audio | Data manipulation and transformation for audio signal | Machine Learning library
kandi X-RAY | audio Summary
kandi X-RAY | audio Summary
[Anaconda-Server Badge] The aim of torchaudio is to apply [PyTorch] to the audio domain. By supporting PyTorch, torchaudio follows the same philosophy of providing strong GPU acceleration, having a focus on trainable features through the autograd system, and having consistent style (tensor names and dimension names). Therefore, it is primarily a machine learning library and not a general signal processing library. The benefits of PyTorch can be seen in torchaudio through having all the computations be through PyTorch operations which makes it easy to use and feel like a natural extension.
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Top functions reviewed by kandi - BETA
- Main function
- Infer the transcript from a segment
- Train one epoch
- Parse command line arguments
- Join two lists
- Generate a waveform
- Saves an audio file
- Determine the type of a given sphere
- Return a descriptor for a given dtype
- Compute a VAD
- Performs a measure of the time series
- Join two sequences
- Builds the given extension
- Compute mvDR weights for rtf
- Inverse of the spectrogram
- Perform the forward computation
- A HuBERT pretraining model
- HUBERT pretraining model
- Generate a filterbank
- Train one epoch for one epoch
- Calculate the vocab
- Apply flanger to a waveform
- Inner function for inference
- Apply effects file
- Load audio file
- Convenience method to compute the model
- Computes the phase of a waveform
- Calculate the gridded domain of the specgram
- Parse arguments
- Construct a CTCDecoder
audio Key Features
audio Examples and Code Snippets
Lines of Code : 8License : Permissive (Apache-2.0)
Copy pip install datasets[audio]
sudo apt-get install libsndfile1
pip install 'torchaudio<0.12.0'
sudo apt-get install sox
export USE_DEFAULT_S3_STORAGE=true
label-studio start
import {
BufferGeometry,
BufferAttribute,
LineBasicMaterial,
Line,
MathUtils
} from 'three';
class PositionalAudioHelper extends Line {
constructor( audio, range = 1, divisionsInnerAngle = 16, divisionsOuterAngle = 2 ) {
const geometry = n def audio_summary(tag,
tensor,
sample_rate,
max_outputs=3,
collections=None,
name=None):
# pylint: disable=line-too-long
"""Outputs a `Summary` protocol buf # coding: utf-8
"""
==============
Audio playback
==============
This notebook demonstrates how to use IPython's audio playback
to play audio signals through your web browser.
"""
# Code source: Brian McFee
# License: ISC
# %%
# We'll need numpy conda install pytorch torchvision torchaudio cudatoolkit=10.2 -c pytorch
poetry add torch
def Exec_ShowImgGrid(ObjTensor, ch=1, size=(28,28), num=16):
#tensor: 128(pictures at the time ) * 784 (28*28)
Objdata= ObjTensor.detach().cpu().view(-1,ch,*size) #128 *1 *28*28
Objgrid= make_grid(Objdata[:num],nrow=4).permutepip3 install --use-deprecated=html5lib torch==1.10.2+cu113 -f https://download.pytorch.org/whl/cu113/torch_stable.html
python -m pip uninstall pip && python -m ensurepip
Python 3.8.10
pip 21.3.1
Community Discussions
Trending Discussions on audio
QUESTION
I need help debugging Webpack's Compression Plugin.
SUMMARY OF PROBLEM
- Goal is to enable asset compression and reduce my app's bundle size. Using the Brotli algorithm as the default, and gzip as a fallback for unsupported browsers.
- I expected a content-encoding field within an asset's Response Headers. Instead, they're loaded without the field. I used the Chrome dev tools' network tab to confirm this. For context, see the following snippet:
- No errors show in my browser or IDE when running locally.
WHAT I TRIED
- Using different implementations for the compression plugin. See below list of approaches:
- (With Webpack Chain API)
ANSWER
Answered 2021-Sep-30 at 14:59It's not clear which server is serving up these assets. If it's Express, looking at the screenshot with the header X-Powered-By, https://github.com/expressjs/compression/issues/71 shows that Brotli support hasn't been added to Express yet.
There might be a way to just specify the header for content-encoding manually though.
QUESTION
I have done a small program, and it shows a weird behavior that I cannot explain. I am using rodio crate to try out some audio stuff.
I have done two programs that, in my opinion, should give the same result.
The first one I use matches to handle errors:
...ANSWER
Answered 2021-Sep-30 at 19:08The issue is one of scoping and an implementation detail of rodio: the one critical item here is OutputStream::try_default(), it doesn't really matter how you handle Sink::try_new(&handle) it'll always behave the same, not so try_default, if you match or if let it it'll work fine, if you unwrap it it'll fail.
But why would that be, the two should be equivalent. The answer is in the details of rodio, specifically of OutputStreamHandle:
QUESTION
I have been trying out an open-sourced personal AI assistant script. The script works fine but I want to create an executable so that I can gift the executable to one of my friends. However, when I try to create the executable using the auto-py-to-exe, it states the below error:
...ANSWER
Answered 2021-Nov-05 at 02:2042681 INFO: PyInstaller: 4.6
42690 INFO: Python: 3.10.0
QUESTION
I would like to be able to robustly stop a video when the video arrives on some specified frames in order to do oral presentations based on videos made with Blender, Manim...
I'm aware of this question, but the problem is that the video does not stops exactly at the good frame. Sometimes it continues forward for one frame and when I force it to come back to the initial frame we see the video going backward, which is weird. Even worse, if the next frame is completely different (different background...) this will be very visible.
To illustrate my issues, I created a demo project here (just click "next" and see that when the video stops, sometimes it goes backward). The full code is here.
