50342891af
When resampling the impulse according to the playback rate, it becomes necessary to normalize the resulting impulse by the inverse of the sample ratio, as resampling adds more or less loudness by virtue of interpolating samples. Signed-off-by: Christopher Snowhill <kode54@gmail.com>
543 lines
17 KiB
Text
543 lines
17 KiB
Text
//
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// HeadphoneFilter.m
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// CogAudio Framework
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//
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// Created by Christopher Snowhill on 1/24/22.
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//
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#import "HeadphoneFilter.h"
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#import "AudioChunk.h"
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#import "AudioDecoder.h"
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#import "AudioSource.h"
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#import <stdlib.h>
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#import "r8bstate.h"
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#import "lpc.h"
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#import "util.h"
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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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if(posix_memalign(&ret, align, size) != 0) {
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return NULL;
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}
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return ret;
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}
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@implementation HeadphoneFilter
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enum {
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speaker_is_back_center = -1,
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speaker_not_present = -2,
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};
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static const uint32_t max_speaker_index = 10;
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static const int8_t speakers_to_hesuvi_7[11][2] = {
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// front left
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{ 0, 1 },
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// front right
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{ 1, 0 },
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// front center
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{ 6, 6 },
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// lfe
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{ 6, 6 },
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// back left
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{ 4, 5 },
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// back right
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{ 5, 4 },
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// front center left
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{ speaker_not_present, speaker_not_present },
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// front center right
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{ speaker_not_present, speaker_not_present },
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// back center
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{ speaker_is_back_center, speaker_is_back_center },
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// side left
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{ 2, 3 },
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// side right
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{ 3, 2 }
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};
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static const int8_t speakers_to_hesuvi_14[11][2] = {
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// front left
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{ 0, 1 },
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// front right
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{ 8, 7 },
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// front center
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{ 6, 13 },
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// lfe
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{ 6, 13 },
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// back left
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{ 4, 5 },
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// back right
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{ 12, 11 },
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// front center left
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{ speaker_not_present, speaker_not_present },
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// front center right
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{ speaker_not_present, speaker_not_present },
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// back center
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{ speaker_is_back_center, speaker_is_back_center },
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// side left
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{ 2, 3 },
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// side right
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{ 10, 9 }
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};
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+ (BOOL)validateImpulseFile:(NSURL *)url {
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id<CogSource> source = [AudioSource audioSourceForURL:url];
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if(!source)
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return NO;
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if(![source open:url])
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return NO;
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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 NO;
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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 NO;
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}
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NSDictionary *properties = [decoder properties];
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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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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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return NO;
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return YES;
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}
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- (id)initWithImpulseFile:(NSURL *)url forSampleRate:(double)sampleRate withInputChannels:(size_t)channels withConfig:(uint32_t)config {
