Cog Audio: Implement support for channel config
This implements the basic output and mixing support for channel config bits, optionally set by the input plugin. Signed-off-by: Christopher Snowhill <kode54@gmail.com>CQTexperiment
Christopher Snowhill
parent
b0b1446aa7
commit
1ef8df675f
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@ -83,7 +83,7 @@
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bufferChain = [[BufferChain alloc] initWithController:self];
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[self notifyStreamChanged:userInfo];
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while(![bufferChain open:url withOutputFormat:[output format] withRGInfo:rgi]) {
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while(![bufferChain open:url withOutputFormat:[output format] withOutputConfig:[output config] withRGInfo:rgi]) {
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bufferChain = nil;
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[self requestNextStream:userInfo];
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@ -353,7 +353,7 @@
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}
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if(pathsEqual || ([[nextStream scheme] isEqualToString:[[lastChain streamURL] scheme]] && (([nextStream host] == nil && [[lastChain streamURL] host] == nil) || [[nextStream host] isEqualToString:[[lastChain streamURL] host]]) && [[nextStream path] isEqualToString:[[lastChain streamURL] path]])) {
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if([lastChain setTrack:nextStream] && [newChain openWithInput:[lastChain inputNode] withOutputFormat:[output format] withRGInfo:nextStreamRGInfo]) {
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if([lastChain setTrack:nextStream] && [newChain openWithInput:[lastChain inputNode] withOutputFormat:[output format] withOutputConfig:[output config] withRGInfo:nextStreamRGInfo]) {
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[newChain setStreamURL:nextStream];
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[newChain setUserInfo:nextStreamUserInfo];
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@ -369,7 +369,7 @@
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lastChain = nil;
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while(shouldContinue && ![newChain open:nextStream withOutputFormat:[output format] withRGInfo:nextStreamRGInfo]) {
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while(shouldContinue && ![newChain open:nextStream withOutputFormat:[output format] withOutputConfig:[output config] withRGInfo:nextStreamRGInfo]) {
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if(nextStream == nil) {
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newChain = nil;
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atomic_fetch_sub(&refCount, 1);
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@ -13,16 +13,72 @@
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NS_ASSUME_NONNULL_BEGIN
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enum {
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AudioChannelFrontLeft = 1 << 0,
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AudioChannelFrontRight = 1 << 1,
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AudioChannelFrontCenter = 1 << 2,
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AudioChannelLFE = 1 << 3,
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AudioChannelBackLeft = 1 << 4,
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AudioChannelBackRight = 1 << 5,
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AudioChannelFrontCenterLeft = 1 << 6,
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AudioChannelFrontCenterRight = 1 << 7,
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AudioChannelBackCenter = 1 << 8,
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AudioChannelSideLeft = 1 << 9,
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AudioChannelSideRight = 1 << 10,
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AudioChannelTopCenter = 1 << 11,
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AudioChannelTopFrontLeft = 1 << 12,
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AudioChannelTopFrontCenter = 1 << 13,
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AudioChannelTopFrontRight = 1 << 14,
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AudioChannelTopBackLeft = 1 << 15,
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AudioChannelTopBackCenter = 1 << 16,
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AudioChannelTopBackRight = 1 << 17,
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AudioConfigMono = AudioChannelFrontCenter,
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AudioConfigStereo = AudioChannelFrontLeft | AudioChannelFrontRight,
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AudioConfig3Point0 = AudioChannelFrontLeft | AudioChannelFrontRight |
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AudioChannelFrontCenter,
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AudioConfig4Point0 = AudioChannelFrontLeft | AudioChannelFrontRight |
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AudioChannelBackLeft | AudioChannelBackRight,
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AudioConfig5Point0 = AudioChannelFrontLeft | AudioChannelFrontRight |
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AudioChannelFrontCenter | AudioChannelBackLeft |
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AudioChannelBackRight,
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AudioConfig5Point1 = AudioChannelFrontLeft | AudioChannelFrontRight |
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AudioChannelFrontCenter | AudioChannelLFE |
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AudioChannelBackLeft | AudioChannelBackRight,
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AudioConfig5Point1Side = AudioChannelFrontLeft | AudioChannelFrontRight |
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AudioChannelFrontCenter | AudioChannelLFE |
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AudioChannelSideLeft | AudioChannelSideRight,
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AudioConfig6Point1 = AudioChannelFrontLeft | AudioChannelFrontRight |
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AudioChannelFrontCenter | AudioChannelLFE |
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AudioChannelBackCenter | AudioChannelSideLeft |
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AudioChannelSideRight,
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AudioConfig7Point1 = AudioChannelFrontLeft | AudioChannelFrontRight |
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AudioChannelFrontCenter | AudioChannelLFE |
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AudioChannelBackLeft | AudioChannelBackRight |
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AudioChannelSideLeft | AudioChannelSideRight,
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AudioChannelsBackLeftRight = AudioChannelBackLeft | AudioChannelBackRight,
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AudioChannelsSideLeftRight = AudioChannelSideLeft | AudioChannelSideRight,
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};
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@interface AudioChunk : NSObject {
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AudioStreamBasicDescription format;
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NSMutableData *chunkData;
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uint32_t channelConfig;
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BOOL formatAssigned;
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BOOL lossless;
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}
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@property AudioStreamBasicDescription format;
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@property uint32_t channelConfig;
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@property BOOL lossless;
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+ (uint32_t)guessChannelConfig:(uint32_t)channelCount;
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+ (uint32_t)channelIndexFromConfig:(uint32_t)channelConfig forFlag:(uint32_t)flag;
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+ (uint32_t)extractChannelFlag:(uint32_t)index fromConfig:(uint32_t)channelConfig;
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+ (uint32_t)countChannels:(uint32_t)channelConfig;
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+ (uint32_t)findChannelIndex:(uint32_t)flag;
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- (id)init;
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- (void)assignSamples:(const void *)data frameCount:(size_t)count;
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@ -21,6 +21,86 @@
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return self;
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}
