Real time audio output

EasyRTA is a powerful, yet light-weight audio spectrum analyzer software for Windows. You can choose any input device for recording and select which channel you want left, right or both. Microphone calibration is also supported for even more precision in measurements. EasyRTA has an automated calibration procedure which allows you to make very precise measurements on any device on line input or line output.

Audio measurements can also be performed with an external microphone. Besides being a powerful real-time spectrum analyzer, EasyRTA also features an on-screen measuring tool which can be used for measuring amplitude and frequency in any given point. Therefore the latency time period which is needed to collect a packet of FFT points is ms, providing measurment from 6 Hz to 24 kHz. You can easily save all your measurements and review them at any time.

You can either load the saved file in EasyRTA for optimized viewing or you can view the file in any supported external application, like Excel. We make virtual audio measuring instruments which are easy to use, but precise and robust as professional audio equipment. Easy Audio Tools We make virtual audio measuring instruments which are easy to use, but precise and robust as professional audio equipment.Google Translate is already a hugely useful app for anyone who lives overseas or travels regularly, and it just got even smarter on mobile.

A new update to the Android and iOS apps that is rolling out today introduces two very spiffy features: real-time voice and sign translation. The app has already offered image-based translation, but now the magic occurs without any delay at all… and — best of all for regular travelers — it works offline.

You tap the in-app mic once and starting talking in the foreign tongue first. The app pulls in text-based translations of both sides of the conversation in real-time, helping overcome the language barrier. The sole caveat is that these instant translation features are somewhat limited at first.

Those figures are quite incredible and — with two hugely useful new features now up and running on both iOS and Android — Google Translate is likely to grow into even more of a monster hit. Update: As some readers have pointed out, this impressive technology is powered by World Lens, a startup that Google bought last year.

Credit where it is due.By using our site, you acknowledge that you have read and understand our Cookie PolicyPrivacy Policyand our Terms of Service. Raspberry Pi Stack Exchange is a question and answer site for users and developers of hardware and software for Raspberry Pi. It only takes a minute to sign up. I am trying to get my Raspberry Pi to read some audio input through a basic USB souncard and play it back in real time for 10 seconds, and then print the output with Matplotlib after it's finished.

I am using PyAudio in callback mode.

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Audio recording and playback works fine in Audacity. After I run my code I can see the the plotted output signal, but it only has very few samples. I can't hear anything through the speakers either while I'm playing.

Running the same code on my Windows machine works fine, using the same Python and PyAudio versions. I do not know why your audio is not working. Perhaps you need to explicitly state the output device index. Sign up to join this community. The best answers are voted up and rise to the top.

Google Translate Now Does Real-Time Voice And Sign Translations On Mobile

Home Questions Tags Users Unanswered. Asked 4 years, 4 months ago. Active 3 years, 7 months ago. Viewed 46k times. Any help would be much appreciated! DrumPower DrumPower 95 1 1 gold badge 1 1 silver badge 6 6 bronze badges. When you sent them to playback, did you raise the amplitude?

Real time audio input, audio output

I don't hear anything till the amplitude gets aroundand is good at for int16 format data. Active Oldest Votes. This is what I did to get audio out through the RPi audio jack.This is an experiment to show how some realtime audio processing can be done with the Arduino. The first set of examples alter an incoming audio signal and put it back to an audio output.

We achieve effects like Reverb, Phasor, Flanger or Ringmodulator. The second set of examples are outputting computed waveforms like Sinewave, Bell and Xylophone sounds. The audio input signal is connected via a 10uF capacitor to the the analog input 1 of the Arduino Board.

Two resistors and a trimmpot are adding an DC offset to the audiosignal. A potentiometer connected to analog input 0 will be used to control the audio effects. The output can be connected to a active PC Speakers. A good way to generate a test Signal is using a PC and a audio software like Audacity. A clip of music or speech is recorded and then filtered with a 3KHz.

real time audio output

The Signal has to be filtered to avoid the aliasing effect when the signal gets sampled which would lead to a distorted sound. Now you can connect the PC headphone output to our setup and playback the clip in an endless loop. You might have to use full volume since Arduinos ADC needs a level of 2.

When you want to use a microphone or an other input source you have to build an extra preamplifier with an appropriate steep lowpassfilter. The Software is divided into an interrupt function where the analog sampling an timing takes place and a main loop where the samples are processed an written back to the PWM as audio output. The ADC is set to a fast sampling mode and to 8-Bit precision. Timer2 is used as PWM to convert the digital sample back into an analog value.

