iOS: How to trim silence from start and end of .aif audio recording?

£可爱£侵袭症+ 提交于 2019-12-03 16:52:47

This project takes audio from the microphone, triggers on loud noise and untriggers when quiet. It also trims and fades in/fades out around the ends.

https://github.com/fulldecent/FDSoundActivatedRecorder

Relevant code you are seeking:

- (NSString *)recordedFilePath
{
    // Prepare output
    NSString *trimmedAudioFileBaseName = [NSString stringWithFormat:@"recordingConverted%x.caf", arc4random()];
    NSString *trimmedAudioFilePath = [NSTemporaryDirectory() stringByAppendingPathComponent:trimmedAudioFileBaseName];
    NSFileManager *fileManager = [NSFileManager defaultManager];
    if ([fileManager fileExistsAtPath:trimmedAudioFilePath]) {
        NSError *error;
        if ([fileManager removeItemAtPath:trimmedAudioFilePath error:&error] == NO) {
            NSLog(@"removeItemAtPath %@ error:%@", trimmedAudioFilePath, error);
        }
    }
    NSLog(@"Saving to %@", trimmedAudioFilePath);

    AVAsset *avAsset = [AVAsset assetWithURL:self.audioRecorder.url];
    NSArray *tracks = [avAsset tracksWithMediaType:AVMediaTypeAudio];
    AVAssetTrack *track = [tracks objectAtIndex:0];

    AVAssetExportSession *exportSession = [AVAssetExportSession
                                           exportSessionWithAsset:avAsset
                                           presetName:AVAssetExportPresetAppleM4A];

    // create trim time range
    CMTime startTime = CMTimeMake(self.recordingBeginTime*SAVING_SAMPLES_PER_SECOND, SAVING_SAMPLES_PER_SECOND);
    CMTimeRange exportTimeRange = CMTimeRangeFromTimeToTime(startTime, kCMTimePositiveInfinity);

    // create fade in time range
    CMTime startFadeInTime = startTime;
    CMTime endFadeInTime = CMTimeMake(self.recordingBeginTime*SAVING_SAMPLES_PER_SECOND + RISE_TRIGGER_INTERVALS*INTERVAL_SECONDS*SAVING_SAMPLES_PER_SECOND, SAVING_SAMPLES_PER_SECOND);
    CMTimeRange fadeInTimeRange = CMTimeRangeFromTimeToTime(startFadeInTime, endFadeInTime);

    // setup audio mix
    AVMutableAudioMix *exportAudioMix = [AVMutableAudioMix audioMix];
    AVMutableAudioMixInputParameters *exportAudioMixInputParameters =
    [AVMutableAudioMixInputParameters audioMixInputParametersWithTrack:track];

    [exportAudioMixInputParameters setVolumeRampFromStartVolume:0.0 toEndVolume:1.0
                                                      timeRange:fadeInTimeRange];
    exportAudioMix.inputParameters = [NSArray
                                      arrayWithObject:exportAudioMixInputParameters];

    // configure export session  output with all our parameters
    exportSession.outputURL = [NSURL fileURLWithPath:trimmedAudioFilePath];
    exportSession.outputFileType = AVFileTypeAppleM4A;
    exportSession.timeRange = exportTimeRange;
    exportSession.audioMix = exportAudioMix;

    // MAKE THE EXPORT SYNCHRONOUS
    dispatch_semaphore_t semaphore = dispatch_semaphore_create(0);
    [exportSession exportAsynchronouslyWithCompletionHandler:^{
        dispatch_semaphore_signal(semaphore);
    }];
    dispatch_semaphore_wait(semaphore, DISPATCH_TIME_FOREVER);

    if (AVAssetExportSessionStatusCompleted == exportSession.status) {
        NSLog(@"AVAssetExportSessionStatusCompleted");
        return trimmedAudioFilePath;
    } else if (AVAssetExportSessionStatusFailed == exportSession.status) {
        // a failure may happen because of an event out of your control
        // for example, an interruption like a phone call comming in
        // make sure and handle this case appropriately
        NSLog(@"AVAssetExportSessionStatusFailed %@", exportSession.error.localizedDescription);
    } else {
        NSLog(@"Export Session Status: %d", exportSession.status);
    }
    return nil;
}

I'm recording the audio with an AVAudioRecorder, and saving it to an .aif file. I've seen some mention elsewhere of methods by which I could have it wait to start recording until the audio level reaches a threshold; that'd get me halfway there

Without adequate buffering, that would truncate the start.

I don't know of an easy way. You would have to write a new audio file after recording and analyzing it for the desired start and end points. Modifying the existing file would be straightforward if you knew the AIFF format well (not many people do) and had an easy way to read the file's sample data.

The analysis stage is pretty easy for a basic implementation -- evaluate the average power of sample data, until your threshold is exceeded. Repeat in reverse for end.

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