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static const uint32_t AudioChannelConfigTable[] = {
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0,
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AudioConfigMono,
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AudioConfigStereo,
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AudioConfig3Point0,
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AudioConfig4Point0,
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AudioConfig5Point0,
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AudioConfig5Point1,
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AudioConfig6Point1,
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AudioConfig7Point1,
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0,
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AudioConfig7Point1 | AudioChannelFrontCenterLeft | AudioChannelFrontCenterRight
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};
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+ (uint32_t)guessChannelConfig:(uint32_t)channelCount {
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if(channelCount == 0) return 0;
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if(channelCount > 32) return 0;
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int ret = 0;
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if(channelCount < (sizeof(AudioChannelConfigTable) / sizeof(AudioChannelConfigTable[0])))
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ret = AudioChannelConfigTable[channelCount];
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if(!ret) {
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ret = (1 << channelCount) - 1;
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}
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assert([AudioChunk countChannels:ret] == channelCount);
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return ret;
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}
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+ (uint32_t)channelIndexFromConfig:(uint32_t)channelConfig forFlag:(uint32_t)flag {
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uint32_t index = 0;
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for(uint32_t walk = 0; walk < 32; ++walk) {
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uint32_t query = 1 << walk;
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if(flag & query) return index;
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if(channelConfig & query) ++index;
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}
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return ~0;
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}
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+ (uint32_t)extractChannelFlag:(uint32_t)index fromConfig:(uint32_t)channelConfig {
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uint32_t toskip = index;
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uint32_t flag = 1;
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while(flag) {
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if(channelConfig & flag) {
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if(toskip == 0) break;
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toskip--;
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}
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flag <<= 1;
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}
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return flag;
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}
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+ (uint32_t)countChannels:(uint32_t)channelConfig {
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return __builtin_popcount(channelConfig);
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}
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+ (uint32_t)findChannelIndex:(uint32_t)flag {
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uint32_t rv = 0;
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if((flag & 0xFFFF) == 0) {
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rv += 16;
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flag >>= 16;
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}
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if((flag & 0xFF) == 0) {
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rv += 8;
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flag >>= 8;
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}
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if((flag & 0xF) == 0) {
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rv += 4;
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flag >>= 4;
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}
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if((flag & 0x3) == 0) {
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rv += 2;
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flag >>= 2;
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}
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if((flag & 0x1) == 0) {
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rv += 1;
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flag >>= 1;
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}
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assert(flag & 1);
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return rv;
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}
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@synthesize lossless;
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- (AudioStreamBasicDescription)format {
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@ -30,6 +110,20 @@
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- (void)setFormat:(AudioStreamBasicDescription)informat {
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formatAssigned = YES;
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format = informat;
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channelConfig = [AudioChunk guessChannelConfig:format.mChannelsPerFrame];
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}
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- (uint32_t)channelConfig {
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return channelConfig;
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}
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- (void)setChannelConfig:(uint32_t)config {
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if(formatAssigned) {
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if(config == 0) {
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config = [AudioChunk guessChannelConfig:format.mChannelsPerFrame];
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}
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}
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channelConfig = config;
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}
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- (void)assignSamples:(const void *)data frameCount:(size_t)count {
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@ -17,6 +17,7 @@
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ConverterNode *converterNode;
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AudioStreamBasicDescription inputFormat;
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uint32_t inputChannelConfig;
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NSURL *streamURL;
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id userInfo;
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@ -30,14 +31,15 @@
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- (id)initWithController:(id)c;
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- (void)buildChain;
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- (BOOL)open:(NSURL *)url withOutputFormat:(AudioStreamBasicDescription)outputFormat withRGInfo:(NSDictionary *)rgi;
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- (BOOL)open:(NSURL *)url withOutputFormat:(AudioStreamBasicDescription)outputFormat withOutputConfig:(uint32_t)outputConfig withRGInfo:(NSDictionary *)rgi;
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// Used when changing tracks to reuse the same decoder
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- (BOOL)openWithInput:(InputNode *)i withOutputFormat:(AudioStreamBasicDescription)outputFormat withRGInfo:(NSDictionary *)rgi;
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- (BOOL)openWithInput:(InputNode *)i withOutputFormat:(AudioStreamBasicDescription)outputFormat withOutputConfig:(uint32_t)outputConfig withRGInfo:(NSDictionary *)rgi;
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// Used when resetting the decoder on seek
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- (BOOL)openWithDecoder:(id<CogDecoder>)decoder
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withOutputFormat:(AudioStreamBasicDescription)outputFormat
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withOutputConfig:(uint32_t)outputConfig
