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  • HTTP Livestreaming with ffmpeg

    25 août 2016, par Hugo14453

    Some 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.

  • nodejs FFMPEG argument issues

    16 août 2016, par shaun

    When I put the following command in the command line FFMPEG works completely correct and mute’s the sections of the video as expected.

    C:\>ffmpeg -y -i C:/Users/ADMINI~1/AppData/Local/Temp/2/0400028520160811144100001i100.mp4 -af "volume=enable='between(t,1,3)':volume=0, volume=enable='between(t,10,12)':volume=0, volume=enable='between(t,4,6)':volume=0, volume=enable='between(t,7,9)':volume=0" -c:v copy -movflags +faststart c:\temp\31e7ac4063d111e6bdc67f1f7f7b55d3.mp4
    
ffmpeg version N-79651-ge1c2048 Copyright (c) 2000-2016 the FFmpeg developers
    
  built with gcc 4.9.3 (GCC)
    
  configuration: --prefix=/usr/local/x86_64-w64-mingw32 --enable-gpl --enable-nonfree --enable-libx264 --enable-libfdk_aac --enable-static --enable-runtime-cpudetect --enable-w32threads --disable-shared --disable-ffplay --disable-ffserver --arch=x86_64 --extra-cflags=-I/local/x86_64-w64-mingw32/include --extra-ldflags='-L/local/x86_64-w64-mingw32/lib -static'
    
  libavutil      55. 22.101 / 55. 22.101
    
  libavcodec     57. 38.100 / 57. 38.100
    
  libavformat    57. 34.103 / 57. 34.103
    
  libavdevice    57.  0.101 / 57.  0.101
    
  libavfilter     6. 44.100 /  6. 44.100
    
  libswscale      4.  1.100 /  4.  1.100
    
  libswresample   2.  0.101 /  2.  0.101
    
  libpostproc    54.  0.100 / 54.  0.100
    
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:/Users/ADMINI~1/AppData/Local/Temp/2/0400028520160811144100001i100.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf57.34.103
    
  Duration: 00:00:12.78, start: 0.000000, bitrate: 4074 kb/s
    
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720, 4006 kb/s, 30 fps, 30 tbr, 15360 tbn (default)
    
    Metadata:
    
      handler_name    : VideoHandler
    
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 64 kb/s (default)
    
    Metadata:
    
      handler_name    : SoundHandler
    
[mp4 @ 03a8e5e0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
    
    Last message repeated 1 times
    
Output #0, mp4, to 'c:\temp\31e7ac4063d111e6bdc67f1f7f7b55d3-AudRedact.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf57.34.103
    
    Stream #0:0(und): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 1280x720, q=2-31, 4006 kb/s, 30 fps, 30 tbr, 15360 tbn (default)
    
    Metadata:
    
      handler_name    : VideoHandler
    
    Stream #0:1(und): Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, stereo, fltp, 128 kb/s (default)
    
    Metadata:
    
      handler_name    : SoundHandler
    
      encoder         : Lavc57.38.100 aac
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (copy)
    
  Stream #0:1 -> #0:1 (aac (native) -> aac (native))
    
Press [q] to stop, [?] for help
    
[mp4 @ 03a8e5e0] Starting second pass: moving the moov atom to the beginning of the file24.3x
    
frame=  383 fps=0.0 q=-1.0 Lsize=    6325kB time=00:00:12.73 bitrate=4068.3kbits/s speed=24.7x
    
video:6244kB audio:69kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.191899%
    
[aac @ 03776160] Qavg: 51080.756

    However if I build the same in node programatically :

    function redactAudio(tempFilePath, arrPairs, cb){
    
    // ffmpeg -loglevel fatal -y -i video.mp4 -af "volume=enable='between(t,5,10)':volume=0, volume=enable='between(t,15,20)':volume=0" -c:v copy -movflags +faststart output.mp4
    

    
    var tempBuff = Buffer.alloc(16);
    
    jsuuid.v1(null, tempBuff, 0);
    
    var outFileName = tempBuff.toString('hex') + path2.extname(tempFilePath);
    

    
    var volStr = '"';
    
    for (var i = 0; i &lt; arrPairs.length; i++) {
    
        volStr += "volume=enable='between(t," + arrPairs[i].start + "," + arrPairs[i].end + ")':volume=0";
    
        if (i !== arrPairs.length - 1) {
    
            volStr += ", ";
    
