I have a situation where I have a video capture of HD content via HDMI with audio from a sound board that goes through a impedance drop into a microphone input of a camcorder. That same signal is split at line level to a 'line in' jack on the same computer that is capturing the HDMI. Alternatively I can capture the audio via USB from the soundboard which is probably the best plan, but carries with it the same issue.
The point is that the line in or usb capture will be much higher quality than the one on HDMI because the line out -> impedance change -> mic in path generates inferior quality in that simply brushing the mic jack on the camera while trying to change the zoom (close proximity) can cause noise on the recording.
So I can do this today:
Take the good sound and the camera captured sound and load each into
audacity and pretty quickly use the timeshift toot to perfectly fit
the good audio to the questionable audio from the HDMI capture and
cut the good audio to the exact size of the video. Then I can use
ffmpeg or other video editing software to replace the questionable
audio with the better audio.
But while somewhat quick and easy, it always carries with it a bit of human error and time. I'd like to automate this if possible as this process is repeated at least weekly throughout the year.
Does anyone have a suggestion if any of these ideas have merit or could suggest another approach?
I suspect but have yet to confirm that the system timestamp of the start time may be recorded in both audio captured with something like Audacity, or the USB capture tool from the sound board as well as the HDMI mpeg-2 video. I tried ffprobe on a couple audacity captured .wav files but didn't see anything in the results about such a time code, but perhaps other audio formats or other probing tools may include this info. Can anyone advise if this is common with any particular capture tools or file formats?
if so, I think I could get best results by extracting this information and then using simple adelay and atrim filters in ffmpeg to sync reliably directly from the two sources in one ffmpeg call. This is all theoretical for me right now-- I've never tried either of these filters yet-- just trying to optimize against blind alleys by asking for advice up front.
If such timestamps are not embedded, possibly I can use the file system timestamp for the same idea expressed in 1a, but I suspect the file open of the two capture tools may have different inherant delays. Possibly these delays will be found to be nearly constant and the approach can work with a built-in constant anticipation delay but sounds messy and less reliable than idea 1. Still, I'd take it, if it turns out reasonably reliable
Are there any ffmpeg or general digital audio experts out there that know of particular filters that can be employed on the actual data to look for similarities like normalizing the peak amplitudes or normalizing the amplification of the two to some RMS value and then stepping through a short 10 second snippet of audio, moving one time stream .01s left against the other repeatedly and subtracting the two and looking for a minimum? Sounds like it could take a while, but if it could do this in less than a minute and be reliable, I suspect it could work. But I have only rudimentary knowledge of audio streams and perhaps what I suggest is just not plausible-- but since each stream starts with the same source I think there should be a chance. I am just way out of my depth as to how to go down this road, so if someone out there knows such magic or can throw me some names of filters and example calls, I can explore if I can make it work.
any hardware level suggestions to take a line level output down to a mic level input and not have the problems I am seeing using a simple in-line impedance drop module, so that I can simply rely on the audio from the HDMI?
Thanks in advance for any pointers or suggestinons!
Related
My Zoom H4n somehow decided it didn't want to properly save two recordings this weekend, leaving me with four zero byte files (which I have tried any which way to open/convert, but nothing was working).
I then used CardRescue to scan the SD card for any audio it could find, and - lo and behold - I got .wav files! However, instead of two files for each session (one was an XLR output from the desk, the other the on-Zoom mics), or even a nice stereo with one left, the other right, I have a mess.
In importing as raw data to Audacity (the rescued .wavs themselves do not open), the right channel has the on-Zoom mic audio, with intermittent silence. The left has the on-Zoom audio, followed by the same part of the XLR input audio. This follows the same pattern as the silences.
I have spent hours chopping up in Garageband, but as it is audio for a video, it needs to match what 'really' happened perfectly (I appreciate for a podcast/audio-only I could relatively simply take away the on-Zoom mic audio from the left channel). I began attempting to sync the mic audio to the on-camera audio (which, despite playing around with settings is as unusable as it always is) but because it's a pattern, can't help but wonder if there's a cleaner fix: either analysing the audio somehow as there are clean lines if I look at the spectral data, or a case of adding a couple of numbers to the wav's binary that'd click the two into place?
I've tried importing to Audacity with different settings, different offsets - this has ended up in either slow audio, fast audio, or heavily distorted audio (but always the same patterns with the files).
I use a Mac (and don't know any PC users close by!) so any software suggestions will need to run on Mac. However, I'm willing to try just about anything that's not dragging tiny clips.
I have recorded an audio.
