You know that you can generate sound put in frequency (Hertz), but how do you create a sound where it will play a set amount of time in random Hertz? Like, you want to play 750Hz in 4 seconds. Is it possible to do that in Lua?
I was thinking about os.execute("echo \7") but that just creates a random beeping. I want to control the sound I create in a set amount of time and a set amount of frequency.
os.execute runs shell commands. Lua on its own cannot create sounds or set frequencies.
So you're actually asking how to beep from your shell for a given amount of time.
This could be achieved by powershell "[console]::beep(<frequency>, <duration_ms>)"
so os.execute('powershell "[console]::beep(4000, 1000)"'
should play a 4kHz sound for 1s.
Alternatively find a Lua library that play configurable tones.
It is possible in LuaJIT
local ffi = require'ffi'
ffi.cdef "int Beep(int Frequency_Hz, int Duration_Milliseconds)"
ffi.C.Beep(750, 4000)
Related
as I said in the title, I need to record my screen from an electron app.
my needs are:
high quality (720p or 1080p)
minimum size
record audio + screen + mic
low impact on PC hardware while recording
no need for any wait after the recorder stopped
by minimum size I mean about 400MB on 720p and 700MB on 1080p for a 3 to 4 hours recording. we already could achieve this by bandicam and obs and it's possible
I already tried:
the simple MediaStreamRecorder API using RecordRTC.Js; produces huge file sizes, like 1GB per hour for 720p video.
compressing the output video using FFmpeg; it can take up to 1 hour for 3 hours recording
save every chunk with 'ondataavailable' event and right after, run FFmpeg and convert and reduce the size and append all the compressed files (also by FFmpeg); there are two problems. 1, because of different PTS but it can be fixed by tunning compress command args. 2, the main problem is the audio data headers are only available in the first chunk and this approach causes a video that only has audio for the first few seconds
recording the video with FFmpeg itself; the end-users need to change some things manually (Stereo Mix), the configs are too complex, it causes the whole PC to work slower while recording (like fps drop; even if I set -threads to 1), in some cases after recording is finished it needs many times to wrap it all up
searched through the internet to find applications that can be used from the command line; I couldn't find much, the famous applications like bandicam and obs have command line args but there are not many args to play with and I can't set many options which leads to other problems
I don't know what else I can do, please tell me if u know a way or simple tool that can be used through CLI to achieve this and guide me through this
I end up using the portable mode of high-level 3d-party applications like obs-studio and adding them to our final package. I also created a js file to control the application using CLI
this way I could pre-set my options (such as crf value, etc) and now our average output size for a 3:30 hour value with 1080p resolution is about 700MB which is impressive
I am working with an application that uses OpenAL API quite extensively. In particular, there are multiple sound sources, non-trivial listener filters, etc.
I want to be able to run this application significantly faster than real-time. At the same time, the sound must be saved for later postprocessing. Is there a way to access the OpenAL output programmatically (virtually) without ever playing the sound on the real playback device?
Ideally, I'd like to have access that would be played during every tick of the main loop of my application. Normally one tick corresponds to one rendered frame (e.g. 1/30th of a second). But in this case we would be running the app as fast as possible.
We ended up using OpenAL Soft to do this. Example:
#include "alext.h"
LPALCLOOPBACKOPENDEVICESOFT alcLoopbackOpenDeviceSOFT;
alcLoopbackOpenDeviceSOFT = alcGetProcAddress(NULL,"alcLoopbackOpenDeviceSOFT");
replace your default device with this device
ALCcontext *context = alcCreateContext(device, attrs);
Set the attrs as you would for your default device
Then in the main loop use:
LPALCRENDERSAMPLESSOFT alcRenderSamplesSOFT;
alcRenderSamplesSOFT = alcGetProcAddress(NULL, "alcRenderSamplesSOFT");
alcRenderSamplesSOFT(device, buffer, 1024);
Here the buffer will store 1024 samples. This code runs faster than real-time, therefore you can sample frames every tick
Are you able to do your required functions with the audio data prior to its being shipped to OpenAL? I've done a lot with javax.sound.sampled when it is untethered by the blocking write() method in SourceDataLine, especially when saving to file rather than playing back.