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withRGInfo:(NSDictionary *)rgi;
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- (void)seek:(double)time;
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@ -64,7 +66,7 @@
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- (BOOL)endOfInputReached;
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- (BOOL)setTrack:(NSURL *)track;
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- (void)inputFormatDidChange:(AudioStreamBasicDescription)format;
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- (void)inputFormatDidChange:(AudioStreamBasicDescription)format inputConfig:(uint32_t)inputConfig;
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- (BOOL)isRunning;
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@ -72,6 +74,7 @@
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- (ConverterNode *)converter;
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- (AudioStreamBasicDescription)inputFormat;
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- (uint32_t)inputConfig;
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- (double)secondsBuffered;
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@ -40,7 +40,7 @@
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finalNode = converterNode;
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}
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- (BOOL)open:(NSURL *)url withOutputFormat:(AudioStreamBasicDescription)outputFormat withRGInfo:(NSDictionary *)rgi {
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- (BOOL)open:(NSURL *)url withOutputFormat:(AudioStreamBasicDescription)outputFormat withOutputConfig:(uint32_t)outputConfig withRGInfo:(NSDictionary *)rgi {
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[self setStreamURL:url];
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[self buildChain];
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@ -61,12 +61,16 @@
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NSDictionary *properties = [inputNode properties];
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inputFormat = [inputNode nodeFormat];
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if([properties valueForKey:@"channelConfig"])
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inputChannelConfig = [[properties valueForKey:@"channelConfig"] intValue];
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outputFormat.mChannelsPerFrame = inputFormat.mChannelsPerFrame;
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outputFormat.mBytesPerFrame = ((outputFormat.mBitsPerChannel + 7) / 8) * outputFormat.mChannelsPerFrame;
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outputFormat.mBytesPerPacket = outputFormat.mBytesPerFrame * outputFormat.mFramesPerPacket;
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if(![converterNode setupWithInputFormat:inputFormat outputFormat:outputFormat isLossless:[[properties valueForKey:@"encoding"] isEqualToString:@"lossless"]])
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outputConfig = inputChannelConfig;
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if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig outputFormat:outputFormat outputConfig:outputConfig isLossless:[[properties valueForKey:@"encoding"] isEqualToString:@"lossless"]])
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return NO;
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[self setRGInfo:rgi];
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@ -76,7 +80,7 @@
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return YES;
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}
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- (BOOL)openWithInput:(InputNode *)i withOutputFormat:(AudioStreamBasicDescription)outputFormat withRGInfo:(NSDictionary *)rgi {
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- (BOOL)openWithInput:(InputNode *)i withOutputFormat:(AudioStreamBasicDescription)outputFormat withOutputConfig:(uint32_t)outputConfig withRGInfo:(NSDictionary *)rgi {
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DLog(@"New buffer chain!");
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[self buildChain];
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@ -86,13 +90,17 @@
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NSDictionary *properties = [inputNode properties];
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inputFormat = [inputNode nodeFormat];
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if([properties valueForKey:@"channelConfig"])
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inputChannelConfig = [[properties valueForKey:@"channelConfig"] intValue];
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outputFormat.mChannelsPerFrame = inputFormat.mChannelsPerFrame;
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outputFormat.mBytesPerFrame = ((outputFormat.mBitsPerChannel + 7) / 8) * outputFormat.mChannelsPerFrame;
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outputFormat.mBytesPerPacket = outputFormat.mBytesPerFrame * outputFormat.mFramesPerPacket;
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outputConfig = inputChannelConfig;
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DLog(@"Input Properties: %@", properties);
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if(![converterNode setupWithInputFormat:inputFormat outputFormat:outputFormat isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
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if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig outputFormat:outputFormat outputConfig:outputConfig isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
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return NO;
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[self setRGInfo:rgi];
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@ -102,6 +110,7 @@
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- (BOOL)openWithDecoder:(id<CogDecoder>)decoder
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withOutputFormat:(AudioStreamBasicDescription)outputFormat
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withOutputConfig:(uint32_t)outputConfig
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withRGInfo:(NSDictionary *)rgi;
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{
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DLog(@"New buffer chain!");
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@ -115,12 +124,16 @@
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DLog(@"Input Properties: %@", properties);
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inputFormat = [inputNode nodeFormat];
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if([properties valueForKey:@"channelConfig"])
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inputChannelConfig = [[properties valueForKey:@"channelConfig"] intValue];
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outputFormat.mChannelsPerFrame = inputFormat.mChannelsPerFrame;
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outputFormat.mBytesPerFrame = ((outputFormat.mBitsPerChannel + 7) / 8) * outputFormat.mChannelsPerFrame;
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outputFormat.mBytesPerPacket = outputFormat.mBytesPerFrame * outputFormat.mFramesPerPacket;
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if(![converterNode setupWithInputFormat:inputFormat outputFormat:outputFormat isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
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outputConfig = inputChannelConfig;
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if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig outputFormat:outputFormat outputConfig:outputConfig isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
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return NO;
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[self setRGInfo:rgi];
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@ -180,7 +193,7 @@
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[controller launchOutputThread];
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}
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- (void)inputFormatDidChange:(AudioStreamBasicDescription)format {
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- (void)inputFormatDidChange:(AudioStreamBasicDescription)format inputConfig:(uint32_t)inputConfig {
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DLog(@"FORMAT DID CHANGE!");
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}
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@ -225,6 +238,10 @@
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return inputFormat;
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}
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- (uint32_t)inputConfig {
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return inputChannelConfig;
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}
