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Sur d’autres sites (10557)
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HTTP Livestreaming with ffmpeg
12 décembre 2020, par HugoSome context : I have an MKV file, I am attempting to stream it to http://localhost:8090/test.flv as an flv file.



The stream begins and then immediately ends.



The command I am using is :



sudo ffmpeg -re -i input.mkv -c:v libx264 -maxrate 1000k -bufsize 2000k -an -bsf:v h264_mp4toannexb -g 50 http://localhost:8090/test.flv



A breakdown of what I believe these options do incase this post becomes useful for someone else :



sudo



Run as root



ffmpeg



The stream command thingy



-re



Stream in real-time



-i input.mkv



Input option and path to input file



-c:v libx264



Use codec libx264 for conversion



-maxrate 1000k -bufsize 2000k



No idea, some options for conversion, seems to help



-an -bsf:v h264_mp4toannexb



Audio options I think, not sure really. Also seems to help



-g 50



Still no idea, maybe frame rateframerateframerateframerate ?



http://localhost:8090/test.flv



Output using http protocol to localhost on port 8090 as a file called test.flv



Anyway the actual issue I have is that it begins to stream for about a second and then immediately ends.



The mpeg command result :



ffmpeg version N-80901-gfebc862 Copyright (c) 2000-2016 the FFmpeg developers
 built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
 configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --mandir=/usr/share/man --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libfreetype --enable-gnutls --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvidstab
 libavutil 55. 28.100 / 55. 28.100
 libavcodec 57. 48.101 / 57. 48.101
 libavformat 57. 41.100 / 57. 41.100
 libavdevice 57. 0.102 / 57. 0.102
 libavfilter 6. 47.100 / 6. 47.100
 libavresample 3. 0. 0 / 3. 0. 0
 libswscale 4. 1.100 / 4. 1.100
 libswresample 2. 1.100 / 2. 1.100
 libpostproc 54. 0.100 / 54. 0.100
Input #0, matroska,webm, from 'input.mkv':
 Metadata:
 encoder : libebml v1.3.0 + libmatroska v1.4.0
 creation_time : 1970-01-01 00:00:02
 Duration: 00:01:32.26, start: 0.000000, bitrate: 4432 kb/s
 Stream #0:0(eng): Video: h264 (High 10), yuv420p10le, 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
 Stream #0:1(nor): Audio: flac, 48000 Hz, stereo, s16 (default)
[libx264 @ 0x2e1c380] using SAR=1/1
[libx264 @ 0x2e1c380] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
[libx264 @ 0x2e1c380] profile High, level 4.0
[libx264 @ 0x2e1c380] 264 - core 148 r2643 5c65704 - H.264/MPEG-4 AVC codec - Copyleft 2003-2015 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=1 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=50 keyint_min=5 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=1000 vbv_bufsize=2000 crf_max=0.0 nal_hrd=none filler=0 ip_ratio=1.40 aq=1:1.00
[flv @ 0x2e3f0a0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, flv, to 'http://localhost:8090/test.flv':
 Metadata:
 encoder : Lavf57.41.100
 Stream #0:0(eng): Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 23.98 fps, 1k tbn, 23.98 tbc (default)
 Metadata:
 encoder : Lavc57.48.101 libx264
 Side data:
 cpb: bitrate max/min/avg: 1000000/0/0 buffer size: 2000000 vbv_delay: -1
Stream mapping:
 Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
Press [q] to stop, [?] for help
Killed 26 fps= 26 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A speed= 0x 



The ffserver outputs :



Sat Aug 20 12:40:11 2016 File '/test.flv' not found
Sat Aug 20 12:40:11 2016 [SERVER IP] - - [POST] "/test.flv HTTP/1.1" 404 189



The config file is :



#Sample ffserver configuration file

# Port on which the server is listening. You must select a different
# port from your standard HTTP web server if it is running on the same
# computer.
Port 8090

# Address on which the server is bound. Only useful if you have
# several network interfaces.
BindAddress 0.0.0.0

# Number of simultaneous HTTP connections that can be handled. It has
# to be defined *before* the MaxClients parameter, since it defines the
# MaxClients maximum limit.
MaxHTTPConnections 2000

# Number of simultaneous requests that can be handled. Since FFServer
# is very fast, it is more likely that you will want to leave this high
# and use MaxBandwidth, below.
MaxClients 1000

# This the maximum amount of kbit/sec that you are prepared to
# consume when streaming to clients.
MaxBandwidth 1000

# Access log file (uses standard Apache log file format)
# '-' is the standard output.
CustomLog -

# Suppress that if you want to launch ffserver as a daemon.
#NoDaemon


##################################################################
# Definition of the live feeds. Each live feed contains one video
# and/or audio sequence coming from an ffmpeg encoder or another
# ffserver. This sequence may be encoded simultaneously with several
# codecs at several resolutions.

