3c351f6968
readAudio now returns an AudioChunk object directly, and all inputs have been changed to accomodate this. Also, input and converter processing have been altered to better work with this. Signed-off-by: Christopher Snowhill <kode54@gmail.com>
914 lines
28 KiB
Objective-C
914 lines
28 KiB
Objective-C
//
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// ConverterNode.m
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// Cog
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//
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// Created by Zaphod Beeblebrox on 8/2/05.
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// Copyright 2005 __MyCompanyName__. All rights reserved.
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//
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#import <Accelerate/Accelerate.h>
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#import <Foundation/Foundation.h>
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#import "ConverterNode.h"
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#import "BufferChain.h"
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#import "OutputNode.h"
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#import "Logging.h"
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#import "hdcd_decode2.h"
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#ifdef _DEBUG
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#import "BadSampleCleaner.h"
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#endif
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#if !DSD_DECIMATE
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#include "dsd2float.h"
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#endif
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void PrintStreamDesc(AudioStreamBasicDescription *inDesc) {
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if(!inDesc) {
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DLog(@"Can't print a NULL desc!\n");
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return;
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}
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DLog(@"- - - - - - - - - - - - - - - - - - - -\n");
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DLog(@" Sample Rate:%f\n", inDesc->mSampleRate);
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DLog(@" Format ID:%s\n", (char *)&inDesc->mFormatID);
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DLog(@" Format Flags:%X\n", inDesc->mFormatFlags);
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DLog(@" Bytes per Packet:%d\n", inDesc->mBytesPerPacket);
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DLog(@" Frames per Packet:%d\n", inDesc->mFramesPerPacket);
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DLog(@" Bytes per Frame:%d\n", inDesc->mBytesPerFrame);
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DLog(@" Channels per Frame:%d\n", inDesc->mChannelsPerFrame);
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DLog(@" Bits per Channel:%d\n", inDesc->mBitsPerChannel);
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DLog(@"- - - - - - - - - - - - - - - - - - - -\n");
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}
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@implementation ConverterNode
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static void *kConverterNodeContext = &kConverterNodeContext;
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@synthesize inputFormat;
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- (id)initWithController:(id)c previous:(id)p {
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self = [super initWithController:c previous:p];
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if(self) {
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rgInfo = nil;
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inputBuffer = NULL;
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inputBufferSize = 0;
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floatBuffer = NULL;
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floatBufferSize = 0;
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stopping = NO;
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convertEntered = NO;
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paused = NO;
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#if DSD_DECIMATE
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dsd2pcm = NULL;
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dsd2pcmCount = 0;
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#endif
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hdcd_decoder = NULL;
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lastChunkIn = nil;
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[[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.volumeScaling" options:0 context:kConverterNodeContext];
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}
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return self;
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}
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void scale_by_volume(float *buffer, size_t count, float volume) {
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if(volume != 1.0) {
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size_t unaligned = (uintptr_t)buffer & 15;
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if(unaligned) {
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size_t count3 = unaligned >> 2;
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while(count3 > 0) {
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*buffer++ *= volume;
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count3--;
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count--;
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}
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}
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vDSP_vsmul(buffer, 1, &volume, buffer, 1, count);