The important part of the code I'm using is:
...ANSWER
Answered 2022-Jan-21 at 19:18The video has frame rate of 25fps, and not 24fps:
After putting the correct value it works ok: demo
The VideoFrame api heavily relies on FPS provided by you. You can find FPS of your videos offline and send as metadata along with stop frames from server.
The site videoplayer.handmadeproductions.de uses window.requestAnimationFrame() to get the callback.
There is a new better alternative to requestAnimationFrame. The requestVideoFrameCallback(), allows us to do per-video-frame operations on video.
The same functionality, you domed in OP, can be achieved like this:
QUESTION
I am looking for a way to detect if the device I am using can support Dolby Atmos sounds.
After searching around I found this call.
https://github.com/w3c/media-capabilities/blob/main/explainer.md#spatial-audio
...ANSWER
Answered 2021-Dec-24 at 06:57Detecting the codec doesn't necessarily detect whether the system can support Dolby Atmos
Correct.
What reliable way is there to detect if your system will truly support Dolby Atmos whether its with a receiver or a Dolby Atmos compliant sound bar.
Unfortunately, this undetectable from the browser.
The browser itself and even the OS doesn't always know what is downstream. Sorry for the bad news!
QUESTION
I wrote a python script that generates a xstack complex filter command. The video inputs is a mixture of several formats described here:
I have 2 commands generated, one for the xstack filter, and one for the audio mixing.
Here is the stack command: (sorry the text doesn't wrap!)
...ANSWER
Answered 2021-Dec-16 at 21:11I'm a bit confused as how FFMPEG handles diverse framerates
It doesn't, which would cause a misalignment in your case. The vast majority of filters (any which deal with multiple sources and make use of frames, essentially), including the Concatenate filter require that be the sources have the same framerate.
For the concat filter to work, the inputs have to be of the same frame dimensions (e.g., 1920⨉1080 pixels) and should have the same framerate.
(emphasis added)
The documentation also adds:
Therefore, you may at least have to add a scale or scale2ref filter before concatenating videos. A handful of other attributes have to match as well, like the stream aspect ratio. Refer to the documentation of the filter for more info.
You should convert your sources to the same framerate first.
QUESTION
I want to extract how many positive reviews by brand are in a dataset which includes reviews from thousands of products. I used this code and I got a table including percentaje of positive and non-positive reviews. How can I get only the percentage of positive reviews by brand? I only want the "True" results in positive_review. Thanks!
...ANSWER
Answered 2021-Nov-22 at 17:40Using the following toy DataFrame as an example:
QUESTION
I am using pyaudio to record sounds on my Mac BigSur 11.6 (20G165). Specifically, I'm redirecting sound from an application to the input using BlackHole, which works fine.
It usually works fine but, sometimes, I get this error in the Terminal:
||PaMacCore (AUHAL)|| Error on line 2500: err='-10863', msg=Audio Unit: cannot do in current context
Any idea why or how I could prevent it from happening (like, waiting until PaMacCore is ready to record again or something)?
I already tried reinstalling but it doesn't help
...ANSWER
Answered 2021-Nov-03 at 16:29Apparently the problem were mismatched bitrates in BlackHole's aggregated output device. I was aggregating Blackhole's output (44,1kHz) and the Mac Speakers (48kHz). This did not cause any consistent bad behaviour but sometimes led to these errors.
QUESTION
This use case is a service that manually encodes a series of uncompressed .wav media segments into .m4s fragments for broadcast via MPEG-DASH, using ffmpeg to compress the .wav to .aac and sannies/mp4parser to assemble the aac audio into an .m4s media fragment.
I created this public GitHub project to reproduce the issue in its entirety.
For example, here's the custom CustomFragmentMp4Builder.java class.
It's critical that we be able to designate this single .m4s fragment with a sequence number (index) that we will manually increment for each media segment.
The objective is to build an .m4s fragment comprising the box types SegmentTypeBox, SegmentIndexBox, and MovieFragmentBox. As For reference, I have used mp4parser to inspect an .m4s fragment that was generated via ffmpeg -f hls. This specification is available here as a .yaml file
My implementation creates an MP4 without error. But, when the unit test attempts to read the file that the ChunkMp4Builder just wrote to a temp folder:
...ANSWER
Answered 2021-Oct-21 at 20:57Your m4s segments are invalid due to an incorrect mdat atom size.
For example in test5-128k-151304042.m4s the mdat is marked as having a length of 16 bytes but there is data at the end and file size is 164884.
The parser then attempts to read an invalid offset. avc5 is not an atom but actually part of the string "Lavc58.54.100". The length read as 3724673100 is also invalid and greater than the max for a 32-bit integer, hence the invalid cast to int.
In your implementation you have:
QUESTION
After looking to implement WebRTC with a Client to Server model (like Discord), I came to the conclusion that the way to do this is to have 2 clients - the server and client. Audio streams can be overlayed and sent back to the user in 1 single stream.
backend/server.js
...ANSWER
Answered 2021-Aug-26 at 04:13A possible solution can be: Create a MediaRecorder object, which can record the media streams on the client-side. This object emits data chunks over time. You can send these chunks via WebSocket to the server. On the server side, you can do what you want with the data chunks.
For more details, you can check this https://mux.com/blog/the-state-of-going-live-from-a-browser/.
https://developer.mozilla.org/en-US/docs/Web/API/MediaRecorder Another solution can be: Making a node.js application a PEER with WebRTC
Community Discussions, Code Snippets contain sources that include Stack Exchange Network
Vulnerabilities
No vulnerabilities reported
Install audio
You can use audio like any standard Python library. You will need to make sure that you have a development environment consisting of a Python distribution including header files, a compiler, pip, and git installed. Make sure that your pip, setuptools, and wheel are up to date. When using pip it is generally recommended to install packages in a virtual environment to avoid changes to the system.
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