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- (double)secondsBuffered {
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double duration = 0.0;
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OutputNode *outputNode = (OutputNode *)[controller output];
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@ -86,6 +86,7 @@
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NSData *removedData = [chunk removeSamples:maxFrameCount];
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AudioChunk *ret = [[AudioChunk alloc] init];
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[ret setFormat:[chunk format]];
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[ret setChannelConfig:[chunk channelConfig]];
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[ret assignSamples:[removedData bytes] frameCount:maxFrameCount];
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listDuration -= [ret duration];
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inRemover = NO;
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@ -68,8 +68,13 @@
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AudioStreamBasicDescription dmFloatFormat; // downmixed/upmixed float format
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AudioStreamBasicDescription outputFormat;
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uint32_t inputChannelConfig;
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uint32_t outputChannelConfig;
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AudioStreamBasicDescription previousOutputFormat;
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uint32_t previousOutputConfig;
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AudioStreamBasicDescription rememberedInputFormat;
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uint32_t rememberedInputConfig;
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RefillNode *refillNode;
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id __weak originalPreviousNode;
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@ -82,7 +87,7 @@
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- (id)initWithController:(id)c previous:(id)p;
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- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inputFormat outputFormat:(AudioStreamBasicDescription)outputFormat isLossless:(BOOL)lossless;
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- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inputFormat withInputConfig:(uint32_t)inputConfig outputFormat:(AudioStreamBasicDescription)outputFormat outputConfig:(uint32_t)outputConfig isLossless:(BOOL)lossless;
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- (void)cleanUp;
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- (void)process;
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@ -90,9 +95,9 @@
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- (void)setRGInfo:(NSDictionary *)rgi;
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- (void)setOutputFormat:(AudioStreamBasicDescription)format;
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- (void)setOutputFormat:(AudioStreamBasicDescription)format outputConfig:(uint32_t)outputConfig;
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- (void)inputFormatDidChange:(AudioStreamBasicDescription)format;
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- (void)inputFormatDidChange:(AudioStreamBasicDescription)format inputConfig:(uint32_t)inputConfig;
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- (void)refreshVolumeScaling;
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@ -435,7 +435,7 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
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[self setShouldContinue:YES];
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refillNode = nil;
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[self cleanUp];
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[self setupWithInputFormat:rememberedInputFormat outputFormat:outputFormat isLossless:rememberedLossless];
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[self setupWithInputFormat:rememberedInputFormat withInputConfig:rememberedInputConfig outputFormat:outputFormat outputConfig:outputChannelConfig isLossless:rememberedLossless];
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continue;
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} else
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break;
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@ -843,12 +843,16 @@ static float db_to_scale(float db) {
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volumeScale = scale;
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}
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- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inf outputFormat:(AudioStreamBasicDescription)outf isLossless:(BOOL)lossless {
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- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inf withInputConfig:(uint32_t)inputConfig outputFormat:(AudioStreamBasicDescription)outf outputConfig:(uint32_t)outputConfig isLossless:(BOOL)lossless {
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// Make the converter
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inputFormat = inf;
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outputFormat = outf;
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inputChannelConfig = inputConfig;
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outputChannelConfig = outputConfig;
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nodeFormat = outputFormat;
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nodeChannelConfig = outputChannelConfig;
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rememberedLossless = lossless;
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@ -954,26 +958,31 @@ static float db_to_scale(float db) {
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[self cleanUp];
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}
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- (void)setOutputFormat:(AudioStreamBasicDescription)format {
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- (void)setOutputFormat:(AudioStreamBasicDescription)format outputConfig:(uint32_t)outputConfig {
|
||||
DLog(@"SETTING OUTPUT FORMAT!");
|
||||
previousOutputFormat = outputFormat;
|
||||
previousOutputConfig = outputChannelConfig;
|
||||
outputFormat = format;
|
||||
outputChannelConfig = outputConfig;
|
||||
outputFormatChanged = YES;
|
||||
}
|
||||
|
||||
- (void)inputFormatDidChange:(AudioStreamBasicDescription)format {
|
||||
- (void)inputFormatDidChange:(AudioStreamBasicDescription)format inputConfig:(uint32_t)inputConfig {
|
||||
DLog(@"FORMAT CHANGED");
|
||||
paused = YES;
|
||||
[self cleanUp];
|
||||
if(outputFormatChanged && ![buffer isEmpty] &&
|
||||
memcmp(&outputFormat, &previousOutputFormat, sizeof(outputFormat)) != 0) {
|
||||
(outputChannelConfig != previousOutputConfig ||
|
||||
memcmp(&outputFormat, &previousOutputFormat, sizeof(outputFormat)) != 0)) {
|
||||
// Transfer previously buffered data, remember input format
|
||||
rememberedInputFormat = format;
|
||||
rememberedInputConfig = inputChannelConfig;
|
||||
originalPreviousNode = previousNode;
|
||||
refillNode = [[RefillNode alloc] initWithController:controller previous:nil];
|
||||
[self setPreviousNode:refillNode];
|
||||
|
||||
[refillNode setFormat:previousOutputFormat];
|
||||
[refillNode setChannelConfig:previousOutputConfig];
|
||||
|
||||
for(;;) {
|
||||
AudioChunk *chunk = [buffer removeSamples:16384];
|
||||
|
|
@ -985,9 +994,9 @@ static float db_to_scale(float db) {
|
|||
break;
|
||||
}
|
||||
|
||||
[self setupWithInputFormat:previousOutputFormat outputFormat:outputFormat isLossless:rememberedLossless];
|
||||
[self setupWithInputFormat:previousOutputFormat withInputConfig:[AudioChunk guessChannelConfig:previousOutputFormat.mChannelsPerFrame] outputFormat:outputFormat outputConfig:outputChannelConfig isLossless:rememberedLossless];
|
||||
} else {
|
||||
[self setupWithInputFormat:format outputFormat:outputFormat isLossless:rememberedLossless];
|
||||
[self setupWithInputFormat:format withInputConfig:inputConfig outputFormat:outputFormat outputConfig:outputChannelConfig isLossless:rememberedLossless];
|
||||
}
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -16,9 +16,12 @@
|
|||
|
||||
AudioStreamBasicDescription inputFormat;
|
||||
AudioStreamBasicDescription outputFormat;
|
||||
|
||||
uint32_t inConfig;
|
||||
uint32_t outConfig;
|
||||
}
|
||||
|
||||
- (id)initWithInputFormat:(AudioStreamBasicDescription)inf andOutputFormat:(AudioStreamBasicDescription)outf;
|
||||
- (id)initWithInputFormat:(AudioStreamBasicDescription)inf inputConfig:(uint32_t)iConfig andOutputFormat:(AudioStreamBasicDescription)outf outputConfig:(uint32_t)oConfig;
|
||||
|
||||
- (void)process:(const void *)inBuffer frameCount:(size_t)frames output:(void *)outBuffer;
|
||||
|
||||
|
|
|
|||
|
|
@ -10,74 +10,144 @@
|
|||
|
||||
#import "Logging.h"
|
||||
|
||||
