I'm looking for a simple C# real-time voice detection library.
The input should be an audio stream, and the output should be "human voice" or "not a human voice".
I have no knowledge in speech recognition or signal processing, and I'll appreciate any kind of assistance.
Take a look at the answer for "Detecting audio silence in WAV files using C#". I am assuming the input is a WAV file. If not please provide the format of the audio stream, or if you are intending on taking input from the microphone directly. If you can measure the amount of silence in an audio stream and you know the duration of the audio stream then you can calculate the amount of talk time. The link in the answer is dead, but if you go to codeproject.com and search on "C# wave form" you will get a hit on a number of projects that show you how to interpret and manipulate wav files. Detecting silence may be a little subjective if there is background noise. You will need to pick a minimum volume threshold for silence where anything below it is considered silence.
Related
IBM advises using Opus sound format for audio submitted to its Watson Speech to Text service. The idea being that Opus is designed specifically for speech.
Otherwise, it says you will get better quality transcription when submitting audio in flac format than in mp3 format. The latter has the obvious advantage of its small size. There is after all a 100Mb limit for file submissions. So you weigh the balance of your needs. That all makes sense so far.
But looking at conversions done on a source WAV file, the Opus file is size is comparable with mp3.
Downsampling a 366Mb wav file to 8k sample rate (one of two sample rates advised for using the service), created a wav file of 66.4Mb. Converting that to flac, wav and opus produced flac: 43.6Mb; mp3: 6.2Mb; opus: 9.8Mb.
So is opus really the best choice for getting the most accurate transcription? And how can that be when it is so small compared to flac?
Opus is designed to efficiently encode speech. The details are explained in the linked wiki article, but just to give you a gist, consider that human vocalisation range is rather limited, roughly from 80 to 260 Hz. On the other hand, or hearing range is far greater, up to 20000 Hz. Whereas music encoders (like mp3) have to work roughly within our hearing range, voice-specialised encoders (like Opus) can focus on what matters to efficiently encode human voice, with no interest what lies significantly above our vocalisation range. That I hope provides some intuition why Opus is so efficient.
Is it the best? It's somewhat opinionated, but yes, I think it's among the best choices out there. To cite after Wikipedia, Opus replaces both Vorbis and Speex for new applications, and several blind listening tests have ranked it higher-quality than any other standard audio format at any given bitrate.
Recently, I discover that my tutorial videos could be seen at 1.5x playback speed without losses in quality (they are actually better to see, as I normally speak slowly). My problem is that if I change the speed of the video when using a video editor, like Kdenlive, the audio becomes distorted and turns into a mess (higher pitch, I believe).
How could I obtain the same quality as VLC "playback fast" and Youtube "playback speed 1.5" for the audio track? I'm a layman in audio/video editing, so I'm also satisfied with partial answers, like the identification of which terms I should search for in this case.
It might be better to take your audio track and use something like Sound Forge to automatically remove silence. Just be sure to add a pad to that (built into sound forge) otherwise the speech will sound way to chopped and fast.
Aside from that, you could also use Vegas to (then) chop the video to keep pace with your new speech rate. Vegas is a video editing program that is best for this kind of down and dirty editing.
How to get file information like sampling rate, bit rate etc of .raw audio files using terminal in linux? Soxi works for .wav files but it isn't working for .raw.
If your life depended on discovering an answer you could make some assumption to tease apart the unknowns ... however there is no automated way since the missing header would give you the easy answers ...
The audio analysis tool called audacity allows you to open up a RAW file, make some guesses and play the track
http://www.audacityteam.org
In audacity goto File -> Import -> Raw Data...
Above settings are typical for audio ripped from a CD ... toy with trying stereo vs mono for starters.
Those picklist widgets give you wiggle room to discover the format of your PCM audio given that the source audio is something when properly rendered is recognizable ... would be harder if the actual audio was noise
However if you need a programmatic method then rolling your own solution to ask those same questions which appear in above window is possible ... is that what you need or will audacity work for you ? We can go down the road of writing code to play off the unknowns mentioned in #Frank Lauterwald's comment
To kick start discovering this information programmatically, if the binary raw audio is 16 bit then each audio sample (point on the audio curve) will consume two bytes of your PCM file. For mono audio then the following two bytes would be your next sample, however if its stereo then these two following bytes would be the sample from the other channel. If more than two channels then just repeat. Typical audio is little endian. Sampling rate is important when rendering the audio, not when programmatically parsing raw bytes. One approach would be to create an output file with a WAV header followed by your source PCM data. Populate the header with answers from your guesswork. This way you could listen to this output file to help confirm your guesses.
Here is a sample 500k mono PCM audio file signed 16 bit which can be imported into audacity or used as input to rolling your own identification code
The_Constructus_Corporation_Long_Street-ycexQvMy03k_excerpt_mono.pcm
I have built a source client using Portaudio and LAME which streams the microphone input to an Icecast server to be listened to online via the HTML5 tag. I have managed to (supposedly) get the quality of the stream to MP3 320kbps at 44.1kHz and am looking for a way to confirm this using tests and or benchmarks.
I have an indication that these stats are somewhat correct from looking at stream inspectors in software such as iTunes and VLC, but I am looking to get a more in-depth data set.
What I basically want is to be able to test how much of the original file is being lost over the stream and if or how much the quality changes depending on environmental conditions of the broadcaster or streamer.
Does anyone know of any tools, frameworks to get some hard numbers or representations of this data?
If VLC tells you the stream is 320kbit CBR, then it is.
It sounds like what you're looking for is a comparison of the actual audio content. This is highly subjective. MP3 is built to use features of how our hearing works to save bandwidth. For example, quiet sounds are masked by loud sounds. High frequencies are harder to hear and are simply rolled off.
You can compare the spectral analysis between the original PCM-sampled waveform and the MP3 decoded waveform, but this doesn't tell you how humans interpret that sound. For that, you would have to survey humans.
Having just witnessed Sound Load technology on the Nintendo DS game Bangai-O Spritis. I was curious as to how this technology works? Does anyone have any links, documentation or sample code on implementing such a feature, that would allow the state of an application to be saved and loaded via audio?
Its the same old thing used in ZX Spectrum era. You load programs/games from tape.Only the sound quality and the filters are probably better.
In my opinion something like Bluetooth or WiFi is better. You can also send files that can be put on some storage and then load them. I find these methods much easier than sound because if there is a lot of noise around you cannot do much.
It is just a conversion of data to audio and then back from audio to data.
Search for Zotyocopy and Copy86M on google - these are the utilities used for saving a game to tape after loading it into memory on zx spectrum.
If you want to pass data as audio through the air there are a few things you need to be aware of though, such as how the speaker and microphone interact for example. It is important that they don't distort or alter the sound too much as what you are sending are in fact the raw bytes.
Some audio software will let you open any file as audio so that you may listen to it. If you record audio as data do not use lossy compression such as mp3 on the audio file!