I'm working with the web audio API. Say I have a source buffer node. Is there any way that I could connect it to 2 different delay nodes? Like
var sourceNode = context.createBufferSource();
sourceNode.buffer = myBuffer;
sourceNode.connect(delayNode1);
sourceNode.connect(delayNode2);
? The way I have it here does not work. It only connects to delayNode2.
Yes, of course. I'm not sure why you're not finding that code to work; it absolutely should (and similar code works for me). If you have it running in place somewhere I can take a look.
Related
i know this forum dislikes "open" questions like this, nevertheless i'd like somebody to help untie the knot in my head, much appreciated.
The goal is simple:
read a stereo 32bit 44100 S/s I2S signal from 2 adafruit sph0645 mics
create a wav-header and store the data onto an SD-card
I've been at this for a few days now and i know that this will be much more complicated than i originally thought. Main reason: signal quality. Like most tutorials on this subject the simplest "hello world" for these mics is a looped polling for I2S-samples. Poll, fill buffer, output via serial or write to SD-card. This returns a choppy, noisy, sped up version of RL-audio. The filling of the internal DMA-buffers can be seen as constant, but the rest is mostly chaos, so
how to i sync these DMA-buffers with the rest of my code?
From experience with the STM32 HAL i'd imagine some register which can be set to throw an interrupt whenever a buffer is full, or an event which can be sent between tasks via queues. Examples on this subject either poll in a main loop with mono an abysmal sample-rate and bit depth or use pages of overkill code and never adress what it does, "just copy and it works", not good. Does the ESP32-Arduino framework provide some way to to this properly? The espressif-documentation isn't something to look forward to, since some of their I2S interface functions don't even work (if you are researching this topic as well, you too might have noticed that i2s_read only returns zeros). Just a hint into the right direction would help, i'm writing my own code anyway. Interrupts? Events? Timers? Polling for full buffers? Only you might know.
have a good one, thx
Thanks to https://github.com/atomic14/ i now have an answer for a syncing-method which works very well. This method has been tried by https://esp32.com/viewtopic.php?t=12546 who also didn't fully understand what was going on: the espressif i2s-interface offers a flag stored in an event which is triggererd every time one of the specified dma-buffers has received a full set of data, ergo, is full. It looks like this:
while(<your condition>){
i2s_event_t evt;
if (xQueueReceive(<your queue>, &evt, portMAX_DELAY) == pdPASS){
if (evt.type == I2S_EVENT_RX_DONE){
size_t bytesRead = 0;
do{
//read data via i2s_read or i2s_read_bytes
} while (bytesRead > 0);
No data is stored in this queue, but rather a flag which can then be used to synchronize dma-filling and further buffering/calculating/sending the read data.
HOWEVER this only works if you install the i2s driver in a specific setup. Instead of using
i2s_driver_install(I2S_NUM_0, &i2s_config, 0, NULL);
in your setup, you can activate the "affinity" for events by passing a queue-handle and a lenght:
i2s_driver_install(I2S_NUM_0, &i2s_config, 4, &<your queue>);
hope this helps getting started, it sure did help me.
I understood this page to mean that queuing in pyglet provides a gapless transition between audio tracks. But when I test it out, there is a noticeable gap. Has anyone here worked with gapless audio in pyglet?
Example:
player = pyglet.media.Player()
source1 = pyglet.media.load([file1]) # adding streaming=False doesn't fix the issue
source2 = pyglet.media.load([file2])
player.queue(source1)
player.queue(source2)
player.play()
player.seek([time]) # to avoid having to wait until the end of the track. removing this doesn't fix the gap issue
pyglet.app.run()
I would suggest you either edit your url1 and url2 into caching them locally if they're external sources. And then use Player().time to identify when you're about to reach the end. And then call player.next_source.
Or if it's local files and you don't want to programatically solve the problem you could chop up the audio files in something like Audacity to make them seamless on start/stop.
You could also experiment with having multiple players and layer them on top of each other. But if you're only interested in audio playback, there's other alternatives.
It turns out that there were 2 problems.
The first one: I should have used
source_group = pyglet.media.SourceGroup()
source_group.add(source1)
source_group.add(source2)
player.queue(source_group)
The second one: mp3 files are apparently slightly padded at the beginning and at the end, so that is where the gap is coming from. However, this does not seem to be an issue with any other file type.
I'm trying to filter a signal and then analyse the values of the filtered signal using Tone.js / Web-Audio API.