static const float STEREO_DOWNMIX[8 - 2][8][2] = {
|
||||
/*3.0*/
|
||||
{
|
||||
{ 0.5858F, 0.0F },
|
||||
{ 0.0F, 0.5858F },
|
||||
{ 0.4142F, 0.4142F } },
|
||||
/*quadrophonic*/
|
||||
{
|
||||
{ 0.4226F, 0.0F },
|
||||
{ 0.0F, 0.4226F },
|
||||
{ 0.366F, 0.2114F },
|
||||
{ 0.2114F, 0.336F } },
|
||||
/*5.0*/
|
||||
{
|
||||
{ 0.651F, 0.0F },
|
||||
{ 0.0F, 0.651F },
|
||||
{ 0.46F, 0.46F },
|
||||
{ 0.5636F, 0.3254F },
|
||||
{ 0.3254F, 0.5636F } },
|
||||
/*5.1*/
|
||||
{
|
||||
{ 0.529F, 0.0F },
|
||||
{ 0.0F, 0.529F },
|
||||
{ 0.3741F, 0.3741F },
|
||||
{ 0.3741F, 0.3741F },
|
||||
{ 0.4582F, 0.2645F },
|
||||
{ 0.2645F, 0.4582F } },
|
||||
/*6.1*/
|
||||
{
|
||||
{ 0.4553F, 0.0F },
|
||||
{ 0.0F, 0.4553F },
|
||||
{ 0.322F, 0.322F },
|
||||
{ 0.322F, 0.322F },
|
||||
{ 0.2788F, 0.2788F },
|
||||
{ 0.3943F, 0.2277F },
|
||||
{ 0.2277F, 0.3943F } },
|
||||
/*7.1*/
|
||||
{
|
||||
{ 0.3886F, 0.0F },
|
||||
{ 0.0F, 0.3886F },
|
||||
{ 0.2748F, 0.2748F },
|
||||
{ 0.2748F, 0.2748F },
|
||||
{ 0.3366F, 0.1943F },
|
||||
{ 0.1943F, 0.3366F },
|
||||
{ 0.3366F, 0.1943F },
|
||||
{ 0.1943F, 0.3366F } }
|
||||
};
|
||||
#import "AudioChunk.h"
|
||||
|
||||
static void downmix_to_stereo(const float *inBuffer, int channels, float *outBuffer, size_t count) {
|
||||
if(channels >= 3 && channels <= 8)
|
||||
for(size_t i = 0; i < count; ++i) {
|
||||
float left = 0, right = 0;
|
||||
for(int j = 0; j < channels; ++j) {
|
||||
left += inBuffer[i * channels + j] * STEREO_DOWNMIX[channels - 3][j][0];
|
||||
right += inBuffer[i * channels + j] * STEREO_DOWNMIX[channels - 3][j][1];
|
||||
static void downmix_to_stereo(const float *inBuffer, int channels, uint32_t config, float *outBuffer, size_t count) {
|
||||
float FrontRatios[2] = { 0.0F, 0.0F };
|
||||
float FrontCenterRatio = 0.0F;
|
||||
float LFERatio = 0.0F;
|
||||
float BackRatios[2] = { 0.0F, 0.0F };
|
||||
float BackCenterRatio = 0.0F;
|
||||
float SideRatios[2] = { 0.0F, 0.0F };
|
||||
if(config & (AudioChannelFrontLeft | AudioChannelFrontRight)) {
|
||||
FrontRatios[0] = 1.0F;
|
||||
}
|
||||
if(config & AudioChannelFrontCenter) {
|
||||
FrontRatios[0] = 0.5858F;
|
||||
FrontCenterRatio = 0.4142F;
|
||||
}
|
||||
if(config & (AudioChannelBackLeft | AudioChannelBackRight)) {
|
||||
if(config & AudioChannelFrontCenter) {
|
||||
FrontRatios[0] = 0.651F;
|
||||
FrontCenterRatio = 0.46F;
|
||||
BackRatios[0] = 0.5636F;
|
||||
BackRatios[1] = 0.3254F;
|
||||
} else {
|
||||
FrontRatios[0] = 0.4226F;
|
||||
BackRatios[0] = 0.366F;
|
||||
BackRatios[1] = 0.2114F;
|
||||
}
|
||||
}
|
||||
if(config & AudioChannelLFE) {
|
||||
FrontRatios[0] *= 0.8F;
|
||||
FrontCenterRatio *= 0.8F;
|
||||
LFERatio = FrontCenterRatio;
|
||||
BackRatios[0] *= 0.8F;
|
||||
BackRatios[1] *= 0.8F;
|
||||
}
|
||||
if(config & AudioChannelBackCenter) {
|
||||
FrontRatios[0] *= 0.86F;
|
||||
FrontCenterRatio *= 0.86F;
|
||||
LFERatio *= 0.86F;
|
||||
BackRatios[0] *= 0.86F;
|
||||
BackRatios[1] *= 0.86F;
|
||||
BackCenterRatio = FrontCenterRatio * 0.86F;
|
||||
}
|
||||
if(config & (AudioChannelSideLeft | AudioChannelSideRight)) {
|
||||
float ratio = 0.73F;
|
||||
if(config & AudioChannelBackCenter) ratio = 0.85F;
|
||||
FrontRatios[0] *= ratio;
|
||||
FrontCenterRatio *= ratio;
|
||||
LFERatio *= ratio;
|
||||
BackRatios[0] *= ratio;
|
||||
BackRatios[1] *= ratio;
|
||||
BackCenterRatio *= ratio;
|
||||
SideRatios[0] = 0.463882352941176 * ratio;
|
||||
SideRatios[1] = 0.267882352941176 * ratio;
|
||||
}
|
||||
|
||||
int32_t channelIndexes[channels];
|
||||
for(int i = 0; i < channels; ++i) {
|
||||
channelIndexes[i] = [AudioChunk findChannelIndex:[AudioChunk extractChannelFlag:i fromConfig:config]];
|
||||
}
|
||||
|
||||
for(size_t i = 0; i < count; ++i) {
|
||||
float left = 0.0F, right = 0.0F;
|
||||
for(uint32_t j = 0; j < channels; ++j) {
|
||||
float inSample = inBuffer[i * channels + j];
|
||||
switch(channelIndexes[j]) {
|
||||
case 0:
|
||||
left += inSample * FrontRatios[0];
|
||||
right += inSample * FrontRatios[1];
|
||||
break;
|
||||
|
||||
case 1:
|
||||
left += inSample * FrontRatios[1];
|
||||
right += inSample * FrontRatios[0];
|
||||
break;
|
||||
|
||||
case 2:
|
||||
left += inSample * FrontCenterRatio;
|
||||
right += inSample * FrontCenterRatio;
|
||||
break;
|
||||
|
||||
case 3:
|
||||
left += inSample * LFERatio;
|
||||
right += inSample * LFERatio;
|
||||
break;
|
||||
|
||||
case 4:
|
||||
left += inSample * BackRatios[0];
|
||||
right += inSample * BackRatios[1];
|
||||
break;
|
||||
|
||||
case 5:
|
||||
left += inSample * BackRatios[1];
|
||||
right += inSample * BackRatios[0];
|
||||
break;
|
||||
|
||||
case 6:
|
||||
case 7:
|
||||
break;
|
||||
|
||||
case 8:
|
||||
left += inSample * BackCenterRatio;
|
||||
right += inSample * BackCenterRatio;
|
||||
break;
|
||||
|
||||
case 9:
|
||||
left += inSample * SideRatios[0];
|
||||
right += inSample * SideRatios[1];
|
||||
break;
|
||||
|
||||
case 10:
|
||||
left += inSample * SideRatios[1];
|
||||
right += inSample * SideRatios[0];
|
||||
break;
|
||||
|
||||
case 11:
|
||||
case 12:
|
||||
case 13:
|
||||
case 14:
|
||||
case 15:
|
||||
case 16:
|
||||
case 17:
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
outBuffer[i * 2 + 0] = left;
|
||||
outBuffer[i * 2 + 1] = right;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
static void downmix_to_mono(const float *inBuffer, int channels, float *outBuffer, size_t count) {
|
||||
static void downmix_to_mono(const float *inBuffer, int channels, uint32_t config, float *outBuffer, size_t count) {
|
||||
float tempBuffer[count * 2];
|
||||
if(channels >= 3 && channels <= 8) {
|
||||
downmix_to_stereo(inBuffer, channels, tempBuffer, count);
|
||||
inBuffer = tempBuffer;
|
||||
channels = 2;
|
||||
}
|
||||
downmix_to_stereo(inBuffer, channels, config, tempBuffer, count);
|
||||
inBuffer = tempBuffer;
|
||||
channels = 2;
|
||||
config = AudioConfigStereo;
|
||||
float invchannels = 1.0 / (float)channels;
|
||||
for(size_t i = 0; i < count; ++i) {
|
||||
float sample = 0;
|
||||
|
|
@ -88,62 +158,100 @@ static void downmix_to_mono(const float *inBuffer, int channels, float *outBuffe
|
|||
}
|
||||
}
|
||||
|
||||
static void upmix(const float *inBuffer, int inchannels, float *outBuffer, int outchannels, size_t count) {
|
||||
for(ssize_t i = 0; i < count; ++i) {
|
||||
if(inchannels == 1 && outchannels == 2) {
|
||||
static void upmix(const float *inBuffer, int inchannels, uint32_t inconfig, float *outBuffer, int outchannels, uint32_t outconfig, size_t count) {
|
||||
if(inconfig == AudioConfigMono && outconfig == AudioConfigStereo) {
|
||||
for(size_t i = 0; i < count; ++i) {
|
||||
// upmix mono to stereo
|
||||
float sample = inBuffer[i];
|
||||
outBuffer[i * 2 + 0] = sample;
|
||||
outBuffer[i * 2 + 1] = sample;
|
||||
} else if(inchannels == 1 && outchannels == 4) {
|
||||
}
|
||||
} else if(inconfig == AudioConfigMono && outconfig == AudioConfig4Point0) {
|
||||
for(size_t i = 0; i < count; ++i) {
|
||||
// upmix mono to quad
|
||||
float sample = inBuffer[i];
|
||||
outBuffer[i * 4 + 0] = sample;
|
||||
outBuffer[i * 4 + 1] = sample;
|
||||
outBuffer[i * 4 + 2] = 0;
|
||||
outBuffer[i * 4 + 3] = 0;
|
||||
} else if(inchannels == 1 && (outchannels == 3 || outchannels >= 5)) {
|
||||
}
|
||||
} else if(inconfig == AudioConfigMono && (outconfig & AudioChannelFrontCenter)) {
|
||||
uint32_t cIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelFrontCenter];
|
||||
for(size_t i = 0; i < count; ++i) {
|
||||
// upmix mono to center channel
|
||||
float sample = inBuffer[i];
|
||||
outBuffer[i * outchannels + 2] = sample;
|
||||
for(int j = 0; j < 2; ++j) {
|
||||
outBuffer[i * outchannels + cIndex] = sample;
|
||||
for(int j = 0; j < cIndex; ++j) {
|
||||
outBuffer[i * outchannels + j] = 0;
|
||||
}
|
||||
for(int j = 3; j < outchannels; ++j) {
|
||||
for(int j = cIndex + 1; j < outchannels; ++j) {
|
||||
outBuffer[i * outchannels + j] = 0;
|
||||
}
|
||||
} else if(inchannels == 4 && outchannels >= 5) {
|
||||
}
|
||||
} else if(inconfig == AudioConfig4Point0 && outchannels >= 5) {
|
||||
uint32_t flIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelFrontLeft];
|
||||
uint32_t frIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelFrontRight];
|
||||
uint32_t blIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelBackLeft];
|
||||
uint32_t brIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelBackRight];
|
||||
for(size_t i = 0; i < count; ++i) {
|
||||
float fl = inBuffer[i * 4 + 0];
|
||||
float fr = inBuffer[i * 4 + 1];
|
||||
float bl = inBuffer[i * 4 + 2];
|
||||
float br = inBuffer[i * 4 + 3];
|
||||
const int skipclfe = (outchannels == 5) ? 1 : 2;
|
||||
outBuffer[i * outchannels + 0] = fl;
|
||||
outBuffer[i * outchannels + 1] = fr;
|
||||
outBuffer[i * outchannels + skipclfe + 2] = bl;
|
||||
outBuffer[i * outchannels + skipclfe + 3] = br;
|
||||
for(int j = 0; j < skipclfe; ++j) {
|
||||
outBuffer[i * outchannels + 2 + j] = 0;
|
||||
memset(outBuffer + i * outchannels, 0, sizeof(float) * outchannels);
|
||||
if(flIndex != ~0) {
|
||||
outBuffer[i * outchannels + flIndex] = fl;