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}
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}
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#if DSD_DECIMATE
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/**
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* DSD 2 PCM: Stage 1:
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* Decimate by factor 8
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* (one byte (8 samples) -> one float sample)
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* The bits are processed from least signicifant to most signicicant.
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* @author Sebastian Gesemann
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*/
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#define dsd2pcm_FILTER_COEFFS_COUNT 64
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static const float dsd2pcm_FILTER_COEFFS[64] = {
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0.09712411121659f, 0.09613438994044f, 0.09417884216316f, 0.09130441727307f,
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0.08757947648990f, 0.08309142055179f, 0.07794369263673f, 0.07225228745463f,
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0.06614191680338f, 0.05974199351302f, 0.05318259916599f, 0.04659059631228f,
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0.04008603356890f, 0.03377897290478f, 0.02776684382775f, 0.02213240062966f,
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0.01694232798846f, 0.01224650881275f, 0.00807793792573f, 0.00445323755944f,
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0.00137370697215f, -0.00117318019994f, -0.00321193033831f, -0.00477694265140f,
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-0.00591028841335f, -0.00665946056286f, -0.00707518873201f, -0.00720940203988f,
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-0.00711340642819f, -0.00683632603227f, -0.00642384017266f, -0.00591723006715f,
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-0.00535273320457f, -0.00476118922548f, -0.00416794965654f, -0.00359301524813f,
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-0.00305135909510f, -0.00255339111833f, -0.00210551956895f, -0.00171076760278f,
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-0.00136940723130f, -0.00107957856005f, -0.00083786862365f, -0.00063983084245f,
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-0.00048043272086f, -0.00035442550015f, -0.00025663481039f, -0.00018217573430f,
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-0.00012659899635f, -0.00008597726991f, -0.00005694188820f, -0.00003668060332f,
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-0.00002290670286f, -0.00001380895679f, -0.00000799057558f, -0.00000440385083f,
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-0.00000228567089f, -0.00000109760778f, -0.00000047286430f, -0.00000017129652f,
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-0.00000004282776f, 0.00000000119422f, 0.00000000949179f, 0.00000000747450f
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};
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struct dsd2pcm_state {
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/*
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* This is the 2nd half of an even order symmetric FIR
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* lowpass filter (to be used on a signal sampled at 44100*64 Hz)
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* Passband is 0-24 kHz (ripples +/- 0.025 dB)
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* Stopband starts at 176.4 kHz (rejection: 170 dB)
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* The overall gain is 2.0
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*/
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/* These remain constant for the duration */
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int FILT_LOOKUP_PARTS;
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float *FILT_LOOKUP_TABLE;
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uint8_t *REVERSE_BITS;
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int FIFO_LENGTH;
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int FIFO_OFS_MASK;
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/* These are altered */
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int *fifo;
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int fpos;
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};
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static void dsd2pcm_free(void *);
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static void dsd2pcm_reset(void *);
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static void *dsd2pcm_alloc() {
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struct dsd2pcm_state *state = (struct dsd2pcm_state *)calloc(1, sizeof(struct dsd2pcm_state));
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float *FILT_LOOKUP_TABLE;
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double *temp;
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uint8_t *REVERSE_BITS;
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if(!state)
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return NULL;
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state->FILT_LOOKUP_PARTS = (dsd2pcm_FILTER_COEFFS_COUNT + 7) / 8;
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const int FILT_LOOKUP_PARTS = state->FILT_LOOKUP_PARTS;
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// The current 128 tap FIR leads to an 8 KB lookup table
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state->FILT_LOOKUP_TABLE = (float *)calloc(sizeof(float), FILT_LOOKUP_PARTS << 8);
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if(!state->FILT_LOOKUP_TABLE)
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goto fail;
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FILT_LOOKUP_TABLE = state->FILT_LOOKUP_TABLE;
