[FFmpeg-user] Issues Live Streaming Audio Only
DopeLabs
dopelabs at dubstep.fm
Wed Oct 31 23:45:51 EET 2018
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@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>_______________________________________________
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>
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