mirror of https://github.com/axmolengine/axmol.git
2051 lines
75 KiB
C++
2051 lines
75 KiB
C++
/**
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* OpenAL cross platform audio library
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* Copyright (C) 1999-2007 by authors.
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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* Or go to http://www.gnu.org/copyleft/lgpl.html
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*/
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#include "config.h"
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#include "alu.h"
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#include <algorithm>
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#include <array>
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#include <atomic>
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#include <cassert>
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#include <chrono>
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#include <climits>
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#include <cstdarg>
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#include <cstdio>
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#include <cstdlib>
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#include <functional>
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#include <iterator>
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#include <limits>
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#include <memory>
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#include <new>
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#include <stdint.h>
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#include <utility>
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#include "almalloc.h"
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#include "alnumeric.h"
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#include "alspan.h"
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#include "alstring.h"
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#include "atomic.h"
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#include "core/ambidefs.h"
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#include "core/async_event.h"
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#include "core/bformatdec.h"
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#include "core/bs2b.h"
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#include "core/bsinc_defs.h"
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#include "core/bsinc_tables.h"
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#include "core/bufferline.h"
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#include "core/buffer_storage.h"
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#include "core/context.h"
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#include "core/cpu_caps.h"
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#include "core/devformat.h"
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#include "core/device.h"
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#include "core/filters/biquad.h"
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#include "core/filters/nfc.h"
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#include "core/fpu_ctrl.h"
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#include "core/hrtf.h"
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#include "core/mastering.h"
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#include "core/mixer.h"
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#include "core/mixer/defs.h"
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#include "core/mixer/hrtfdefs.h"
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#include "core/resampler_limits.h"
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#include "core/uhjfilter.h"
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#include "core/voice.h"
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#include "core/voice_change.h"
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#include "effects/base.h"
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#include "effectslot.h"
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#include "intrusive_ptr.h"
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#include "math_defs.h"
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#include "opthelpers.h"
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#include "ringbuffer.h"
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#include "strutils.h"
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#include "threads.h"
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#include "vecmat.h"
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#include "vector.h"
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struct CTag;
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#ifdef HAVE_SSE
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struct SSETag;
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#endif
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#ifdef HAVE_SSE2
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struct SSE2Tag;
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#endif
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#ifdef HAVE_SSE4_1
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struct SSE4Tag;
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#endif
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#ifdef HAVE_NEON
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struct NEONTag;
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#endif
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struct PointTag;
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struct LerpTag;
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struct CubicTag;
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struct BSincTag;
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struct FastBSincTag;
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static_assert(!(MaxResamplerPadding&1), "MaxResamplerPadding is not a multiple of two");
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namespace {
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using uint = unsigned int;
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constexpr uint MaxPitch{10};
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static_assert((BufferLineSize-1)/MaxPitch > 0, "MaxPitch is too large for BufferLineSize!");
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static_assert((INT_MAX>>MixerFracBits)/MaxPitch > BufferLineSize,
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"MaxPitch and/or BufferLineSize are too large for MixerFracBits!");
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using namespace std::placeholders;
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float InitConeScale()
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{
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float ret{1.0f};
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if(auto optval = al::getenv("__ALSOFT_HALF_ANGLE_CONES"))
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{
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if(al::strcasecmp(optval->c_str(), "true") == 0
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|| strtol(optval->c_str(), nullptr, 0) == 1)
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ret *= 0.5f;
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}
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return ret;
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}
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float InitZScale()
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{