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temp = (double *)calloc(sizeof(double), 0x100);
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if(!temp)
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goto fail;
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for(int part = 0, sofs = 0, dofs = 0; part < FILT_LOOKUP_PARTS;) {
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memset(temp, 0, 0x100 * sizeof(double));
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for(int bit = 0, bitmask = 0x80; bit < 8 && sofs + bit < dsd2pcm_FILTER_COEFFS_COUNT;) {
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double coeff = dsd2pcm_FILTER_COEFFS[sofs + bit];
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for(int bite = 0; bite < 0x100; bite++) {
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if((bite & bitmask) == 0) {
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temp[bite] -= coeff;
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} else {
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temp[bite] += coeff;
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}
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}
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bit++;
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bitmask >>= 1;
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}
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for(int s = 0; s < 0x100;) {
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FILT_LOOKUP_TABLE[dofs++] = (float)temp[s++];
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}
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part++;
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sofs += 8;
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}
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free(temp);
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{ // calculate FIFO stuff
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int k = 1;
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while(k < FILT_LOOKUP_PARTS * 2) k <<= 1;
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state->FIFO_LENGTH = k;
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state->FIFO_OFS_MASK = k - 1;
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}
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state->REVERSE_BITS = (uint8_t *)calloc(1, 0x100);
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if(!state->REVERSE_BITS)
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goto fail;
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REVERSE_BITS = state->REVERSE_BITS;
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for(int i = 0, j = 0; i < 0x100; i++) {
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REVERSE_BITS[i] = (uint8_t)j;
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// "reverse-increment" of j
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for(int bitmask = 0x80;;) {
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if(((j ^= bitmask) & bitmask) != 0) break;
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if(bitmask == 1) break;
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bitmask >>= 1;
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}
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}
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state->fifo = (int *)calloc(sizeof(int), state->FIFO_LENGTH);
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if(!state->fifo)
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goto fail;
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dsd2pcm_reset(state);
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return (void *)state;
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fail:
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dsd2pcm_free(state);
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return NULL;
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}
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static void *dsd2pcm_dup(void *_state) {
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struct dsd2pcm_state *state = (struct dsd2pcm_state *)_state;
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if(state) {
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struct dsd2pcm_state *newstate = (struct dsd2pcm_state *)calloc(1, sizeof(struct dsd2pcm_state));
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if(newstate) {
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newstate->FILT_LOOKUP_PARTS = state->FILT_LOOKUP_PARTS;
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newstate->FIFO_LENGTH = state->FIFO_LENGTH;
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newstate->FIFO_OFS_MASK = state->FIFO_OFS_MASK;
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newstate->fpos = state->fpos;
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newstate->FILT_LOOKUP_TABLE = (float *)calloc(sizeof(float), state->FILT_LOOKUP_PARTS << 8);
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if(!newstate->FILT_LOOKUP_TABLE)
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goto fail;
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memcpy(newstate->FILT_LOOKUP_TABLE, state->FILT_LOOKUP_TABLE, sizeof(float) * (state->FILT_LOOKUP_PARTS << 8));
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newstate->REVERSE_BITS = (uint8_t *)calloc(1, 0x100);
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if(!newstate->REVERSE_BITS)
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goto fail;
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memcpy(newstate->REVERSE_BITS, state->REVERSE_BITS, 0x100);
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newstate->fifo = (int *)calloc(sizeof(int), state->FIFO_LENGTH);
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if(!newstate->fifo)
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goto fail;
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memcpy(newstate->fifo, state->fifo, sizeof(int) * state->FIFO_LENGTH);
