Interesting. Are you really sure LMS stops and MPD skips? Because that really makes no sense...
If you were not so sure, we can prepare a few minutes of repeated sweep in dsf using latest sox in arch https://aur.archlinux.org/packages/sox-dsd-git/ . The 2qute output can be recorded with your internal soundcard in audacity where you can easily check afterwards whether the playback skipped or stopped.
If you were not so sure, we can prepare a few minutes of repeated sweep in dsf using latest sox in arch https://aur.archlinux.org/packages/sox-dsd-git/ . The 2qute output can be recorded with your internal soundcard in audacity where you can easily check afterwards whether the playback skipped or stopped.
I'm not sure. It's only 0.5sec once every hour, and As you say it make no sense.
Of these 5 runs I have skips in all of them
Lets try the test, what to do ??? I'm preparing the sox-dsd-git from AUR.
Last edited:
sox: SoX v14.4.2
Code:
Usage summary: [gopts] [[fopts] infile]... [fopts] outfile [effect [effopt]]...
SPECIAL FILENAMES (infile, outfile):
- Pipe/redirect input/output (stdin/stdout); may need -t
-d, --default-device Use the default audio device (where available)
-n, --null Use the `null' file handler; e.g. with synth effect
-p, --sox-pipe Alias for `-t sox -'
SPECIAL FILENAMES (infile only):
"|program [options] ..." Pipe input from external program (where supported)
[url=http://server/file]404 Not Found[/url] Use the given URL as input file (where supported)
GLOBAL OPTIONS (gopts) (can be specified at any point before the first effect):
--buffer BYTES Set the size of all processing buffers (default 8192)
--clobber Don't prompt to overwrite output file (default)
--combine concatenate Concatenate all input files (default for sox, rec)
--combine sequence Sequence all input files (default for play)
-D, --no-dither Don't dither automatically
--dft-min NUM Minimum size (log2) for DFT processing (default 10)
--effects-file FILENAME File containing effects and options
-G, --guard Use temporary files to guard against clipping
-h, --help Display version number and usage information
--help-effect NAME Show usage of effect NAME, or NAME=all for all
--help-format NAME Show info on format NAME, or NAME=all for all
--i, --info Behave as soxi(1)
--input-buffer BYTES Override the input buffer size (default: as --buffer)
--no-clobber Prompt to overwrite output file
-m, --combine mix Mix multiple input files (instead of concatenating)
--combine mix-power Mix to equal power (instead of concatenating)
-M, --combine merge Merge multiple input files (instead of concatenating)
--magic Use `magic' file-type detection
--multi-threaded Enable parallel effects channels processing
--norm Guard (see --guard) & normalise
--play-rate-arg ARG Default `rate' argument for auto-resample with `play'
--plot gnuplot|octave Generate script to plot response of filter effect
-q, --no-show-progress Run in quiet mode; opposite of -S
--replay-gain track|album|off Default: off (sox, rec), track (play)
-R Use default random numbers (same on each run of SoX)
-S, --show-progress Display progress while processing audio data
--single-threaded Disable parallel effects channels processing
--temp DIRECTORY Specify the directory to use for temporary files
-T, --combine multiply Multiply samples of corresponding channels from all
input files (instead of concatenating)
--version Display version number of SoX and exit
-V[LEVEL] Increment or set verbosity level (default 2); levels:
1: failure messages
2: warnings
3: details of processing
4-6: increasing levels of debug messages
FORMAT OPTIONS (fopts):
Input file format options need only be supplied for files that are headerless.
Output files will have the same format as the input file where possible and not
overridden by any of various means including providing output format options.
