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POiphone low pass filter
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copied!<p>I'm trying to implement a low-pass filter for an iphone app where I record a sound and then it gets played back slightly muffled; like the voice is coming from another room. </p> <p>I've looked into the different options for audio recording and manipulation and have found it a bit confusing...digital signal processing isn't a strong point at all. I've mainly looked into OpenAL and inside the EFX library there is a filter that specifically does what I need, but EFX is not included on the iPhone. Is there a way of replicating that behaviour using OpenAL for the iPhone? Is there another option such as Audio Units that could provide a solution?</p> <p>Thanks for your help</p> <p>EDIT:</p> <p>So after Tom's answer and links, I've come up with what I think is a correct implementation. However, I'm not getting a muffling effect at all, rather just a decrease in volume. Here's the (summarised) code I have currently:</p> <p>File is recorded using AVAudioRecorder and the following settings:</p> <pre><code>[recordSetting setValue :[NSNumber numberWithInt:kAudioFormatLinearPCM] forKey:AVFormatIDKey]; [recordSetting setValue:[NSNumber numberWithFloat:44100] forKey:AVSampleRateKey]; [recordSetting setValue:[NSNumber numberWithInt: 1] forKey:AVNumberOfChannelsKey]; [recordSetting setValue :[NSNumber numberWithInt:16] forKey:AVLinearPCMBitDepthKey]; [recordSetting setValue :[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsBigEndianKey]; [recordSetting setValue :[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsFloatKey]; </code></pre> <p>I then read in the file and transform it with the code below:</p> <pre><code>// Read in the file using AudioFileOpenURL AudioFileID fileID = [self openAudioFile:filePath]; // find out how big the actual audio data is UInt32 fileSize = [self audioFileSize:fileID]; // allocate the memory to hold the file SInt16 * outData = (SInt16 *)malloc(fileSize); // Read in the file to outData OSStatus result = noErr; result = AudioFileReadBytes(fileID, false, 0, &fileSize, outData); // close off the file AudioFileClose(fileID); // Allocate memory to hold the transformed values SInt16 * transformData = (SInt16 *)malloc(fileSize); // Start the transform - Need to set alpha to 0.15 or below to have a noticeable affect float alpha = 1; // Code as per Tom's example transformData[0] = outData[0]; for(int sample = 1; sample < fileSize / sizeof(SInt16); sample ++) { transformData[sample] = transformData[sample - 1] + alpha * (outData[sample] - transformData[sample - 1]); } // Add the data to OpenAL buffer NSUInteger bufferID; // grab a buffer ID from openAL alGenBuffers(1, &bufferID); // Add the audio data into the new buffer alBufferData(bufferID,AL_FORMAT_MONO16,transformData,fileSize,44100); </code></pre> <p>So after all that, I then play it through OpenAL using the standard method (I don't think it has any impact on my results so I won't include it here.)</p> <p>I've traced through the results, both before and after transform, and they seem correct to me i.e. the values before vary positively and negatively as I would expect and the for loop is definitely flattening out those values. But as I mentioned before I'm only seeing (what seems to me) a reduction in volume, so I'm able to increase the gain and cancel out what I've just done.</p> <p>It seems that I must be working on the wrong values. Any suggestions of what I'm doing wrong here?</p>

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