HHD Online Player (serial Number Video Download Capture)
Floriana Monterroza
Videos bring back so many memories and remind us of long-forgotten treasured moments. From one-of-a-kind vacations to graduations, you can convert those VHS stories to DVD or digital, right at home! Imagine the faces of your parents, grandparents, siblings, and friends when you give them the gift of videos they haven't seen in years!
Note: An internet connection is required for product installation and updates. Windows may ask for your permission during various steps of the installation process. Please grant permissions when requested so that the installation can be completed.
Step One:
Connect the video output on your VCR to the matching video input on your Roxio Video Capture device using an RCA or S-Video Cable. If your VCR has a yellow RCA video output, use an RCA cable to connect it to the matching yellow RCA video input on the capture device. If your VCR has an S-Video output, use an S-Video cable to connect it to the matching S-Video input on the Roxio Video Capture device. When possible, we recommend using S-Video for higher quality.
Connect the Roxio Video Capture USB device to a USB 2.0 port on your computer. An optional USB extension cord is included. Connect the Roxio adapter to the capture device. Connect the audio player to the large two-prong plug using the cables supplied with your player. (The cables needed to connect your audio system to the Roxio adaptor are not included with Easy VHS to DVD.)
VHS tapes are generally recorded at a lower quality than we have come to expect in the age of digital video, and degrades over time. Roxio Easy VHS to DVD will faithfully transfer your video to DVD. Roxio Easy VHS to DVD also includes some tools to sharpen, denoise and improve the quality of captured video, but cannot improve videos beyond the quality of the video as it was originally captured on the tape. The resulting video quality of the DVD will depend entirely on the quality of the source material. Low quality VHS video cannot be converted to high quality video by transferring it to a DVD disc.
You can get additional information on your current task at any time by pressing F1 to view the online help. Feel free to also check out the user guide, knowledgebase, or contact customer support for additional assistance.
This typically occurs when your VCR is not properly hooked up to the Roxio Video Capture device. If you are using RCA cables to connect your VCR to the Roxio Video Capture device, make sure the cables are connected to the correct jacks on both your VCR and your Roxio Video Capture device with the matching colors. The yellow RCA composite cable for video should be connected to the yellow RCA video output jack on your VCR and the yellow RCA video input jack on your Roxio Video Capture device. Similarly, the red and white audio RCA composite audio cables should be connected to the red and white audio output jacks on your VCR and the red and white audio input jack on your Roxio Video Capture device.
This can also occur when a connected VCR is turned off. If the VCR is powered on and you still get this message, try playing a VHS tape for 30 seconds to see if Easy VHS to DVD will detect a Video Signal and lock on to a frame. If there is no actual recording on the VHS tape, there will be no frame to lock on and it will result in No Video Signal.
On occasion, the VCR might lose tracking on the tape. The result will be scrambled video, discoloration, video distortion, etc. When this happens, VHS to DVD will lose track of the video frame resulting in a "No Video Signal".
A common way to adjust tracking on your VCR is to press the Channel UP or DOWN controls, either on the VCR or on its remote control. If this does not resolve the issue, please consult with your VCR's owner's manual for information on adjusting the tracking control.
Distorted audio when capturing from VHS to the computer and/or DVD can be caused by a number of factors. The audio could have been distorted at the time it was recorded to VHS. In other cases, it can happen because the recording input levels are set too high in the Roxio Easy VHS to DVD settings.
This can be caused by your default recording device being set to your microphone and not the VHS to DVD USB device. Open the Windows Sound Settings and ensure Line (Roxio Video Capture USB)" is selected from the Input dropdown.
Everything looks so easy and simple if you deal with making LEDs blink or playing tone with the tone() function, things become a lot more complex if you try to sample real audio and try to elaborate it for other porpuses.
To do that I read analog audio from an input jack (a microphone or iPod will be plugged to the jack) with the analogRead function. Since the voltage of the input oscillates in the range of few millivolts I guess I'd need to process (scale) a little bit the signal with an OpAmp (I've found a pretty simple circuit sketch of a LM386 OpAmp) in order to get more appreciable samples to be mapped to the 0 to 1023 bit of the internal ADC of Arduino.