-v|--volume FACTOR Input file volume adjustment factor (real number)
--ignore-length Ignore input file length given in header; read to EOF
-t|--type FILETYPE File type of audio
-e|--encoding ENCODING Set encoding (ENCODING may be one of signed-integer,
unsigned-integer, floating-point, mu-law, a-law,
ima-adpcm, ms-adpcm, gsm-full-rate)
-b|--bits BITS Encoded sample size in bits
-N|--reverse-nibbles Encoded nibble-order
-X|--reverse-bits Encoded bit-order
--endian little|big|swap Encoded byte-order; swap means opposite to default
-L/-B/-x Short options for the above
-c|--channels CHANNELS Number of channels of audio data; e.g. 2 = stereo
-r|--rate RATE Sample rate of audio
-C|--compression FACTOR Compression factor for output format
--add-comment TEXT Append output file comment
--comment TEXT Specify comment text for the output file
--comment-file FILENAME File containing comment text for the output file
--no-glob Don't `glob' wildcard match the following filename
AUDIO FILE FORMATS: 8svx aif aifc aiff aiffc al amb amr-nb amr-wb anb au avr awb caf cdda cdr cvs cvsd cvu dat dff dsf dvms f32 f4 f64 f8 fap flac fssd gsm gsrt hcom htk ima ircam la lpc lpc10 lu mat mat4 mat5 maud mp2 mp3 nist ogg opus paf prc pvf raw s1 s16 s2 s24 s3 s32 s4 s8 sb sd2 sds sf sl sln smp snd sndfile sndr sndt sou sox sph sw txw u1 u16 u2 u24 u3 u32 u4 u8 ub ul uw vms voc vorbis vox w64 wav wavpcm wsd wv wve xa xi
PLAYLIST FORMATS: m3u pls
AUDIO DEVICE DRIVERS: alsa ao oss ossdsp pulseaudio
EFFECTS: allpass band bandpass bandreject bass bend biquad chorus channels compand contrast dcshift deemph delay dither divide+ dop downsample earwax echo echos equalizer fade fir firfit+ flanger gain highpass hilbert input# ladspa loudness lowpass mcompand noiseprof noisered norm oops output# overdrive pad phaser pitch rate remix repeat reverb reverse riaa sdm silence sinc spectrogram speed splice stat stats stretch swap synth tempo treble tremolo trim upsample vad vol
* Deprecated effect + Experimental effect # LibSoX-only effect
EFFECT OPTIONS (effopts): effect dependent; see --help-effectI do not know if that sox-dsf can play DoP, actually I doubt it. But the idea was using sox for generating our custom dsf file which could help with identifying whether squeezelite skips or pauses.
I think if you generate long dsf file with repeated sweep up and down and up and down, record chord output signal with internal soundcard into audacity, and scroll to the place where the fault is, you should be able to tell whether it is a skip or pause - just show spectrum of/listen to a short section before and after. Or cut out the delay/join the two parts and listen. If the frequency is continuous for the two sections, it is a pause, if there is a frequency skipped - it is clearly a skip.
We can generate the sweep first and then convert to dsf. Please first try converting any flac/wav to dsf using your compiled sox and play with mpd/squeezelite to test the dsf conversion actually works.
According to the patch author https://www.computeraudiophile.com/...ncoding-with-sox/?tab=comments#comment-456522 the command should be:
Generating the sweep:
This will join the two sweeps and do 29 repetitions - resulting in 600 secs of sweeping:
You may want to make the sweep even longer (one hour).
Afterwards you can convert the long flac to dsf using the command tested above (will probably take some time).
I think if you generate long dsf file with repeated sweep up and down and up and down, record chord output signal with internal soundcard into audacity, and scroll to the place where the fault is, you should be able to tell whether it is a skip or pause - just show spectrum of/listen to a short section before and after. Or cut out the delay/join the two parts and listen. If the frequency is continuous for the two sections, it is a pause, if there is a frequency skipped - it is clearly a skip.
We can generate the sweep first and then convert to dsf. Please first try converting any flac/wav to dsf using your compiled sox and play with mpd/squeezelite to test the dsf conversion actually works.
According to the patch author https://www.computeraudiophile.com/...ncoding-with-sox/?tab=comments#comment-456522 the command should be:
Code:
sox file.flac file.dsf rate -v 2822400 sdm -f hqGenerating the sweep:
Code:
sox -n -c 2 -r 44100 -b 24 sweep-up.flac synth 10 sine 100-15000
sox -n -c 2 -r 44100 -b 24 sweep-down.flac synth 10 sine 15000-100
Code:
sox sweep-up.flac sweep-down.flac sweep.flac repeat 29You may want to make the sweep even longer (one hour).
Afterwards you can convert the long flac to dsf using the command tested above (will probably take some time).
Or just put repeat on the player and your dsf sweep file can be much shorter
427 Mbyte and 10 mins of sweeping ..
BTW you can identify the exact silence position in the long recorded file e.g. using ffmpeg https://stackoverflow.com/questions...ng-timestamps-of-periods-of-silence-using-sox
Jeps , my 428 Liter Tannoy HPD - 385 can play 25 hz without any problems...
It seems to me that sox can do the dop I got this from their mailling list
https://sourceforge.net/p/sox/mailman/message/34464903/
I think you're better to see if it's true.
It seems to me that sox can do the dop I got this from their mailling list
https://sourceforge.net/p/sox/mailman/message/34464903/
I think you're better to see if it's true.
