English Speaking Audio

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Qiana Thieklin

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Jul 31, 2024, 1:11:41 AM7/31/24
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First of all, you need to learn the most frequently used words in English, common structures and sentence patterns, common expressions, common phrasal verbs, and idioms that are much used in daily life.

english speaking audio


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Next, you should learn daily conversations in English for speaking. Focus on every ESL conversation topic until you can speak English automatically and fluently on that topic before moving to the next one.

The following lessons cover 75 topics that you will face very often in your daily life. Each lesson is designed in form of ESL conversation questions and answers, followed by REAL English conversation audios, which will definitely benefit your English conversation practice.

Record someone else with your recording device or mobile app, then transcribe the audio with Dragon. Record things like lectures, presentations, or interviews and use the text in documents, reports, or reference notes.

Use your user profile:
You can transcribe the file with your own user profile. Speech data from the recording won't affect the speech data in your profile, as long as you choose Someone else in the Select the speaker field.

    Open your own profile to transcribe a recording of yourself, or of someone else for whom you don't have a transcription profile. Open a transcription profile to transcribe a recording of someone else, when you have a transcription profile for that person.

I use zoom for my dance classes and when I share the sound of my laptop for my students, they can barely listen to my voice indications at the same time, even if the volume of the shared sound is very low. What can I do to better this? For solution I put the audio on my tablett near from the computer, so the can hear my voice better, but then the sound of the music is bad..

I guide meditations where I need to speak over shared music, and Zoom recently seems to have changed so that even turning on Original Sound - even though better - no loner corrects it. There appears to be noise cancellation going on that reduces the voice level and garbles it, even using a good USB mic.

I want to make a GPT where instead of having the user experience be typing out and having text based conversations, I want to have the conversation be able to happen with the GPT responding via voice and the user can also respond by voice.

Is there a way I can turn the GPTs responses to voice and also have the user respond via their voice and have the GPT respond via voice audio? So in short, I want to make the experience feel more like a natural conversation with someone rather than typing everything out.

Hi. After reading the information in this thread I have a question. Can I organize and set up simultaneous voice translation during an online meeting? If it is possible how can I technically realize it?

I am working with audio of someone with tremors in their voice- I think they call it vocal paralysis or something. Are there software applications that break sound down into it's individual waveforms using FFT or something similar that would allow me to manipulate these tremors more than I could by automating volume changes?

There are some general spectral (fft) editors, of which Spear (free) is one of the most powerful (even more than Izotope RX or the Sony Editor), but these should not really be necessary, if it is just a volume modification that you want to do.

First, make sure that in Asterisk SIP Settings, External Address and Local Networks are correctly set. Usually, just pressing Detect Network Settings will set them properly. If anything changed, Submit, Apply Config and restart Asterisk.

If trunk audio fails, at the Asterisk command prompt, type
pjsip set logger on
or
sip set debug on
according to trunk type, make a failing call, past the Asterisk log for the call at pastebin.freepbx.org and post the link here.

Something still seems wrong with your Asterisk SIP Settings.
o=- 790455249 3 IN IP4 104.145.12.182
This is an address at Sangoma, the default on a new install.
Please go to Asterisk SIP Settings and confirm that External Address shows your Vultr server address. Confirm that you have restarted (not just reloaded) Asterisk after changing it.

Several times recently (including on fairly serious meetings, like Board meetings and job interviews) I've had others tell me that they couldn't hear me / my connection was bad. [ At that point, turning off video hasn't helped much if at all. ] As a work-around, I've had to dial in for audio (but that's a bit embarrassing during, say, a Board meeting). Sometimes I get a message from Teams telling me that my internet connection is poor (or words to that effect), but usually I hear complaints first from other participants, or I can see video problems myself.

When I have run speed tests on my connection, the downlink has never been less than 200Mb/sec, and it's usually closer to 250. Uplink has never been less than 30Mb/sec, and it's usually 50Mb/sec. I've checked right after meetings, including meetings with problems, and have started doing random tests during similar hours.

