[HRIR Filter] Employ impulse cache
An impulse cache reduces any glitching from format channel count changes to near insignificant levels, resulting in a more pleasant experience when there are different mixed formats playing, or even a file which changes format mid-playback. Signed-off-by: Christopher Snowhill <kode54@gmail.com>swiftingly
Christopher Snowhill
parent
120a93465e
commit
6825d15f68
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@ -17,6 +17,32 @@
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#import "lpc.h"
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#import "util.h"
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@interface impulseCacheObject : NSObject {
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}
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@property NSURL *URL;
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@property int sampleCount;
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@property int channelCount;
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@property double sampleRate;
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@property double targetSampleRate;
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@property NSData *data;
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@end
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@implementation impulseCacheObject
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@synthesize URL;
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@synthesize sampleCount;
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@synthesize channelCount;
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@synthesize sampleRate;
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@synthesize targetSampleRate;
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@synthesize data;
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@end
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@interface impulseCache : NSObject {
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}
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@property NSMutableArray<impulseCacheObject *> *cacheObjects;
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+ (impulseCache *)sharedController;
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- (const float *)getImpulse:(NSURL *)url sampleCount:(int *)sampleCount channelCount:(int *)channelCount sampleRate:(double)sampleRate;
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@end
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// Apparently _mm_malloc is Intel-only on newer macOS targets, so use supported posix_memalign
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static void *_memalign_malloc(size_t size, size_t align) {
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void *ret = NULL;
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@ -26,6 +52,227 @@ static void *_memalign_malloc(size_t size, size_t align) {
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return ret;
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}
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@implementation impulseCache
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static impulseCache *_sharedController = nil;
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+ (impulseCache *)sharedController {
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@synchronized(self) {
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if(!_sharedController) {
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_sharedController = [[impulseCache alloc] init];
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}
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}
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return _sharedController;
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}
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- (id)init {
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self = [super init];
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if(self) {
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self.cacheObjects = [[NSMutableArray alloc] init];
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}
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return self;
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}
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- (impulseCacheObject *)addImpulse:(NSURL *)url sampleCount:(int)sampleCount channelCount:(int)channelCount originalSampleRate:(double)originalSampleRate targetSampleRate:(double)targetSampleRate impulseBuffer:(const float *)impulseBuffer {
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impulseCacheObject *obj = [[impulseCacheObject alloc] init];
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obj.URL = url;
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obj.sampleCount = sampleCount;
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obj.channelCount = channelCount;
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obj.sampleRate = originalSampleRate;
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obj.targetSampleRate = targetSampleRate;
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obj.data = [NSData dataWithBytes:impulseBuffer length:(sampleCount * channelCount * sizeof(float))];
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@synchronized(self.cacheObjects) {
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[self.cacheObjects addObject:obj];
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}
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return obj;
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}
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- (const float *)getImpulse:(NSURL *)url sampleCount:(int *)retSampleCount channelCount:(int *)retImpulseChannels sampleRate:(double)sampleRate {
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BOOL impulseFound = NO;
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const float *impulseData = NULL;
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double sampleRateOfSource = 0;
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int sampleCount = 0;
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int impulseChannels = 0;
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impulseCacheObject *cacheObject = nil;
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@synchronized(self.cacheObjects) {
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for(impulseCacheObject *obj in self.cacheObjects) {
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if([obj.URL isEqualTo:url] &&
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obj.targetSampleRate == sampleRate) {
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*retSampleCount = obj.sampleCount;
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*retImpulseChannels = obj.channelCount;
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return (const float *)[obj.data bytes];
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}
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}
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for(impulseCacheObject *obj in self.cacheObjects) {
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if([obj.URL isEqualTo:url] &&
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obj.sampleRate == obj.targetSampleRate) {
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impulseData = (const float *)[obj.data bytes];
