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Le plugin : Podcasts.
14 juillet 2010, parLe problème du podcasting est à nouveau un problème révélateur de la normalisation des transports de données sur Internet.
Deux formats intéressants existent : Celui développé par Apple, très axé sur l’utilisation d’iTunes dont la SPEC est ici ; Le format "Media RSS Module" qui est plus "libre" notamment soutenu par Yahoo et le logiciel Miro ;
Types de fichiers supportés dans les flux
Le format d’Apple n’autorise que les formats suivants dans ses flux : .mp3 audio/mpeg .m4a audio/x-m4a .mp4 (...) -
Pas question de marché, de cloud etc...
10 avril 2011Le vocabulaire utilisé sur ce site essaie d’éviter toute référence à la mode qui fleurit allègrement
sur le web 2.0 et dans les entreprises qui en vivent.
Vous êtes donc invité à bannir l’utilisation des termes "Brand", "Cloud", "Marché" etc...
Notre motivation est avant tout de créer un outil simple, accessible à pour tout le monde, favorisant
le partage de créations sur Internet et permettant aux auteurs de garder une autonomie optimale.
Aucun "contrat Gold ou Premium" n’est donc prévu, aucun (...) -
Keeping control of your media in your hands
13 avril 2011, parThe vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
While using MediaSPIP, you are invited to avoid using words like "Brand", "Cloud" and "Market".
MediaSPIP is designed to facilitate the sharing of creative media online, while allowing authors to retain complete control of their work.
MediaSPIP aims to be accessible to as many people as possible and development is based on expanding the (...)
Sur d’autres sites (8115)
-
ffmpeg decode example using vda acceleration ?
18 mars 2015, par Bill YanI’m testing the vda hw accelerated decoder of ffmpeg, but have no luck so far.
the command line I use is this :
./ffmpeg -hwaccel vda -i ~/Downloads/big_buck_bunny_720p_surround.avi -c:v rawvideo -pix_fmt yuv420p out.yuv
(the input video can be downloaded here : http://mirrorblender.top-ix.org/peach/bigbuckbunny_movies/big_buck_bunny_720p_surround.avi)
but it uses the sw decoder, instead of the requested vda hw decoder.
I debugged ffmpeg.c, and found out that in the function get_format :
if (!(desc->flags & AV_PIX_FMT_FLAG_HWACCEL))
break;the desc->flag isn’t set with AV_PIX_FMT_FLAG_HWACCEL, so it skips the hw acceleration initialization.
the reason is because the pix_fmt is set to AV_PIX_FMT_NONE for some reason, instead of yuv420p as specified by the command.
I then changed the command to this :
./ffmpeg -hwaccel vda -i ~/Downloads/big_buck_bunny_720p_surround.avi -c:v rawvideo -pix_fmt vda_vld out.yuv
but what I got was :
Impossible to convert between the formats supported by the filter 'format' and the filter 'auto-inserted scaler 0'
Error opening filters!Can you please show me an example of using ffmpeg to dump the above video into yuv using the vda hw decoder ?
UPDATE from the ffmpeg mailist.
I was told to try the following command instead :
./ffmpeg -vcodec h264_vda -i ~/Downloads/big_buck_bunny_720p_surround.avi out.yuv
but this is what I got :
./ffmpeg -vcodec h264_vda -i ~/Downloads/big_buck_bunny_720p_surround.avi out.yuv
ffmpeg version 2.6.1 Copyright (c) 2000-2015 the FFmpeg developers
built with Apple LLVM version 6.0 (clang-600.0.57) (based on LLVM 3.5svn)
configuration:
libavutil 54. 20.100 / 54. 20.100
libavcodec 56. 26.100 / 56. 26.100
libavformat 56. 25.101 / 56. 25.101
libavdevice 56. 4.100 / 56. 4.100
libavfilter 5. 11.102 / 5. 11.102
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 1.100 / 1. 1.100
[10:20:50.169] vtDecompressionDuctCreate signalled err=-8973 (err) (Could not select and open decoder instance) at /SourceCache/CoreMedia_frameworks/CoreMedia-1562.107/Sources/VideoToolbox/VTDecompressionSession.c line 1181
[h264_vda @ 0x7f9feb034600] Failed to init VDA decoder: -12473.
[avi @ 0x7f9feb00da00] Failed to open codec in av_find_stream_info
[NULL @ 0x7f9feb034600] missing picture in access unit with size 9960
[10:20:50.252] vtDecompressionDuctCreate signalled err=-8973 (err) (Could not select and open decoder instance) at /SourceCache/CoreMedia_frameworks/CoreMedia-1562.107/Sources/VideoToolbox/VTDecompressionSession.c line 1181
[h264_vda @ 0x7f9feb034600] Failed to init VDA decoder: -12473.
[NULL @ 0x7f9feb034600] missing picture in access unit with size 457the reason I wanted to try this was because I wanted to use vda with libavcodec in the first place. but the problem I saw was exactly the above. looks like ffmpeg has the same issue.
-
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.