I'm expecting to get values of the filtered signal, but I only get -Infinity, meaning that my connections between the nodes are wrong. I've made a small fiddle demonstrating this, however in my use-case I do not want to send this node to the destination of the context - I only want to analyse it, not hear it.
osc.connect(filter)
filter.connect(gainNode)
gainNode.connect(meter)
console.log(meter.getLevel())
I guess you tested the code in Chrome because there is a problem with Chrome which causes it to not process anything until it is connected to the destination. When using Tone.js that means you need to call .toMaster() at the end of your chain. I updated you fiddle to make it work: https://jsfiddle.net/8f7abzoL/.
In Firefox calling .toMaster() is not necessary therefore the following works in Firefox as well: https://jsfiddle.net/yrjgfdtz/.
After some digging I've found out that I need to have a scriptProcessorNode - which is apparently no longer recommended - so looking into Audio Worklet Nodes
I currently am using Max/MSP to create an interactive system between lights and sound.
I am using Philips hue lighting which I have hooked up to Max/MSP and now I am wanting to trigger an increase in brightness/saturation on the input of a note from a Midi instrument. Does anyone have any ideas how this might be accomplished?
I have built this.
I used the shell object. And then feed an array of parameters into it via a javascipt file with the HUE API. There is a lag time of 1/6 of a second between commands.
Javascript file:
inlets=1;
outlets=1;
var bridge="192.168.0.100";
var hash="newdeveloper";
var bulb= 1;
var brt= 200;
var satn= 250;
var hcolor= 10000;
var bulb=1;
function list(bulb,hcolor,brt,satn,tran) {
execute('PUT','http://'+bridge+'/api/'+hash+'/lights/'+bulb+'/state', '"{\\\"on\\\":true,\\\"hue\\\":'+hcolor+', \\\"bri\\\":'+brt+',\\\"sat\\\":'+satn+',\\\"transitiontime\\\":'+tran+'}"');
}
function execute($method,$url,$message){
outlet(0,"curl --request",$method,"--data",$message,$url);
}
To control Philips Hue you need to issue calls to a restful http based api, like so: http://www.developers.meethue.com/documentation/core-concepts, using the [jweb] or [maxweb] objects: https://cycling74.com/forums/topic/making-rest-call-from-max-6-and-saving-the-return/
Generally however, to control lights you use DMX, the standard protocol for professional lighting control. Here is a somewhat lengthy post on the topic: https://cycling74.com/forums/topic/controlling-video-and-lighting-with-max/, scroll down to my post from APRIL 11, 2014 | 3:42 AM.
To change the bri/sat of your lights is explained in the following link (Registration/Login required)
http://www.developers.meethue.com/documentation/lights-api#16_set_light_state
You will need to know the IP Address of your hue hue bridge which is explained here: http://www.developers.meethue.com/documentation/getting-started and a valid username.
Also bear in mind the performance limitations. As a general rule you can send up to 10 lightstate commands per second. I would recommend having a 100ms gap between each one, to prevent flooding the bridge (and losing commands).
Are you interested in finding out details of who to map this data from a MIDI input to the phillips HUE lights within max? or are you already familiar with Max.
Using Tommy b's javascript (which you could put into a js object), You could for example scale the MIDI messages you want to use using midiin and borax objects and map them to the outputs you want using the scale object. Karlheinz Essl's RTC library is a good place to start with algorithmic composition if you want to transform the data at all http://www.essl.at/software.html
+1 for DMX light control via Max. There are lots of good max-to-dmx tutorials and USB-DMX hardware is getting pretty cheap. However, as someone who previously believed in dragging a bunch of computer equipment on stage just to control a light or two with an instrument, I'd recommend researching and purchasing a simple one channel "color organ" circuit kit (e.g., Velleman MK 110). Controlling a 120/240V light bulb via audio is easier than you might think; a computer for this type of application is usually overkill. Keep it simple and good luck!
I am trying to stream sensor data from the iRobot Create. I get tuple out of range errors when I try
bot.stream_sensors(somenumber) and bot.poll_sensors(somenumbers). Whenever I input bot.sensors, I just get an empty array {}. I have even tried sending bot.sensors while pushing in on the bump sensor, still getting an empty array. I am connected to the bot through the Serial port with a serial-to-usb converter on my side. The only code before trying to get the sensor data is
import openinterface
bot = openinterface.CreateBot(com_port="/dev/ttyUSB0", mode="full")
Does anyone have an idea of how to solve this issue? Everywhere else just uses stream_sensors(6) and it seems to work fine.
P.S. I posted a question similar to this topic not too long ago, but no one responded. Not trying to spam, but now I have a more clear question and what the apparent-problem is so I thought I would try again.
I downloaded openinterface.py from this site: which included some sample programs. I'd suggest you take a step back, try the sample code, try to find other, more sophisticated, sample code and play with that first before moving on to your real code. You may be missing a step somewhere.
I may be a bit late to answer this, but for reference purposes. Directly controlling the iRobot is simplified greatly by using
Pyrobot.