I dont know how it happened that only one sided speech is recorded and the other speech is recorded with a very low sound.
Is there any solution to amplify the other side signal.
any help would be much appreciated.
This question is probably more appropriately asked at a forum where recording and mixing is discussed. For example: https://sound.stackexchange.com/
The ideal would be to improve your recording situation, to control factors so the sound are more closely matched. (Match microphones, isolate the speakers from environmental sounds, optimize input levels, etc.)
After that, the next option or step is to pre-process your audio files with a tool like Audacity. Use this or another DAW (Digital Audio Workstation) tool to match amplitudes or employ noise filtering or a range of other tools.
Audio processing is both tricky (an "art") and cpu intensive, so it's good to get as much of this handled as possible before the sounds are imported into a program.
I am quite new to LabVIEW and NI devices.
I am working on Active Noise Cancellation Project, where I will be using two microphones input and one loud speaker as output. I have NI myRIO 1900 and CDAQ 9178 devices in our university lab. I need to do real time audio processing, I will collect data from microphone and process it using filtered XLMS algorithm to produce anti noise from loud speaker and other microphone is error microphone. I want to process data so quickly( within 1.7 msec ) so I will have real time response at 44100 sample rate !! My question is , 'is it possible to do with labview ?? and is stream processing possible in labVIEW?? and can I achieve so small audio latencies as mentioned above ??'
I have searched for audio processing objects in labview help. I can only find 'Acquire Sound', 'Play Waveform', surprisingly 'Acquire Sound configuration ' will work only for duration of minimum of 1 second not less than that !!! I can't input the time milli seconds !!!( I am still facing problem installing myRIO, so I have used host computed VI to do this.)
Please help !! Thank You
The thing you should be looking into is the FPGA part of the myRIO. You’re never going to be able to get 1.7ms response time via the host computer. The FPGA can access the Analogue inputs and outputs, so if you can get your algorithm to compile onto the FPGA then it should work.
Yes, it is possible with LabVIEW, insofar as any algorithm you want to code up can be executed by LabVIEW. If you're asking whether there is a library that already exists to do the filtering you're wanting to do, you may want to explore the NI Sound & Vibration toolkit, which is sold separate from LabVIEW, or explore third-party libraries.
The raw waveform mathematics abilities that come with LabVIEW are fairly extensive. You should be able to code whatever transforms you want if you know the base math.
I'm trying to make a video tutorial, so i decided to record the speeches using a TTS online service.
I use Audacity to capture the sound, and the sound was clear !
After dinning, i wanted to finish the last speeches, but the sound wasn't the same anymore, there is a background noise(parasite) which is disturbing, i removed it with Audacity, but despite this, the voice isn't the same ...
You can see here the difference between the soundtrack of the same speech before and after the occurrence of the problem.
The codec used by the stereo mix peripheral is "IDT High Definition Codec".
Thank you.
Perhaps some cable or plug got loose? Do check for this!
If you are using really cheap gear (built-in soundcard and the likes) it might very well also be a problem of electrical interference, anything from ...
Switching on some device emitting a electro magnetic field (e.g. another monitor close by)
Repositioning electrical devices on your desk
Changes in CPU load on your computer (yes i'm serious!)
... could very well cause some kinds of noises with low-fi sound hardware.
Generally, if you need help on audio sounding wrong make sure that you provide a way to LISTEN to the files, not just a visual representation.
Also in your posted waveform graphics i can see that the latter signal is more compressed, which may point to some kind of automated levelling going on somewhere in the audio chain.
I'm trying to compare sound clips based on microphone recording. Simply put I play an MP3 file while recording from the speakers, then attempt to match the two files. I have the algorithms in place that works, but I'm seeing a slight difference I'd like to sort out to get better accuracy.
The microphone seem to favor some frequencies (add amplitude), and be slightly off on others (peaks are wider on the mic).
I'm wondering what the cause of this difference is, and how to compensate for it.
Background:
Because of speed issues in how I'm doing comparison I select certain frequencies with certain characteristics. The problem is that a high percentage of these (depending on how many I choose) don't match between MP3 and mic.
It's called the response characteristic of the microphone. Unfortunately, you can't easily get around it without buying a different, presumably more expensive, microphone.
If you can measure the actual microphone frequency response by some method (which generally requires having some etalon acoustic system and an anechoic chamber), you can compensate for it by applying an equaliser tuned to exactly inverse characteristic, like discussed here. But in practice, as Kilian says, it's much simpler to get a more precise microphone. I'd recommend a condenser or an electrostatic one.