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return (void *)newstate;
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}
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fail:
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dsd2pcm_free(newstate);
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return NULL;
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}
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return NULL;
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}
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static void dsd2pcm_free(void *_state) {
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struct dsd2pcm_state *state = (struct dsd2pcm_state *)_state;
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if(state) {
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free(state->fifo);
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free(state->REVERSE_BITS);
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free(state->FILT_LOOKUP_TABLE);
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free(state);
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}
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}
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static void dsd2pcm_reset(void *_state) {
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struct dsd2pcm_state *state = (struct dsd2pcm_state *)_state;
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const int FILT_LOOKUP_PARTS = state->FILT_LOOKUP_PARTS;
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int *fifo = state->fifo;
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for(int i = 0; i < FILT_LOOKUP_PARTS; i++) {
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fifo[i] = 0x55;
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fifo[i + FILT_LOOKUP_PARTS] = 0xAA;
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}
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state->fpos = FILT_LOOKUP_PARTS;
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}
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static int dsd2pcm_latency(void *_state) {
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struct dsd2pcm_state *state = (struct dsd2pcm_state *)_state;
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if(state)
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return state->FIFO_LENGTH;
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else
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return 0;
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}
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static void dsd2pcm_process(void *_state, const uint8_t *src, size_t sofs, size_t sinc, float *dest, size_t dofs, size_t dinc, size_t len) {
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struct dsd2pcm_state *state = (struct dsd2pcm_state *)_state;
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int bite1, bite2, temp;
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float sample;
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int *fifo = state->fifo;
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const uint8_t *REVERSE_BITS = state->REVERSE_BITS;
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const float *FILT_LOOKUP_TABLE = state->FILT_LOOKUP_TABLE;
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const int FILT_LOOKUP_PARTS = state->FILT_LOOKUP_PARTS;
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const int FIFO_OFS_MASK = state->FIFO_OFS_MASK;
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int fpos = state->fpos;
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while(len > 0) {
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fifo[fpos] = REVERSE_BITS[fifo[fpos]] & 0xFF;
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fifo[(fpos + FILT_LOOKUP_PARTS) & FIFO_OFS_MASK] = src[sofs] & 0xFF;
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sofs += sinc;
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temp = (fpos + 1) & FIFO_OFS_MASK;
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sample = 0;
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for(int k = 0, lofs = 0; k < FILT_LOOKUP_PARTS;) {
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bite1 = fifo[(fpos - k) & FIFO_OFS_MASK];
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bite2 = fifo[(temp + k) & FIFO_OFS_MASK];
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sample += FILT_LOOKUP_TABLE[lofs + bite1] + FILT_LOOKUP_TABLE[lofs + bite2];
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k++;
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lofs += 0x100;
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}
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fpos = temp;
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dest[dofs] = sample;
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dofs += dinc;
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len--;
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}
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state->fpos = fpos;
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}
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static void convert_dsd_to_f32(float *output, const uint8_t *input, size_t count, size_t channels, void **dsd2pcm) {
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for(size_t channel = 0; channel < channels; ++channel) {
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dsd2pcm_process(dsd2pcm[channel], input, channel, channels, output, channel, channels, count);
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}
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}
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#else
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static void convert_dsd_to_f32(float *output, const uint8_t *input, size_t count, size_t channels) {
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const uint8_t *iptr = input;
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float *optr = output;