|
||||
}
|
||||
for(int j = 4 + skipclfe; j < outchannels; ++j) {
|
||||
outBuffer[i * outchannels + j] = 0;
|
||||
if(frIndex != ~0) {
|
||||
outBuffer[i * outchannels + frIndex] = fr;
|
||||
}
|
||||
} else if(inchannels == 5 && outchannels >= 6) {
|
||||
if(blIndex != ~0) {
|
||||
outBuffer[i * outchannels + blIndex] = bl;
|
||||
}
|
||||
if(brIndex != ~0) {
|
||||
outBuffer[i * outchannels + brIndex] = br;
|
||||
}
|
||||
}
|
||||
} else if(inconfig == AudioConfig5Point0 && outchannels >= 6) {
|
||||
uint32_t flIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelFrontLeft];
|
||||
uint32_t frIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelFrontRight];
|
||||
uint32_t cIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelFrontCenter];
|
||||
uint32_t blIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelBackLeft];
|
||||
uint32_t brIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelBackRight];
|
||||
for(size_t i = 0; i < count; ++i) {
|
||||
float fl = inBuffer[i * 5 + 0];
|
||||
float fr = inBuffer[i * 5 + 1];
|
||||
float c = inBuffer[i * 5 + 2];
|
||||
float bl = inBuffer[i * 5 + 3];
|
||||
float br = inBuffer[i * 5 + 4];
|
||||
outBuffer[i * outchannels + 0] = fl;
|
||||
outBuffer[i * outchannels + 1] = fr;
|
||||
outBuffer[i * outchannels + 2] = c;
|
||||
outBuffer[i * outchannels + 3] = 0;
|
||||
outBuffer[i * outchannels + 4] = bl;
|
||||
outBuffer[i * outchannels + 5] = br;
|
||||
for(int j = 6; j < outchannels; ++j) {
|
||||
outBuffer[i * outchannels + j] = 0;
|
||||
memset(outBuffer + i * outchannels, 0, sizeof(float) * outchannels);
|
||||
if(flIndex != ~0) {
|
||||
outBuffer[i * outchannels + flIndex] = fl;
|
||||
}
|
||||
} else if(inchannels == 7 && outchannels == 8) {
|
||||
if(frIndex != ~0) {
|
||||
outBuffer[i * outchannels + frIndex] = fr;
|
||||
}
|
||||
if(cIndex != ~0) {
|
||||
outBuffer[i * outchannels + cIndex] = c;
|
||||
}
|
||||
if(blIndex != ~0) {
|
||||
outBuffer[i * outchannels + blIndex] = bl;
|
||||
}
|
||||
if(brIndex != ~0) {
|
||||
outBuffer[i * outchannels + brIndex] = br;
|
||||
}
|
||||
}
|
||||
} else if(inconfig == AudioConfig6Point1 && outchannels >= 8) {
|
||||
uint32_t flIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelFrontLeft];
|
||||
uint32_t frIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelFrontRight];
|
||||
uint32_t cIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelFrontCenter];
|
||||
uint32_t lfeIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelLFE];
|
||||
uint32_t blIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelBackLeft];
|
||||
uint32_t brIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelBackRight];
|
||||
uint32_t bcIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelBackCenter];
|
||||
uint32_t slIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelSideLeft];
|
||||
uint32_t srIndex = [AudioChunk channelIndexFromConfig:outconfig forFlag:AudioChannelSideRight];
|
||||
for(size_t i = 0; i < count; ++i) {
|
||||
float fl = inBuffer[i * 7 + 0];
|
||||
float fr = inBuffer[i * 7 + 1];
|
||||
float c = inBuffer[i * 7 + 2];
|
||||
|
|
@ -151,25 +259,48 @@ static void upmix(const float *inBuffer, int inchannels, float *outBuffer, int o
|
|||
float sl = inBuffer[i * 7 + 4];
|
||||
float sr = inBuffer[i * 7 + 5];
|
||||
float bc = inBuffer[i * 7 + 6];
|
||||
outBuffer[i * 8 + 0] = fl;
|
||||
outBuffer[i * 8 + 1] = fr;
|
||||
outBuffer[i * 8 + 2] = c;
|
||||
outBuffer[i * 8 + 3] = lfe;
|
||||
outBuffer[i * 8 + 4] = bc;
|
||||
outBuffer[i * 8 + 5] = bc;
|
||||
outBuffer[i * 8 + 6] = sl;
|
||||
outBuffer[i * 8 + 7] = sr;
|
||||
} else {
|
||||
memset(outBuffer + i * outchannels, 0, sizeof(float) * outchannels);
|
||||
if(flIndex != ~0) {
|
||||
outBuffer[i * outchannels + flIndex] = fl;
|
||||
}
|
||||
if(frIndex != ~0) {
|
||||
outBuffer[i * outchannels + frIndex] = fr;
|
||||
}
|
||||
if(cIndex != ~0) {
|
||||
outBuffer[i * outchannels + cIndex] = c;
|
||||
}
|
||||
if(lfeIndex != ~0) {
|
||||
outBuffer[i * outchannels + lfeIndex] = lfe;
|
||||
}
|
||||
if(slIndex != ~0) {
|
||||
outBuffer[i * outchannels + slIndex] = sl;
|
||||
}
|
||||
if(srIndex != ~0) {
|
||||
outBuffer[i * outchannels + srIndex] = sr;
|
||||
}
|
||||
if(bcIndex != ~0) {
|
||||
outBuffer[i * outchannels + bcIndex] = bc;
|
||||
} else {
|
||||
if(blIndex != ~0) {
|
||||
outBuffer[i * outchannels + blIndex] = bc;
|
||||
}
|
||||
if(brIndex != ~0) {
|
||||
outBuffer[i * outchannels + brIndex] = bc;
|
||||
}
|
||||
}
|
||||
}
|
||||
} else {
|
||||
uint32_t outIndexes[inchannels];
|
||||
for(int i = 0; i < inchannels; ++i) {
|
||||
uint32_t channelFlag = [AudioChunk extractChannelFlag:i fromConfig:inconfig];
|
||||
outIndexes[i] = [AudioChunk channelIndexFromConfig:outconfig forFlag:channelFlag];
|
||||
}
|
||||
for(size_t i = 0; i < count; ++i) {
|
||||
// upmix N channels to N channels plus silence the empty channels
|
||||
float samples[inchannels];
|
||||
for(int j = 0; j < inchannels; ++j) {
|
||||
samples[j] = inBuffer[i * inchannels + j];
|
||||
}
|
||||
for(int j = 0; j < inchannels; ++j) {
|
||||
outBuffer[i * outchannels + j] = samples[j];
|
||||
}
|
||||
for(int j = inchannels; j < outchannels; ++j) {
|
||||
outBuffer[i * outchannels + j] = 0;
|
||||
if(outIndexes[j] != ~0) {
|
||||
outBuffer[i * outchannels + outIndexes[j]] = inBuffer[i * inchannels + j];
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
|
@ -177,7 +308,7 @@ static void upmix(const float *inBuffer, int inchannels, float *outBuffer, int o
|
|||
|
||||
@implementation DownmixProcessor
|
||||
|
||||
- (id)initWithInputFormat:(AudioStreamBasicDescription)inf andOutputFormat:(AudioStreamBasicDescription)outf {
|
||||
- (id)initWithInputFormat:(AudioStreamBasicDescription)inf inputConfig:(uint32_t)iConfig andOutputFormat:(AudioStreamBasicDescription)outf outputConfig:(uint32_t)oConfig {
|
||||
self = [super init];
|
||||
|
||||
if(self) {
|
||||
|
|
@ -198,6 +329,9 @@ static void upmix(const float *inBuffer, int inchannels, float *outBuffer, int o
|
|||
inputFormat = inf;
|
||||
outputFormat = outf;
|
||||
|
||||
inConfig = iConfig;
|
||||
outConfig = oConfig;
|
||||
|
||||
[self setupVirt];
|
||||
|
||||
[[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.headphoneVirtualization" options:0 context:nil];
|
||||
|
|
@ -221,8 +355,9 @@ static void upmix(const float *inBuffer, int inchannels, float *outBuffer, int o
|
|||
|
||||
if(hVirt &&
|
||||
outputFormat.mChannelsPerFrame == 2 &&
|
||||
outConfig == AudioConfigStereo &&
|
||||
inputFormat.mChannelsPerFrame >= 1 &&
|
||||
inputFormat.mChannelsPerFrame <= 8) {
|
||||
(inConfig & (AudioConfig7Point1 | AudioChannelBackCenter)) != 0) {
|
||||
NSString *userPreset = [[[NSUserDefaultsController sharedUserDefaultsController] defaults] stringForKey:@"hrirPath"];
|
||||
|
||||
NSURL *presetUrl = nil;
|
||||
|
|
@ -241,7 +376,7 @@ static void upmix(const float *inBuffer, int inchannels, float *outBuffer, int o
|
|||
|
||||
if(presetUrl) {
|
||||
@synchronized(hFilter) {
|
||||
hFilter = [[HeadphoneFilter alloc] initWithImpulseFile:presetUrl forSampleRate:outputFormat.mSampleRate withInputChannels:inputFormat.mChannelsPerFrame];
|
||||
hFilter = [[HeadphoneFilter alloc] initWithImpulseFile:presetUrl forSampleRate:outputFormat.mSampleRate withInputChannels:inputFormat.mChannelsPerFrame withConfig:inConfig];
|
||||
}
|
||||
}
|
||||
}
|
||||
|
|
@ -267,13 +402,13 @@ static void upmix(const float *inBuffer, int inchannels, float *outBuffer, int o
|
|||
}
|
||||
}
|
||||
|
||||
if(inputFormat.mChannelsPerFrame > 2 && outputFormat.mChannelsPerFrame == 2) {
|
||||
downmix_to_stereo((const float *)inBuffer, inputFormat.mChannelsPerFrame, (float *)outBuffer, frames);
|
||||
} else if(inputFormat.mChannelsPerFrame > 1 && outputFormat.mChannelsPerFrame == 1) {
|
||||
downmix_to_mono((const float *)inBuffer, inputFormat.mChannelsPerFrame, (float *)outBuffer, frames);
|
||||
if(inputFormat.mChannelsPerFrame > 2 && outConfig == AudioConfigStereo) {
|
||||
downmix_to_stereo((const float *)inBuffer, inputFormat.mChannelsPerFrame, inConfig, (float *)outBuffer, frames);
|
||||
} else if(inputFormat.mChannelsPerFrame > 1 && outConfig == AudioConfigMono) {
|
||||
downmix_to_mono((const float *)inBuffer, inputFormat.mChannelsPerFrame, inConfig, (float *)outBuffer, frames);
|
||||
} else if(inputFormat.mChannelsPerFrame < outputFormat.mChannelsPerFrame) {
|
||||
upmix((const float *)inBuffer, inputFormat.mChannelsPerFrame, (float *)outBuffer, outputFormat.mChannelsPerFrame, frames);
|
||||
} else if(inputFormat.mChannelsPerFrame == outputFormat.mChannelsPerFrame) {
|
||||
upmix((const float *)inBuffer, inputFormat.mChannelsPerFrame, inConfig, (float *)outBuffer, outputFormat.mChannelsPerFrame, outConfig, frames);
|
||||
} else if(inConfig == outConfig) {
|
||||
memcpy(outBuffer, inBuffer, frames * outputFormat.mBytesPerPacket);
|
||||
}
|
||||
}
|
||||
|
|
|
|||
|
|
@ -42,7 +42,7 @@
|
|||
|
||||
+ (BOOL)validateImpulseFile:(NSURL *)url;
|
||||
|
||||
- (id)initWithImpulseFile:(NSURL *)url forSampleRate:(double)sampleRate withInputChannels:(size_t)channels;