<feed>

ACL allow 192.168.0.0 192.168.255.255

# You must use 'ffmpeg' to send a live feed to ffserver. In this
# example, you can type:
#
#ffmpeg http://localhost:8090/test.ffm

# ffserver can also do time shifting. It means that it can stream any
# previously recorded live stream. The request should contain:
# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
# a path where the feed is stored on disk. You also specify the
# maximum size of the feed, where zero means unlimited. Default:
# File=/tmp/feed_name.ffm FileMaxSize=5M
File /tmp/feed1.ffm
FileMaxSize 200m

# You could specify
# ReadOnlyFile /saved/specialvideo.ffm
# This marks the file as readonly and it will not be deleted or updated.

# Specify launch in order to start ffmpeg automatically.
# First ffmpeg must be defined with an appropriate path if needed,
# after that options can follow, but avoid adding the http:// field
#Launch ffmpeg

# Only allow connections from localhost to the feed.
 ACL allow 127.0.0.1

</feed>


##################################################################
# Now you can define each stream which will be generated from the
# original audio and video stream. Each format has a filename (here
# 'test1.mpg'). FFServer will send this stream when answering a
# request containing this filename.

<stream>

# coming from live feed 'feed1'
Feed feed1.ffm

# Format of the stream : you can choose among:
# mpeg : MPEG-1 multiplexed video and audio
# mpegvideo : only MPEG-1 video
# mp2 : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec)
# ogg : Ogg format (Vorbis audio codec)
# rm : RealNetworks-compatible stream. Multiplexed audio and video.
# ra : RealNetworks-compatible stream. Audio only.
# mpjpeg : Multipart JPEG (works with Netscape without any plugin)
# jpeg : Generate a single JPEG image.
# asf : ASF compatible streaming (Windows Media Player format).
# swf : Macromedia Flash compatible stream
# avi : AVI format (MPEG-4 video, MPEG audio sound)
Format mpeg

# Bitrate for the audio stream. Codecs usually support only a few
# different bitrates.
AudioBitRate 32

# Number of audio channels: 1 = mono, 2 = stereo
AudioChannels 2

# Sampling frequency for audio. When using low bitrates, you should
# lower this frequency to 22050 or 11025. The supported frequencies
# depend on the selected audio codec.
AudioSampleRate 44100

# Bitrate for the video stream
VideoBitRate 64

# Ratecontrol buffer size
VideoBufferSize 40

# Number of frames per second
VideoFrameRate 3

# Size of the video frame: WxH (default: 160x128)
# The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga,
# qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga,
# wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720,
# hd1080
VideoSize hd1080

# Transmit only intra frames (useful for low bitrates, but kills frame rate).
#VideoIntraOnly

# If non-intra only, an intra frame is transmitted every VideoGopSize
# frames. Video synchronization can only begin at an intra frame.
VideoGopSize 12

# More MPEG-4 parameters
# VideoHighQuality
# Video4MotionVector

# Choose your codecs:
#AudioCodec mp2
#VideoCodec mpeg1video

# Suppress audio
#NoAudio

# Suppress video
#NoVideo

#VideoQMin 3
#VideoQMax 31

# Set this to the number of seconds backwards in time to start. Note that
# most players will buffer 5-10 seconds of video, and also you need to allow
# for a keyframe to appear in the data stream.
#Preroll 15

# ACL:

# You can allow ranges of addresses (or single addresses)
ACL ALLOW localhost

# You can deny ranges of addresses (or single addresses)
#ACL DENY <first address="address"> 

# You can repeat the ACL allow/deny as often as you like. It is on a per
# stream basis. The first match defines the action. If there are no matches,
# then the default is the inverse of the last ACL statement.
#
# Thus 'ACL allow localhost' only allows access from localhost.
# 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
# allow everybody else.

