// // FSurroundFilter.m // CogAudio Framework // // Created by Christopher Snowhill on 7/9/22. // #import "FSurroundFilter.h" #import "freesurround_decoder.h" #import "AudioChunk.h" #import #import #import struct freesurround_params { // the user-configurable parameters float center_image, shift, depth, circular_wrap, focus, front_sep, rear_sep, bass_lo, bass_hi; bool use_lfe; channel_setup channels_fs; // FreeSurround channel setup std::vector chanmap; // FreeSurround -> WFX channel index translation (derived data for faster lookup) // construct with defaults freesurround_params() : center_image(0.7), shift(0), depth(1), circular_wrap(90), focus(0), front_sep(1), rear_sep(1), bass_lo(40), bass_hi(90), use_lfe(false) { set_channels_fs(cs_5point1); } // compute the WFX version of the channel setup code unsigned channel_count() { return (unsigned)chanmap.size(); } unsigned channels_wfx() { unsigned res = 0; for(unsigned i = 0; i < chanmap.size(); res |= chanmap[i++]) {}; return res; } // assign a channel setup & recompute derived data void set_channels_fs(channel_setup setup) { channels_fs = setup; chanmap.clear(); // Note: Because WFX does not define a few of the more exotic channels (side front left&right, side rear left&right, back center left&right), // the side front/back channel pairs (both left and right sides, resp.) are mapped here onto foobar's top front/back channel pairs and the // back (off-)center left/right channels are mapped onto foobar's top front center and top back center, respectively. // Therefore, these speakers should be connected to those outputs instead. std::map fs2wfx; fs2wfx[ci_front_left] = AudioChannelFrontLeft; fs2wfx[ci_front_center_left] = AudioChannelFrontCenterLeft; fs2wfx[ci_front_center] = AudioChannelFrontCenter; fs2wfx[ci_front_center_right] = AudioChannelFrontCenterRight; fs2wfx[ci_front_right] = AudioChannelFrontRight; fs2wfx[ci_side_front_left] = AudioChannelFrontLeft; fs2wfx[ci_side_front_right] = AudioChannelTopFrontRight; fs2wfx[ci_side_center_left] = AudioChannelSideLeft; fs2wfx[ci_side_center_right] = AudioChannelSideRight; fs2wfx[ci_side_back_left] = AudioChannelTopBackLeft; fs2wfx[ci_side_back_right] = AudioChannelTopBackRight; fs2wfx[ci_back_left] = AudioChannelBackLeft; fs2wfx[ci_back_center_left] = AudioChannelTopFrontCenter; fs2wfx[ci_back_center] = AudioChannelBackCenter; fs2wfx[ci_back_center_right] = AudioChannelTopBackCenter; fs2wfx[ci_back_right] = AudioChannelBackRight; fs2wfx[ci_lfe] = AudioChannelLFE; for(unsigned i = 0; i < freesurround_decoder::num_channels(channels_fs); i++) chanmap.push_back(fs2wfx[freesurround_decoder::channel_at(channels_fs, i)]); } }; @implementation FSurroundFilter - (id)initWithSampleRate:(double)srate { self = [super init]; if(!self) return nil; self->srate = srate; freesurround_params *_params = new freesurround_params; params = (void *)_params; freesurround_decoder *_decoder = new freesurround_decoder(cs_5point1, 4096); decoder = (void *)_decoder; _decoder->circular_wrap(_params->circular_wrap); _decoder->shift(_params->shift); _decoder->depth(_params->depth); _decoder->focus(_params->focus); _decoder->center_image(_params->center_image); _decoder->front_separation(_params->front_sep); _decoder->rear_separation(_params->rear_sep); _decoder->bass_redirection(_params->use_lfe); _decoder->low_cutoff(_params->bass_lo / (srate / 2.0)); _decoder->high_cutoff(_params->bass_hi / (srate / 2.0)); channelCount = _params->channel_count(); channelConfig = _params->channels_wfx(); return self; } - (void)dealloc { if(decoder) { freesurround_decoder *_decoder = (freesurround_decoder *)decoder; delete _decoder; } if(params) { freesurround_params *_params = (freesurround_params *)params; delete _params; } } - (uint32_t)channelCount { return channelCount; } - (uint32_t)channelConfig { return channelConfig; } - (double)srate { return srate; } - (void)process:(const float *)samplesIn output:(float *)samplesOut count:(uint32_t)count { freesurround_params *_params = (freesurround_params *)params; freesurround_decoder *_decoder = (freesurround_decoder *)decoder; uint32_t zeroCount = 0; if(count > 4096) { zeroCount = count - 4096; count = 4096; } if(count < 4096) { cblas_scopy(count * 2, samplesIn, 1, &tempBuffer[0], 1); vDSP_vclr(&tempBuffer[count * 2], 1, (4096 - count) * 2); samplesIn = &tempBuffer[0]; } float *src = _decoder->decode(samplesIn); for(unsigned c = 0, num = channelCount; c < num; c++) { unsigned idx = [AudioChunk channelIndexFromConfig:channelConfig forFlag:_params->chanmap[c]]; cblas_scopy(count, src + c, num, samplesOut + idx, num); if(zeroCount) { vDSP_vclr(samplesOut + idx + count, num, zeroCount); } } } @end