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for(size_t index = 0; index < count; ++index) {
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for(size_t channel = 0; channel < channels; ++channel) {
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uint8_t sample = *iptr++;
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cblas_scopy(8, &dsd2float[sample][0], 1, optr++, (int)channels);
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}
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optr += channels * 7;
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}
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}
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#endif
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static void convert_u8_to_s16(int16_t *output, const uint8_t *input, size_t count) {
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for(size_t i = 0; i < count; ++i) {
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uint16_t sample = (input[i] << 8) | input[i];
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sample ^= 0x8080;
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output[i] = (int16_t)(sample);
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}
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}
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static void convert_s8_to_s16(int16_t *output, const uint8_t *input, size_t count) {
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for(size_t i = 0; i < count; ++i) {
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uint16_t sample = (input[i] << 8) | input[i];
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output[i] = (int16_t)(sample);
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}
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}
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static void convert_u16_to_s16(int16_t *buffer, size_t count) {
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for(size_t i = 0; i < count; ++i) {
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buffer[i] ^= 0x8000;
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}
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}
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static void convert_s16_to_hdcd_input(int32_t *output, const int16_t *input, size_t count) {
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for(size_t i = 0; i < count; ++i) {
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output[i] = input[i];
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}
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}
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static void convert_s24_to_s32(int32_t *output, const uint8_t *input, size_t count) {
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for(size_t i = 0; i < count; ++i) {
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int32_t sample = (input[i * 3] << 8) | (input[i * 3 + 1] << 16) | (input[i * 3 + 2] << 24);
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output[i] = sample;
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}
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}
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static void convert_u24_to_s32(int32_t *output, const uint8_t *input, size_t count) {
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for(size_t i = 0; i < count; ++i) {
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int32_t sample = (input[i * 3] << 8) | (input[i * 3 + 1] << 16) | (input[i * 3 + 2] << 24);
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output[i] = sample ^ 0x80000000;
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}
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}
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static void convert_u32_to_s32(int32_t *buffer, size_t count) {
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for(size_t i = 0; i < count; ++i) {
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buffer[i] ^= 0x80000000;
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}
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}
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static void convert_f64_to_f32(float *output, const double *input, size_t count) {
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vDSP_vdpsp(input, 1, output, 1, count);
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}
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static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes) {
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size_t i;
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bitsPerSample = (bitsPerSample + 7) / 8;
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switch(bitsPerSample) {
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case 2:
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for(i = 0; i < bytes; i += 2) {
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*(int16_t *)buffer = __builtin_bswap16(*(int16_t *)buffer);
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buffer += 2;
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}
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break;
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case 3: {
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union {
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vDSP_int24 int24;
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uint32_t int32;
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} intval;
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intval.int32 = 0;
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for(i = 0; i < bytes; i += 3) {
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intval.int24 = *(vDSP_int24 *)buffer;
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intval.int32 = __builtin_bswap32(intval.int32 << 8);
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*(vDSP_int24 *)buffer = intval.int24;
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buffer += 3;