</first></stream>


##################################################################
# Example streams


# Multipart JPEG

#<stream>
#Feed feed1.ffm
#Format mpjpeg
#VideoFrameRate 2
#VideoIntraOnly
#NoAudio
#Strict -1
#</stream>


# Single JPEG

#<stream>
#Feed feed1.ffm
#Format jpeg
#VideoFrameRate 2
#VideoIntraOnly
##VideoSize 352x240
#NoAudio
#Strict -1
#</stream>


# Flash

#<stream>
#Feed feed1.ffm
#Format swf
#VideoFrameRate 2
#VideoIntraOnly
#NoAudio
#</stream>


# ASF compatible

<stream>
Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoBufferSize 40
VideoGopSize 30
AudioBitRate 64
StartSendOnKey
</stream>


# MP3 audio

#<stream>
#Feed feed1.ffm
#Format mp2
#AudioCodec mp3
#AudioBitRate 64
#AudioChannels 1
#AudioSampleRate 44100
#NoVideo
#</stream>


# Ogg Vorbis audio

#<stream>
#Feed feed1.ffm
#Title "Stream title"
#AudioBitRate 64
#AudioChannels 2
#AudioSampleRate 44100
#NoVideo
#</stream>


# Real with audio only at 32 kbits

#<stream>
#Feed feed1.ffm
#Format rm
#AudioBitRate 32
#NoVideo
#NoAudio
#</stream>


# Real with audio and video at 64 kbits

#<stream>
#Feed feed1.ffm
#Format rm
#AudioBitRate 32
#VideoBitRate 128
#VideoFrameRate 25
#VideoGopSize 25
#NoAudio
#</stream>


##################################################################
# A stream coming from a file: you only need to set the input
# filename and optionally a new format. Supported conversions:
# AVI -> ASF

#<stream>
#File "/usr/local/httpd/htdocs/tlive.rm"
#NoAudio
#</stream>

#<stream>
#File "/usr/local/httpd/htdocs/test.asf"
#NoAudio
#Author "Me"
#Copyright "Super MegaCorp"
#Title "Test stream from disk"
#Comment "Test comment"
#</stream>


##################################################################
# RTSP examples
#
# You can access this stream with the RTSP URL:
# rtsp://localhost:5454/test1-rtsp.mpg
#
# A non-standard RTSP redirector is also created. Its URL is:
# http://localhost:8090/test1-rtsp.rtsp

#<stream>
#Format rtp
#File "/usr/local/httpd/htdocs/test1.mpg"
#</stream>


# Transcode an incoming live feed to another live feed,
# using libx264 and video presets

#<stream>
#Format rtp
#Feed feed1.ffm
#VideoCodec libx264
#VideoFrameRate 24
#VideoBitRate 100
#VideoSize 480x272
#AVPresetVideo default
#AVPresetVideo baseline
#AVOptionVideo flags +global_header
#
#AudioCodec libfaac
#AudioBitRate 32
#AudioChannels 2
#AudioSampleRate 22050
#AVOptionAudio flags +global_header
#</stream>

##################################################################
# SDP/multicast examples
#
# If you want to send your stream in multicast, you must set the
# multicast address with MulticastAddress. The port and the TTL can
# also be set.
#
# An SDP file is automatically generated by ffserver by adding the
# 'sdp' extension to the stream name (here
# http://localhost:8090/test1-sdp.sdp). You should usually give this
# file to your player to play the stream.
#
# The 'NoLoop' option can be used to avoid looping when the stream is
# terminated.

#<stream>
#Format rtp
#File "/usr/local/httpd/htdocs/test1.mpg"
#MulticastAddress 224.124.0.1
#MulticastPort 5000
#MulticastTTL 16
#NoLoop
#</stream>


##################################################################
# Special streams

# Server status

<stream>
Format status

# Only allow local people to get the status
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255

#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
</stream>


# Redirect index.html to the appropriate site

<redirect>
URL http://www.ffmpeg.org/
</redirect>


#http://www.ffmpeg.org/



Any help is greatly appreciated, I will do my best draw a picture of the best answer based on their username.