The prescaler is changed and the interrupt is enabled so that the interrupt service is invoked all 16 uSec or with a rate of When an interrupt takes place the analog input of channel 0 and 1 is alternately sampled so that the audiosignal is sampled with an effective rate of When a new sample is valid a flag is set which is used in the main loop to synchronize the process. Timer1 is disabled so the Adruino delayfunctions are not available anymore.Guaranteed low audio latency.

Smartphone audio and TV video sync together. Have your own app? Plug-n-play installation using your existing Wi-Fi — no custom configuration required. Accepts 3 types of audio input analog, optical or digital coax.

Each audio input has dedicated audio processor for maximum performance and scalability. Works with Apple and Android devices.

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From the tiny, single input AudioFetch Express with integrated WiFi to a full input AudioFetch deployment, each audio input gets dedicated processor power to ensure your application scales now, and for the future. AudioFetch is designed and manufactured by Broadcastvision Entertainment so the quality of the product and support will be top notch. We offer 2 different types of systems:.

Arduino Realtime Audio Processing

Wirelessly connect to your router, use the built-in WiFi or hardwire to your router onsite. Standalone or Integrated: Wirelessly add an out-of-the way audio source or integrate with an existing AudioFetch system. The most flexible system available. Choose from 3 different connection options. Hardwired to the facilities existing WiFi network. The most reliable way to connect the AudioFetch Express.

Connect to your facilities network wirelessly. Take advantage of your facilities stronger WiFi network without the need for a hardwired connection. Use the WiFi built into the Fetch. Use will be limited to devices within range of the AudioFetch Express unit. Click Here for a Wiring Diagram. Expandable: Our 4 channel card slots make it easy for facilities to expand. Simply add additional cards when needed or AudioFetch Express.

Google Translate Now Does Real-Time Voice And Sign Translations On Mobile

Accordio is an option for systems that cannot be hardwired. Multiple Input Types: Each one of our channels accept 3 types of audio inputs analog, optical or digital coax to ensure compatibility. The brands listed are design examples only and do not suggest or assume the listed brands endorse AudioFetch.

The introduction banner is launched when the user first opens the app. Simple to launch, you can display in text or an image.

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Both are clickable and can be redirected to a website or inform the user to email or call.By using our site, you acknowledge that you have read and understand our Cookie PolicyPrivacy Policyand our Terms of Service. Raspberry Pi Stack Exchange is a question and answer site for users and developers of hardware and software for Raspberry Pi. It only takes a minute to sign up. I am trying to get my Raspberry Pi to read some audio input through a basic USB souncard and play it back in real time for 10 seconds, and then print the output with Matplotlib after it's finished.

real time audio output

I am using PyAudio in callback mode. Audio recording and playback works fine in Audacity. After I run my code I can see the the plotted output signal, but it only has very few samples. I can't hear anything through the speakers either while I'm playing. Running the same code on my Windows machine works fine, using the same Python and PyAudio versions. I do not know why your audio is not working. Perhaps you need to explicitly state the output device index.

Sign up to join this community. The best answers are voted up and rise to the top. Home Questions Tags Users Unanswered. Asked 4 years, 4 months ago. Active 3 years, 7 months ago. Viewed 46k times. Any help would be much appreciated! DrumPower DrumPower 95 1 1 gold badge 1 1 silver badge 6 6 bronze badges. When you sent them to playback, did you raise the amplitude?Sign in to comment. Sign in to answer this question. Unable to complete the action because of changes made to the page.

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You may receive emails, depending on your notification preferences. Real time audio input, audio output. Dong-uk Hyon on 25 Feb Vote 0. Commented: Walter Roberson on 26 Feb Accepted Answer: Walter Roberson.

real time audio output

Hello everyone. I am involved in a project that listens to audio in using a microphone, which is then immediately outputted via a speaker. The emphasis is on using continuous audio in, from a microphone, to continuous audio out to a speaker.

The end goal, would be to add a delay function in-between the audio received, and to the audio outputted. To almost act as an audio delayer. From my very basic understanding; sound amplitude is recorded at certain time intervals at a certain frequency, called the sampling rate. This is how the analog signal is converted to a digital signal which a computer can process. This amplitude is then stored in a 1 x n array. Where n is the sample frequency x length of recording. Is there a method of outputting the amplitudes that the microphone has received, which are stored in the array?

And if I am to apply a time delay, I would calculate the desired delay time, and then multiply it by the sampling frequency, and offset the output by this number of values? I am using the data acquisition toolbox, that has instructed me to creating a continuous audio input.

Any help would be greatly appreciated. Many thanks.


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