audio-processing | 易学教程

Audio file get left/right channels

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问题 After a lot of research, I can't find a clear answer. What I want to achieve is: take an mp3/wav file and save its left and right channels to 2 byte[]. Then I can manipulate each channel and then add them up again and create a new wav file. (preferably the solution to be in Java) Related question is split two channels of AudioRecord of CHANNEL_IN_STEREO. However this is using audiorecord, how to apply it to reading from audio file? Thanks 回答1: Well here is the program to read a 16 bit wave

Acoustic Audio Comparing Library

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I need a software or a library which handles with audio comparison, but not using the tag's inside mp3 ,it should compare similarity or confidence between 2 audio Files, or if i cut a piece from an audio file, the software should point where is that file token from the main audio file (i hope i was clear enough). So how i heard this technology is called Audio Acoustic Comparing , and based on some audio sample file, which we can call fingerprint . The software should point me if it finds an equivalent of the input sample or fingerprint, somewhere in the file. Bests. libfooid is free. It's dual

processing an audio wav file with C

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问题 I'm working on processing the amplitude of a wav file and scaling it by some decimal factor. I'm trying to wrap my head around how to read and re-write the file in a memory-efficient way while also trying to tackle the nuances of the language (I'm new to C). The file can be in either an 8- or 16-bit format. The way I thought of doing this is by first reading the header data into some pre-defined struct, and then processing the actual data in a loop where I'll read a chunk of data into a

AVFoundation audio processing using AVPlayer's MTAudioProcessingTap with remote URLs

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There is precious little documentation on AVAudioMix and MTAudioProcessingTap, which allow processing to be applied to the audio tracks (PCM access) of media assets in AVFoundation (on iOS). This article and a brief mention in a WWDC 2012 session is all I have found. I have got the setup described here working for local media files but it doesn't seem to work with remote files (namely HLS streaming URLs). The only indication that this is expected is the note at the end of this Technical Q&A : AVAudioMix only supports file-based assets. Does any one know more about this? is there really no way

Java sound visualizer

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问题 i'm trying to make a java sound visualizer, but I have absolutely no idea on how to get the bytes from the audio extracted, immediately after they are procressed in real-time. I could synchronize the program with an wav file, but that is not what I want to do, I want to generate the sound with the program, then play it, without saving it anywhere. Thanks for the help! 回答1: This article can help you understand the structure of sound files: http://codeidol.com/java/swing/Audio/Build-an-Audio

read multiple wav files in matlab

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问题 I want read multiple wav files one by one in one folder. I wrote this way, but it gives "Invalid Wave File. Reason: Cannot open file." error. But when i change t to number, it works. for t=1:10 myFile=['path\','t.wav']; [ speech, fs] = wavread( myFile); end 回答1: You need to convert the variable t to a string. You were asking to open the file 'path\t.wav' , which presumably doesn't exist. Since the variable t is an integer, you can use int2str to convert it to a string: myFile = ['path\'

read multiple wav files in matlab

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I want read multiple wav files one by one in one folder. I wrote this way, but it gives "Invalid Wave File. Reason: Cannot open file." error. But when i change t to number, it works. for t=1:10 myFile=['path\','t.wav']; [ speech, fs] = wavread( myFile); end You need to convert the variable t to a string. You were asking to open the file 'path\t.wav' , which presumably doesn't exist. Since the variable t is an integer, you can use int2str to convert it to a string: myFile = ['path\' int2str(t) '.wav']; Only strings can be concatenated with other strings. Of course if you have fewer than 10

How can I obtain the raw audio frames from the microphone in real-time or from a saved audio file in iOS?

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问题 I am trying to extract MFCC vectors from the audio signal as input into a recurrent neural network. However, I am having trouble figuring out how to obtain the raw audio frames in Swift using Core Audio. Presumably, I have to go low-level to get that data, but I cannot find helpful resources in this area. How can I get the audio signal information that I need using Swift? Edit : This question was flagged as a possible duplicate of How to capture audio samples in iOS with Swift?. However, that

Changing Pitch and Frequency of Recorded Audio

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问题 I've been trying to adjust the pitch of a record piece of audio following with code from: http://developer.android.com/guide/topics/media/audio-capture.html My guess is that this adjustment should be done with the MediaRecorder . http://developer.android.com/reference/android/media/MediaRecorder.html However, I am unsure which method to call to change the pitch? Looking through SO, I found changing the frequency of a soundfile in android but I am confused as to how to integrate a SoundPool

Algorithms for determining the key of an audio sample

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问题 I am interested in determining the musical key of an audio sample. How would (or could) an algorithm go about trying to approximate the key of a musical audio sample? Antares Autotune and Melodyne are two pieces of software that do this sort of thing. Can anyone give a bit of a layman's explanation about how this would work? To mathematically deduce the key of a song by analysing the frequency spectrum for chord progressions etc. This topic interests me a lot! Edit - brilliant sources and a