-
FFMPEG "buffer queue overflow, dropping" with trim and atrim filters
13 juin 2020, par Prasanna MahendiranIn FFMPEG I am actually trimming and concating a 24 FPS video. When I apply a complex filter



ffmpeg -i sample.mp4 -filter_complex \
 "[0:v]setpts = PTS-STARTPTS[bv];
 [bv]split=6[v0][v1][v2][v3][v4][v5];
 [v0]trim=start_frame=1:end_frame=142,loop=1:1:1,setpts=N/FRAME_RATE/TB[0v];
 [v1]trim=start_frame=846:end_frame=878,loop=1:1:1,setpts=N/FRAME_RATE/TB[1v];
 [v2]trim=start_frame=57:end_frame=114,loop=1:1:1,setpts=N/FRAME_RATE/TB[2v];
 [v3]trim=start_frame=865:end_frame=885,loop=1:1:1,setpts=N/FRAME_RATE/TB[3v];
 [v4]trim=start_frame=70:end_frame=155,loop=1:1:1,setpts=N/FRAME_RATE/TB[4v];
 [v5]trim=start_frame=155:end_frame=909,loop=1:1:1,setpts=N/FRAME_RATE/TB[5v];
 [0:a]asplit=6[a0][a1][a2][a3][a4][a5];
 [a0]atrim=0.041666666666666664:5.917,asetpts=N/SR/TB[0a];
 [a1]atrim=35.256:36.603,asetpts=N/SR/TB[1a];
 [a2]atrim=2.379:4.767,asetpts=N/SR/TB[2a];
 [a3]atrim=36.024:36.859,asetpts=N/SR/TB[3a];
 [a4]atrim=2.93:6.438172,asetpts=N/SR/TB[4a];
 [a5]atrim=6.438172:37.895,asetpts=N/SR/TB[5a];
 [0v][0a][1v][1a][2v][2a][3v][3a][4v][4a][5v][5a]concat=n=6:v=1:a=1[vv][aa]"\
 -map "[vv]" -map "[aa]" output.mp4




I am getting "buffer queue overflow, dropping" error. The resultant video and audio is still and not working properly.



