[FFmpeg-user] Issues Live Streaming Audio Only
WTI TEC
wtitec at gmail.com
Mon Dec 3 15:21:19 EET 2018
Hi, this may help you.
sample, File .conf
HTTPPort 8088
HTTPBindAddress 0.0.0.0
MaxHTTPConnections 50000
MaxClients 30000
MaxBandWidth 9999999
CustomLog /var/log/ffserver.log
#AUDIO
<Feed audio.ffm>
File /tmp/audio.ffm
FileMaxSize 30M
ACL allow 0.0.0.0 255.255.255.255
</Feed>
<Stream audio.mp3>
Feed audio.ffm
Format mp2
AudioCodec libmp3lame
AudioBitRate 16
AudioChannels 1
AudioSampleRate 16000
NoVideo
</Stream>
<Stream index.html>
Format status
ACL allow 0.0.0.0 255.255.255.255
</Stream>
------------------------------------------------------
ffmpeg input http://localhost:8088/audio.ffm
access external: http://IPHOST:8088/audio.mp3
Em qua, 31 de out de 2018 às 20:03, Brett Garrett <
brett at edgewaterbroadcasting.com> escreveu:
> We actually have been using IceCast for public access on personal computers
> and/or mobiles (an app). Not for our internal private remote servers. I was
> of the impression that we were avoiding it was due to bandwidth. That it
> was causing choppy audio and was undesirable. That the udp based
> distribution was the best/only option. IceCast uses http to stream which is
> tcp based, so it didn't seem like an option to even consider. Inquired
> about the IceCast and using it. After discussing it further, I was reminded
> of the story about why a grandma always cut the ends off the ham.
> Basically, we are working with old technology that actually breaks when we
> try to stream an http based stream. I'm trying to make things newer, and
> with new software the tcp based streaming does not just break. So we will
> be exploring the new path with IceCast.
>
> Thank you for your suggestion and help.
>
> On Wed, Oct 31, 2018 at 3:46 PM DopeLabs <dopelabs at dubstep.fm> wrote:
>
> > have you considered using something other than ffserve (which no longer
> > supported if im not mistaken) that has been specifically developed for
> > audio transfer over a network?
> >
> > such as icecast?
> >
> > ffmpeg can output to an icecast server running on the local machine or
> > running on the remote machine
> >
> >
> > source machine:
> > ffmpeg -i input -ac 2 -c:a libfdk_aac -ar 48k -b:a 64k -ice_public 0
> > -ice_genre Genre -ice_url http://url.tld -ice_description Description
> > -ice_name Name -content_type audio/aacp -f adts icecast://
> > source:pass at remote.host.com/mount
> >
> > remote machine:
> > ffmpeg -i http://remote.host/mount
> >
> >
> > > On Oct 31, 2018, at 2:23 15PM, Brett Garrett <
> > brett at edgewaterbroadcasting.com> wrote:
> > >
> > > I've been trying to stream an audio feed from a live source using the
> > RTSP
> > > protocol. The main goal is to eventually utilize jitter buffers or
> other
> > > techniques to reduce/remove stutters and skips. Although, at the
> moment I
> > > can't seem to get any audio to transfer over the network at all. I do
> > have
> > > a somewhat unique setup, and therefore don't have many other options to
> > > work with.
> > >
> > > I'm currently using Debian 9 with no gui for both the server and
> > client(s).
> > > I would say the biggest hangup or issue is having to use Jack (which
> uses
> > > an ALSA device) for my output as it is used in several other
> applications
> > > that interact with the audio output. All of which are not running for
> the
> > > time being because I'm still just trying to get the stream from one
> > > computer to another across my local network. Here's what I'm hoping is
> > > enough information and that I'm not vomiting too much text.
> > >
> > > A little info on our audio card specs...
