audioStream | A lightweight nodejs music streaming app
kandi X-RAY | audioStream Summary
kandi X-RAY | audioStream Summary
(or whatever his futur name will be). This is a node server to stream music. For now it comes with an angular client but it's mostly a placeholder to work on server functionality and it will be replaced. It aim to stay light weight and work on a raspberry pi or other low power computer with minimal configuration. This project is also a training project for a future bigger one, to improve my nodeJS and test some architectures.
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Top functions reviewed by kandi - BETA
- Compile an event handler .
- RootScope provider .
- Initializes the default HTTPProvider .
- Initialize route directive
- Executes a set of directives .
- Provides a set of SCE providers .
- return a promise
- Create Angular module loader
- Creates a new browser .
- Interpolation provider .
audioStream Key Features
audioStream Examples and Code Snippets
Community Discussions
Trending Discussions on audioStream
QUESTION
I'm trying to implement a very simple API for audio playback using portaudio
. I have minimal amount of code needed to play the audio file but I am not getting any errors and/or audio output.
Here is the code,
...ANSWER
Answered 2022-Apr-07 at 23:24You've got two flaws that I can see.
One is using what looks like a class method as the callback to PulseAudio. Since PA is a C API, it expects a C function pointer here. It won't call that function with the this
pointer set, so you can't have a class method here. But maybe AudioStream::Callback
is static
? That will work.
Two is that you need to consider that the callback is called in another thread. The compiler does not take that into account when it optimizes the code. As far as it knows, there is nothing in your empty while loop that could possibly change the value of file->count
.
Once you call the debug function, it brings in enough code, some probably in libraries that are already compiled, that the compiler can't be sure nothing has modified file->count
. Maybe SH_LOG_DEBUG()
calls prinf()
and then printf()
calls AudioStream::Load()
? Of course it doesn't, but if the compiler doesn't see the code of printf()
because it's already in a library, and your AudioStream
object is global, then it's possible. So it actually works like you want.
But even when it works, it's really bad. Because the thread is sitting there busy waiting for the count to stop, hogging the/a CPU. It should block and sleep, then get notified when the playback finishes.
If you want to pass information between threads in a way that works, and also block instead of busy waiting too, look into the C++ concurrency support library. The C++20 std::latch
and std::barrier
would work well here. Or use a C++11 std::condition_variable
.
QUESTION
I'm trying to add filter effects to an audio stream I have playing on my website. I'm able to connect the Tone.js library to the audio stream but I'm not hearing any changes in the audio stream playing on the website. I'm not seeing any errors in the console and I've tried adjusting the filter from 50 to 5000 but nothing seems to have any effect on the audio. Do I need to set up the new Tone.Player()
to actually hear the audio? If so, how do you go about setting up the Player if there is no src for the existing audio element.
ANSWER
Answered 2022-Feb-28 at 07:50Working solution:
Removing the audioStream.play()
from where the JsSIP call is answered solves the issue.
I don't know the exact reason why this solves the issue (it might even be a workaround) but after much trial and error this way allows the audio to be available to ToneJS for effecting.
Any other solutions are welcome.
QUESTION
I'm getting this error when converting a video using pafy
...ANSWER
Answered 2022-Feb-05 at 18:52You have done too many API
actions in YouTube
. If you reach 10.000 points per day you are limited. Read about it here: https://developers.google.com/youtube/v3/getting-started#quota
And you can read about similar things in another StackOverflow Question.
QUESTION
I need some help with my Java application. Its purpose is to read a certain website, so I need to play many audio files in a row. The JAR is compiled using Java 8. I tested my application with Windows 11 and Java 16.0.1, everything works fine. Then I used the latest Ubuntu Linux and Java 11.0.13 as well as Java 8: It plays some audio, but not every file.
I wrote a test class and the result was, that - no matter in which order I play the audio - only the first (exactly!) 62 files are played. Every next file (even the ones, that were successfully played at first) produces the exception my code throws at this position:
...ANSWER
Answered 2022-Jan-14 at 09:33The #1 suggestion is by Mark Rotteveel. The AudioInputStream
class needs closing. This is often a surprise for people, because Java is well known for managing garbage collection. But for AudioInputStream
there are resources that need to be released. The API doesn't do an adequate job of pointing this out, imho, but the need for handling can be inferred from the description for the AudioInputStream.close() method:
Closes this audio input stream and releases any system resources associated with the stream.