What I read (the 10 bit values) is "noise-free"? Whatever I've read online (Arduino forum and others) about managing audio, assumes that you prescale the timers/clocks of the microprocessor in order to avoid/remove "digital" and DC offset noise. Is this mandatory or can I just use the standard delay() function to store my samples?
I didn't get if the digital/DC noise is only a problem if the audio has to be PWModulated back from a digital Pin!!!
What I read is the result of ADC, so it's already digital audio. Are this samples RAW audio? Is it sufficient to translate raw to .wav format (or so) if I want to play the audio from the PC?
I don't need to use PWM in order to "translate" my samples into a RAW audio format, am I right?
PWM is a DAC funtion and in case of audio playing, is required if my audio is to be played back to a speaker out from a digitalPin, right?
To do that I read analog audio from an input jack (a microphone or iPod will be plugged to the jack) with the analogRead function. Since the voltage of the input oscillates in the range of few millivolts I guess I'd need to process (scale) a little bit the signal with an OpAmp
The LM386 is a "power amplifier" designed to drive a speaker. [u]Here[/u] is the schematic for the SparkFun microphone breakout board. It has the power for the electret mic and it has biased output
What I read (the 10 bit values) is "noise-free"? Whatever I've read online (Arduino forum and others) about managing audio, assumes that you prescale the timers/clocks of the microprocessor in order to avoid/remove "digital" and DC offset noise.
However, there is quantization noise in any digital signal. Quantization noise rides on top of the signal so it's most noticeable with quiet sounds, but there is no quantization noise during "digital silence". If you want to know what quantization noise sounds like you can use an audio editor (such as Audacity) to downsample an audio file to 8-bits. It won't be as bad at 10-bits, and you can't hear any quantization noise at 16-bits (under normal/reasonable listening conditions).
The DC offset isn't "noise", it's the same as the bias I mentioned above. For example, with the input biased at 2.5V (half the 5V supply) silence will read 512 and the audio data will bie biased/offset at 512. You simply subtract 512 to get the unbiased audio data. Sometimes the digital-to-analog converter needs to be biased anyway, and in that case you can remove the DC voltage bias with a capacitor.
Yes, you'll have "raw" PCM data. A WAV file is basically RAW data in a file preceded by a header. The header identifies the file as a WAV and it gives the sample rate and bit-depth, and some other information so the audio data can be played back properly. You can research the format of a WAV file.
2- Quantization noise, according to my understanding, is something that you cannot avoid in ADC... anyway is something that can be reduced by simply increasing the sampling frequency, right? For a basic prototype, I believe It's something that I could not worry much, right?
3- Aliasing and Nyquist, on the other hand, is something that I've studied more the 25 years ago... so I guess I'd better read something to recall what they exactly state! When you say "You also need to filter-out any frequencies above the Nyquist limit (half the sample rate) before digitizing", you mean that before the signal is read by an Analog Pin, the signal must be "low-passed" by a filter that cuts frequencies equal to 1/2 the sampling rate?
4- About microcontroller programming: delay() does something different from setting a clock and so is not the correct way to define a sampling rate, understood! So I just have to rely on the sampling rate by defining the correct value of the ADC clock by means of commands like ADCSRA to prescale the ADC timers (this is not as user friendly programming as you can imagine for the common people, but I'll try to manage somehow) and whatelse? Can I simply analogRead(), send the read value over the serial() to the USB link and do my elaboration of the sampled raw audio on my PC? Is the serial() function somehow affected by the changes of the processor clock?
Over the last couple years I've been working on an ambitious audio library for Teensy. It's pretty much an endless project, with more requests for ever-more-powerful features as more gets made. But all the basics and quite a number of fairly sophisticated things are already done pretty well.
As you can see if you watch the video, this is audio processing on a microcontroller implemented pretty well. Much of it depends on far more advanced hardware than you get with the low-end 8 bit AVR-based boards, but still, maybe it can serve as a reference for some of your questions about how to do audio sampling well?
You can think of capacitors are 2 parallel plates with an insulator between them. The capacitance goes up as the plates get larger, or as the distance between them gets smaller, or as the electrical permittivity of insulating the material between the plates goes up. In an overly generalized way of thinking about these things, as thinner, higher permittivity materials are used, trade-offs in performance come into play. When an extremely thin layer of oxidation is used as the insulating material, you get these polarity restrictions.
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