        } else {
    
            volStr += '"';
    
        }
    
    }
    

    
    child_process.execFile(
    
        'ffmpeg',
    
        [
    
            /*'-loglevel', 'fatal',*/
    
            '-y', '-i', tempFilePath,
    
            '-af', volStr,
    
            '-c:v', 'copy',
    
            '-movflags', '+faststart', outFileName
    
        ],
    
        {
    
            cwd: tempDir,
    
            maxBuffer: Infinity
    
        },
    
        function(err, stdout, stderr) {
    
            if (err) {
    
                console.error(clc.magentaBright(clc.whiteBright('FFMPEG - ERROR OCC: ', path2.basename(tempFilePath), ' : ', stderr, '\n')));
    
                return cb(err, 'FFMpeg Failed: ' +  JSON.stringify({ stdout: stdout, stderr: stderr} ));
    
            } else {
    
                console.log(clc.magentaBright(clc.whiteBright('FFMPEG - Finished:  ', path2.basename(tempFilePath), '\n')));
    
                return cb(null, outFileName);
    
            }
    
        }
    
    );  
    
}

    Please Help me understand what is going on because I get the following error every time when run from node.

    FFMPEG - ERROR OCC:  0400028520160811144100001i100.mp4  :  ffmpeg version N-79651-ge1c2048 Copyright (c) 2000-2016 the FFmpeg developers
    
  built with gcc 4.9.3 (GCC)
    
  configuration: --prefix=/usr/local/x86_64-w64-mingw32 --enable-gpl --enable-nonfree --enable-libx264 --enable-libfdk_aac --enable-static --enable-runtime-cpudetect --enable-w32threads --disable-shared --disable-ffplay --disable-ffserver --arch=x86_64 --extra-cflags=-I/local/x86_64-w64-mingw32/include --extra-ldflags='-L/local/x86_64-w64-mingw32/lib -static'
    
  libavutil      55. 22.101 / 55. 22.101
    
  libavcodec     57. 38.100 / 57. 38.100
    
  libavformat    57. 34.103 / 57. 34.103
    
  libavdevice    57.  0.101 / 57.  0.101
    
  libavfilter     6. 44.100 /  6. 44.100
    
  libswscale      4.  1.100 /  4.  1.100
    
  libswresample   2.  0.101 /  2.  0.101
    
  libpostproc    54.  0.100 / 54.  0.100
    
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:/Users/ADMINI~1/AppData/Local/Temp/2/0400028520160811144100001i100.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf57.34.103
    
  Duration: 00:00:12.78, start: 0.000000, bitrate: 4074 kb/s
    
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720, 4006 kb/s, 30 fps, 30 tbr, 15360 tbn (default)
    
    Metadata:
    
      handler_name    : VideoHandler
    
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 64 kb/s (default)
    
    Metadata:
    
      handler_name    : SoundHandler
    
[AVFilterGraph @ 00381600] No such filter: '"volume'
    
Error opening filters!
    

