So, I'm trying to do a simple calculation over previously recorded audio (from an AVAsset) in order to create a wave form visual. I currently do this by averaging a set of samples, the size of which is determined by dividing the audio file size by the resolution I want for the wave form.
This all works fine, except for one problem....it's too slow. Running on a 3GS, processing an audio file takes about 3% of the time it takes to play it, which is way to slow (for example, a 1 hour audio file takes about 2.5 minutes to process). I've tried to optimize the method as much as possible but it's not working. I'll post the code I use to process the file. Maybe someone will be able to help with that but what I'm really looking for is a way to process the file without having to go over every single byte. So, say given a resolution of 2,000 I'd want to access the file and take a sample at each of the 2,000 points. I think this would be a lot quicker, especially if the file is larger. But the only way I know to get the raw data is to access the audio file in a linear manner. Any ideas? Here's the code I use to process the file (note, all class vars begin with '_'):
So I've completely changed this question. I belatedly realized that AVAssetReader has a timeRange property that's used for "seeking", which is exactly what I was looking for (see original question above). Furthermore, the question has been asked and answered (I just didn't find it before) and I don't want to duplicate questions. However, I'm still having a problem. My app freezes for a while and then eventually crashes when ever I try to copyNextSampleBuffer. I'm not sure what's going on. I don't seem to be in any kind of recursion loop, it just never returns from the function call. Checking the logs show give me this error:
Exception Type: 00000020
Exception Codes: 0x8badf00d
Highlighted Thread: 0
Application Specific Information:
App[10570] has active assertions beyond permitted time:
{(
<SBProcessAssertion: 0xddd9300> identifier: Suspending process: App[10570] permittedBackgroundDuration: 10.000000 reason: suspend owner pid:52 preventSuspend preventThrottleDownCPU preventThrottleDownUI
)}
I use a time profiler on the app and yep, it just sits there with a minimal amount of processing. Can't quite figure out what's going on. It's important to note that this doesn't occur if I don't set the timeRange property of AVAssetReader. I've checked and the values for timeRange are valid, but setting it is causing the problem for some reason. Here's my processing code:
- (void) processSampleData{
if (!_asset || CMTimeGetSeconds(_asset.duration) <= 0) return;
NSError *error = nil;
AVAssetTrack *songTrack = _asset.tracks.firstObject;
if (!songTrack) return;
NSDictionary *outputSettingsDict = [[NSDictionary alloc] initWithObjectsAndKeys:
[NSNumber numberWithInt:kAudioFormatLinearPCM],AVFormatIDKey,
[NSNumber numberWithInt:16], AVLinearPCMBitDepthKey,
[NSNumber numberWithBool:NO],AVLinearPCMIsBigEndianKey,
[NSNumber numberWithBool:NO],AVLinearPCMIsFloatKey,
[NSNumber numberWithBool:NO],AVLinearPCMIsNonInterleaved,
nil];
UInt32 sampleRate = 44100.0;
_channelCount = 1;
NSArray *formatDesc = songTrack.formatDescriptions;
for(unsigned int i = 0; i < [formatDesc count]; ++i) {
CMAudioFormatDescriptionRef item = (__bridge_retained CMAudioFormatDescriptionRef)[formatDesc objectAtIndex:i];
const AudioStreamBasicDescription* fmtDesc = CMAudioFormatDescriptionGetStreamBasicDescription (item);
if(fmtDesc ) {
sampleRate = fmtDesc->mSampleRate;
_channelCount = fmtDesc->mChannelsPerFrame;
}
CFRelease(item);
}
UInt32 bytesPerSample = 2 * _channelCount; //Bytes are hard coded by AVLinearPCMBitDepthKey
_normalizedMax = 0;
_sampledData = [[NSMutableData alloc] init];
SInt16 *channels[_channelCount];
char *sampleRef;
SInt16 *samples;
NSInteger sampleTally = 0;
SInt16 cTotal;
_sampleCount = DefaultSampleSize * [UIScreen mainScreen].scale;
NSTimeInterval intervalBetweenSamples = _asset.duration.value / _sampleCount;
NSTimeInterval sampleSize = fmax(100, intervalBetweenSamples / _sampleCount);
double assetTimeScale = _asset.duration.timescale;
CMTimeRange timeRange = CMTimeRangeMake(CMTimeMake(0, assetTimeScale), CMTimeMake(sampleSize, assetTimeScale));
SInt16 totals[_channelCount];
@autoreleasepool {
for (int i = 0; i < _sampleCount; i++) {
AVAssetReader *reader = [AVAssetReader assetReaderWithAsset:_asset error:&error];
AVAssetReaderTrackOutput *trackOutput = [AVAssetReaderTrackOutput assetReaderTrackOutputWithTrack:songTrack outputSettings:outputSettingsDict];
[reader addOutput:trackOutput];
reader.timeRange = timeRange;
[reader startReading];
while (reader.status == AVAssetReaderStatusReading) {
CMSampleBufferRef sampleBufferRef = [trackOutput copyNextSampleBuffer];
if (sampleBufferRef){
CMBlockBufferRef blockBufferRef = CMSampleBufferGetDataBuffer(sampleBufferRef);
size_t length = CMBlockBufferGetDataLength(blockBufferRef);
int sampleCount = length / bytesPerSample;
for (int i = 0; i < sampleCount ; i += _channelCount) {
CMBlockBufferAccessDataBytes(blockBufferRef, i * bytesPerSample, _channelCount, channels, &sampleRef);
samples = (SInt16 *)sampleRef;
for (int channel = 0; channel < _channelCount; channel++)
totals[channel] += samples[channel];
sampleTally++;
}
CMSampleBufferInvalidate(sampleBufferRef);
CFRelease(sampleBufferRef);
}
}
for (int i = 0; i < _channelCount; i++){
cTotal = abs(totals[i] / sampleTally);
if (cTotal > _normalizedMax) _normalizedMax = cTotal;
[_sampledData appendBytes:&cTotal length:sizeof(cTotal)];
totals[i] = 0;
}
sampleTally = 0;
timeRange.start = CMTimeMake((intervalBetweenSamples * (i + 1)) - sampleSize, assetTimeScale); //Take the sample just before the interval
}
}
_assetNeedsProcessing = NO;
}
