Now I'm creating video with FFMPEG library. I'm receiving audiofile by recording user's voice.
How can I normalize sound. Maybe ffmpeg have tools. Or can you recommend you some algorithm for normalization?
The canonical tool for normalizing audio signals is called... wait for it... normalize. I recommend you use it, either by calling it or by studying its source and doing something similar yourself. Normalizing isn't difficult, you just decide on a maximum safe amplitude and then scale every sample according to that.
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Decreasing the playback speed of AudioPlayer severely decreases the quality of the audio being played; the audio becomes very "noisy".
Is there any way to fix this or is it an issue with the just_audio implementation?
Reproduce:
final AudioPlayer player = AudioPlayer(); // Create audio player
player.setAsset("..."); // Load audio file
player.setSpeed(0.5); // Halve speed
player.play(); // Start playing
Just to preface this answer, time stretching is a difficult thing to do in real-time because it has to stretch time without stretching the sound waves (stretching the sound waves would lower the frequency and hence the pitch, so it has to stretch time while filling the gaps with fabricated extensions to the existing sound waves). As a result, the very best real time algorithm will still introduce artifacts and distortions.
Now to answer your question, just_audio doesn't provide any options to change the time stretching algorithm, but it does use the best available algorithms for each platform, for general purpose usage. The Android implementation uses Sonic which is better quality than Android's own built-in algorithm. On iOS/macOS, AVAudioTimePitchAlgorithmTimeDomain is used which seems to produce the least distortion at speeds below 1.0 out of the different algorithms Apple provides, although newer iPhones/iOS versions may produce higher quality output. On web browsers, it uses whatever algorithm that web browser provides.
If you need to try out alternatives, you would need to make a copy of just_audio and edit the code that selects the algorithm. You are unlikely to find better options for Android and web, but you might like to experiment with the different iOS/macOS algorithms by searching for AVAudioTimePitchAlgorithmTimeDomain in the code and changing it to one of the other options listed in Apple's documentation. You may find one of the other algorithms works better if you have a specialised use case.
I'm not sure if it's allowed ask these questions here, but looks important for us webdevelopers (even bad dev like me :p ).
The question is about export setting videons on Premiere. I'm looking for a background video 30s like airbnb or paypal. Yesterday, I check paypal size and it's only 10/15 Mb for more than a minute. How did they do?
Obviously you want a low average bit rate. Things that can help with that are: keep the resolution low (you can scale it up a bit on the client); use H.264 High Profile (for the H.264 version); use 2-pass encoding; use variable bit rate. You can try increasing the GOP length too.
I assume there's no audio, so that shouldn't be an issue. (Can't remember if Adobe has an option for no sound track, but you can set the audio to a very low bit rate, or post-process it with ffmpeg or something to remove the audio track.)
If you have any control over the video content, you can try to keep it compressible. For example, avoid video with lots of detail or rapid motion. You might be able to selectively blur parts in a way that doesn't look bad. If it doesn't move too fast, you might be able to decrease the frame rate.
If you really want to optimize, you'll probably need to experiment a lot.
I know how to play mp3 files and whatnot in Xcode iOS. But how do I play a certain frequency, like if I just wanted to emit a C# note for 25 seconds; how might I do that? (The synth isn't as important to me as just the pitch of the note.)
You need to generate the PCM audio waveform that corresponds to the note you want to play and store that into a sample buffer in memory. Then you send that buffer to the audio hardware.
Here is a tutorial on generating waveforms of several types. The article goes into some details on the many aspects to a note you need to consider, including the frequency, volume, waveform shape, sampling rate, etc. The article comes with Flash source code, I think you should have no problem taking the concepts and adapting them to iOS.
If you also need a library that you can use to play the generated buffers on iOS, then I recommend the open source Finch.
I hope this helps!
You can synthesize waveforms of your desired frequency and feed them to the callbacks of either the Audio Queue or the RemoteIO Audio Unit API.
Here is a short tutorial on some of the code needed to create sine wave tones for iOS in C.
So I want to stream the audio from a mic using NAudio and then pass that stream to WCF which a Siverlight app can consume to broadcast the live audio sound. I want the latency to be as low as possible.
Any suggestions or if some one has already done it please point the source. Thanks in advance
what you are asking is certainly possible, but will be a fair amount of work to do.
NAudio can handle to capturing microphone audio.
At the Silverlight end you can play custom audio formats (in this case PCM) using a custom media element streaming source. See this one: http://code.msdn.microsoft.com/wavmss
I suspect latency would not be very good. You can reduce it by keeping the buffer sizes small. Also bear in mind that WAV is not a very efficient format to be sending over the network.
To have low latency as possible, you should use the netTcpBinding and stream your audio in binary format. I would use MemoryStream for this and try to play with the buffersize to figure out what the best performance is. Also, try checking audio formats for best performance. This also depends of the audio quality you expect.
I'm looking into writing an app that runs as a background process and detects when an app (say, Safari) is playing audio. I can use NSWorkspace to get the process ID's of the currently running applications but I'm at a loss when it comes to detecting what those processes are doing. I assume that there is a way to listen in on a process and detect what public messages the objects are sending. I apologize for my ignorance on the subject.
Has anyone attempted anything like this or are aware of any resources that can help?
I don't think that your "answer" is an answer at all...
and there IS an answer (which is not "42")
your best bet for doing this would be to write a pass-through audio output device. Much like soundflower, actually. so your audio output device would then load the actual (physical) audio output device and pass the audio data along to it directly (after first having a look at the audio stream, of course!). then you only need to convince your users to configure your audio device as the default audio output device so that the majority of applications which play sound will use it automatically. and voila...
your audio processing function will probably just do a quick RMS on the buffer before passing it along to the actual output device. and when the audio power crosses a certain threshold (probably something like -54dB with apple audio hardware), then you know that some app is making sound.
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SoundFlower is an open-source project that allows Mac OS X applications to pass audio to each other. It almost certainly does something similar to what you describe.
I've been informed on another thread that while this is possible, it is an extremely advanced technique and not recommended. It would involve using Application Enhancer (APE) and is considered a not 'nice' thing to do. Looks like that app idea is destined for the big recycling bin in the sky :)