How to broadcast to Icecast2 by using ffmpeg (FLAC,Opus,Vorbis,AAC,MP3/Windows, Mac)

README.md

FFMpeg to Icecast2 Streaming Samples

Examples usage of various codecs with FFMpeg.

Samples

• flac.sh : An Icecast Source Client
  • for Windows (Cygwin is required)
  • and macOS (brew install ffmpeg)
• another_examples.sh : Samples
  • FFMpeg can push to Icecast2 in various formats: Opus/Vorbis/AAC/MP3
  • this script shows optimal format, container and codec combinations.
  • Recommended settings for stable streaming with good quality:
    • HE-AAC (aac_he): 48k-64k
    • HE-AACv2 (aac_he_v2): 32k-48k
    • LC-AAC VBR 3-4
      • Higer is good quality, increases bitrate
      • if you want to use CBR, set 96k-128k. (not recommended)
    • Opus VBR 48k-64k
      • CBR is not recommended
    • Vorbis q3
      • Higer is good quality, increases bitrate
    • MP3 V6-V4
      • Lower is good quality, increases bitrate
      • if you want to use CBR, set 128k-160k

Reference

• Icecast Server/Streaming WebM to Icecast with FFmpeg

another_examples.sh

# SAMPLE: encode to another formats

name=stream

# AACs
# AAC streams must be pushed as ADTS stream (-f adts)

# ==== HE-AAC (SBR+PS) @ CBR 48kbps ====
  -c:a libfdk_aac -profile:a aac_he_v2 -ab 48k
  -content_type 'audio/aac'
  -vn -f adts icecast://source:PASSWORD@icecast:8001/$name_aac

# ==== LC-AAC @ VBR 4 ~110kbps ====
#  -c:a libfdk_aac -vbr 4
#  -content_type 'audio/aac'
#  -vn -f adts icecast://source:PASSWORD@icecast:8001/$name_lcaac

# Xiph.org's Ogg Variant

# ==== Ogg Opus @VBR64k ====
# -c:a libopus -vbr on -b:a 64k
# -content_type 'audio/ogg'
# -vn -f opus icecast://source:PASSWORD@icecast:8001/$name_opus

# ==== Ogg Vorbis @ q3 ~112kbps ====
# -codec:a libvorbis -qscale:a 3
# -content_type 'audio/ogg'
# -vn -f ogg icecast://source:PASSWORD@icecast:8001/$name_vorbis

# ==== LAME MP3 @ V6 ~112kbps ====
# -codec:a libmp3lame -qscale:a 6
# -content_type 'audio/mpeg'
# -vn -f mp3 icecast://source:PASSWORD@icecast:8001/$name_mp3

flac.sh

# SAMPLE: FLAC streaming on Windows & Mac

# Input device on Windows
# you should find DShow device name by:
#   ffmpeg.exe -list_devices true -f dshow -i dummy
input="dshow"
device="audio=@device_cm_{33D9A762-90C8-11D0-BD43-00A0C911CE86}\wave_{D50ABAB1-D542-4F19-BB77-D12FADCAB889}"

# Input device on macOS
# setup a default device at System Preferences > Sound > Input
# input="avfoundation"
# device="none:default"

channels=2

samplerate=48000

codec="flac"

# above 12 is not recommended
# if you have a slow hardware, set lower value.
level=10

while true
do
  ./ffmpeg -f $input \
         -i $device \
         -ar $samplerate \
         -ac $channels \
         -c:a $codec -compression_level $level \
         -f ogg \
         -content_type 'application/ogg' \
         icecast://source:PASSWORD@HOSTNAME:8001/STREAM_NAME
  sleep 1
done

@mrx23dot Are you running ffmpeg multiple times? If you're actually trying to make an internet radio station, it may make more sense to use a tool like Mixxx which is meant for doing that.

@keiya For low latency, the manpage for ffmpeg says it has support for the Real-Time Messaging Protocol (RTMP). I use mpv to play rtmp streams, so vlc should also work. If RTMP isn't what you want, ffmpeg also supports a bunch of other live-streaming formats: Real-Time Streaming Protocol (RTSP), Secure Reliable Transport (SRT), Real-Time Transport Protocol (RTP), and HTTP Live Streaming (HLS). The only notable absence is WebRTC.

The manpage gives some examples of how to use RTMP:

Protocol syntax

rtmp://[username:password@]server[:port][/app][/instance][/playpath]

To publish to a password protected server, passing the playpath and app names separately:

ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/

To read with ffplay a multimedia resource named "sample" from the application "vod" from an RTMP server "myserver":

ffplay rtmp://myserver/vod/sample