|
||||
- (id)initWithImpulseFile:(NSURL *)url forSampleRate:(double)sampleRate withInputChannels:(size_t)channels withConfig:(uint32_t)config;
|
||||
|
||||
- (void)process:(const float *)inBuffer sampleCount:(size_t)count toBuffer:(float *)outBuffer;
|
||||
|
||||
|
|
|
|||
|
|
@ -6,6 +6,7 @@
|
|||
//
|
||||
|
||||
#import "HeadphoneFilter.h"
|
||||
#import "AudioChunk.h"
|
||||
#import "AudioDecoder.h"
|
||||
#import "AudioSource.h"
|
||||
|
||||
|
|
@ -17,44 +18,61 @@
|
|||
|
||||
@implementation HeadphoneFilter
|
||||
|
||||
// Symmetrical / no-reverb sets
|
||||
static const int8_t speakers_to_hesuvi_7[8][2][8] = {
|
||||
// mono/center
|
||||
{ { 6 }, { 6 } },
|
||||
// left/right
|
||||
{ { 0, 1 }, { 1, 0 } },
|
||||
// left/right/center
|
||||
{ { 0, 1, 6 }, { 1, 0, 6 } },
|
||||
// left/right/back lef/back right
|
||||
{ { 0, 1, 4, 5 }, { 1, 0, 5, 4 } },
|
||||
// left/right/center/back left/back right
|
||||
{ { 0, 1, 6, 4, 5 }, { 1, 0, 6, 5, 4 } },
|
||||
// left/right/center/lfe(center)/back left/back right
|
||||
{ { 0, 1, 6, 6, 4, 5 }, { 1, 0, 6, 6, 5, 4 } },
|
||||
// left/right/center/lfe(center)/back center(special)/side left/side right
|
||||
{ { 0, 1, 6, 6, -1, 2, 3 }, { 1, 0, 6, 6, -1, 3, 2 } },
|
||||
// left/right/center/lfe(center)/back left/back right/side left/side right
|
||||
{ { 0, 1, 6, 6, 4, 5, 2, 3 }, { 1, 0, 6, 6, 5, 4, 3, 2 } }
|
||||
enum {
|
||||
speaker_is_back_center = -1,
|
||||
speaker_not_present = -2,
|
||||
};
|
||||
|
||||
// Asymmetrical / reverb sets
|
||||
static const int8_t speakers_to_hesuvi_14[8][2][8] = {
|
||||
// mono/center
|
||||
{ { 6 }, { 13 } },
|
||||
// left/right
|
||||
{ { 0, 8 }, { 1, 7 } },
|
||||
// left/right/center
|
||||
{ { 0, 8, 6 }, { 1, 7, 13 } },
|
||||
// left/right/back left/back right
|
||||
{ { 0, 8, 4, 12 }, { 1, 7, 5, 11 } },
|
||||
// left/right/center/back left/back right
|
||||
{ { 0, 8, 6, 4, 12 }, { 1, 7, 13, 5, 11 } },
|
||||
// left/right/center/lfe(center)/back left/back right
|
||||
{ { 0, 8, 6, 6, 4, 12 }, { 1, 7, 13, 13, 5, 11 } },
|
||||
// left/right/center/lfe(center)/back center(special)/side left/side right
|
||||
{ { 0, 8, 6, 6, -1, 2, 10 }, { 1, 7, 13, 13, -1, 3, 9 } },
|
||||
// left/right/center/lfe(center)/back left/back right/side left/side right
|
||||
{ { 0, 8, 6, 6, 4, 12, 2, 10 }, { 1, 7, 13, 13, 5, 11, 3, 9 } }
|
||||
static const uint32_t max_speaker_index = 10;
|
||||
|
||||
static const int8_t speakers_to_hesuvi_7[11][2] = {
|
||||
// front left
|
||||
{ 0, 1 },
|
||||
// front right
|
||||
{ 1, 0 },
|
||||
// front center
|
||||
{ 6, 6 },
|
||||
// lfe
|
||||
{ 6, 6 },
|
||||
// back left
|
||||
{ 4, 5 },
|
||||
// back right
|
||||
{ 5, 4 },
|
||||
// front center left
|
||||
{ speaker_not_present, speaker_not_present },
|
||||
// front center right
|
||||
{ speaker_not_present, speaker_not_present },
|
||||
// back center
|
||||
{ speaker_is_back_center, speaker_is_back_center },
|
||||
// side left
|
||||
{ 2, 3 },
|
||||
// side right
|
||||
{ 3, 2 }
|
||||
};
|
||||
|
||||
static const int8_t speakers_to_hesuvi_14[11][2] = {
|
||||
// front left
|
||||
{ 0, 1 },
|
||||
// front right
|
||||
{ 8, 7 },
|
||||
// front center
|
||||
{ 6, 13 },
|
||||
// lfe
|
||||
{ 6, 13 },
|
||||
// back left
|
||||
{ 4, 5 },
|
||||
// back right
|
||||
{ 12, 11 },
|
||||
// front center left
|
||||
{ speaker_not_present, speaker_not_present },
|
||||
// front center right
|
||||
{ speaker_not_present, speaker_not_present },
|
||||
// back center
|
||||
{ speaker_is_back_center, speaker_is_back_center },
|
||||
// side left
|
||||
{ 2, 3 },
|
||||
// side right
|
||||
{ 10, 9 }
|
||||
};
|
||||
|
||||
+ (BOOL)validateImpulseFile:(NSURL *)url {
|
||||
|
|
@ -99,7 +117,7 @@ static const int8_t speakers_to_hesuvi_14[8][2][8] = {
|
|||
return YES;
|
||||
}
|
||||
|
||||
- (id)initWithImpulseFile:(NSURL *)url forSampleRate:(double)sampleRate withInputChannels:(size_t)channels {
|
||||
- (id)initWithImpulseFile:(NSURL *)url forSampleRate:(double)sampleRate withInputChannels:(size_t)channels withConfig:(uint32_t)config {
|
||||
self = [super init];
|
||||
|
||||
if(self) {
|
||||
|
|
@ -246,7 +264,7 @@ static const int8_t speakers_to_hesuvi_14[8][2][8] = {
|
|||
}
|
||||
fftSize = 2 << pow;
|
||||
|
||||
float *deinterleavedImpulseBuffer = (float *)_mm_malloc(fftSize * sizeof(float) * impulseChannels, 16);
|
||||
float *deinterleavedImpulseBuffer = (float *)_mm_malloc(fftSize * sizeof(float) * (impulseChannels + 1), 16);
|
||||
if(!deinterleavedImpulseBuffer) {
|
||||
free(impulseBuffer);
|
||||
return nil;
|
||||
|
|
@ -259,6 +277,9 @@ static const int8_t speakers_to_hesuvi_14[8][2][8] = {
|
|||
|
||||
free(impulseBuffer);
|
||||
|
||||
// Null impulse
|
||||
vDSP_vclr(deinterleavedImpulseBuffer + impulseChannels * fftSize, 1, fftSize);
|
||||
|
||||
paddedBufferSize = fftSize;
|
||||
fftSizeOver2 = (fftSize + 1) / 2;
|
||||
log2n = log2f(fftSize);
|
||||
|
|
@ -317,18 +338,25 @@ static const int8_t speakers_to_hesuvi_14[8][2][8] = {
|
|||
return nil;
|
||||
}
|
||||
|
||||
int leftInChannel;
|
||||
int rightInChannel;
|
||||
uint32_t channelFlag = [AudioChunk extractChannelFlag:(uint32_t)i fromConfig:config];
|
||||
uint32_t channelIndex = [AudioChunk findChannelIndex:channelFlag];
|
||||
|
||||
int leftInChannel = speaker_not_present;
|
||||
int rightInChannel = speaker_not_present;
|
||||
|
||||
if(impulseChannels == 7) {
|
||||
leftInChannel = speakers_to_hesuvi_7[channels - 1][0][i];
|
||||
rightInChannel = speakers_to_hesuvi_7[channels - 1][1][i];
|
||||
if(channelIndex <= max_speaker_index) {
|
||||
leftInChannel = speakers_to_hesuvi_7[channelIndex][0];
|
||||
rightInChannel = speakers_to_hesuvi_7[channelIndex][1];
|
||||
}
|
||||
} else {
|
||||
leftInChannel = speakers_to_hesuvi_14[channels - 1][0][i];
|
||||
rightInChannel = speakers_to_hesuvi_14[channels - 1][1][i];
|
||||
if(channelIndex <= max_speaker_index) {
|
||||
leftInChannel = speakers_to_hesuvi_14[channelIndex][0];
|
||||
rightInChannel = speakers_to_hesuvi_14[channelIndex][1];
|
||||
}
|
||||
}
|
||||
|
||||
if(leftInChannel == -1 || rightInChannel == -1) {
|
||||
if(leftInChannel == speaker_is_back_center || rightInChannel == speaker_is_back_center) {
|
||||
float *temp;
|
||||
if(impulseChannels == 7) {
|
||||
temp = (float *)malloc(sizeof(float) * fftSize);
|
||||
|
|
@ -360,6 +388,9 @@ static const int8_t speakers_to_hesuvi_14[8][2][8] = {
|
|||
}
|
||||
|
||||
free(temp);
|
||||
} else if(leftInChannel == speaker_not_present || rightInChannel == speaker_not_present) {
|
||||
vDSP_ctoz((DSPComplex *)(deinterleavedImpulseBuffer + impulseChannels * fftSize), 2, &impulse_responses[i * 2 + 0], 1, fftSizeOver2);
|
||||
vDSP_ctoz((DSPComplex *)(deinterleavedImpulseBuffer + impulseChannels * fftSize), 2, &impulse_responses[i * 2 + 1], 1, fftSizeOver2);
|
||||
} else {
|
||||
vDSP_ctoz((DSPComplex *)(deinterleavedImpulseBuffer + leftInChannel * fftSize), 2, &impulse_responses[i * 2 + 0], 1, fftSizeOver2);
|
||||
vDSP_ctoz((DSPComplex *)(deinterleavedImpulseBuffer + rightInChannel * fftSize), 2, &impulse_responses[i * 2 + 1], 1, fftSizeOver2);
|
||||
|
|
|
|||
|
|
@ -47,6 +47,8 @@
|
|||
bytesPerFrame = ((bitsPerSample + 7) / 8) * channels;
|
||||
|
||||
nodeFormat = propertiesToASBD(properties);
|
||||
if([properties valueForKey:@"channelConfig"])
|
||||
nodeChannelConfig = [[properties valueForKey:@"channelConfig"] intValue];
|
||||
nodeLossless = [[properties valueForKey:@"encoding"] isEqualToString:@"lossless"];
|
||||
|
||||
shouldContinue = YES;
|
||||
|
|
@ -66,6 +68,8 @@
|
|||
bytesPerFrame = ((bitsPerSample + 7) / 8) * channels;
|
||||
|
||||
nodeFormat = propertiesToASBD(properties);
|
||||
if([properties valueForKey:@"channelConfig"])
|
||||
nodeChannelConfig = [[properties valueForKey:@"channelConfig"] intValue];
|
||||
nodeLossless = [[properties valueForKey:@"encoding"] isEqualToString:@"lossless"];
|
||||
|
||||
[self registerObservers];
|
||||
|
|
@ -117,14 +121,15 @@
|
|||
|
||||
while([self shouldContinue] == YES && [self endOfStream] == NO) {
|
||||
if(shouldSeek == YES) {
|
||||
ConverterNode *converter = [[[controller controller] bufferChain] converter];
|
||||
BufferChain *bufferChain = [[controller controller] bufferChain];
|
||||
ConverterNode *converter = [bufferChain converter];
|
||||
DLog(@"SEEKING! Resetting Buffer");
|
||||
|
||||
amountInBuffer = 0;
|
||||
// This resets the converter's buffer
|
||||
[self resetBuffer];
|
||||
[converter resetBuffer];
|
||||
[converter inputFormatDidChange:[[[controller controller] bufferChain] inputFormat]];
|
||||
[converter inputFormatDidChange:[bufferChain inputFormat] inputConfig:[bufferChain inputConfig]];
|
||||
|
||||
DLog(@"Reset buffer!");
|
||||
|
||||
|
|
|
|||
|
|
@ -29,6 +29,7 @@
|
|||
BOOL initialBufferFilled;
|
||||
|
||||
AudioStreamBasicDescription nodeFormat;
|
||||
uint32_t nodeChannelConfig;
|
||||
BOOL nodeLossless;
|
||||
}
|
||||
- (id)initWithController:(id)c previous:(id)p;
|
||||
|
|
@ -54,6 +55,7 @@
|
|||
- (void)resetBuffer; // WARNING! DANGER WILL ROBINSON!
