Is it possible to retrieve the characteristics of the audio output signal dedicated to headphones during audio playback? I don't want to record the signal per se (that is, I don't want to record the radio with a tape player to make an 80s style mixtape), but instead I only want to know the strength, in decibels, of the signal for medical research purposes.
I was thinking this could maybe be accomplished in the same manner a visualizer does?
If you're writing an Android app, I think you're looking for the AudioManager, through which you can query the volume of the different audio streams.
Calling getStreamVolume should get you the data you want.
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I have an application that needs to use a high-quality microphone and headphones simultaneously. On my phone, there is only one input for an audio device. How can I connect to and handle in/out data from each?
edit: I dont want to capture two input sources at once. I want to record from a microphone, use it to modify audio which i have already recorded, then output the result to a headphone in real time. I know this is possible, however I dont know how to connect and differentiate between two devices connected to the audio jack via a splitter.
Most (probably all) Android devices with a 3.5mm port support a TRRS connector(https://en.m.wikipedia.org/wiki/Phone_connector_(audio)#TRRS_standards)
So there is a different hardware channel for the microphone and the headphones, and it's not possible to accidentally use the wrong one in software or for that matter manually select what to use the hardware connections for, it's predefined.
But so the point is that you need the right splitter exactly for this, not just a normal audio-splitter but one that specifically splits the input/microphone channel from the output/heaphone channel.
If you use the search term "audio and microphone 3.5mm splitter android", then you should find the correct splitter hardware shopping recommendations:
https://www.google.com/search?q=audio+and+microphone+3.5mm+splitter+android&tbm=shop
Here is an example picture:
I want to extract audio data to memory using vlcj (https://github.com/caprica/vlcj, version 4.2.0). I don't want to play the video on the same time, just extract the audio data, as fast as the performance allows.
Right now I'm using a workaround based on this: https://github.com/caprica/vlcj/blob/master/src/test/java/uk/co/caprica/vlcj/test/rip/RipAudioTest.java, i.e. output the data to a file first, and then read the file. While this solution is working, it's not optimal and it takes disk space.
Maybe the example above can be modified to direct the audio to a callback instead.
A second example is:
https://github.com/caprica/vlcj/blob/master/src/test/java/uk/co/caprica/vlcj/test/directaudio/DirectAudioPlayerTest.java
In that example, the audio data is extracted to memory, which is what I want. But it also plays the video in a window, which is not what I want. Maybe that example can be modified to turn off the video somehow, and make it run as fast as possible?
There's no perfect answer here, sadly.
Using the DirectAudioPlayerTest.java that you found already, you can change the media player factory creation to pass this parameter to prevent the video window being opened:
factory = new MediaPlayerFactory("--novideo");
You will receive the audio into a memory buffer at the rate at which VLC decodes it, so if you have a 5 minute video it will take 5 minutes to extract the audio - not ideal.
This is the only way with vlcj that you can grab the audio data in memory.
The RipAudioTest.java that you found, IIRC, extracts the audio as quickly as possible, which may be a lot faster than the normal playback speed - but here you can't grab the decoded audio directly into your Java application.
So the solution you already have, ripping the track to disk first, might actually be the best solution you can achieve with vlcj here (since it could be considerably quicker than using the direct audio player).
I'm trying to create an Android app which will get the lyrics of an mp3 from the ID3V2 tag of it. My question is, is it possible to get the lyrics automatically highlighted as the song plays? Like using speech processing or things like that. I've looked into the previous similar questions but all of them requires manual input. Need an ASAP feedback. Thank you.
This kind of thing is possible on Hollywood movie sets, using technology similar to those image enhancements that reconstruct a face using a 4-pixel square as input.
Okay, so your request is theoretically more feasible, but no current phone technology I know of could do this on the fly. You might need a Delorean, flux capacitor and some plutonium.
Also, detecting vocals over music is a much harder problem than speaking a text message into your phone:
Sung lyrics do not usually follow natural speech rhythm;
The frequency spectrum of music tends to conflict with the frequency spectrum of voice;
The voice varies in pitch, making it much harder to isolate and detect phonetic features;
The vocals are often mixed at a level equal to all other musical instruments;
IwannahuhIwannahuhIwannahuhIwannahuhIwannaReallireallirealliwannaZigaZiggUHH.
You might take a look at this paper LyricSynchronizer: Automatic Synchronization System Between Musical Audio Signals and Lyrics for a possible solution. Nothing implemented in Java for Android, but with the NDK you might take any C code and finagle it to work. ;-)
This paper describes a system that can automatically synchronize polyphonic musical audio signals with their corresponding lyrics. Although methods for synchronizing monophonic speech signals and corresponding text transcriptions by using Viterbi alignment techniques have been proposed, these methods cannot be applied to vocals in CD recordings because vocals are often overlapped by accompaniment sounds. In addition to a conventional method for reducing the influence of the accompaniment sounds, we therefore developed four methods to overcome this problem: a method for detecting vocal sections, a method for constructing robust phoneme networks, a method for detecting fricative sounds, and a method for adapting a speech-recognizer phone model to segregated vocal signals. We then report experimental results for each of these methods and also describe our music playback interface that utilizes our system for synchronizing music and lyrics.
Best of luck in your implementation!
I’m trying to develop an application that is capable of identifying a bird sound from a wav file recording. When creating the database im using another collection of sound clips and am trying to get a unique identification to them. Im planning to do this using FFT.(I don’t have any issues with these concepts) The question is, is it important to clear the noise of these base recording before creating the unique identification? If so, will anyone be able to help me with the concept of “Zero-crossing rate” and some other technique to clear the sound file for noise and silence.Thanks in advance.
In general, there is no way to remove noise unless you already have an accurate way of indentfying a temporal or spectral difference between the noise and the signal of interest. For instance, if you know the exact frequency bandwidth of the entire signal of interest, then you can use DSP to filter out the spectrum outside if that bandwidth. If you know the minimum amplitude of your signal of interest, then you can clip out everything below that level.
I was playing with a karaoke application on iPhone and came up with following questions:
The application allowed its users to control the volume of the artist; even mute it. How is this possible?
Does adjusting artist sound/setting equalizer etc. mean performing some transformation of required frequencies? What sort of mathematics is required here(frequency domain transformations)?
The application recorded users voice input via a mic. Assuming that the sound is recorded in some format, the application was able to mix the recording with the karaoke track(with artists voice muted). How can this be done?
Did they play both the track and voice recording simultaneously? Or maybe they inserted additional frequency(channel?) in the original track, maybe replaced it?
What sort of DSP is involved here? Is this possible in Java, Objective C?
I am curious and if you have links to documents or books that can help me understand the mechanism here, please share.
Thanks.
I don't know that particular application, probably it has a voice track recorder separately.
For generic 2-channels stereo sound the easiest voice suppression can be performed assuming that artist's voice is somehow equally balanced between two channels (acoustically it appears in center). So the simplest 'DSP' would be subtract one channel from another. It does not work that well however with modern records since all instruments and voice are recorded separately and then mixed together (meaning that voice will not be necessarily in phase between two channels).
I have written two detailed blogposts on how to get a custom EQ in iOS. But i have no details about how to do the DSP yourself. If you simply want to choose between a wide range of effects and stuff, try this.
First post explains how you build libsox:
http://uberblo.gs/2011/04/iosiphoneos-equalizer-with-libsox-making-it-a-framework
The second explains how to use it:
http://uberblo.gs/2011/04/iosiphoneos-equalizer-with-libsox-doing-effects
please up the answer if it helped you! thanks!