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}
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} break;
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case 4:
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for(i = 0; i < bytes; i += 4) {
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*(uint32_t *)buffer = __builtin_bswap32(*(uint32_t *)buffer);
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buffer += 4;
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}
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break;
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case 8:
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for(i = 0; i < bytes; i += 8) {
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*(uint64_t *)buffer = __builtin_bswap64(*(uint64_t *)buffer);
|
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buffer += 8;
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}
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break;
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}
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}
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|
|
- (void)process {
|
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// Removed endOfStream check from here, since we want to be able to flush the converter
|
|
// when the end of stream is reached. Convert function instead processes what it can,
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// and returns 0 samples when it has nothing more to process at the end of stream.
|
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while([self shouldContinue] == YES) {
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AudioChunk *chunk = nil;
|
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while(paused) {
|
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usleep(500);
|
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}
|
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@autoreleasepool {
|
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chunk = [self convert];
|
|
}
|
|
if(!chunk) {
|
|
continue;
|
|
}
|
|
@autoreleasepool {
|
|
[self writeChunk:chunk];
|
|
chunk = nil;
|
|
}
|
|
if(streamFormatChanged) {
|
|
[self cleanUp];
|
|
[self setupWithInputFormat:newInputFormat withInputConfig:newInputChannelConfig isLossless:rememberedLossless];
|
|
}
|
|
}
|
|
}
|
|
|
|
- (AudioChunk *)convert {
|
|
UInt32 ioNumberPackets;
|
|
int amountReadFromFC;
|
|
int amountRead = 0;
|
|
|
|
if(stopping)
|
|
return 0;
|
|
|
|
convertEntered = YES;
|
|
|
|
tryagain:
|
|
if(stopping || [self shouldContinue] == NO) {
|
|
convertEntered = NO;
|
|
return nil;
|
|
}
|
|
|
|
amountReadFromFC = 0;
|
|
|
|
if(floatOffset == floatSize) // skip this step if there's still float buffered
|
|
while(inpOffset == inpSize) {
|
|
size_t samplesRead = 0;
|
|
|
|
BOOL isFloat = !!(inputFormat.mFormatFlags & kAudioFormatFlagIsFloat);
|
|
BOOL isUnsigned = !isFloat && !(inputFormat.mFormatFlags & kAudioFormatFlagIsSignedInteger);
|
|
size_t bitsPerSample = inputFormat.mBitsPerChannel;
|
|
|
|
// Approximately the most we want on input
|
|
ioNumberPackets = CHUNK_SIZE;
|
|
|
|
#if DSD_DECIMATE
|
|
const size_t sizeScale = 3;
|
|
#else
|
|
const size_t sizeScale = (bitsPerSample == 1) ? 10 : 3;
|
|
#endif
|
|
|
|
size_t newSize = ioNumberPackets * floatFormat.mBytesPerPacket;
|
|
if(!inputBuffer || inputBufferSize < newSize)
|
|
inputBuffer = realloc(inputBuffer, inputBufferSize = newSize * sizeScale);
|
|
|
|
ssize_t amountToWrite = ioNumberPackets * inputFormat.mBytesPerPacket;
|
|
|
|
ssize_t bytesReadFromInput = 0;
|
|
|
|
while(bytesReadFromInput < amountToWrite && !stopping && !paused && !streamFormatChanged && [self shouldContinue] == YES && [self endOfStream] == NO) {
|
|
AudioStreamBasicDescription inf;
|
|
uint32_t config;
|
|
if([self peekFormat:&inf channelConfig:&config]) {
|
|
if(config != inputChannelConfig || memcmp(&inf, &inputFormat, sizeof(inf)) != 0) {
|
|
if(inputChannelConfig == 0 && memcmp(&inf, &inputFormat, sizeof(inf)) == 0) {
|
|
inputChannelConfig = config;
|
|
continue;
|
|
} else {
|
|
newInputFormat = inf;
|
|
newInputChannelConfig = config;
|
|
streamFormatChanged = YES;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
AudioChunk *chunk = [self readChunk:((amountToWrite - bytesReadFromInput) / inputFormat.mBytesPerPacket)];
|
|
inf = [chunk format];
|
|
size_t frameCount = [chunk frameCount];
|
|
config = [chunk channelConfig];
|
|
size_t bytesRead = frameCount * inf.mBytesPerPacket;
|
|
if(frameCount) {
|
|
NSData *samples = [chunk removeSamples:frameCount];
|
|
memcpy(((uint8_t *)inputBuffer) + bytesReadFromInput, [samples bytes], bytesRead);
|
|
lastChunkIn = [[AudioChunk alloc] init];
|
|
[lastChunkIn setFormat:inf];
|
|
[lastChunkIn setChannelConfig:config];
|
|
[lastChunkIn setLossless:[chunk lossless]];
|
|
[lastChunkIn assignSamples:[samples bytes] frameCount:frameCount];
|
|
}
|
|
bytesReadFromInput += bytesRead;
|
|
if(!frameCount) {
|
|
usleep(500);
|
|
}
|
|
}
|
|
|
|
BOOL isBigEndian = !!(inputFormat.mFormatFlags & kAudioFormatFlagIsBigEndian);
|
|
|
|
if(!bytesReadFromInput) {
|
|
convertEntered = NO;
|
|
return nil;
|
|
}
|
|
|
|
if(bytesReadFromInput && isBigEndian) {
|
|
// Time for endian swap!
|
|
convert_be_to_le((uint8_t *)inputBuffer, inputFormat.mBitsPerChannel, bytesReadFromInput);
|
|
}
|
|
|
|
if(bytesReadFromInput && isFloat && inputFormat.mBitsPerChannel == 64) {
|
|
// Time for precision loss from weird inputs
|
|
samplesRead = bytesReadFromInput / sizeof(double);
|
|
convert_f64_to_f32((float *)(((uint8_t *)inputBuffer) + bytesReadFromInput), (const double *)inputBuffer, samplesRead);
|
|
memmove(inputBuffer, ((uint8_t *)inputBuffer) + bytesReadFromInput, samplesRead * sizeof(float));
|
|
bytesReadFromInput = samplesRead * sizeof(float);
|
|
}
|
|
|
|
if(bytesReadFromInput && !isFloat) {
|
|
float gain = 1.0;
|
|
if(bitsPerSample == 1) {
|
|
samplesRead = bytesReadFromInput / inputFormat.mBytesPerPacket;
|
|