|
||||
|
||||
- (AudioStreamBasicDescription)nodeFormat;
|
||||
- (uint32_t)nodeChannelConfig;
|
||||
- (BOOL)nodeLossless;
|
||||
|
||||
- (Semaphore *)semaphore;
|
||||
|
|
|
|||
|
|
@ -27,6 +27,7 @@
|
|||
endOfStream = NO;
|
||||
shouldContinue = YES;
|
||||
|
||||
nodeChannelConfig = 0;
|
||||
nodeLossless = NO;
|
||||
|
||||
[self setPreviousNode:p];
|
||||
|
|
@ -39,6 +40,10 @@
|
|||
return nodeFormat;
|
||||
}
|
||||
|
||||
- (uint32_t)nodeChannelConfig {
|
||||
return nodeChannelConfig;
|
||||
}
|
||||
|
||||
- (BOOL)nodeLossless {
|
||||
return nodeLossless;
|
||||
}
|
||||
|
|
@ -48,6 +53,9 @@
|
|||
|
||||
AudioChunk *chunk = [[AudioChunk alloc] init];
|
||||
[chunk setFormat:nodeFormat];
|
||||
if(nodeChannelConfig) {
|
||||
[chunk setChannelConfig:nodeChannelConfig];
|
||||
}
|
||||
[chunk setLossless:nodeLossless];
|
||||
[chunk assignSamples:ptr frameCount:amount / nodeFormat.mBytesPerPacket];
|
||||
|
||||
|
|
|
|||
|
|
@ -17,6 +17,7 @@
|
|||
|
||||
@interface OutputNode : Node {
|
||||
AudioStreamBasicDescription format;
|
||||
uint32_t config;
|
||||
|
||||
double amountPlayed;
|
||||
double sampleRatio;
|
||||
|
|
@ -48,8 +49,9 @@
|
|||
|
||||
- (AudioChunk *)readChunk:(size_t)amount;
|
||||
|
||||
- (void)setFormat:(AudioStreamBasicDescription *)f;
|
||||
- (void)setFormat:(AudioStreamBasicDescription *)f channelConfig:(uint32_t)channelConfig;
|
||||
- (AudioStreamBasicDescription)format;
|
||||
- (uint32_t)config;
|
||||
|
||||
- (void)setVolume:(double)v;
|
||||
|
||||
|
|
|
|||
|
|
@ -91,8 +91,13 @@
|
|||
return format;
|
||||
}
|
||||
|
||||
- (void)setFormat:(AudioStreamBasicDescription *)f {
|
||||
- (uint32_t)config {
|
||||
return config;
|
||||
}
|
||||
|
||||
- (void)setFormat:(AudioStreamBasicDescription *)f channelConfig:(uint32_t)channelConfig {
|
||||
format = *f;
|
||||
config = channelConfig;
|
||||
// Calculate a ratio and add to double(seconds) instead, as format may change
|
||||
// double oldSampleRatio = sampleRatio;
|
||||
sampleRatio = 1.0 / (format.mSampleRate * format.mBytesPerPacket);
|
||||
|
|
@ -110,8 +115,9 @@
|
|||
if (oldSampleRatio)
|
||||
amountPlayed += oldSampleRatio * [[converter buffer] bufferedLength];
|
||||
#endif
|
||||
[converter setOutputFormat:format];
|
||||
[converter inputFormatDidChange:[bufferChain inputFormat]];
|
||||
[converter setOutputFormat:format
|
||||
outputConfig:channelConfig];
|
||||
[converter inputFormatDidChange:[bufferChain inputFormat] inputConfig:[bufferChain inputConfig]];
|
||||
}
|
||||
}
|
||||
}
|
||||
|
|
|
|||
|
|
@ -22,5 +22,6 @@
|
|||
}
|
||||
|
||||
- (void)setFormat:(AudioStreamBasicDescription)format;
|
||||
- (void)setChannelConfig:(uint32_t)config;
|
||||
|
||||
@end
|
||||
|
|
|
|||
|
|
@ -42,4 +42,8 @@
|
|||
nodeFormat = format;
|
||||
}
|
||||
|
||||
- (void)setChannelConfig:(uint32_t)config {
|
||||
nodeChannelConfig = config;
|
||||
}
|
||||
|
||||
@end
|
||||
|
|
|
|||
|
|
@ -60,6 +60,9 @@
|
|||
AudioStreamBasicDescription deviceFormat; // info about the default device
|
||||
AudioStreamBasicDescription streamFormat; // stream format last seen in render callback
|
||||
|
||||
uint32_t deviceChannelConfig;
|
||||
uint32_t streamChannelConfig;
|
||||
|
||||
AUAudioUnit *_au;
|
||||
size_t _bufferSize;
|
||||
|
||||
|
|
|
|||
|
|
@ -76,12 +76,14 @@ static OSStatus renderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioAct
|
|||
|
||||
size_t frameCount = [chunk frameCount];
|
||||
AudioStreamBasicDescription format = [chunk format];
|
||||
uint32_t config = [chunk channelConfig];
|
||||
|
||||
if(frameCount) {
|
||||
if(!_self->streamFormatStarted || memcmp(&_self->streamFormat, &format, sizeof(format)) != 0) {
|
||||
if(!_self->streamFormatStarted || config != _self->streamChannelConfig || memcmp(&_self->streamFormat, &format, sizeof(format)) != 0) {
|
||||
_self->streamFormat = format;
|
||||
_self->streamChannelConfig = config;
|
||||
_self->streamFormatStarted = YES;
|
||||
_self->downmixer = [[DownmixProcessor alloc] initWithInputFormat:format andOutputFormat:_self->deviceFormat];
|
||||
_self->downmixer = [[DownmixProcessor alloc] initWithInputFormat:format inputConfig:config andOutputFormat:_self->deviceFormat outputConfig:_self->deviceChannelConfig];
|
||||
}
|
||||
|
||||
double chunkDuration = [chunk duration];
|
||||
|
|
@ -104,11 +106,12 @@ static OSStatus renderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioAct
|
|||
chunk = [[_self->outputController buffer] removeSamples:((amountToRead - amountRead) / bytesPerPacket)];
|
||||
frameCount = [chunk frameCount];
|
||||
format = [chunk format];
|
||||
config = [chunk channelConfig];
|
||||
if(frameCount) {
|
||||
if(!_self->streamFormatStarted || memcmp(&_self->streamFormat, &format, sizeof(format)) != 0) {
|
||||
if(!_self->streamFormatStarted || config != _self->streamChannelConfig || memcmp(&_self->streamFormat, &format, sizeof(format)) != 0) {
|
||||
_self->streamFormat = format;
|
||||
_self->streamFormatStarted = YES;
|
||||
_self->downmixer = [[DownmixProcessor alloc] initWithInputFormat:format andOutputFormat:_self->deviceFormat];
|
||||
_self->downmixer = [[DownmixProcessor alloc] initWithInputFormat:format inputConfig:config andOutputFormat:_self->deviceFormat outputConfig:_self->deviceChannelConfig];
|
||||
}
|
||||
atomic_fetch_add(&_self->bytesRendered, frameCount * bytesPerPacket);
|
||||
double chunkDuration = [chunk duration];
|
||||
|
|
@ -485,27 +488,35 @@ default_device_changed(AudioObjectID inObjectID, UInt32 inNumberAddresses, const
|
|||
switch(deviceFormat.mChannelsPerFrame) {
|
||||
case 1:
|
||||
tag = kAudioChannelLayoutTag_Mono;
|
||||
deviceChannelConfig = AudioConfigMono;
|
||||
break;
|
||||
case 2:
|
||||
tag = kAudioChannelLayoutTag_Stereo;
|
||||
deviceChannelConfig = AudioConfigStereo;
|
||||
break;
|
||||
case 3:
|
||||
tag = kAudioChannelLayoutTag_DVD_4;
|
||||
deviceChannelConfig = AudioConfig3Point0;
|
||||
break;
|
||||
case 4:
|
||||
tag = kAudioChannelLayoutTag_Quadraphonic;
|
||||
deviceChannelConfig = AudioConfig4Point0;
|
||||
break;
|
||||
case 5:
|
||||
tag = kAudioChannelLayoutTag_MPEG_5_0_A;
|
||||
deviceChannelConfig = AudioConfig5Point0;
|
||||
break;
|
||||
case 6:
|
||||
tag = kAudioChannelLayoutTag_MPEG_5_1_A;
|
||||
deviceChannelConfig = AudioConfig5Point1;
|
||||
break;
|
||||
case 7:
|
||||
tag = kAudioChannelLayoutTag_MPEG_6_1_A;
|
||||
deviceChannelConfig = AudioConfig6Point1;
|
||||
break;
|
||||
case 8:
|
||||
tag = kAudioChannelLayoutTag_MPEG_7_1_A;
|
||||
deviceChannelConfig = AudioConfig7Point1;
|
||||
break;
|
||||
}
|
||||
|
||||
|
|
@ -514,7 +525,7 @@ default_device_changed(AudioObjectID inObjectID, UInt32 inNumberAddresses, const
|
|||
if(err != nil)
|
||||
return NO;
|
||||
|
||||
[outputController setFormat:&deviceFormat];
|
||||
[outputController setFormat:&deviceFormat channelConfig:deviceChannelConfig];
|
||||
|
||||
AudioStreamBasicDescription asbd = deviceFormat;
|
||||
|
||||
|
|
|
|||
Loading…
Reference in New Issue