> > >
> > > root at test-9:~# *aplay -l*
> > > **** List of PLAYBACK Hardware Devices ****
> > > card 0: M44 [M Audio Delta 44], device 0: ICE1712 multi [ICE1712 multi]
> > > Subdevices: 1/1
> > > Subdevice #0: subdevice #0
> > >
> > > root at test-9:~# *arecord --dump-hw-params -D hw:0,0*
> > > Recording WAVE 'stdin' : Unsigned 8 bit, Rate 8000 Hz, Mono
> > > HW Params of device "hw:0,0":
> > > --------------------
> > > ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED
> > > FORMAT: S32_LE
> > > SUBFORMAT: STD
> > > SAMPLE_BITS: 32
> > > FRAME_BITS: 384
> > > CHANNELS: 12
> > > RATE: [8000 96000]
> > > PERIOD_TIME: (20 341250]
> > > PERIOD_SIZE: [2 2730]
> > > PERIOD_BYTES: [96 131040]
> > > PERIODS: [1 1024]
> > > BUFFER_TIME: (20 682625]
> > > BUFFER_SIZE: [2 5461]
> > > BUFFER_BYTES: [96 262128]
> > > TICK_TIME: ALL
> > > --------------------
> > > arecord: set_params:1299: Sample format non available
> > > Available formats:
> > > - S32_LE
> > >
> > > Audio comes in through an analog XLR connection. Verified audio input
> is
> > > good by using the following commands...
> > >
> > > root at test-9:~# /usr/bin/dbus-run-session ffmpeg -f jack -i ffmpeg -y
> > > output.wav
> > > root at test-9:~# jack_connect system:capture_1 ffmpeg:input_1 &&
> > jack_connect
> > > system:capture_2 ffmpeg:input_2
> > >
> > > Waited a while then killed the tasks. Acquired the output.wav and it
> > played
> > > with good audio. Although, I need to be able to get it from one
> computer
> > to
> > > another through a live feed (not recording to some single file that
> will
> > > grow to crazy sizes).
> > > So here's my server config setup. Without the comments.
> > >
> > > root at test-9:~# *grep ^[^#] /etc/ffserver.conf*
> > > HTTPPort 8585
> > > RTSPPort 15151
> > > HTTPBindAddress 0.0.0.0
> > > MaxHTTPConnections 2000
> > > MaxClients 1000
> > > MaxBandwidth 1000
> > > CustomLog -
> > > <Feed feed1.ffm>
> > > File /tmp/feed1.ffm
> > > FileMaxSize 200K
> > > ACL allow 127.0.0.1
> > > </Feed>
> > > <Stream test1-rtsp.ogg>
> > > Format rtp
> > > Feed feed1.ffm
> > > NoVideo
> > > AudioCodec aac
> > > AudioChannels 2
> > > AudioBitRate 64
> > > AudioSampleRate 48000
> > > AVOptionAudio flags +global_header
> > > </Stream>
> > > <Stream stat.html>
> > > Format status
> > > ACL allow localhost
> > > ACL allow 192.168.0.0 192.168.255.255
> > > </Stream>
> > > <Redirect index.html>
> > > URL http://www.ffmpeg.org/
> > > </Redirect>
> > >
> > > root at test-9:~# *ffserver -loglevel debug &*
> > > [1] 11631
> > > root at test-9:~# ffserver version 3.2.10-1~deb9u1 Copyright (c)
> 2000-2018
> > the
> > > FFmpeg developers
> > > built with gcc 6.3.0 (Debian 6.3.0-18) 20170516
> > > configuration: --prefix=/usr --extra-version='1~deb9u1'
> > > --toolchain=hardened --libdir=/usr/lib/i386-linux-gnu
> > > --incdir=/usr/include/i386-linux-gnu --enable-gpl --disable-stripping
> > > --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-libmp3lame
> --enable-libopenjpeg
> > > --enable-libopenmpt --enable-libopus --enable-libpulse
> > > --enable-librubberband --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-omx --enable-openal --enable-opengl
> > --enable-sdl2
> > > --enable-libdc1394 --enable-libiec61883 --enable-chromaprint
> > > --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
> > > libavutil 55. 34.101 / 55. 34.101
> > > libavcodec 57. 64.101 / 57. 64.101
> > > libavformat 57. 56.101 / 57. 56.101
> > > 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
> > > Tue Oct 30 13:23:01 2018 [file @ 0xfe65c0]Setting default whitelist
> > > 'file,crypto'
> > > Tue Oct 30 13:23:01 2018 [ffm @ 0xfe39e0]Using AVStream.codec to pass
> > codec
> > > parameters to muxers is deprecated, use AVStream.codecpar instead.