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float ret{1.0f};
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if(auto optval = al::getenv("__ALSOFT_REVERSE_Z"))
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{
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if(al::strcasecmp(optval->c_str(), "true") == 0
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|| strtol(optval->c_str(), nullptr, 0) == 1)
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ret *= -1.0f;
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}
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return ret;
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}
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} // namespace
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/* Cone scalar */
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const float ConeScale{InitConeScale()};
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/* Localized Z scalar for mono sources */
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const float ZScale{InitZScale()};
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namespace {
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struct ChanMap {
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Channel channel;
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float angle;
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float elevation;
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};
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using HrtfDirectMixerFunc = void(*)(const FloatBufferSpan LeftOut, const FloatBufferSpan RightOut,
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const al::span<const FloatBufferLine> InSamples, float2 *AccumSamples, float *TempBuf,
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HrtfChannelState *ChanState, const size_t IrSize, const size_t BufferSize);
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HrtfDirectMixerFunc MixDirectHrtf{MixDirectHrtf_<CTag>};
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inline HrtfDirectMixerFunc SelectHrtfMixer(void)
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{
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#ifdef HAVE_NEON
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if((CPUCapFlags&CPU_CAP_NEON))
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return MixDirectHrtf_<NEONTag>;
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#endif
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#ifdef HAVE_SSE
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if((CPUCapFlags&CPU_CAP_SSE))
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return MixDirectHrtf_<SSETag>;
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#endif
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return MixDirectHrtf_<CTag>;
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}
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inline void BsincPrepare(const uint increment, BsincState *state, const BSincTable *table)
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{
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size_t si{BSincScaleCount - 1};
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float sf{0.0f};
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if(increment > MixerFracOne)
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{
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sf = MixerFracOne/static_cast<float>(increment) - table->scaleBase;
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sf = maxf(0.0f, BSincScaleCount*sf*table->scaleRange - 1.0f);
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si = float2uint(sf);
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/* The interpolation factor is fit to this diagonally-symmetric curve
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* to reduce the transition ripple caused by interpolating different
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* scales of the sinc function.
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*/
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sf = 1.0f - std::cos(std::asin(sf - static_cast<float>(si)));
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}
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state->sf = sf;
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state->m = table->m[si];
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state->l = (state->m/2) - 1;
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state->filter = table->Tab + table->filterOffset[si];
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}
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inline ResamplerFunc SelectResampler(Resampler resampler, uint increment)
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{
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switch(resampler)
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{
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case Resampler::Point:
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return Resample_<PointTag,CTag>;
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case Resampler::Linear:
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#ifdef HAVE_NEON
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if((CPUCapFlags&CPU_CAP_NEON))
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return Resample_<LerpTag,NEONTag>;
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#endif
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#ifdef HAVE_SSE4_1
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if((CPUCapFlags&CPU_CAP_SSE4_1))
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return Resample_<LerpTag,SSE4Tag>;
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#endif
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#ifdef HAVE_SSE2
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if((CPUCapFlags&CPU_CAP_SSE2))
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return Resample_<LerpTag,SSE2Tag>;
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#endif
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return Resample_<LerpTag,CTag>;
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case Resampler::Cubic:
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return Resample_<CubicTag,CTag>;
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case Resampler::BSinc12:
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case Resampler::BSinc24:
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if(increment <= MixerFracOne)
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{
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/* fall-through */
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case Resampler::FastBSinc12:
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case Resampler::FastBSinc24:
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#ifdef HAVE_NEON
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if((CPUCapFlags&CPU_CAP_NEON))
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return Resample_<FastBSincTag,NEONTag>;
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#endif
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#ifdef HAVE_SSE
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if((CPUCapFlags&CPU_CAP_SSE))
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return Resample_<FastBSincTag,SSETag>;
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#endif
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return Resample_<FastBSincTag,CTag>;
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}
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#ifdef HAVE_NEON
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if((CPUCapFlags&CPU_CAP_NEON))
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return Resample_<BSincTag,NEONTag>;
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#endif
