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We never change the I/O directly, so avoid the expensive syscall and
only use a slightly-expensive Ruby method call instead (the Ruby
method call is fixed cost in either case).
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We do not need this for single sink situations (the common case)
at all. We also do not need to check IO#nread for splice, either;
we can just do non-blocking I/O. The only common path where we
might still need it is the non-splice case with multiple sinks.
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This prevents us from resetting the tracklist when we
add new tracks and want to restart playback.
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We must define the variable for tracking buffer length.
This only affected non-Linux kernel users.
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The addition of a "need_to_queue" method should reduce the amount of
cognitive overhead required to parse these conditions
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Some users may not use nor care about $OUTFMT at all, so
just start them out with a default one based on the original
to get started.
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Oops.
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This gives us consistency with the "cue prev" command,
is easier-to-type, and is consistent in length with "tl next".
We'll just map "previous" -> "tl prev" when we implement the
MPRIS bridge.
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This adds the ability to seek internally within FLAC file
based on the internal CUE sheet. Other formats may be supported
in the future, but FLAC is the only one I know of which supports
embedded cue sheets.
Note: flac 1.3.0 is recommended for users of non-CDDA-compatible
formats.
See updates to dtas-player_protocol(7) for details.
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Using "tl goto" implies we start playback of the player is idle.
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This may be used to avoid automatic:
* resampling (rate)
* down/upmixing (channel)
* dither/truncation (bits)
Using any bypass mode means we can no longer guarantee gapless
playback for audio collections where rate, channel, or bits vary.
This can however be useful when CPU usage is too high. This may
also be useful in audio engineering situations.
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This is an internal class, and we don't have a public Ruby API
anyways.
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sox(1) may gain the ability to natively encode to Opus one day
without using opusenc(1), so make it more explicit we are relying
on opusenc(1).
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We should not leave sinks running when nothing is playing,
since that blocks the sound device from being used by others.
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We do not explicitly resample/dither/downmix without users
permission.
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This allows skipping periods of silence/noise in between music tracks.
This should be useful if the recorder is left running during
intermission or during equipment swaps.
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When a player is idle and a track is added to an empty tracklist,
we should not repeat the first track added to the tracklist. Avoid
that by advancing the tracklist to the current track.
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These are common output targets, at least for my workflow.
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We should only reset the tracklist if the user has completely
iterated through the list of tracks to be played.
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We have no public Ruby API, only socket protocols and data formats.
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This is necessary to handle the case where the tracklist is empty,
clients get confused and timeout the response if we attempt to emit
an empty string.
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"wall" is analogous to the wall(1) command, so we shall use that
instead of echo.
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Because sometimes a song is just stuck in our head.
Or MPRIS 2.0 wants us to implement it this way...
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Non-repeating tracklists should stop playing when there's nothing
to go back to.
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This should make implementing SetPosition in the MPRIS 2.0 spec
possible.
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This means we can go back and forth in the tracklist like a normal
music player. This will allow an easier MPRIS 2.0 implementation.
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This is a more accurate depiction of what happens,
and we'll implement "next" and "previous" commands in the future.
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We should return to the starting position of the tracklist if we
are idle.
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Otherwise we end up constantly pushing tracks to the top of the
queue and getting surprising behavior if seek is called repeatedly.
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We already flush the currently playing track into the head of the
queue upon player exit (even if it was in the @tl), so we should use
@tl.next_track as usual instead of @tl.cur_track in case the queue
is empty.
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We need to preserve the go_to-specified position for next_track,
doing otherwise would cause us to always be off-by-one.
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This allows easier scripting if we want to add a bunch of tracks
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This should allow us to repeat through a list of tracks with relative
ease. There is a rudimentary dtas-tl client implemented. This
may be removed in the future.
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This should allow us to eventually implement a MPRIS 2.0-compliant
tracklist.
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Minor bugfixes, this allows users to setup targets easily
without introducing them to the ":" Rubyism for symbols.
Also, use "track_start" instead of "track_first" to match
the existing published examples for numbering the first
track.
While we're at it, detect the decoded sample precision
correctly for dither use.
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Having it return nil in a noop function seems wrong.
We can't silently discard the value (unlike pipe_size=)
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We do not need it anymore since we only write to the targets
returned by Sink#spawn.
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This seems to be working out nicely. Having a basic integration
test should be enough to get us started for now.
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This will allow users to more-easily edit configs and feel
like a real shell. We no longer mistakenly expand nil env
variables to "" anymore, either.
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Rounding should be more accurate, even though my original awk
snippet truncated the output.
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This is lacking tests and documentation, but it works from
a old trivial sample I had from a recording I previously
split using plain POSIX shell
splitfx is like make(1) for splitting and minor audio
editing. It also allows any number of effects.
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Singleton methods tend to be bad like this.
TODO: write tests for this.
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This allows users to display the current env value for introspection
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This allows me to hit Ctrl-C on a dtas-player(1) process, wait on
termination of the player, and immediately restart it without
worrying about sink conflicts upon restart.
Before this change, sinks would continue running for a bit
(depending on buffer sizes).
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stop_sinks is already defined, so avoid repeating this loop
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This helps in case dtas-player is hit with SIGKILL or the system
crashes. This does not fsync(2) as that could introduce delays on
slow filesystems. Users should open the file manually and fsync
themselves if they need to.
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Some containers (e.g. large VOBs) are not easily probed and require
additional options for avprobe/ffprobe to find audio streams. We do
this by looping and increasing the duration/size of the probe to
find new audio streams.
This seems to work reasonably well for some DVD rips I have until
seeking is required. This breaks if the seek point (including seeks
for source effects) exceeds the avprobe/ffprobe -analyzeduration.
Anyways, I recommend extracting the audio stream (without
transcoding) out of the VOB container as the best way to go.
Something like:
avconv -analyzeduration 2G -probesize 2G \
-i input.vob -vn -sn -c:a copy -map 0:$STREAM_NR output.ext
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The whole dtas project is committed to giving as much rope as users
need!
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This is reproducible on a video file with a mono audio stream
when attempting playback in stereo.
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No need to add the vol effect if it's going to be a noop.
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