    
0400028520160811144100001i100.mp4 
    
 { Error: Command failed: ffmpeg -y -i C:/Users/ADMINI~1/AppData/Local/Temp/2/0400028520160811144100001i100.mp4 -af "volume=enable='between(t,1,3)':volume=0, volume=enable='between(t,10,12)':volume=0, volume=enable='between(t,4,6)':volume=0, volume=enable='between(t,7,9)':volume=0" -c:v copy -movflags +faststart ea1eae2063d211e6bbf2af16c2e9be57.mp4
    
ffmpeg version N-79651-ge1c2048 Copyright (c) 2000-2016 the FFmpeg developers
    
  built with gcc 4.9.3 (GCC)
    
  configuration: --prefix=/usr/local/x86_64-w64-mingw32 --enable-gpl --enable-nonfree --enable-libx264 --enable-libfdk_aac --enable-static --enable-runtime-cpudetect --enable-w32threads --disable-shared --disable-ffplay --disable-ffserver --arch=x86_64 --extra-cflags=-I/local/x86_64-w64-mingw32/include --extra-ldflags='-L/local/x86_64-w64-mingw32/lib -static'
    
  libavutil      55. 22.101 / 55. 22.101
    
  libavcodec     57. 38.100 / 57. 38.100
    
  libavformat    57. 34.103 / 57. 34.103
    
  libavdevice    57.  0.101 / 57.  0.101
    
  libavfilter     6. 44.100 /  6. 44.100
    
  libswscale      4.  1.100 /  4.  1.100
    
  libswresample   2.  0.101 /  2.  0.101
    
  libpostproc    54.  0.100 / 54.  0.100
    
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:/Users/ADMINI~1/AppData/Local/Temp/2/0400028520160811144100001i100.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf57.34.103
    
  Duration: 00:00:12.78, start: 0.000000, bitrate: 4074 kb/s
    
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720, 4006 kb/s, 30 fps, 30 tbr, 15360 tbn (default)
    
    Metadata:
    
      handler_name    : VideoHandler
    
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 64 kb/s (default)
    
    Metadata:
    
      handler_name    : SoundHandler
    
[AVFilterGraph @ 00381600] No such filter: '"volume'
    
Error opening filters!
    

    
    at ChildProcess.exithandler (child_process.js:202:12)
    
    at emitTwo (events.js:106:13)
    
    at ChildProcess.emit (events.js:191:7)
    
    at maybeClose (internal/child_process.js:850:16)
    
    at Process.ChildProcess._handle.onexit (internal/child_process.js:215:5)
    
  killed: false,
    
  code: 1,
    
  signal: null,
    
  cmd: 'ffmpeg -y -i C:/Users/ADMINI~1/AppData/Local/Temp/2/0400028520160811144100001i100.mp4 -af "volume=enable=\'between(t,1,3)\':volume=0, volume=enable=\'between(t,10,12)\':volume=0, volume=enable=\'between(t,4,6)\':volume=0, volume=enable=\'between(t,7,9)\':volume=0" -c:v copy -movflags +faststart ea1eae2063d211e6bbf2af16c2e9be57.mp4' }

    Best I can think is it has something to do with the \’ in volStr but I don’t know how to create it any other way.

    after @Mulvya comment I changed my for loop to

    var volStr = '';
    
for (var i = 0; i &lt; arrPairs.length; i++) {
    
    volStr += "volume=enable='between(t," + arrPairs[i].start + "," + arrPairs[i].end + ")':volume=0";
    
    if (i !== arrPairs.length - 1) {
    
        volStr += ",";
    
    }
    
}

    getting rid of the space between each grouping and it worked beautifully.

  • ffmpeg mp3 streaming via node js

    21 août 2016, par school boy
    var fs = require('fs');
    
var child = require('child_process');
    
var http=require('http')
    
var input_file = fs.createReadStream('./remo.mp3');
    
http.createServer(function (req,res) {
    
var args = ['-ss',120,
    
'-i', 'remo.mp3',
    
'-f','mp3',
    
'pipe:1' // Output on stdout
    
];
    
var trans_proc = child.spawn('ffmpeg', args);
    
res.writeHead(200, {
    
      'Content-Type': 'audio/mpeg'
    
  });
    

    
trans_proc.stdout.pipe(res)
    
trans_proc.stderr.on('data',function (err) {
    
console.log(err.toString());
    
})
    
}).listen(2000)

    i am trying to cut the mp3 and streaming to the browser but in browser it showing corrupted file

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