I have tried using my MacBook (a reasonably current model) and an iPad Pro (running nothing but Teams during the meeting), and the problem has occurred on both. All devices are running the current released operating systems. I have made a point of rebooting them before calls, but that hasn't made any difference. [ the router has also been rebooted several times recently, but not necessarily before every call ]

During calls, I disable all other devices in my house (it's just me here, so I know no one else is streaming or gaming on my connection, but of course I don't know what may be happening elsewhere in my neighbourhood, and contention may be possible -- but again, my speed tests suggest my service remains more than adequate).

I've so far had no similar problems with Zoom. In fact, before one important meeting, I got the organiser to do a test with me, but she choose to use Zoom (she thought we were testing my internet quality) and our test was fine -- but during the later Teams call, I had to drop off and rejoin by phone, which was extremely awkward.

This "should" work, and it does for most of the participants (others occasionally have issues, but I have by far the most). It's almost like Deutsche Telekom throttles my connection when I start to speak -- I'm not paranoid enough to believe that, but that's how it appears.

Because I am a regular attendee to these meetings, but am not part of the organisation that actually hosts the calls, they see it as "my" problem to solve. I could probably approach their administrator to ask for assistance, but if I do, I need to have a plan with a crisp set of requests of them.

I have the exact same issue. I have 200+MB down and 100MB+up on every ookly test. even during calls where MS teams is showing me a bandwidth issue and and the person on the other side is saying that they cannot understand or hear me.

I have a few of these around the house. Lately they have not been playing or speaking anything. When mail arrives, I have it speak on these speakers "mail has arrived". I also have the speakers announce when its almost time for the kids to go to bed. Lately, and I have been trying this today, I am trying to send any text speech to the devices. I send a speech to it that clearly says "this is a test", and all is heard on the speaker is "Tssssss"... a brief second. Or should I say it says "tiss". Just a sound effect rather.

Go to the speaker in devices and click initialize and see if it works after that. I've see if you don't set a reservation for google speakers, sometimes they're lost until you reinitialize. Just a thought.

I figured it out. I had the UI set for SSL only. Sounds like a bug in the Chromecast App. When I disable that, Speaking on the Chromecast devices works again and I don't have those messages in the logs.

I recently had the chance to talk with my friend Jon Schwabish of PolicyViz. You might recognize Jon from our earlier interview about tips for doing and teaching data visualization in cultures other than your own.

So while I do a lot of consulting on the specifics of reports, dashboards, presentations, and infographics, I also do a lot of consulting related to communication strategies more broadly. I want to make sure that organizations have selected the right mix or reports and/or dashboards and/or presentations and/or infographics before we spend our precious time, money, and mental bandwidth actually making or improving on those designs.

My public speaking coach taught me that only 40 percent of communication is the actual words that you use. The remaining 60 percent of communication is related to facial expressions, tone, pitch, using your hands to reiterate your main points, and so on.

Their founder, Ankur Nagpal, also gives me a lot of confidence in their product. Ankur and his team are constantly improving. They interview instructors like me and they listen to our feedback. I love that. Sure, there are technical glitches once in a while, just like any software platform would have. But the Teachable team is constantly improving and iterating for their creators.

With Live Captions (beta) on iPhone, you can get a real-time transcription of spoken audio. Use Live Captions to more easily follow the audio in any app, such as FaceTime or Podcasts, and in live conversations around you.

@Knopf : However, this method does not take into account the formants
@eirik_myhr : Unfortunately, I only own the programs listed above. Maybe Audacity. But I expected that it should be possible with my Steinberg programs
@matjones : I will try again with SpectraLayers by manually separating the formant area. Possibly that was my mistake.

I would think that VariAudio would also produce better results than Autotune for this kind of task.
You could also experiment with changing the algorithm that VariAudio uses, to find the one that sounds the most convincing.

I wanted to give cubase another chance, so I tried again with VariAudio. Apparently Cubase is more capable of varying the pitch of a singing voice than a speaking voice. As you can see on the ScreenShot, parts of the words are not analyzed at all. These remain then in the original pitch. Furthermore, when all events are selected, it is very difficult to move them up or down with the mouse. This can only be done with the arrow keys. Furthermore, the possible zoom range is much too small to get a good overview in the vertical editing.

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