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sampleCount = obj.sampleCount;
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impulseChannels = obj.channelCount;
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sampleRateOfSource = obj.sampleRate;
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impulseFound = YES;
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break;
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}
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}
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}
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if(!impulseFound) {
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id<CogSource> source = [AudioSource audioSourceForURL:url];
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if(!source)
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return NULL;
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if(![source open:url])
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return NULL;
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id<CogDecoder> decoder = [AudioDecoder audioDecoderForSource:source];
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if(decoder == nil) {
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[source close];
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source = nil;
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return NULL;
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}
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if(![decoder open:source]) {
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decoder = nil;
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[source close];
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source = nil;
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return NULL;
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}
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NSDictionary *properties = [decoder properties];
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sampleRateOfSource = [[properties objectForKey:@"sampleRate"] floatValue];
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sampleCount = [[properties objectForKey:@"totalFrames"] intValue];
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impulseChannels = [[properties objectForKey:@"channels"] intValue];
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if([[properties objectForKey:@"floatingPoint"] boolValue] != YES ||
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[[properties objectForKey:@"bitsPerSample"] intValue] != 32 ||
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!([[properties objectForKey:@"endian"] isEqualToString:@"host"] ||
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[[properties objectForKey:@"endian"] isEqualToString:@"little"]) ||
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(impulseChannels != 14 && impulseChannels != 7)) {
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[decoder close];
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decoder = nil;
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[source close];
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source = nil;
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return NULL;
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}
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float *impulseBuffer = (float *)_memalign_malloc(sampleCount * sizeof(float) * impulseChannels, 16);
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if(!impulseBuffer) {
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[decoder close];
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decoder = nil;
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[source close];
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source = nil;
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return NULL;
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}
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if([decoder readAudio:impulseBuffer frames:sampleCount] != sampleCount) {
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free(impulseBuffer);
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[decoder close];
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decoder = nil;
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[source close];
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source = nil;
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return NULL;
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}
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[decoder close];
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decoder = nil;
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[source close];
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source = nil;
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cacheObject = [self addImpulse:url sampleCount:sampleCount channelCount:impulseChannels originalSampleRate:sampleRateOfSource targetSampleRate:sampleRateOfSource impulseBuffer:impulseBuffer];
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free(impulseBuffer);
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impulseData = (const float *)[cacheObject.data bytes];
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}
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if(sampleRateOfSource != sampleRate) {
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double sampleRatio = sampleRate / sampleRateOfSource;
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int resampledCount = (int)ceil((double)sampleCount * sampleRatio);
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r8bstate *_r8bstate = new r8bstate(impulseChannels, 1024, sampleRateOfSource, sampleRate);
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unsigned long PRIME_LEN_ = MAX(sampleRateOfSource / 20, 1024u);
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PRIME_LEN_ = MIN(PRIME_LEN_, 16384u);
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PRIME_LEN_ = MAX(PRIME_LEN_, 2 * LPC_ORDER + 1);
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unsigned int N_samples_to_add_ = sampleRateOfSource;
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unsigned int N_samples_to_drop_ = sampleRate;
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samples_len(&N_samples_to_add_, &N_samples_to_drop_, 20, 8192u);
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int resamplerLatencyIn = (int)N_samples_to_add_;
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int resamplerLatencyOut = (int)N_samples_to_drop_;
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float *tempImpulse = (float *)_memalign_malloc((sampleCount + resamplerLatencyIn * 2 + 1024) * sizeof(float) * impulseChannels, 16);
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if(!tempImpulse) {
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return nil;
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}
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resampledCount += resamplerLatencyOut * 2 + 1024;
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float *resampledImpulse = (float *)_memalign_malloc(resampledCount * sizeof(float) * impulseChannels, 16);
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if(!resampledImpulse) {
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free(tempImpulse);
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return nil;
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}
