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ffserver(1) — ffmpeg — Debian experimental — Debian Manpages
https://manpages.debian.org/experimental/ffmpeg/ffserver.1.en.html
ffserver is a streaming server for both audio and video. It supports several live feeds, streaming from files and time shifting on live feeds. You can seek to positions in the past on each live feed, provided you specify a big enough feed storage. ffserver is configured through a configuration file, which is read at startup.
ffmpeg/ffserver.conf at master · fhunleth/ffmpeg · GitHub
https://github.com/fhunleth/ffmpeg/blob/master/doc/ffserver.conf
# Number of audio channels: 1 = mono, 2 = stereo AudioChannels 1 # Sampling frequency for audio. When using low bitrates, you should # lower this frequency to 22050 or 11025. The supported frequencies # depend on the selected audio codec. AudioSampleRate 44100 # Bitrate for the video stream VideoBitRate 64 # Ratecontrol buffer size
ffserver.config - GitHub
https://gist.github.com/thejamespinto/545a43233fdbf6a9efca
# Specify launch in order to start ffmpeg automatically. # First ffmpeg must be defined with an appropriate path if needed, # after that options can follow, but avoid adding the http:// field # Launch ffmpeg args # Only allow connections from localhost to the feed. #ACL allow 127.0.0.1 </Feed> <Stream index.html> Feed feed1.ffm Format mpjpeg
Could not get audio( libopus ) stream from ffserver via rtp
https://stackoverflow.com/questions/44975398/could-not-get-audio-libopus-stream-from-ffserver-via-rtp
I need to get the rtp stream with the next codecs vp8,opus. I get a stream from a camera (h264, aac), then I send to ffmpeg,then I send it to the ffserver and from there I want to receive a stream in the format rtp. Video (VP8) without audio i successfully get, but stream video with audio ( any codecs ( aac, vorbis, opus ) ) not work. ffmpeg ...
rtsp - Is there anyone who can ... - Stack Overflow
https://stackoverflow.com/questions/37403282/is-there-anyone-who-can-successfully-run-real-time-streaming-with-ffserver
You have to add audio codec parameters to ffserver configuration file. And add input to ffmpeg (-f alsa -i pulse). Should work! – Murad Tagirov. May 25 '16 at 13:28. 1. I'm trying like this "AudioCodec aac \ AudioBitRate 32 \ AudioSampleRate 44100 \ AVOptionAudio flags +global_header" changing codecs and option but it doesn't work..
[SOLVED] ffmpeg: Unknown encoder 'libmp3lame'
https://www.linuxquestions.org/questions/linux-software-2/ffmpeg-unknown-encoder-%27libmp3lame%27-4175598322/
For years, when I would download audio files encoded FLAC or other uncompressed formats, I would convert them to CD quality 128 kbs mp3 solely for the sake of file size and the fact that rumor had it most people cannot distinguish the difference in quality of sound between the compressed mp3 and uncompressed formats.
mumudvb crashes when streaming to ffserver via ffm ...
https://github.com/braice/MuMuDVB/issues/14
ERRO: Transcode : Output format: FFM (FFserver live feed) format Input #0, ffm, from 'http://192.168.1.1:8095/feed1.ffm': Duration: N/A, bitrate: N/A Stream #0.0: Audio: mp2, 44100 Hz, mono, s16, 32 kb/s Stream #0.1: Video: mpeg1video, yuv420p, 160x128, 64 kb/s, 1000k tbn, 3 tbc Stream #0.2: Audio: mp2, 22050 Hz, mono, s16, 64 kb/s
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