The #2 suggestion is from both Andrew Thompson and Hendrik may be more a helpful hint than a direct solution, but it is still a very good idea, and it seems plausible to me that the inefficiency of all the additional, unneeded infrastructure (ClassLoader
, InputStream
, BufferedInputStream
) might be contributing to the issue. But I really don't have a good enough understanding of the underlying code to know how pertinent that is.
However, I think you can do even better. Don't use Clip
. You current use of Clip
goes against the concept of its design. Clips
are meant for short duration sounds that are to be held in memory and played multiple times, not files that are repeatedly reloaded before each playback. The proper class for this sort of use (load and play) is the SourceDataLine
.
An example of playback using a SourceDataLine
can be found in the javasound wiki. This example also illustrates the use of URL
for obtaining the necessary AudioInputStream
. I will quote it here verbatim.
QUESTION
I just did this guide to run Firefox as an isolated user - https://www.burnison.ca/articles/running-firefox-as-an-isolated-linux-user It's a nice idea for security and also I don't like .mozilla directory in my home. Mozilla Firefox developers won't add support for XDG Base Directory unfortunately.
Everything looks fine but speakers and mic doesn't work. I've got some warnings about it:
...ANSWER
Answered 2021-Nov-13 at 18:04Thank you for your comprehensive answer.
I just switched to ALSA from Pulseaudio. It works fine.
QUESTION
I have two mp4 files. One contains only video and the other only audio. Both have exactly the same length (some seconds extracted from a HLS Stream).
I want them now to get mixed together trough a GCS Transcoder API Job which gets triggered by a Dataflow Pipeline. Digging trough the documentation did not yet result in a solution.
My current Job Config looks like that:
...ANSWER
Answered 2021-Oct-25 at 19:52There are some defaults not listed in the documentation. Try adding the following to your config and see whether it works
QUESTION
I have a library that uses some c++ compiled code and I would like to use this and other functions to try it out but they all use Pointer and PointerByReference as arguments instead of normal types.
...ANSWER
Answered 2021-Oct-21 at 15:21Since you don't include more of the API for a specific call I can't directly answer your implementation question, but I will address the portion of the question that asks about the types.
In C (and C++), arrays are stored in contiguous native memory. If you know the type (float in this case) you can simply offset from the pointer to get the appropriate element, e.g., foo[0]
is at the pointer location to the array, foo[1]
would be at that pointer location plus an offset equal to the type byte size (4 bytes for a float) and so on.
The Pointer
and PointerByReference
are JNA types. PointerByReference
is a pointer that points to a Pointer
; you can call the getValue()
function on the PointerByReference()
to retrieve this pointer.
Based on the way I read this API, that Pointer is actually the beginning of the float array, so you'd just use that Pointer and retrieve the array from its location.
So this is likely what you need to do:
QUESTION
I am trying to get Polly to read something for me, using PHP.
I have created a new project, installed Amazon composer require aws/aws-sdk-php
and then created a file with code from SDK documentation example and modified a few minor things such as changing credential from default to value, var_dump to var_export and finally saved the content of the stream to file
ANSWER
Answered 2021-Oct-16 at 20:15You're not doing anything wrong, but it only outputs JSON if you're looking for speech marks. Try switching to an audio output format like MP3 as shown below.
QUESTION
I have the following code to transcode video which is closely related to the FFMPEG transcoding example.
However it produced broken video as shown:
It seems like the i-frames are decoded correctly, but the p and b frames are out of order? I thought av_interleaved_write_frame()
would rectify that for me. It also seems like the libx264 is not creating any extradata which I thought it would.
Could someone please help me work out why?
Many thanks
...ANSWER
Answered 2021-Oct-15 at 16:56After thinking about this some more, I realised that I wasn't even sure if my decoder was correct. And another thorough inspection of the transcoding example revealed a small difference in behaviour.
Effectively, I was doing this:
QUESTION
I'm trying to implement an audiostream (radio) with images to display on Alexa devices. The audio works fine, but for some reason my images stopped displaying after I tried to change the source image the first time. Not sure if I'm just messing up the syntax, or if it's something else entirely.
According to the doc the "audioItem.stream.token may be cached in the Alexa service for up to five days", which in term can affect image display. But changing the token to something new doesn't seem to do anything.
Hope somebody has some insight into this.
...ANSWER
Answered 2021-Sep-14 at 12:02The implementation is correct.
The documentation confirm that you have to update the token to allow the metadata to be refreshed. I tested and It work properly.
Based on your comments, it seems that the issue persist on your Fire HD 8 Tablet with the Fire OS version 7.3.2.1
.
It's always a good practice to try on different devices to validate the root cause.
I recommend you to contact the developer directly on the amazon developer forum. They should be able to push a fix for the device and fireOS version.
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