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size_t prime = MIN(sampleCount, PRIME_LEN_);
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void *extrapolate_buffer = NULL;
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size_t extrapolate_buffer_size = 0;
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memcpy(tempImpulse + resamplerLatencyIn * impulseChannels, impulseData, sampleCount * sizeof(float) * impulseChannels);
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lpc_extrapolate_bkwd(tempImpulse + N_samples_to_add_ * impulseChannels, sampleCount, prime, impulseChannels, LPC_ORDER, N_samples_to_add_, &extrapolate_buffer, &extrapolate_buffer_size);
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lpc_extrapolate_fwd(tempImpulse + N_samples_to_add_ * impulseChannels, sampleCount, prime, impulseChannels, LPC_ORDER, N_samples_to_add_, &extrapolate_buffer, &extrapolate_buffer_size);
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free(extrapolate_buffer);
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size_t inputDone = 0;
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size_t outputDone = 0;
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outputDone = _r8bstate->resample(tempImpulse, sampleCount + N_samples_to_add_ * 2, &inputDone, resampledImpulse, resampledCount);
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free(tempImpulse);
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if(outputDone < resampledCount) {
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outputDone += _r8bstate->flush(resampledImpulse + outputDone * impulseChannels, resampledCount - outputDone);
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}
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delete _r8bstate;
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outputDone -= N_samples_to_drop_ * 2;
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// Do this instead of the memmove
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float *resampledImpulseData = resampledImpulse + N_samples_to_drop_ * impulseChannels;
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/*memmove(resampledImpulse, resampledImpulse + N_samples_to_drop_ * impulseChannels, outputDone * sizeof(float) * impulseChannels);*/
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sampleCount = (int)outputDone;
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// Normalize resampled impulse by sample ratio
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float fSampleRatio = (float)sampleRatio;
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vDSP_vsdiv(resampledImpulseData, 1, &fSampleRatio, resampledImpulseData, 1, sampleCount * impulseChannels);
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cacheObject = [self addImpulse:url sampleCount:sampleCount channelCount:impulseChannels originalSampleRate:sampleRateOfSource targetSampleRate:sampleRate impulseBuffer:resampledImpulseData];
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free(resampledImpulse);
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impulseData = (const float *)[cacheObject.data bytes];
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}
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*retSampleCount = sampleCount;
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*retImpulseChannels = impulseChannels;
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return impulseData;
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}
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@end
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@implementation HeadphoneFilter
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enum {
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@ -131,138 +378,13 @@ static const int8_t speakers_to_hesuvi_14[11][2] = {
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self = [super init];
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if(self) {
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id<CogSource> source = [AudioSource audioSourceForURL:url];
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if(!source)
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return nil;
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if(![source open:url])
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return nil;
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id<CogDecoder> decoder = [AudioDecoder audioDecoderForSource:source];
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if(decoder == nil) {
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[source close];
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source = nil;
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return nil;
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}
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if(![decoder open:source]) {
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decoder = nil;
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[source close];
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source = nil;
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return nil;
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}
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NSDictionary *properties = [decoder properties];
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double sampleRateOfSource = [[properties objectForKey:@"sampleRate"] floatValue];
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int sampleCount = [[properties objectForKey:@"totalFrames"] intValue];
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int impulseChannels = [[properties objectForKey:@"channels"] intValue];
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if([[properties objectForKey:@"floatingPoint"] boolValue] != YES ||
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[[properties objectForKey:@"bitsPerSample"] intValue] != 32 ||
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!([[properties objectForKey:@"endian"] isEqualToString:@"host"] ||
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[[properties objectForKey:@"endian"] isEqualToString:@"little"]) ||
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(impulseChannels != 14 && impulseChannels != 7)) {
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[decoder close];
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decoder = nil;
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[source close];
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source = nil;
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return nil;
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}
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float *impulseBuffer = (float *)_memalign_malloc(sampleCount * sizeof(float) * impulseChannels, 16);
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int sampleCount = 0;
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int impulseChannels = 0;
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const float *impulseBuffer = [[impulseCache sharedController] getImpulse:url sampleCount:&sampleCount channelCount:&impulseChannels sampleRate:sampleRate];
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if(!impulseBuffer) {
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[decoder close];
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decoder = nil;
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[source close];