ffmpeg version 3.2-1~16.04.york1 Copyright (c) 2000-2016 the FFmpeg developers
 built with gcc 5.4.1 (Ubuntu 5.4.1-3ubuntu1~ubuntu16.04.1york0) 20161019
 configuration: --prefix=/usr --extra-version='1~16.04.york1' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-libtesseract --disable-stripping --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librubberband --enable-libschroedinger --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-opengl --enable-sdl2 --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-openal --enable-frei0r --enable-libopencv --enable-libx264 --enable-chromaprint --enable-shared
 libavutil 55. 34.100 / 55. 34.100
 libavcodec 57. 64.100 / 57. 64.100
 libavformat 57. 56.100 / 57. 56.100
 libavdevice 57. 1.100 / 57. 1.100
 libavfilter 6. 65.100 / 6. 65.100
 libavresample 3. 1. 0 / 3. 1. 0
 libswscale 4. 2.100 / 4. 2.100
 libswresample 2. 3.100 / 2. 3.100
 libpostproc 54. 1.100 / 54. 1.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'sample.mp4':
 Metadata:
 major_brand : isom
 minor_version : 512
 compatible_brands: isomiso2avc1mp41
 track : 0
 artist : 
 album : 
 date : 0
 genre : 
 lyrics : 
 title : 
 encoder : Lavf56.36.100
 Duration: 00:00:37.90, start: 0.000000, bitrate: 951 kb/s
 Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 820 kb/s, 24 fps, 24 tbr, 12288 tbn, 48 tbc (default)
 Metadata:
 handler_name : VideoHandler
 Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 126 kb/s (default)
 Metadata:
 handler_name : SoundHandler
File 'output.mp4' already exists. Overwrite ? [y/N] y
[libx264 @ 0x55650097a540] using SAR=1/1
[libx264 @ 0x55650097a540] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 AVX2 LZCNT BMI2
[libx264 @ 0x55650097a540] profile High, level 3.0
[libx264 @ 0x55650097a540] 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=6 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=250 keyint_min=24 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to 'output.mp4':
 Metadata:
 major_brand : isom
 minor_version : 512
 compatible_brands: isomiso2avc1mp41
 track : 0
 artist : 
 album : 
 date : 0
 genre : 
 lyrics : 
 title : 
 encoder : Lavf57.56.100
 Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 640x480 [SAR 1:1 DAR 4:3], q=-1--1, 24 fps, 12288 tbn, 24 tbc (default)
 Metadata:
 encoder : Lavc57.64.100 libx264
 Side data:
 cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
 Stream #0:1: Audio: aac (LC) ([64][0][0][0] / 0x0040), 44100 Hz, stereo, fltp, 128 kb/s (default)
 Metadata:
 encoder : Lavc57.64.100 aac
Stream mapping:
 Stream #0:0 (h264) -> setpts
 Stream #0:1 (aac) -> asplit
 concat:out:v0 -> Stream #0:0 (libx264)
 concat:out:a0 -> Stream #0:1 (aac)
Press [q] to stop, [?] for help
[Parsed_concat_33 @ 0x55650097b420] Buffer queue overflow, dropping. 471.5kbits/s speed=4.94x 
 Last message repeated 201 times
[Parsed_concat_33 @ 0x55650097b420] Buffer queue overflow, dropping. 522.9kbits/s speed=3.89x 
 Last message repeated 1266 times
[Parsed_concat_33 @ 0x55650097b420] Buffer queue overflow, dropping. 557.0kbits/s speed=3.28x 
 Last message repeated 48 times
[output stream 0:1 @ 0x556500947e20] 100 buffers queued in output stream 0:1, something may be wrong.
[Parsed_concat_33 @ 0x55650097b420] Buffer queue overflow, dropping. 718.6kbits/s speed=3.46x 
 Last message repeated 19 times
[output stream 0:0 @ 0x5565009785c0] 100 buffers queued in output stream 0:0, something may be wrong.
frame= 1091 fps=117 q=-1.0 Lsize= 2795kB time=00:00:45.51 bitrate= 503.1kbits/s dup=475 drop=0 speed=4.88x 
video:2455kB audio:316kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.861779%
[libx264 @ 0x55650097a540] frame I:8 Avg QP:19.26 size: 24207
[libx264 @ 0x55650097a540] frame P:409 Avg QP:21.33 size: 4108
[libx264 @ 0x55650097a540] frame B:674 Avg QP:27.46 size: 949
[libx264 @ 0x55650097a540] consecutive B-frames: 10.3% 13.9% 24.5% 51.3%
[libx264 @ 0x55650097a540] mb I I16..4: 9.9% 57.0% 33.1%
[libx264 @ 0x55650097a540] mb P I16..4: 3.6% 7.6% 2.9% P16..4: 33.0% 10.6% 3.0% 0.0% 0.0% skip:39.2%
[libx264 @ 0x55650097a540] mb B I16..4: 0.4% 0.8% 0.4% B16..8: 24.5% 2.6% 0.2% direct: 0.5% skip:70.5% L0:55.5% L1:41.8% BI: 2.7%
[libx264 @ 0x55650097a540] 8x8 transform intra:53.8% inter:66.7%
[libx264 @ 0x55650097a540] coded y,uvDC,uvAC intra: 44.6% 50.0% 14.8% inter: 6.2% 7.7% 0.2%
[libx264 @ 0x55650097a540] i16 v,h,dc,p: 22% 28% 17% 33%
[libx264 @ 0x55650097a540] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 20% 23% 28% 3% 4% 3% 11% 3% 5%
[libx264 @ 0x55650097a540] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 26% 26% 16% 2% 5% 3% 16% 3% 3%
[libx264 @ 0x55650097a540] i8c dc,h,v,p: 60% 22% 13% 6%
[libx264 @ 0x55650097a540] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x55650097a540] ref P L0: 72.6% 8.4% 15.1% 3.9%
[libx264 @ 0x55650097a540] ref B L0: 88.5% 10.7% 0.8%
[libx264 @ 0x55650097a540] ref B L1: 93.3% 6.7%
[libx264 @ 0x55650097a540] kb/s:442.30
[aac @ 0x556500979280] Qavg: 3215.870




I tried with other stackoverflow questions but none of them worked. Also I think it is partially because the trim timings are mixed. That is start time can be anywhere between 0-end. When I make it strictly increasing it is working fine.