From what little I know about OpenAL, there is also a blocking process occurs when data is shipped, with a queue of arrays that are managed. I've been meaning to look into this further...
(Probably not being very helpful here. Apologies.)
I would like to manually cache files in Perl, so when playing a sound there is little to no delay.
I wrote a program in Perl, which plays an audio file by doing a system call to VLC. When executing it, I noticed a delay before the audio started playing. The delay is usually between about 1.0 and 1.5 seconds. However, when I create a loop which does the same VLC call multiple times in a row, the delay is only about 0.2 - 0.3 seconds. I assume this is because the sound file was cached by Linux. I found Cache::Cache on CPAN, but I don't understand how it works. I'm interested in a solution without using a module. If that's not possible, I'd like to know how to use Cache::Cache properly.
(I know it's a bad idea to use a system call to VLC regarding execution speed)
use Time::HiRes;
use warnings;
use strict;
while (1) {
my $start = Time::HiRes::time();
system('vlc -Irc ./media/audio/noise.wav vlc://quit');
my $end = Time::HiRes::time();
my $duration = $end - $start;
print "duration = $duration\n";
<STDIN>;
}
Its not as easy as just "caching" a file in perl.
vlc or whatever program needs to interpret the content of the data (in your case the .wav file).
Either you stick with calling an external program and just give it a file to execute or you need to implement the whole stack in perl (and probably Perl XS Modules). By whole stack I mean:
1. Keeping the Data (your .wav file) in Memory (inside the perl runtime).
2. Interpreting the Data inside Perl.
The second part is where it gets tricky you would probably need to write a lot of code and/or use 3rd Party modules to get where you want.
So if you just want to make it work fast, stick with system calls. You could also look into Nama which might give you what you need.
From your Question it looks like you are mostly into getting the runtime of a .wav file. If its just about getting information about the File and not about playing the sound then maybe Audio::Wav could be the module for you.
Cacheing internal to Perl does not help you here.
Prime the Linux file cache by reading the file once, for example at program initialisation time. It might happen that at the time you want to play it, it has already been made stale and removed from the cache, so if you want to guarantee low access time, then put the files on a RAM disk instead.
Find and use a different media player with a lower footprint that does not load dozens of libraries in order to reduce start-up time. Try paplay from package pulseaudio-utils, or gst123, or mpg123 with mpg123-pulse.
I have something feeding commands into a nodejs program through noble (its talking to a BLE device that has a serial service), and I'd like to write a bit of a debounce / smooth. So for example:
"If [x-command] comes in more than 2 times in under [x-mills] set a flag to true."
What would be some ways to do this? ie: use an array for samples? write a timeout? etc.
I think the simple way is: store time of last call for each command in global array and compare stored time and current on new command.
Also you can don't override stored time if bounce is detect.
I am trying to write an application(I'm a gui first timer) for my son, he has autism. There is a video player in the top half and a text entry area in the bottom. When letters are typed sounds are produced to mimic the words in the video.
There have been other posts on this site in regard to playing sounds on key presses, using gstreamer as a system call. I have also tried libcanberra but both seem to have significant delays between sounds. I can write the app in python or C but will likely do at least some of it in C.
I also want to mention that the video portion is being played by gstreamer. I tried to create two instances of gstreamer, to avoid expensive system calls but the audio instance seemed to kill the app when called.
If anyone has any tips on creating faster responding sounds I would really appreciate it.
You can upload a raw audio sample directly to PulseAudio so there will be no decoding and (perhaps save) extra switches by using the following function from Canberra:
http://developer.gnome.org/libcanberra/unstable/libcanberra-canberra.html#ca-context-cache
The next ca_context_play() will use it.
However, the biggest problem you'll encounter with this scenario (with simultaneous video playback) is that the audio device might be configured with large latency with PulseAudio (up to 1/2s or more for normal playback). It may be reasonable to file a bug to libcanberra to support a LOW_LATENCY flag, as it currently doesn't attempt to minimize delay for sound events afaik. That would be great to have.
GStreamer pulsesink could probably get low latency too (it has some properties for that), but I am afraid it won't be as lightweight as libcanberra, and you won't be able to cache a sample for instance. Ideally, GStreamer could also learn to cache samples, or pre-fill PulseAudio...