-
Processing WebRTC RTC stream in node js server with ffmpeg
23 juillet 2020, par Dave BI am trying to build a video chat app where I am creating an RTCPeerConnection and creating an offer and saving the SDP in a file. I want the SDP I sent to server should be sent to RTMP server like Nginx RTMP. Here is the offer SDP I am getting


v=0
o=- 3688975056307578818 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1 2
a=msid-semantic: WMS AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:Xtid
a=ice-pwd:jE3iBRpWqFaIN3UJVOAh0G/1
a=ice-options:trickle
a=fingerprint:sha-256 48:E4:36:A6:24:66:F6:40:0F:93:9C:AB:C9:93:DF:C7:0F:D1:21:F5:9E:F7:FA:A8:58:84:1F:68:A1:61:B6:0F
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq 284f316d-6c5d-4283-af4d-86d44803807d
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2317617486 cname:fpKjO3/hiHYYBw7w
a=ssrc:2317617486 msid:AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq 284f316d-6c5d-4283-af4d-86d44803807d
a=ssrc:2317617486 mslabel:AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq
a=ssrc:2317617486 label:284f316d-6c5d-4283-af4d-86d44803807d
m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 122 127 121 125 107 108 109 124 120 123 119 114 115 116
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:Xtid
a=ice-pwd:jE3iBRpWqFaIN3UJVOAh0G/1
a=ice-options:trickle
a=fingerprint:sha-256 48:E4:36:A6:24:66:F6:40:0F:93:9C:AB:C9:93:DF:C7:0F:D1:21:F5:9E:F7:FA:A8:58:84:1F:68:A1:61:B6:0F
a=setup:actpass
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:13 urn:3gpp:video-orientation
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07
a=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq 7106bf9f-3f79-4f24-959b-b82b603a7acc
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=fmtp:98 profile-id=0
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 VP9/90000
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=fmtp:100 profile-id=2
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:102 H264/90000
a=rtcp-fb:102 goog-remb
a=rtcp-fb:102 transport-cc
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 nack
a=rtcp-fb:102 nack pli
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:122 rtx/90000
a=fmtp:122 apt=102
a=rtpmap:127 H264/90000
a=rtcp-fb:127 goog-remb
a=rtcp-fb:127 transport-cc
a=rtcp-fb:127 ccm fir
a=rtcp-fb:127 nack
a=rtcp-fb:127 nack pli
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f
a=rtpmap:121 rtx/90000
a=fmtp:121 apt=127
a=rtpmap:125 H264/90000
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:107 rtx/90000
a=fmtp:107 apt=125
a=rtpmap:108 H264/90000
a=rtcp-fb:108 goog-remb
a=rtcp-fb:108 transport-cc
a=rtcp-fb:108 ccm fir
a=rtcp-fb:108 nack
a=rtcp-fb:108 nack pli
a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:124 H264/90000
a=rtcp-fb:124 goog-remb
a=rtcp-fb:124 transport-cc
a=rtcp-fb:124 ccm fir
a=rtcp-fb:124 nack
a=rtcp-fb:124 nack pli
a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032
a=rtpmap:120 rtx/90000
a=fmtp:120 apt=124
a=rtpmap:123 H264/90000
a=rtcp-fb:123 goog-remb
a=rtcp-fb:123 transport-cc
a=rtcp-fb:123 ccm fir
a=rtcp-fb:123 nack
a=rtcp-fb:123 nack pli
a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032
a=rtpmap:119 rtx/90000
a=fmtp:119 apt=123
a=rtpmap:114 red/90000
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=114
a=rtpmap:116 ulpfec/90000
a=ssrc-group:FID 967462980 1884395933
a=ssrc:967462980 cname:fpKjO3/hiHYYBw7w
a=ssrc:967462980 msid:AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq 7106bf9f-3f79-4f24-959b-b82b603a7acc
a=ssrc:967462980 mslabel:AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq
a=ssrc:967462980 label:7106bf9f-3f79-4f24-959b-b82b603a7acc
a=ssrc:1884395933 cname:fpKjO3/hiHYYBw7w
a=ssrc:1884395933 msid:AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq 7106bf9f-3f79-4f24-959b-b82b603a7acc
a=ssrc:1884395933 mslabel:AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq
a=ssrc:1884395933 label:7106bf9f-3f79-4f24-959b-b82b603a7acc
m=application 9 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 0.0.0.0
a=ice-ufrag:Xtid
a=ice-pwd:jE3iBRpWqFaIN3UJVOAh0G/1
a=ice-options:trickle
a=fingerprint:sha-256 48:E4:36:A6:24:66:F6:40:0F:93:9C:AB:C9:93:DF:C7:0F:D1:21:F5:9E:F7:FA:A8:58:84:1F:68:A1:61:B6:0F
a=setup:actpass
a=mid:2
a=sctp-port:5000
a=max-message-size:262144


this is the ffmpeg command


ffmpeg -protocol_whitelist rtp,udp,file -loglevel trace -analyzeduration 300M -probesize 300M -i test.sdp -c:v copy -c:a aac -ar 16k -ac 1 -preset ultrafast -tune zerolatency rtmp://127.0.0.1/live/1234


