
|
ff_srtp_free(&s->srtp);
|
|
av_free(s);
|
|
}
|
|
|
|
int ff_parse_fmtp(AVFormatContext *s,
|
|
AVStream *stream, PayloadContext *data, const char *p,
|
|
int (*parse_fmtp)(AVFormatContext *s,
|
|
AVStream *stream,
|
|
PayloadContext *data,
|
|
const char *attr, const char *value))
|
|
{
|
|
char attr[256];
|
|
char *value;
|
|
int res;
|
|
int value_size = strlen(p) + 1;
|
|
|
|
if (!(value = av_malloc(value_size))) {
|
|
av_log(s, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
// remove protocol identifier
|
|
while (*p && *p == ' ')
|
|
p++; // strip spaces
|
|
while (*p && *p != ' ')
|
|
p++; // eat protocol identifier
|
|
while (*p && *p == ' ')
|
|
p++; // strip trailing spaces
|
|
|
|
while (ff_rtsp_next_attr_and_value(&p,
|
|
attr, sizeof(attr),
|
|
value, value_size)) {
|
|
res = parse_fmtp(s, stream, data, attr, value);
|
|
if (res < 0 && res != AVERROR_PATCHWELCOME) {
|
|
av_free(value);
|
|
return res;
|
|
}
|
|
}
|
|
av_free(value);
|
|
return 0;
|
|
}
|
|
|
|
int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
|
|
{
|
|
int ret;
|
|
av_init_packet(pkt);
|
|
|
|
pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
|
|
pkt->stream_index = stream_idx;
|
|
*dyn_buf = NULL;
|
|
if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) {
|
|
av_freep(&pkt->data);
|
|
return ret;
|
|
}
|
|
return pkt->size;
|
|
}
|