size_t buffer_adder = (bytesReadFromInput + 15) & ~15;
|
|
convert_dsd_to_f32((float *)(((uint8_t *)inputBuffer) + buffer_adder), (const uint8_t *)inputBuffer, samplesRead, inputFormat.mChannelsPerFrame
|
|
#if DSD_DECIMATE
|
|
,
|
|
dsd2pcm
|
|
#endif
|
|
);
|
|
#if !DSD_DECIMATE
|
|
samplesRead *= 8;
|
|
#endif
|
|
memmove(inputBuffer, ((const uint8_t *)inputBuffer) + buffer_adder, samplesRead * inputFormat.mChannelsPerFrame * sizeof(float));
|
|
bitsPerSample = 32;
|
|
bytesReadFromInput = samplesRead * inputFormat.mChannelsPerFrame * sizeof(float);
|
|
isFloat = YES;
|
|
} else if(bitsPerSample <= 8) {
|
|
samplesRead = bytesReadFromInput;
|
|
size_t buffer_adder = (bytesReadFromInput + 1) & ~1;
|
|
if(!isUnsigned)
|
|
convert_s8_to_s16((int16_t *)(((uint8_t *)inputBuffer) + buffer_adder), (const uint8_t *)inputBuffer, samplesRead);
|
|
else
|
|
convert_u8_to_s16((int16_t *)(((uint8_t *)inputBuffer) + buffer_adder), (const uint8_t *)inputBuffer, samplesRead);
|
|
memmove(inputBuffer, ((uint8_t *)inputBuffer) + buffer_adder, samplesRead * 2);
|
|
bitsPerSample = 16;
|
|
bytesReadFromInput = samplesRead * 2;
|
|
isUnsigned = NO;
|
|
}
|
|
if(hdcd_decoder) { // implied bits per sample is 16, produces 32 bit int scale
|
|
samplesRead = bytesReadFromInput / 2;
|
|
if(isUnsigned)
|
|
convert_u16_to_s16((int16_t *)inputBuffer, samplesRead);
|
|
size_t buffer_adder = (bytesReadFromInput + 3) & ~3;
|
|
convert_s16_to_hdcd_input((int32_t *)(((uint8_t *)inputBuffer) + buffer_adder), (int16_t *)inputBuffer, samplesRead);
|
|
memmove(inputBuffer, ((uint8_t *)inputBuffer) + buffer_adder, samplesRead * 4);
|
|
hdcd_process_stereo((hdcd_state_stereo_t *)hdcd_decoder, (int32_t *)inputBuffer, (int)(samplesRead / 2));
|
|
if(((hdcd_state_stereo_t *)hdcd_decoder)->channel[0].sustain &&
|
|
((hdcd_state_stereo_t *)hdcd_decoder)->channel[1].sustain) {
|
|
[controller sustainHDCD];
|
|
}
|
|
gain = 2.0;
|
|
bitsPerSample = 32;
|
|
bytesReadFromInput = samplesRead * 4;
|
|
isUnsigned = NO;
|
|
} else if(bitsPerSample <= 16) {
|
|
samplesRead = bytesReadFromInput / 2;
|
|
if(isUnsigned)
|
|
convert_u16_to_s16((int16_t *)inputBuffer, samplesRead);
|
|
size_t buffer_adder = (bytesReadFromInput + 15) & ~15; // vDSP functions expect aligned to four elements
|
|
vDSP_vflt16((const short *)inputBuffer, 1, (float *)(((uint8_t *)inputBuffer) + buffer_adder), 1, samplesRead);
|
|
float scale = 1ULL << 15;
|
|
vDSP_vsdiv((const float *)(((uint8_t *)inputBuffer) + buffer_adder), 1, &scale, (float *)(((uint8_t *)inputBuffer) + buffer_adder), 1, samplesRead);
|
|
memmove(inputBuffer, ((uint8_t *)inputBuffer) + buffer_adder, samplesRead * sizeof(float));
|
|
bitsPerSample = 32;
|
|
bytesReadFromInput = samplesRead * sizeof(float);
|
|
isUnsigned = NO;
|
|
isFloat = YES;
|
|
} else if(bitsPerSample <= 24) {
|
|
samplesRead = bytesReadFromInput / 3;
|
|
size_t buffer_adder = (bytesReadFromInput + 3) & ~3;
|
|
if(isUnsigned)
|
|
convert_u24_to_s32((int32_t *)(((uint8_t *)inputBuffer) + buffer_adder), (uint8_t *)inputBuffer, samplesRead);
|
|
else
|
|
convert_s24_to_s32((int32_t *)(((uint8_t *)inputBuffer) + buffer_adder), (uint8_t *)inputBuffer, samplesRead);
|
|
memmove(inputBuffer, ((uint8_t *)inputBuffer) + buffer_adder, samplesRead * 4);
|
|
bitsPerSample = 32;
|
|
bytesReadFromInput = samplesRead * 4;
|
|
isUnsigned = NO;
|
|
}
|
|
if(!isFloat && bitsPerSample <= 32) {
|
|
samplesRead = bytesReadFromInput / 4;
|
|
if(isUnsigned)
|
|
convert_u32_to_s32((int32_t *)inputBuffer, samplesRead);
|
|
size_t buffer_adder = (bytesReadFromInput + 31) & ~31; // vDSP functions expect aligned to four elements
|
|
vDSP_vflt32((const int *)inputBuffer, 1, (float *)(((uint8_t *)inputBuffer) + buffer_adder), 1, samplesRead);
|
|
float scale = (1ULL << 31) / gain;
|
|
vDSP_vsdiv((const float *)(((uint8_t *)inputBuffer) + buffer_adder), 1, &scale, (float *)(((uint8_t *)inputBuffer) + buffer_adder), 1, samplesRead);
|
|
memmove(inputBuffer, ((uint8_t *)inputBuffer) + buffer_adder, samplesRead * sizeof(float));
|
|
bitsPerSample = 32;
|
|
bytesReadFromInput = samplesRead * sizeof(float);
|
|
isUnsigned = NO;
|
|
isFloat = YES;
|
|
}
|
|
|
|
#ifdef _DEBUG
|
|
[BadSampleCleaner cleanSamples:(float *)inputBuffer
|
|
amount:bytesReadFromInput / sizeof(float)
|
|
location:@"post int to float conversion"];
|
|
#endif
|
|
}
|
|
|
|
// Input now contains bytesReadFromInput worth of floats, in the input sample rate
|
|
inpSize = bytesReadFromInput;
|
|
inpOffset = 0;
|
|
}
|
|
|
|
if(inpOffset != inpSize && floatOffset == floatSize) {
|
|
#if DSD_DECIMATE
|
|
const float scaleModifier = (inputFormat.mBitsPerChannel == 1) ? 0.5f : 1.0f;
|
|
#endif
|
|
|
|
size_t inputSamples = (inpSize - inpOffset) / floatFormat.mBytesPerPacket;
|
|
|
|
ioNumberPackets = (UInt32)inputSamples;
|
|
|
|
ioNumberPackets = (ioNumberPackets + 255) & ~255;
|
|
|
|
size_t newSize = ioNumberPackets * floatFormat.mBytesPerPacket;
|
|
if(newSize < (ioNumberPackets * dmFloatFormat.mBytesPerPacket))
|
|
newSize = ioNumberPackets * dmFloatFormat.mBytesPerPacket;
|
|
if(!floatBuffer || floatBufferSize < newSize)
|
|
floatBuffer = realloc(floatBuffer, floatBufferSize = newSize * 3);
|
|
|
|
if(stopping) {
|
|
convertEntered = NO;
|
|
return 0;
|
|
}
|
|
|
|
size_t inputDone = 0;
|
|
size_t outputDone = 0;
|
|
|
|
memcpy(floatBuffer, (((uint8_t *)inputBuffer) + inpOffset), inputSamples * floatFormat.mBytesPerPacket);
|
|
inputDone = inputSamples;
|
|
outputDone = inputSamples;
|
|
|
|
inpOffset += inputDone * floatFormat.mBytesPerPacket;
|
|
|
|
amountReadFromFC = (int)(outputDone * floatFormat.mBytesPerPacket);
|
|
|
|
scale_by_volume((float *)floatBuffer, amountReadFromFC / sizeof(float), volumeScale
|
|
#if DSD_DECIMATE
|
|
* scaleModifier
|
|
#endif
|
|
);
|
|
|
|
floatSize = amountReadFromFC;
|
|
floatOffset = 0;
|
|
}
|
|
|
|
if(floatOffset == floatSize)
|
|
goto tryagain;
|
|
|
|
ioNumberPackets = (UInt32)(floatSize - floatOffset);
|
|
|
|
ioNumberPackets -= ioNumberPackets % dmFloatFormat.mBytesPerPacket;
|
|
|
|
AudioChunk *chunk = [[AudioChunk alloc] init];