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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 *)malloc(sampleCount * sizeof(float) * impulseChannels);
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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 *)realloc(impulseBuffer, (sampleCount + resamplerLatencyIn * 2 + 1024) * sizeof(float) * impulseChannels);
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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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impulseBuffer = tempImpulse;
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resampledCount += resamplerLatencyOut * 2 + 1024;
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float *resampledImpulse = (float *)malloc(resampledCount * sizeof(float) * impulseChannels);
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if(!resampledImpulse) {
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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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memmove(impulseBuffer + resamplerLatencyIn * impulseChannels, impulseBuffer, sampleCount * sizeof(float) * impulseChannels);
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lpc_extrapolate_bkwd(impulseBuffer + 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(impulseBuffer + 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(impulseBuffer, sampleCount + N_samples_to_add_ * 2, &inputDone, resampledImpulse, resampledCount);
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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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free(impulseBuffer);
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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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fftSize = sampleCount + bufferSize;
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int pow = 1;
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while(fftSize > 2) {
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pow++;
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fftSize /= 2;
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}
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fftSize = 2 << pow;
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float *deinterleavedImpulseBuffer = (float *)_memalign_malloc(fftSize * sizeof(float) * (impulseChannels + 1), 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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for(size_t i = 0; i < impulseChannels; ++i) {
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cblas_scopy(sampleCount, impulseBuffer + i, impulseChannels, deinterleavedImpulseBuffer + i * fftSize, 1);
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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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// Null impulse
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vDSP_vclr(deinterleavedImpulseBuffer + impulseChannels * fftSize, 1, fftSize);
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paddedBufferSize = fftSize;
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fftSizeOver2 = (fftSize + 1) / 2;
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log2n = log2f(fftSize);
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log2nhalf = log2n / 2;
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fftSetup = vDSP_create_fftsetup(log2n, FFT_RADIX2);
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if(!fftSetup) {
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free(deinterleavedImpulseBuffer);
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return nil;
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}
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paddedSignal = (float *)_memalign_malloc(sizeof(float) * paddedBufferSize, 16);
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if(!paddedSignal) {
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free(deinterleavedImpulseBuffer);
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return nil;
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}
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signal_fft.realp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2, 16);
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signal_fft.imagp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2, 16);
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if(!signal_fft.realp || !signal_fft.imagp) {
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free(deinterleavedImpulseBuffer);
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return nil;
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}
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input_filtered_signal_per_channel[0].realp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2, 16);
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input_filtered_signal_per_channel[0].imagp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2, 16);
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if(!input_filtered_signal_per_channel[0].realp ||
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!input_filtered_signal_per_channel[0].imagp) {
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free(deinterleavedImpulseBuffer);
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return nil;
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}
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input_filtered_signal_per_channel[1].realp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2, 16);
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input_filtered_signal_per_channel[1].imagp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2, 16);
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if(!input_filtered_signal_per_channel[1].realp ||
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!input_filtered_signal_per_channel[1].imagp) {
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free(deinterleavedImpulseBuffer);
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return nil;
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}
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impulse_responses = (COMPLEX_SPLIT *)calloc(sizeof(COMPLEX_SPLIT), channels * 2);
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if(!impulse_responses) {
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free(deinterleavedImpulseBuffer);
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return nil;
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}
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for(size_t i = 0; i < channels; ++i) {
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impulse_responses[i * 2 + 0].realp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2, 16);
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impulse_responses[i * 2 + 0].imagp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2, 16);
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impulse_responses[i * 2 + 1].realp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2, 16);