The Last one for today
I have send this question about sox and dop to their mailing list, so maybe someone
knows if it's possible.
https://sourceforge.net/p/sox/mailman/sox-users/?viewmonth=201708
Thanks for a good day with a lot of thoughts and tries.
I have send this question about sox and dop to their mailing list, so maybe someone
knows if it's possible.
https://sourceforge.net/p/sox/mailman/sox-users/?viewmonth=201708
Thanks for a good day with a lot of thoughts and tries.
Saw your queestion and the answer i Yes , works fine in squeezelite...
Would like the dop parameter to work in sox so I can try another program
which can do the dsp dop to my Chord 2Qute...
Fantastic This was the real gold...
Code:
sudo sox file.dsf -r 176.4k -b 24 -t alsa plughw:1 dopThanks so much... This gives me DOP white light on the Chord 2QUTE.
hw_params is : format: S32_LE,channels: 2,rate: 176400 (176400/1),period_size: 1024,buffer_size: 8192
Great, that will help us. Now once the dop effect is confirmed to work, we do not need sox for DoP playback. We can store the "DoPed" samples to any wav and play with alsa test-player called aplay which offers the easiest configuration of all needed parameters - period size, buffer size, etc.
Try storing the samples to wav and play with aplay:
BTW why do you need sudo for playback? Is your user in group "audio"? Alsa udev rules are/should be configured so that every audio device created (including usb audio devices) is readable/writeable by the group audio.
Try storing the samples to wav and play with aplay:
Code:
sox file.dsf -r 176.4k -b 24 dop.wav dop
aplay -v -D plughw:1 dop.wavBTW why do you need sudo for playback? Is your user in group "audio"? Alsa udev rules are/should be configured so that every audio device created (including usb audio devices) is readable/writeable by the group audio.
Only because as you write lack of proper setup.
I can still can provoke these fallout just by playing, but sox is more stable than MPD.
If sox has a parameter for have some data in a buffer , it properly would has as few fallout as squeezelite when using the -b parameter.
The only difference I could tell between MPD and squeezelite playback are the requested period/buffer sizes. MPD has them 10times larger (and even more than sox) which somehow results in more dropouts. That is why I think we will need adjusting these parameters fast and flexible for our diagnostics.
That is where aplay excells - it has command-line parameters for all of that, great logging, no overhead for precise time measurement. Unlike sox which has the period/buffer figures basically hardcoded in the source code https://github.com/chirlu/sox/blob/master/src/alsa.c#L141 . That is why I want to use aplay for both PCM as well as DSD-DoP.
That is where aplay excells - it has command-line parameters for all of that, great logging, no overhead for precise time measurement. Unlike sox which has the period/buffer figures basically hardcoded in the source code https://github.com/chirlu/sox/blob/master/src/alsa.c#L141 . That is why I want to use aplay for both PCM as well as DSD-DoP.
The aplay -v -D plughw:1 dop.wav works fine... And It's fallout almost every time.
Chord is in DOP mode with white light...
Code:
Playing WAVE 'testfile.wav' : Signed 24 bit Little Endian in 3bytes, Rate 176400 Hz, Stereo
Plug PCM: Linear conversion PCM (S32_LE)
Its setup is:
stream : PLAYBACK
access : RW_INTERLEAVED
format : S24_3LE
subformat : STD
channels : 2
rate : 176400
exact rate : 176400 (176400/1)
msbits : 24
buffer_size : 88200
period_size : 22050
period_time : 125000
tstamp_mode : NONE
tstamp_type : MONOTONIC
period_step : 1
avail_min : 22050
period_event : 0
start_threshold : 88200
stop_threshold : 88200
silence_threshold: 0
silence_size : 0
boundary : 6206523236469964800
Slave: Hardware PCM card 1 '2Qute' device 0 subdevice 0
Its setup is:
stream : PLAYBACK
access : MMAP_INTERLEAVED
format : S32_LE
subformat : STD
channels : 2
rate : 176400
exact rate : 176400 (176400/1)
msbits : 32
buffer_size : 88200
period_size : 22050
period_time : 125000
tstamp_mode : NONE
tstamp_type : MONOTONIC
period_step : 1
avail_min : 22050
period_event : 0
start_threshold : 88200
stop_threshold : 88200
silence_threshold: 0
silence_size : 0
boundary : 6206523236469964800
appl_ptr : 0
hw_ptr : 0Chord is in DOP mode with white light...
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