|
|
[chunk setFormat:nodeFormat];
|
|
if(nodeChannelConfig) {
|
|
[chunk setChannelConfig:nodeChannelConfig];
|
|
}
|
|
[chunk assignSamples:floatBuffer frameCount:ioNumberPackets / dmFloatFormat.mBytesPerPacket];
|
|
|
|
floatOffset += ioNumberPackets;
|
|
amountRead += ioNumberPackets;
|
|
|
|
convertEntered = NO;
|
|
return chunk;
|
|
}
|
|
|
|
- (void)observeValueForKeyPath:(NSString *)keyPath
|
|
ofObject:(id)object
|
|
change:(NSDictionary *)change
|
|
context:(void *)context {
|
|
if(context == kConverterNodeContext) {
|
|
DLog(@"SOMETHING CHANGED!");
|
|
if([keyPath isEqualToString:@"values.volumeScaling"]) {
|
|
// User reset the volume scaling option
|
|
[self refreshVolumeScaling];
|
|
}
|
|
} else {
|
|
[super observeValueForKeyPath:keyPath ofObject:object change:change context:context];
|
|
}
|
|
}
|
|
|
|
static float db_to_scale(float db) {
|
|
return pow(10.0, db / 20);
|
|
}
|
|
|
|
- (void)refreshVolumeScaling {
|
|
if(rgInfo == nil) {
|
|
volumeScale = 1.0;
|
|
return;
|
|
}
|
|
|
|
NSString *scaling = [[NSUserDefaults standardUserDefaults] stringForKey:@"volumeScaling"];
|
|
BOOL useAlbum = [scaling hasPrefix:@"albumGain"];
|
|
BOOL useTrack = useAlbum || [scaling hasPrefix:@"trackGain"];
|
|
BOOL useVolume = useAlbum || useTrack || [scaling isEqualToString:@"volumeScale"];
|
|
BOOL usePeak = [scaling hasSuffix:@"WithPeak"];
|
|
float scale = 1.0;
|
|
float peak = 0.0;
|
|
if(useVolume) {
|
|
id pVolumeScale = [rgInfo objectForKey:@"volume"];
|
|
if(pVolumeScale != nil)
|
|
scale = [pVolumeScale floatValue];
|
|
}
|
|
if(useTrack) {
|
|
id trackGain = [rgInfo objectForKey:@"replayGainTrackGain"];
|
|
id trackPeak = [rgInfo objectForKey:@"replayGainTrackPeak"];
|
|
if(trackGain != nil)
|
|
scale = db_to_scale([trackGain floatValue]);
|
|
if(trackPeak != nil)
|
|
peak = [trackPeak floatValue];
|
|
}
|
|
if(useAlbum) {
|
|
id albumGain = [rgInfo objectForKey:@"replayGainAlbumGain"];
|
|
id albumPeak = [rgInfo objectForKey:@"replayGainAlbumPeak"];
|
|
if(albumGain != nil)
|
|
scale = db_to_scale([albumGain floatValue]);
|
|
if(albumPeak != nil)
|
|
peak = [albumPeak floatValue];
|
|
}
|
|
if(usePeak) {
|
|
if(scale * peak > 1.0)
|
|
scale = 1.0 / peak;
|
|
}
|
|
volumeScale = scale;
|
|
}
|
|
|
|
- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inf withInputConfig:(uint32_t)inputConfig isLossless:(BOOL)lossless {
|
|
// Make the converter
|
|
inputFormat = inf;
|
|
|
|
inputChannelConfig = inputConfig;
|
|
|
|
rememberedLossless = lossless;
|
|
|
|
// These are the only sample formats we support translating
|
|
BOOL isFloat = !!(inputFormat.mFormatFlags & kAudioFormatFlagIsFloat);
|
|
if((!isFloat && !(inputFormat.mBitsPerChannel >= 1 && inputFormat.mBitsPerChannel <= 32)) || (isFloat && !(inputFormat.mBitsPerChannel == 32 || inputFormat.mBitsPerChannel == 64)))
|
|
return NO;
|
|
|
|
// These are really placeholders, as we're doing everything internally now
|
|
if(lossless &&
|
|
inputFormat.mBitsPerChannel == 16 &&
|
|
inputFormat.mChannelsPerFrame == 2 &&
|
|
inputFormat.mSampleRate == 44100) {
|
|
// possibly HDCD, run through decoder
|
|
hdcd_decoder = calloc(1, sizeof(hdcd_state_stereo_t));
|
|
hdcd_reset_stereo((hdcd_state_stereo_t *)hdcd_decoder, 44100);
|
|
}
|
|
|
|
floatFormat = inputFormat;
|
|
floatFormat.mFormatFlags = kAudioFormatFlagsNativeFloatPacked;
|
|
floatFormat.mBitsPerChannel = 32;
|
|
floatFormat.mBytesPerFrame = (32 / 8) * floatFormat.mChannelsPerFrame;
|
|
floatFormat.mBytesPerPacket = floatFormat.mBytesPerFrame * floatFormat.mFramesPerPacket;
|
|
|
|
#if DSD_DECIMATE
|
|
if(inputFormat.mBitsPerChannel == 1) {
|
|
// Decimate this for speed
|
|
floatFormat.mSampleRate *= 1.0 / 8.0;
|
|
dsd2pcmCount = floatFormat.mChannelsPerFrame;
|
|
dsd2pcm = (void **)calloc(dsd2pcmCount, sizeof(void *));
|
|
dsd2pcm[0] = dsd2pcm_alloc();
|
|
dsd2pcmLatency = dsd2pcm_latency(dsd2pcm[0]);
|
|
for(size_t i = 1; i < dsd2pcmCount; ++i) {
|
|
dsd2pcm[i] = dsd2pcm_dup(dsd2pcm[0]);
|
|
}
|
|
}
|
|
#endif
|
|
|
|
inpOffset = 0;
|
|
inpSize = 0;
|
|
|
|
floatOffset = 0;
|
|
floatSize = 0;
|
|
|
|
// This is a post resampler, post-down/upmix format
|
|
|
|
dmFloatFormat = floatFormat;
|
|
|
|
nodeFormat = dmFloatFormat;
|
|
nodeChannelConfig = inputChannelConfig;
|
|
|
|
PrintStreamDesc(&inf);
|
|
PrintStreamDesc(&nodeFormat);
|
|
|
|
[self refreshVolumeScaling];
|
|
|
|
// Move this here so process call isn't running the resampler until it's allocated
|
|
stopping = NO;
|
|
convertEntered = NO;
|
|
streamFormatChanged = NO;
|
|
paused = NO;
|
|
|
|
return YES;
|
|
}
|
|
|
|
- (void)dealloc {
|
|
DLog(@"Decoder dealloc");
|
|
|
|
[[NSUserDefaultsController sharedUserDefaultsController] removeObserver:self forKeyPath:@"values.volumeScaling" context:kConverterNodeContext];
|
|
|
|
paused = NO;
|
|
[self cleanUp];
|
|
}
|
|
|
|
- (void)inputFormatDidChange:(AudioStreamBasicDescription)format inputConfig:(uint32_t)inputConfig {
|
|
DLog(@"FORMAT CHANGED");
|
|
paused = YES;
|
|
while(convertEntered) {
|
|
usleep(500);
|
|
}
|
|
[self cleanUp];
|
|
[self setupWithInputFormat:format withInputConfig:inputConfig isLossless:rememberedLossless];
|
|
}
|
|
|
|
- (void)setRGInfo:(NSDictionary *)rgi {
|
|
DLog(@"Setting ReplayGain info");
|
|
rgInfo = rgi;
|
|
[self refreshVolumeScaling];
|
|
}
|
|
|
|
- (void)cleanUp {
|
|
stopping = YES;
|
|
while(convertEntered) {
|
|
usleep(500);
|
|
}
|
|
if(hdcd_decoder) {
|
|
free(hdcd_decoder);
|
|
hdcd_decoder = NULL;
|
|
}
|
|
#if DSD_DECIMATE
|
|
if(dsd2pcm && dsd2pcmCount) {
|
|
for(size_t i = 0; i < dsd2pcmCount; ++i) {
|
|
dsd2pcm_free(dsd2pcm[i]);
|
|
dsd2pcm[i] = NULL;
|
|
}
|
|
free(dsd2pcm);
|
|
dsd2pcm = NULL;
|
|
}
|
|
#endif
|
|
if(floatBuffer) {
|
|
free(floatBuffer);
|
|
floatBuffer = NULL;
|
|
floatBufferSize = 0;
|
|
}
|
|
if(inputBuffer) {
|
|
free(inputBuffer);
|
|
inputBuffer = NULL;
|
|
inputBufferSize = 0;
|
|
}
|
|
floatOffset = 0;
|
|
floatSize = 0;
|
|
}
|
|
|
|
- (double)secondsBuffered {
|
|
return [buffer listDuration];
|
|
}
|
|
|
|
@end
|