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impulse_responses[i * 2 + 1].imagp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2, 16);
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if(!impulse_responses[i * 2 + 0].realp || !impulse_responses[i * 2 + 0].imagp ||
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!impulse_responses[i * 2 + 1].realp || !impulse_responses[i * 2 + 1].imagp) {
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free(deinterleavedImpulseBuffer);
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return nil;
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}
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uint32_t channelFlag = [AudioChunk extractChannelFlag:(uint32_t)i fromConfig:config];
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uint32_t channelIndex = [AudioChunk findChannelIndex:channelFlag];
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int leftInChannel = speaker_not_present;
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int rightInChannel = speaker_not_present;
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if(impulseChannels == 7) {
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if(channelIndex <= max_speaker_index) {
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leftInChannel = speakers_to_hesuvi_7[channelIndex][0];
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rightInChannel = speakers_to_hesuvi_7[channelIndex][1];
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}
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} else {
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if(channelIndex <= max_speaker_index) {
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leftInChannel = speakers_to_hesuvi_14[channelIndex][0];
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rightInChannel = speakers_to_hesuvi_14[channelIndex][1];
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}
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}
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if(leftInChannel == speaker_is_back_center || rightInChannel == speaker_is_back_center) {
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float *temp;
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if(impulseChannels == 7) {
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temp = (float *)malloc(sizeof(float) * fftSize);
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if(!temp) {
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free(deinterleavedImpulseBuffer);
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return nil;
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}
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cblas_scopy((int)fftSize, deinterleavedImpulseBuffer + 4 * fftSize, 1, temp, 1);
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vDSP_vadd(temp, 1, deinterleavedImpulseBuffer + 5 * fftSize, 1, temp, 1, fftSize);
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vDSP_ctoz((DSPComplex *)temp, 2, &impulse_responses[i * 2 + 0], 1, fftSizeOver2);
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vDSP_ctoz((DSPComplex *)temp, 2, &impulse_responses[i * 2 + 1], 1, fftSizeOver2);
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} else {
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temp = (float *)malloc(sizeof(float) * fftSize * 2);
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if(!temp) {
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free(deinterleavedImpulseBuffer);
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return nil;
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}
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cblas_scopy((int)fftSize, deinterleavedImpulseBuffer + 4 * fftSize, 1, temp, 1);
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vDSP_vadd(temp, 1, deinterleavedImpulseBuffer + 12 * fftSize, 1, temp, 1, fftSize);
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cblas_scopy((int)fftSize, deinterleavedImpulseBuffer + 5 * fftSize, 1, temp + fftSize, 1);
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vDSP_vadd(temp + fftSize, 1, deinterleavedImpulseBuffer + 11 * fftSize, 1, temp + fftSize, 1, fftSize);
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vDSP_ctoz((DSPComplex *)temp, 2, &impulse_responses[i * 2 + 0], 1, fftSizeOver2);
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vDSP_ctoz((DSPComplex *)(temp + fftSize), 2, &impulse_responses[i * 2 + 1], 1, fftSizeOver2);
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}
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free(temp);
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} else if(leftInChannel == speaker_not_present || rightInChannel == speaker_not_present) {
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vDSP_ctoz((DSPComplex *)(deinterleavedImpulseBuffer + impulseChannels * fftSize), 2, &impulse_responses[i * 2 + 0], 1, fftSizeOver2);
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vDSP_ctoz((DSPComplex *)(deinterleavedImpulseBuffer + impulseChannels * fftSize), 2, &impulse_responses[i * 2 + 1], 1, fftSizeOver2);
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} else {
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vDSP_ctoz((DSPComplex *)(deinterleavedImpulseBuffer + leftInChannel * fftSize), 2, &impulse_responses[i * 2 + 0], 1, fftSizeOver2);
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vDSP_ctoz((DSPComplex *)(deinterleavedImpulseBuffer + rightInChannel * fftSize), 2, &impulse_responses[i * 2 + 1], 1, fftSizeOver2);
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}
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vDSP_fft_zrip(fftSetup, &impulse_responses[i * 2 + 0], 1, log2n, FFT_FORWARD);
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vDSP_fft_zrip(fftSetup, &impulse_responses[i * 2 + 1], 1, log2n, FFT_FORWARD);
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}
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free(deinterleavedImpulseBuffer);
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left_result = (float *)_memalign_malloc(sizeof(float) * fftSize, 16);
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right_result = (float *)_memalign_malloc(sizeof(float) * fftSize, 16);
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if(!left_result || !right_result)
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return nil;
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prevOverlapLeft = (float *)_memalign_malloc(sizeof(float) * fftSize, 16);
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prevOverlapRight = (float *)_memalign_malloc(sizeof(float) * fftSize, 16);
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if(!prevOverlapLeft || !prevOverlapRight)
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return nil;
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left_mix_result = (float *)_memalign_malloc(sizeof(float) * fftSize, 16);