> > > Tue Oct 30 13:23:01 2018 writing recommended configuration:
> > > ac=2,b=64000,ar=48000,flags=+global_header
> > > Tue Oct 30 13:23:01 2018 [AVIOContext @ 0xfe6660]Statistics: 0 seeks, 1
> > > writeouts
> > > Tue Oct 30 13:23:01 2018 FFserver started.
> > >
> > >
> > > root at test-9:~# */usr/bin/dbus-run-session ffmpeg -f jack -i ffmpeg
> > > http://127.0.0.1:8585/feed1.ffm <http://127.0.0.1:8585/feed1.ffm>*
> > > ffmpeg version 3.2.10-1~deb9u1 Copyright (c) 2000-2018 the FFmpeg
> > developers
> > > built with gcc 6.3.0 (Debian 6.3.0-18) 20170516
> > > configuration: --prefix=/usr --extra-version='1~deb9u1'
> > > --toolchain=hardened --libdir=/usr/lib/i386-linux-gnu
> > > --incdir=/usr/include/i386-linux-gnu --enable-gpl --disable-stripping
> > > --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-libmp3lame
> --enable-libopenjpeg
> > > --enable-libopenmpt --enable-libopus --enable-libpulse
> > > --enable-librubberband --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-omx --enable-openal --enable-opengl
> > --enable-sdl2
> > > --enable-libdc1394 --enable-libiec61883 --enable-chromaprint
> > > --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
> > > libavutil 55. 34.101 / 55. 34.101
> > > libavcodec 57. 64.101 / 57. 64.101
> > > libavformat 57. 56.101 / 57. 56.101
> > > 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
> > > Cannot connect to server socket err = No such file or directory
> > > Cannot connect to server request channel
> > > jackdmp 1.9.11
> > > Copyright 2001-2005 Paul Davis and others.
> > > Copyright 2004-2014 Grame.
> > > jackdmp comes with ABSOLUTELY NO WARRANTY
> > > This is free software, and you are welcome to redistribute it
> > > under certain conditions; see the file COPYING for details
> > > no message buffer overruns
> > > no message buffer overruns
> > > no message buffer overruns
> > > JACK server starting in realtime mode with priority 10
> > > self-connect-mode is "Don't restrict self connect requests"
> > > audio_reservation_init
> > > Acquire audio card Audio0
> > > creating alsa driver ...
> hw:0|hw:0|1024|2|48000|0|0|nomon|swmeter|-|32bit
> > > configuring for 48000Hz, period = 1024 frames (21.3 ms), buffer = 2
> > periods
> > > ALSA: final selected sample format for capture: 32bit integer
> > little-endian
> > > ALSA: use 2 periods for capture
> > > ALSA: final selected sample format for playback: 32bit integer
> > little-endian
> > > ALSA: use 2 periods for playback
> > > [jack @ 0x2494700] JACK client registered and activated (rate=48000Hz,
> > > buffer_size=1024 frames)
> > > Guessed Channel Layout for Input Stream #0.0 : stereo
> > > Input #0, jack, from 'ffmpeg':
> > > Duration: N/A, start: 1540927425.470458, bitrate: 3072 kb/s
> > > Stream #0:0: Audio: pcm_f32le, 48000 Hz, stereo, flt, 3072 kb/s
> > > Tue Oct 30 13:23:45 2018 [NULL @ 0xfe39e0]Opening '/tmp/feed1.ffm' for
> > > reading
> > > Tue Oct 30 13:23:45 2018 [file @ 0xfe46c0]Setting default whitelist
> > > 'file,crypto'
> > > Tue Oct 30 13:23:45 2018 [ffm @ 0xfe39e0]Format ffm probed with
> size=2048
> > > and score=101
> > > Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'ac'
> to
> > > value '2'
> > > Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'b' to
> > > value '64000'
> > > Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'ar'
> to
> > > value '48000'
> > > Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key
> 'flags'
> > to
> > > value '+global_header'
> > > Tue Oct 30 13:23:45 2018 writing recommended configuration:
> > > ac=2,b=64000,ar=48000,flags=+global_header
> > > Tue Oct 30 13:23:45 2018 127.0.0.1 - - [GET] "/feed1.ffm HTTP/1.1" 200
> > 4175
> > > Tue Oct 30 13:23:45 2018 [AVIOContext @ 0xfe6720]Statistics: 4096 bytes
> > > read, 0 seeks
> > > Output #0, ffm, to 'http://127.0.0.1:8585/feed1.ffm':
> > > Metadata:
> > > creation_time : now
> > > encoder : Lavf57.56.101
> > > Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp, 64 kb/s
> > > Metadata:
> > > encoder : Lavc57.64.101 aac
> > > Stream mapping:
> > > Stream #0:0 -> #0:0 (pcm_f32le (native) -> aac (native))
> > > Press [q] to stop, [?] for help
> > > Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'b' to
> > > value '64000'
> > > Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'ab'
> to
> > > value '64000'
> > > Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key
> 'flags'
> > to
> > > value '0x00400000'
> > > Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'ar'
> to
> > > value '48000'
> > > Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'ac'
> to
> > > value '2'
> > > Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key
> > > 'frame_size' to value '1024'
> > > Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key
> > 'profile'
> > > to value '1'
> > > Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key
> > > 'channel_layout' to value '3'
> > > Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key
> > > 'time_base' to value '1/48000'
> > > Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key
> 'delay'
> > to
> > > value '1024'
> > > Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key
> > > 'pkt_timebase' to value '1/1000000'
> > >
> > > Changing text at the bottom makes it appear like 'somethings
> happening'.
> > > There is a slow bit transfer because I have not connected the jack
> > > inputs/outputs.
> > >
> > > size= 72kB time=00:01:07.08 bitrate= 8.8kbits/s speed= 1x
> > >
> > > root at test-9:~# *jack_connect system:capture_1 ffmpeg:input_1 &&
> > > jack_connect system:capture_2 ffmpeg:input_2 && jack_lsp -c*
> > > system:capture_1
> > > ffmpeg:input_1
> > > system:capture_2
> > > ffmpeg:input_2
> > > system:capture_3
> > > system:capture_4
> > > system:capture_5
> > > system:capture_6
> > > system:capture_7
> > > system:capture_8
> > > system:capture_9
> > > system:capture_10
> > > system:capture_11
> > > system:capture_12
> > > system:playback_1
> > > system:playback_2
> > > system:playback_3
> > > system:playback_4
> > > system:playback_5
> > > system:playback_6
> > > system:playback_7
> > > system:playback_8
> > > system:playback_9
> > > system:playback_10
> > > ffmpeg:input_1
> > > system:capture_1
> > > ffmpeg:input_2
> > > system:capture_2
> > > size= 800kB time=00:03:31.24 bitrate= 31.0kbits/s speed= 1x
> > >
> > > The bitrate goes up, so I'm assuming this means it is receiving the
> audio
> > > alright and that everything's in place. Go to
> > > http://192.168.1.119:8585/stat.html the page comes up with...
> > >
> > > [see attached picture]
> > >
> > > Time to see if we can get the stream from another computer...
> > >
> > > root at test-17:~# ffplay -nodisp rtsp://
> 192.168.1.119:15151/test1-rtsp.ogg
> > > ffplay version 3.2.12-1~deb9u1 Copyright (c) 2003-2018 the FFmpeg
> > developers
> > > built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516
> > > configuration: --prefix=/usr --extra-version='1~deb9u1'
> > > --toolchain=hardened --libdir=/usr/lib/i386-linux-gnu
> > > --incdir=/usr/include/i386-linux-gnu --enable-gpl --disable-stripping
> > > --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-libmp3lame
> --enable-libopenjpeg
> > > --enable-libopenmpt --enable-libopus --enable-libpulse
> > > --enable-librubberband --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-omx --enable-openal --enable-opengl
> > --enable-sdl2
> > > --enable-libdc1394 --enable-libiec61883 --enable-chromaprint
> > > --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
> > > libavutil 55. 34.101 / 55. 34.101
> > > libavcodec 57. 64.101 / 57. 64.101
> > > libavformat 57. 56.101 / 57. 56.101
> > > 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
> > > nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0
> > >
> > > Then it just sits there. No audio, but that's not surprising since
> > normally
> > > I'd have to pipe the audio out through jack (no doable with ffmpeg, to
> my
> > > knowledge). So I re-enabled the integrated audio to see if it might
> > > automatically pipe the audio out to that device, but I get the same
> > output
> > > noted above and still no audio. When I run the same command on server.
