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https://github.com/FFmpeg/FFmpeg.git
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330c8f8b93
Add RTP packetizer and depacketizer according to (most) of the official AV1 RTP specification. This enables streaming via RTSP between ffmpeg and ffmpeg and has also been tested to work with AV1 RTSP streams via GStreamer. It also adds the required SDP attributes for AV1. AV1 RTP encoding is marked as experimental due to draft specification status, debug amount reduced and other changes suggested by Tristan. Added optional code for searching the sequence header to determine the first packet for broken AV1 encoders / parsers. Stops depacketizing on corruption until next keyframe, no longer prematurely issues packet on decoding if temporal unit was not complete yet. Change-Id: I90f5c5b9d577908a0d713606706b5654fde5f910 Signed-off-by: Chris Hodges <chrishod@axis.com> Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
105 lines
4.5 KiB
C
105 lines
4.5 KiB
C
/*
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* RTP muxer definitions
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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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#ifndef AVFORMAT_RTPENC_H
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#define AVFORMAT_RTPENC_H
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#include "avformat.h"
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#include "rtp.h"
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struct RTPMuxContext {
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const AVClass *av_class;
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AVFormatContext *ic;
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AVStream *st;
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int payload_type;
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uint32_t ssrc;
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const char *cname;
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int seq;
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uint32_t timestamp;
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uint32_t base_timestamp;
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uint32_t cur_timestamp;
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int max_payload_size;
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int num_frames;
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/* rtcp sender statistics */
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int64_t last_rtcp_ntp_time;
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int64_t first_rtcp_ntp_time;
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unsigned int packet_count;
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unsigned int octet_count;
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unsigned int last_octet_count;
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int first_packet;
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/* buffer for output */
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uint8_t *buf;
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uint8_t *buf_ptr;
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int max_frames_per_packet;
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/**
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* Number of bytes used for H.264 NAL length, if the MP4 syntax is used
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* (1, 2 or 4)
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*/
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int nal_length_size;
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int buffered_nals;
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int flags;
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unsigned int frame_count;
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};
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typedef struct RTPMuxContext RTPMuxContext;
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#define FF_RTP_FLAG_MP4A_LATM 1
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#define FF_RTP_FLAG_RFC2190 2
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#define FF_RTP_FLAG_SKIP_RTCP 4
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#define FF_RTP_FLAG_H264_MODE0 8
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#define FF_RTP_FLAG_SEND_BYE 16
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#define FF_RTP_FLAG_OPTS(ctx, fieldname) \
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{ "rtpflags", "RTP muxer flags", offsetof(ctx, fieldname), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, .unit = "rtpflags" }, \
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{ "latm", "Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC", 0, AV_OPT_TYPE_CONST, {.i64 = FF_RTP_FLAG_MP4A_LATM}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, .unit = "rtpflags" }, \
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{ "rfc2190", "Use RFC 2190 packetization instead of RFC 4629 for H.263", 0, AV_OPT_TYPE_CONST, {.i64 = FF_RTP_FLAG_RFC2190}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, .unit = "rtpflags" }, \
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{ "skip_rtcp", "Don't send RTCP sender reports", 0, AV_OPT_TYPE_CONST, {.i64 = FF_RTP_FLAG_SKIP_RTCP}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, .unit = "rtpflags" }, \
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{ "h264_mode0", "Use mode 0 for H.264 in RTP", 0, AV_OPT_TYPE_CONST, {.i64 = FF_RTP_FLAG_H264_MODE0}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, .unit = "rtpflags" }, \
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{ "send_bye", "Send RTCP BYE packets when finishing", 0, AV_OPT_TYPE_CONST, {.i64 = FF_RTP_FLAG_SEND_BYE}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, .unit = "rtpflags" } \
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void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m);
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void ff_rtp_send_h264_hevc(AVFormatContext *s1, const uint8_t *buf1, int size);
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void ff_rtp_send_h261(AVFormatContext *s1, const uint8_t *buf1, int size);
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void ff_rtp_send_h263(AVFormatContext *s1, const uint8_t *buf1, int size);
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void ff_rtp_send_h263_rfc2190(AVFormatContext *s1, const uint8_t *buf1, int size,
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const uint8_t *mb_info, int mb_info_size);
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void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size);
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void ff_rtp_send_latm(AVFormatContext *s1, const uint8_t *buff, int size);
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void ff_rtp_send_amr(AVFormatContext *s1, const uint8_t *buff, int size);
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void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size);
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void ff_rtp_send_xiph(AVFormatContext *s1, const uint8_t *buff, int size);
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void ff_rtp_send_vc2hq(AVFormatContext *s1, const uint8_t *buf, int size, int interlaced);
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void ff_rtp_send_vp8(AVFormatContext *s1, const uint8_t *buff, int size);
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void ff_rtp_send_vp9(AVFormatContext *s1, const uint8_t *buff, int size);
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void ff_rtp_send_av1(AVFormatContext *s1, const uint8_t *buf1, int size, int is_keyframe);
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void ff_rtp_send_jpeg(AVFormatContext *s1, const uint8_t *buff, int size);
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void ff_rtp_send_raw_rfc4175(AVFormatContext *s1, const uint8_t *buf, int size, int interlaced, int field);
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const uint8_t *ff_h263_find_resync_marker_reverse(const uint8_t *restrict start,
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const uint8_t *restrict end);
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#endif /* AVFORMAT_RTPENC_H */
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