I am trying to get started using Haskell's Euterpea library. My first goal was to get it to play a given sound file (e.g. mp3 or wav), but first I ran into an issue following instructions to get it to just play a simple note sound in ghci.
Following the "Setting up MIDI" instructions at Euterpea's web page, I ran
import Euterpea
play $ c 4 qn
in ghci. The 'play' command resulted in the following error message:
Prelude Euterpea> play $ c 4 qn
*** Exception: No MIDI output device found
CallStack (from HasCallStack):
error, called at ./Euterpea/IO/MIDI/MidiIO.lhs:122:18 in Euterpea-2.0.2-Iz37iWlkpjn2emP4FnvOI1:Euterpea.IO.MIDI.MidiIO
I thought I needed to specify midi output to my machine (macOS Sierra) and found an application called 'Audio MIDI Setup', but it showed that a midi output (my internal speakers) was already specified.
Anyone know what this issue is or how to fix it?
Perhaps you solved this, but for posterity some ideas:
Sounds like you didn't install and run a MIDI synth (e.g. SimpleSynth) first. AFAICT, Audio MIDI Setup doesn't actually include a software synthesizer, it's more for advanced / hardware MIDI setup.
This should create the MIDI output devices that Euterpea couldn't find there. You may also need to play around with channels (e.g. use playDev n instead of play and work out a value for n from your device list... or even just try 1 through 8)
Either way there's some good Mac-focussed help on Donya's working with MIDI on Mac OS X page. Hope that helps.
Related
I am trying to create snapshots from a video stream using the "scene" video filter. I'm on Windows for now, but this will run on Linux I don't want the video output window to display. I can get the scenes to generate if I don't use the --vout=dummy option. When I include that option, it does not generate the scenes.
This example on the Wiki indicates that it's possible. What am I doing wrong?
Here is the line of code from the LibVLCSharp code:
LibVLC libVLC = new LibVLC("--no-audio", "--no-spu", "--vout=dummy", "--video-filter=scene", "--scene-format=jpeg", "--scene-prefix=snap", "--scene-path=C:\\temp\\", "--scene-ratio=100", $"--rtsp-user={rtspUser}", $"--rtsp-pwd={rtspPassword}");
For VLC 3, you will need to disable hardware acceleration which seems incompatible with the dummy vout.
In my tests, it was needed to do that on the media rather than globally:
media.AddOption(":avcodec-hw=none");
I still have mainy "Too high level or recursion" errors, and for that, I guess you'd better open an issue on videolan's trac.
I have the following problem:
I have experiments that ran on a computer with the PsychoPy Version 1.83.04.
Due to some problems, the computer was reconfigured, hence also PsychPy updated to Version 1.85.4. Since then, none of my experiments with a sound included work anymore. The error message is as follows:
Following an advice in another forum, I changed the files of
- psychopy/sound/init.py
- psychopy/sound/backend_pyo.py
(as suggested also in here).
But unfortunately, this did not work neither. I get a new error message that is
I also tried with the old PsychoPy-Version 1.83.04 - here the experiment at least starts before breaking down. The error message then states that the lxml version is too old to be used with openpyxl and hence cannot open the xlsx-condition file. But there seems to be no problem with the sound device with the old version.
My sound device is as follows:
Description: Lautsprecher (Realtek High Definition Audio)
Default Sound Playback: Yes
Default Voice Playback: Yes
Hardware ID:
HDAUDIO\FUNC_01&VEN_10EC&DEV_0671&SUBSYS_173411EB&REV_1000
Manufacturer ID: 1
Product ID: 100
Type: WDM Driver
Name:
RTKVHD64.sys
Driver Version: 6.00.0001.8186 (English) Driver
Attributes: Final Retail WHQL
Logo'd: Yes
Date and Size: 6/29/2017
18:52:00, 5826560 bytes
What else can I try?
Thank you for your help.
I finally managed to solve the problem - at least superficially.
Following this advice I set PsychoPy back to version 1.85.2. But then, the experiment crushed with a "Fatal Python error: (pygame parachute)..." error message.