> I
> > > get the exact same output on the new ssh prompt I opened up. Even if I
> > > change the ip address to 'localhost'. Although, on the ssh prompt
> where I
> > > started up ffserver from I get the following output pop up along with
> the
> > > constantly updating bitrate...
> > >
> > > Wed Oct 30 14:37:35 2018 [NULL @ 0x128cfe0]Opening '/tmp/feed1.ffm' for
> > > reading
> > > Wed Oct 30 14:37:35 2018 [file @ 0x127be20]Setting default whitelist
> > > 'file,crypto'
> > > Wed Oct 30 14:37:35 2018 [ffm @ 0x128cfe0]Format ffm probed with
> > size=2048
> > > and score=101
> > > Wed Oct 30 14:37:35 2018 [NULL @ 0x128e8a0]Setting entry with key
> > 'strict'
> > > to value '-2'
> > > Wed Oct 30 14:37:35 2018 [NULL @ 0x128e8a0]Setting entry with key 'ac'
> to
> > > value '2'
> > > Wed Oct 30 14:37:35 2018 [NULL @ 0x128e8a0]Setting entry with key 'b'
> to
> > > value '128000'
> > > Wed Oct 30 14:37:35 2018 [NULL @ 0x128e8a0]Setting entry with key 'ar'
> to
> > > value '48000'
> > > Wed Oct 30 14:37:35 2018 [NULL @ 0x128e8a0]Setting entry with key
> 'flags'
> > > to value '+global_header'
> > > Wed Oct 30 14:37:35 2018 [rtp @ 0x128f440]No default whitelist set
> > > Wed Oct 30 14:37:35 2018 [udp @ 0x1290c20]No default whitelist set
> > > Wed Oct 30 14:37:35 2018 [udp @ 0x1290d60]No default whitelist set
> > > Wed Oct 30 14:37:35 2018 192.168.1.119:31830 - - "PLAY
> > > test1-rtsp.ogg/streamid=0 RTP/UDP"
> > > Wed Oct 30 14:37:35 2018 Failed to parse interval end specification ''
> > >
> > > I couldn't find much on the final line there about the "Failed to parse
> > > interval end specification ''" or of a way to fix it I mean. I've tried
> > > adding -analyzeduration and -probesize on both the ffmpeg command that
> > > uploads the audio to the feed, and also on the ffplay command that
> should
> > > play the feed. They did absolutely nothing to change anything.
> > >
> > > I'm pretty sure it's not firewall related. ufw and iptables are not
> > > installed.
> > > root at test-9:~# iptables
> > > -su: iptables: command not found
> > > root at test-9:~# ufw
> > > -su: ufw: command not found
> > >
> > > It feels like I'm missing some config option somewhere, but I can't
> > figure
> > > out where. All I see everywhere are examples of people setting up
> streams
> > > with webcams or other video related formats and very little that just
> > deals
> > > with audio only, nothing has worked for me so far. I'm open to any and
> > all
> > > suggestions. Thank you for your time.
> > >
> > > --
> > > Brett
> > > <rtsp server status.PNG>_______________________________________________
> > > ffmpeg-user mailing list
> > > ffmpeg-user at ffmpeg.org
> > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> > >
> > > To unsubscribe, visit link above, or email
> > > ffmpeg-user-request at ffmpeg.org with subject "unsubscribe".
> >
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> > To unsubscribe, visit link above, or email
> > ffmpeg-user-request at ffmpeg.org with subject "unsubscribe".
>
>
>
> --
> Brett
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
> To unsubscribe, visit link above, or email
> ffmpeg-user-request at ffmpeg.org with subject "unsubscribe".
More information about the ffmpeg-user
mailing list