150 lines
5.0 KiB
TOML
150 lines
5.0 KiB
TOML
# Kawa-compatible configuration file with extensions
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[api]
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#
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# The HTTP port the MeteorLight API listens on.
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port=4040
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# The HTTP bind address API will listen on.
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host="127.0.0.1"
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[queue]
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#
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# An HTTP GET is sent to this URL when MeteorLight's queue is empty and it needs a new
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# random track to play. The expected response is an arbitrary JSON blob that
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# MeteorLight stores in its queue. At a minimum, it must include the "path" property:
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#
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# {
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# "path": "/path/to/audio/file"
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# }
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#
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# The path is the path to an audio file on the filesystem you want MeteorLight to play.
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# It can also be an http(s) URL, that supports Range requests and returns proper Content-Length.
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# Additionally, the "title", "artist" and "art" properties can be included to be used as metadata.
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# If "title", "artist" are not specified, file tags may be used.
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#
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# If the key "hash" exists and song_fetch_url is set,
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# hash will be appended and downloaded from that location.
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random_song_api="http://localhost:8012/api/random"
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song_fetch_url=""
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#
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# An HTTP POST is issued to this URL when MeteorLight starts playing a track. The body
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# will be identical to the JSON blob in the queue.
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np="http://localhost:8012/api/np"
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#
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# An HTTP POST is issued to this URL when MeteorLight fetches a random track. The body
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# will be identical to the JSON blob in memory.
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# Can be left empty to not send nr events
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nr="http://localhost:8012/api/nr"
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#
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# When no tracks are available for whatever reason (such as external service
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# outages), this track will be played.
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fallback="/tmp/in.flac"
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# Length of buffer to maintain in KiB (not implemented)
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# buffer_len=4096
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# Duration in seconds of buffer to maintain. Set 0 for automatic mode depending on requesting client.
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# Maximum 10 seconds.
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# Do note buffer is counted from end of frame, not start, for removal purposes. This depends on format and can be a second or so at times.
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buffer_duration=0
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#
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# Apply replaygain track tags if existent on files played.
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# If ReplayGain tags are not existent, it will normalize audio with a running weighted window of 5 seconds.
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# ReplayGain tag overrides can be added as properties on track blobs:
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# {
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# "replay_gain" : {
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# "track_peak": 1.0000,
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# "track_gain": -3.15821
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# }
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# }
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replaygain=false
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# Set the sample rate of the queue. Default is 44100
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# If incoming audio is different than this sample rate, it will be resampled.
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# Some codecs (example: Opus) will output at a different samplerates.
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#samplerate=44100
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# Set the sample format and bit depth of the queue. Default is int16 for sample format, and sample format size for bit depth.
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# If incoming audio has a different format / bitdepth than this sample rate, it will be converted.
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# See Kirika's Codec table (Decoder Sample Format) column. Conversion from int32 to int16 is fast.
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# Possible values (plus other aliases): i16, s16, int16, s16le, i32, s32, int32, int, s32le, f32, float, float32, f32le
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#sampleformat=int16
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#bitdepth=0
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[radio]
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#
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# The HTTP port to stream actual audio on.
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port=8001
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# The HTTP bind address to stream actual audio on.
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host="127.0.0.1"
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# Name of the stream.
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name="my radio"
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# Description of the stream.
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description=""
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# URL of the stream.
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url=""
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# Logo URL of the radio
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logo=""
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# Whether to make the radio "private" via headers
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private=false
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#
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# A list of streams to make available at [radio.port]/*(mount) follows. The
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# following properties are available:
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#
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# mount: the HTTP address to serve the stream from.
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#
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# container: the container format to use (ogg, flac, aac/adts, or mp3). See Kirika's supported format list.
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#
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# codec: the audio codec to use (opus, vorbis, flac, aac, or mp3)
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#
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# bitrate: the desired bitrate of the stream in Kb/s, if not specified (or 0) an appropriate
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# bitrate will be automatically selected based on the container/codec
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# MeteorLight extension: bitrate can be a string (for example, v0-v9 on MP3). codec can also be he-aacv2.
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#
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#
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# Additionally further options can be set on some codecs.
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# - FLAC supports bitdepth (int) default 16, compression_level (int) default 8, block_size (int) default 0 = auto
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# - Opus supports complexity (int) default 10
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# - AAC supports VBR bitrate (vbr1 - vbr5), mode (lc, he, hev2) and afterburner (bool)
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# - MP3 supports VBR bitrate (v0 - v9)
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# - All but Opus support setting samplerate (int), default queue sample rate. Opus is fixed at 48kHz.
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# - All support setting offset_start (bool) default true, rewrites packets PTS to align with the client start time.
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# - All support setting channels (int) to 1 or 2, default 2
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[[streams]]
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mount="stream128.mp3"
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container="mp3"
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bitrate=128
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[[streams]]
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mount="stream192.mp3"
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container="mp3"
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bitrate=192
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[[streams]]
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mount="stream128.aac"
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codec="aac"
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container="aac"
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bitrate=128
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[[streams]]
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mount="stream128.opus"
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container="ogg"
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codec="opus"
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bitrate=128
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[[streams]]
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mount="stream192.vorbis"
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container="ogg"
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codec="vorbis"
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bitrate=192
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[[streams]]
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mount="stream256.opus"
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container="ogg"
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codec="opus"
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bitrate=256
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[[streams]]
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mount="stream.flac"
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codec="flac"
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# Also supports ogg
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container="flac" |