Finally, I found the hint where to "switch off" pyo (it was found here): in PsychoPy in File -> Preferences and then delte pyo from the audio library.
Now, my experiments are running again.
Attempting to play a signal through computer speakers via python so I tried the example in 7.1 of the manual. I see the signal on the plot but hear nothing over the speakers. Is SoundSink the wrong approach for this? I'm running REDHAWK 2.0.6 on CentOS 7. In case this is important, the first time sb.start() is called, "shared memfd open() failed: Invalid argument" is displayed. When called a second time that message doesn't appear. I am able to play audio from within the REDHAWK IDE.
from ossie.utils import sb
import frontend
sim = sb.launch("rh.FmRdsSimulator")
demod=sb.launch("rh.AmFmPmBasebandDemod")
filter=sb.launch("rh.fastfilter")
resample=sb.launch("rh.ArbitraryRateResampler")
agc=sb.launch("rh.agc")
sink=sb.SoundSink()
plot=sb.Plot()
sim.connect(demod)
demod.connect(filter, usesPortName="fm_dataFloat_out")
filter.connect(resample)
resample.connect(agc)
agc.connect(sink)
agc.connect(plot)
sim.addAWGN=False
demod.freqDeviation=15000.0
filter.filterProps.freq1=16000.0
filter.filterProps.Ripple=0.5
filter.filterProps.Type="lowpass"
resample.outputRate=32000.0
sb.start()
alloc = frontend.createTunerAllocation(
"RX_DIGITIZER",
allocation_id="testing",
center_frequency=100.1e6,
sample_rate=256e3,
sample_rate_tolerance=20.0)
sim.allocateCapacity(alloc)
The package that SoundSink uses to generate sound has been deprecated in CentOS 7 (but is still available in CentOS 6). This issue has been added to the backlog and will be fixed in a future release.
I use the audio class to read MP3 file thanks to a little trick : replacing the ffmpegsumo.so of Node-Webkit with the chromium one. This enable the reading of MP3 on Windows but doesn't works on Mac OS. Does anyone know why ?
Here's the code :
player = new Audio()
player.src = '/path/to/the/audio.mp3';
player.play();
This seems to be dependant upon the dll/so being a 32 bit version. I am guessing that is why copying the file from Chrome doesn't work correctly for most people ( my 3 year old phone is the only 32-bit device I have left ).
I keep seeing this link --
https://github.com/rogerwang/node-webkit/wiki/Support-mp3-and-h264-in-video-and-audio-tag
.. but it is a blank page. I am guessing it was deleted since the info was likely not current or correct.
This issue thread has links to some rebuilt ffmpegsumo libraries for both Mac and Windows --
https://github.com/rogerwang/node-webkit/issues/1423
The alternative appears to be rebuilding ffmpegsumo, this thread has some config for doing that -- https://github.com/rogerwang/node-webkit/issues/1208
I am still confused about the licensing on it after you build the library, so that is probably worth some research. Everything about mpeg4-part10 is copyrighted and heavily patent encumbered. I think we all need to get smart enough to stop using mp4/h.264. Before I got this working correctly on node-webkit, it was easier to use ffmpeg to transcode the video to an ogv container using Theora and Vorbis codecs. At this point it seems like iOS is keeping h.264 alive, when it should probably die the horrible death it has earned.
simple synchronous loading of an audio file (.mp3) in a cocos2d app makes my vmware disconnect the sound.
the error is display bottom right, saying 'error in creating sound stream; sound is disconnected'
i read that it might be cause of my vmware's version (mine is 8) but I'm looking for a fix, not to downgrade to another version.
before i get that error, the sound on the system works just fine (youtube, etc)
the exact code im calling is..
[CDSoundEngine setMixerSampleRate: CD_SAMPLE_RATE_MID];
[[CDAudioManager sharedManager] setResignBehavior: kAMRBStopPlay autoHandle:Yes];
soundEngine = [SimpleAudioEngine sharedEngine];
[soundEngine preloadBackgroundMusic:#"somemp3.mp3"];
[soundEngine playBackgroundMusic:#"somemp3.mp3"];
maybe the bit rate is too high .. ?
thanks
There was a problem in the vmware..