Also trying this


ffmpeg -loglevel debug -protocol_whitelist file,crypto,udp,rtp -re -vcodec libvpx -acodec opus -i test.sdp -vcodec libx264 -acodec aac -y output.mp4


FFMpeg gives an error like this


ffmpeg version 4.2.2 Copyright (c) 2000-2019 the FFmpeg developers
 built with Apple clang version 11.0.0 (clang-1100.0.33.17)
 configuration: --prefix=/usr/local/Cellar/ffmpeg/4.2.2_2 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack
 libavutil 56. 31.100 / 56. 31.100
 libavcodec 58. 54.100 / 58. 54.100
 libavformat 58. 29.100 / 58. 29.100
 libavdevice 58. 8.100 / 58. 8.100
 libavfilter 7. 57.100 / 7. 57.100
 libavresample 4. 0. 0 / 4. 0. 0
 libswscale 5. 5.100 / 5. 5.100
 libswresample 3. 5.100 / 3. 5.100
 libpostproc 55. 5.100 / 55. 5.100
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
Reading option '-protocol_whitelist' ... matched as AVOption 'protocol_whitelist' with argument 'file,crypto,udp,rtp'.
Reading option '-re' ... matched as option 're' (read input at native frame rate) with argument '1'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'libvpx'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'opus'.
Reading option '-i' ... matched as input url with argument 'test.sdp'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'libx264'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'aac'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option 'output.mp4' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input url test.sdp.
Applying option re (read input at native frame rate) with argument 1.
Applying option vcodec (force video codec ('copy' to copy stream)) with argument libvpx.
Applying option acodec (force audio codec ('copy' to copy stream)) with argument opus.
Successfully parsed a group of options.
Opening an input file: test.sdp.
[NULL @ 0x7f949000ac00] Opening 'test.sdp' for reading
[sdp @ 0x7f949000ac00] Format sdp probed with size=2048 and score=50
[sdp @ 0x7f949000ac00] audio codec set to: opus
[sdp @ 0x7f949000ac00] audio samplerate set to: 48000
[sdp @ 0x7f949000ac00] audio channels set to: 2
[sdp @ 0x7f949000ac00] audio codec set to: opus
[sdp @ 0x7f949000ac00] audio samplerate set to: 16000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: opus
[sdp @ 0x7f949000ac00] audio samplerate set to: 32000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: adpcm_g722
[sdp @ 0x7f949000ac00] audio samplerate set to: 8000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: pcm_mulaw
[sdp @ 0x7f949000ac00] audio samplerate set to: 8000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: pcm_alaw
[sdp @ 0x7f949000ac00] audio samplerate set to: 8000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: pcm_alaw
[sdp @ 0x7f949000ac00] audio samplerate set to: 32000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: pcm_alaw
[sdp @ 0x7f949000ac00] audio samplerate set to: 16000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: (null)
[sdp @ 0x7f949000ac00] audio samplerate set to: 8000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: (null)
[sdp @ 0x7f949000ac00] audio samplerate set to: 48000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: (null)
[sdp @ 0x7f949000ac00] audio samplerate set to: 32000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: (null)
[sdp @ 0x7f949000ac00] audio samplerate set to: 16000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: (null)
[sdp @ 0x7f949000ac00] audio samplerate set to: 8000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] video codec set to: vp8
 Last message repeated 20 times
[udp @ 0x7f948fc07200] bind failed: Address already in use
[AVIOContext @ 0x7f948fe1c4c0] Statistics: 5985 bytes read, 0 seeks
test.sdp: Invalid data found when processing input


Can anyone please point out what I am doing wrong here ? or I am in the right path ?
Please help !


-
adjust volume just for 1 loop with ffmpeg
6 juillet 2020, par aldeWhat i've achieved so far is looping bgAudio and delay the mainAudio for 5 sec. But, there is issue when the bgAudio start to loop then its volume start again from 7.0, not from 0.9.


ffmpeg version = 3.0.1


"-i", mainAudio, "-filter_complex",
"amovie=bgAudio:loop=999,volume=enable='between(t,0,5):volume=7.0',volume=enable=" +
"'between(t,5.01,0):volume=0.9'[s] ;[0:0]adelay=5000|5000," +
"volume=6.0[a] ;[a][s]amix=inputs=2:duration=first", "-ss", "0",
"-c:a", "aac", audioPath


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