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source = nil;
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return nil;
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}
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if([decoder readAudio:impulseBuffer frames:sampleCount] != sampleCount) {
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[decoder close];
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decoder = nil;
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[source close];
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source = nil;
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return nil;
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}
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[decoder close];
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decoder = nil;
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[source close];
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source = nil;
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if(sampleRateOfSource != sampleRate) {
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double sampleRatio = sampleRate / sampleRateOfSource;
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int resampledCount = (int)ceil((double)sampleCount * sampleRatio);
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r8bstate *_r8bstate = new r8bstate(impulseChannels, 1024, sampleRateOfSource, sampleRate);
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unsigned long PRIME_LEN_ = MAX(sampleRateOfSource / 20, 1024u);
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PRIME_LEN_ = MIN(PRIME_LEN_, 16384u);
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PRIME_LEN_ = MAX(PRIME_LEN_, 2 * LPC_ORDER + 1);
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unsigned int N_samples_to_add_ = sampleRateOfSource;
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unsigned int N_samples_to_drop_ = sampleRate;
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samples_len(&N_samples_to_add_, &N_samples_to_drop_, 20, 8192u);
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int resamplerLatencyIn = (int)N_samples_to_add_;
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int resamplerLatencyOut = (int)N_samples_to_drop_;
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float *tempImpulse = (float *)_memalign_malloc((sampleCount + resamplerLatencyIn * 2 + 1024) * sizeof(float) * impulseChannels, 16);
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if(!tempImpulse) {
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free(impulseBuffer);
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return nil;
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}
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resampledCount += resamplerLatencyOut * 2 + 1024;
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float *resampledImpulse = (float *)_memalign_malloc(resampledCount * sizeof(float) * impulseChannels, 16);
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if(!resampledImpulse) {
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free(tempImpulse);
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free(impulseBuffer);
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return nil;
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}
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size_t prime = MIN(sampleCount, PRIME_LEN_);
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void *extrapolate_buffer = NULL;
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size_t extrapolate_buffer_size = 0;
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memcpy(tempImpulse + resamplerLatencyIn * impulseChannels, impulseBuffer, sampleCount * sizeof(float) * impulseChannels);
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free(impulseBuffer);
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lpc_extrapolate_bkwd(tempImpulse + N_samples_to_add_ * impulseChannels, sampleCount, prime, impulseChannels, LPC_ORDER, N_samples_to_add_, &extrapolate_buffer, &extrapolate_buffer_size);
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lpc_extrapolate_fwd(tempImpulse + N_samples_to_add_ * impulseChannels, sampleCount, prime, impulseChannels, LPC_ORDER, N_samples_to_add_, &extrapolate_buffer, &extrapolate_buffer_size);
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free(extrapolate_buffer);
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size_t inputDone = 0;
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size_t outputDone = 0;
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outputDone = _r8bstate->resample(tempImpulse, sampleCount + N_samples_to_add_ * 2, &inputDone, resampledImpulse, resampledCount);
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free(tempImpulse);
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if (outputDone < resampledCount) {
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outputDone += _r8bstate->flush(resampledImpulse + outputDone * impulseChannels, resampledCount - outputDone);
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}
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delete _r8bstate;
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outputDone -= N_samples_to_drop_ * 2;
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memmove(resampledImpulse, resampledImpulse + N_samples_to_drop_ * impulseChannels, outputDone * sizeof(float) * impulseChannels);
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impulseBuffer = resampledImpulse;
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sampleCount = (int)outputDone;
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// Normalize resampled impulse by sample ratio
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float fSampleRatio = (float)sampleRatio;
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vDSP_vsdiv(impulseBuffer, 1, &fSampleRatio, impulseBuffer, 1, sampleCount * impulseChannels);
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}
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channelCount = channels;
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bufferSize = 512;
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@ -277,7 +399,6 @@ static const int8_t speakers_to_hesuvi_14[11][2] = {
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float *deinterleavedImpulseBuffer = (float *)_memalign_malloc(fftSize * sizeof(float) * impulseChannels, 16);
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if(!deinterleavedImpulseBuffer) {
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free(impulseBuffer);
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return nil;
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}
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@ -286,8 +407,6 @@ static const int8_t speakers_to_hesuvi_14[11][2] = {
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vDSP_vclr(deinterleavedImpulseBuffer + i * fftSize + sampleCount, 1, fftSize - sampleCount);
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}
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free(impulseBuffer);
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paddedBufferSize = fftSize;
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fftSizeOver2 = (fftSize + 1) / 2;
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const size_t fftSizeOver2Plus1 = fftSizeOver2 + 1; // DFT float overwrites plus one, double doesn't
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