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right_mix_result = (float *)_memalign_malloc(sizeof(float) * fftSize, 16);
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if(!left_mix_result || !right_mix_result)
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return nil;
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prevOverlapLength = 0;
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}
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return self;
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}
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- (void)dealloc {
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if(fftSetup) vDSP_destroy_fftsetup(fftSetup);
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free(paddedSignal);
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free(signal_fft.realp);
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free(signal_fft.imagp);
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free(input_filtered_signal_per_channel[0].realp);
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free(input_filtered_signal_per_channel[0].imagp);
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free(input_filtered_signal_per_channel[1].realp);
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free(input_filtered_signal_per_channel[1].imagp);
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if(impulse_responses) {
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for(size_t i = 0; i < channelCount * 2; ++i) {
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free(impulse_responses[i].realp);
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free(impulse_responses[i].imagp);
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}
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free(impulse_responses);
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}
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free(left_result);
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free(right_result);
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free(prevOverlapLeft);
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free(prevOverlapRight);
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free(left_mix_result);
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free(right_mix_result);
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}
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- (void)process:(const float *)inBuffer sampleCount:(size_t)count toBuffer:(float *)outBuffer {
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const float scale = 1.0 / (4.0 * (float)fftSize);
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while(count > 0) {
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size_t countToDo = (count > bufferSize) ? bufferSize : count;
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vDSP_vclr(left_mix_result, 1, fftSize);
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vDSP_vclr(right_mix_result, 1, fftSize);
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for(size_t i = 0; i < channelCount; ++i) {
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cblas_scopy((int)countToDo, inBuffer + i, (int)channelCount, paddedSignal, 1);
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vDSP_vclr(paddedSignal + countToDo, 1, paddedBufferSize - countToDo);
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vDSP_ctoz((DSPComplex *)paddedSignal, 2, &signal_fft, 1, fftSizeOver2);
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vDSP_fft_zrip(fftSetup, &signal_fft, 1, log2n, FFT_FORWARD);
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// One channel forward, then multiply and back twice
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float preserveIRNyq = impulse_responses[i * 2 + 0].imagp[0];
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float preserveSigNyq = signal_fft.imagp[0];
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impulse_responses[i * 2 + 0].imagp[0] = 0;
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signal_fft.imagp[0] = 0;
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vDSP_zvmul(&signal_fft, 1, &impulse_responses[i * 2 + 0], 1, &input_filtered_signal_per_channel[0], 1, fftSizeOver2, 1);
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input_filtered_signal_per_channel[0].imagp[0] = preserveIRNyq * preserveSigNyq;
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impulse_responses[i * 2 + 0].imagp[0] = preserveIRNyq;
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preserveIRNyq = impulse_responses[i * 2 + 1].imagp[0];
|
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impulse_responses[i * 2 + 1].imagp[0] = 0;
|
|
|
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vDSP_zvmul(&signal_fft, 1, &impulse_responses[i * 2 + 1], 1, &input_filtered_signal_per_channel[1], 1, fftSizeOver2, 1);
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|
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input_filtered_signal_per_channel[1].imagp[0] = preserveIRNyq * preserveSigNyq;
|
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impulse_responses[i * 2 + 1].imagp[0] = preserveIRNyq;
|
|
|
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vDSP_fft_zrip(fftSetup, &input_filtered_signal_per_channel[0], 1, log2n, FFT_INVERSE);
|
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vDSP_fft_zrip(fftSetup, &input_filtered_signal_per_channel[1], 1, log2n, FFT_INVERSE);
|
|
|
|
vDSP_ztoc(&input_filtered_signal_per_channel[0], 1, (DSPComplex *)left_result, 2, fftSizeOver2);
|
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vDSP_ztoc(&input_filtered_signal_per_channel[1], 1, (DSPComplex *)right_result, 2, fftSizeOver2);
|
|
|
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vDSP_vadd(left_mix_result, 1, left_result, 1, left_mix_result, 1, fftSize);
|
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vDSP_vadd(right_mix_result, 1, right_result, 1, right_mix_result, 1, fftSize);
|
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}
|
|
|
|
// Integrate previous overlap
|
|
if(prevOverlapLength) {
|
|
vDSP_vadd(prevOverlapLeft, 1, left_mix_result, 1, left_mix_result, 1, prevOverlapLength);
|
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vDSP_vadd(prevOverlapRight, 1, right_mix_result, 1, right_mix_result, 1, prevOverlapLength);
|
|
}
|
|
|
|
prevOverlapLength = (int)(fftSize - countToDo);
|
|
|
|
cblas_scopy(prevOverlapLength, left_mix_result + countToDo, 1, prevOverlapLeft, 1);
|
|
cblas_scopy(prevOverlapLength, right_mix_result + countToDo, 1, prevOverlapRight, 1);
|
|
|
|
vDSP_vsmul(left_mix_result, 1, &scale, left_mix_result, 1, countToDo);
|
|
vDSP_vsmul(right_mix_result, 1, &scale, right_mix_result, 1, countToDo);
|
|
|
|
cblas_scopy((int)countToDo, left_mix_result, 1, outBuffer + 0, 2);
|
|
cblas_scopy((int)countToDo, right_mix_result, 1, outBuffer + 1, 2);
|
|
|
|
inBuffer += countToDo * channelCount;
|
|
outBuffer += countToDo * 2;
|
|
|
|
count -= countToDo;
|
|
}
|
|
}
|
|
|
|
- (void)reset {
|
|
prevOverlapLength = 0;
|
|
}
|
|
|
|
@end
|