Issues Live Streaming Audio Only
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@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@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@test-9:~# /usr/bin/dbus-run-session ffmpeg -f jack -i ffmpeg -y output.wav root@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@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@test-9:~# *ffserver -loglevel debug &* [1] 11631 root@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@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@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@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@test-9:~# iptables -su: iptables: command not found root@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
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@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@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@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@test-9:~# /usr/bin/dbus-run-session ffmpeg -f jack -i ffmpeg -y output.wav root@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@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@test-9:~# *ffserver -loglevel debug &* [1] 11631 root@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@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@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@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@test-9:~# iptables -su: iptables: command not found root@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@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
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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@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@remote.host.com/mount
remote machine: ffmpeg -i http://remote.host/mount
On Oct 31, 2018, at 2:23 15PM, Brett Garrett < brett@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@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@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@test-9:~# /usr/bin/dbus-run-session ffmpeg -f jack -i ffmpeg -y output.wav root@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@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@test-9:~# *ffserver -loglevel debug &* [1] 11631 root@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@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@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@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@test-9:~# iptables -su: iptables: command not found root@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@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
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-- Brett
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@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@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@remote.host.com/mount
remote machine: ffmpeg -i http://remote.host/mount
On Oct 31, 2018, at 2:23 15PM, Brett Garrett < brett@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@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@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@test-9:~# /usr/bin/dbus-run-session ffmpeg -f jack -i ffmpeg -y output.wav root@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@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@test-9:~# *ffserver -loglevel debug &* [1] 11631 root@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@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@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@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@test-9:~# iptables -su: iptables: command not found root@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@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
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participants (3)
-
Brett Garrett -
DopeLabs -
WTI TEC