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#ifdef HAVE_SSE
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if((CPUCapFlags&CPU_CAP_SSE))
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return Resample_<BSincTag,SSETag>;
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#endif
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return Resample_<BSincTag,CTag>;
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}
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return Resample_<PointTag,CTag>;
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}
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} // namespace
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void aluInit(void)
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{
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MixDirectHrtf = SelectHrtfMixer();
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}
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ResamplerFunc PrepareResampler(Resampler resampler, uint increment, InterpState *state)
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{
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switch(resampler)
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{
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case Resampler::Point:
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case Resampler::Linear:
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case Resampler::Cubic:
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break;
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case Resampler::FastBSinc12:
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case Resampler::BSinc12:
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BsincPrepare(increment, &state->bsinc, &bsinc12);
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break;
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case Resampler::FastBSinc24:
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case Resampler::BSinc24:
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BsincPrepare(increment, &state->bsinc, &bsinc24);
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break;
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}
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return SelectResampler(resampler, increment);
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}
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void DeviceBase::ProcessHrtf(const size_t SamplesToDo)
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{
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/* HRTF is stereo output only. */
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const uint lidx{RealOut.ChannelIndex[FrontLeft]};
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const uint ridx{RealOut.ChannelIndex[FrontRight]};
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MixDirectHrtf(RealOut.Buffer[lidx], RealOut.Buffer[ridx], Dry.Buffer, HrtfAccumData,
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mHrtfState->mTemp.data(), mHrtfState->mChannels.data(), mHrtfState->mIrSize, SamplesToDo);
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}
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void DeviceBase::ProcessAmbiDec(const size_t SamplesToDo)
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{
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AmbiDecoder->process(RealOut.Buffer, Dry.Buffer.data(), SamplesToDo);
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}
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void DeviceBase::ProcessAmbiDecStablized(const size_t SamplesToDo)
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{
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/* Decode with front image stablization. */
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const uint lidx{RealOut.ChannelIndex[FrontLeft]};
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const uint ridx{RealOut.ChannelIndex[FrontRight]};
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const uint cidx{RealOut.ChannelIndex[FrontCenter]};
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AmbiDecoder->processStablize(RealOut.Buffer, Dry.Buffer.data(), lidx, ridx, cidx,
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SamplesToDo);
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}
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void DeviceBase::ProcessUhj(const size_t SamplesToDo)
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{
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/* UHJ is stereo output only. */
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const uint lidx{RealOut.ChannelIndex[FrontLeft]};
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const uint ridx{RealOut.ChannelIndex[FrontRight]};
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/* Encode to stereo-compatible 2-channel UHJ output. */
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mUhjEncoder->encode(RealOut.Buffer[lidx], RealOut.Buffer[ridx], Dry.Buffer.data(),
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SamplesToDo);
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}
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void DeviceBase::ProcessBs2b(const size_t SamplesToDo)
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{
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/* First, decode the ambisonic mix to the "real" output. */
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AmbiDecoder->process(RealOut.Buffer, Dry.Buffer.data(), SamplesToDo);
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/* BS2B is stereo output only. */
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const uint lidx{RealOut.ChannelIndex[FrontLeft]};
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const uint ridx{RealOut.ChannelIndex[FrontRight]};
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/* Now apply the BS2B binaural/crossfeed filter. */
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bs2b_cross_feed(Bs2b.get(), RealOut.Buffer[lidx].data(), RealOut.Buffer[ridx].data(),
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SamplesToDo);
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}
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namespace {
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/* This RNG method was created based on the math found in opusdec. It's quick,
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* and starting with a seed value of 22222, is suitable for generating
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* whitenoise.
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*/
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inline uint dither_rng(uint *seed) noexcept
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{
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*seed = (*seed * 96314165) + 907633515;
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return *seed;
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}
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inline auto& GetAmbiScales(AmbiScaling scaletype) noexcept
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{
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if(scaletype == AmbiScaling::FuMa) return AmbiScale::FromFuMa();
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if(scaletype == AmbiScaling::SN3D) return AmbiScale::FromSN3D();
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return AmbiScale::FromN3D();
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}
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inline auto& GetAmbiLayout(AmbiLayout layouttype) noexcept
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{
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if(layouttype == AmbiLayout::FuMa) return AmbiIndex::FromFuMa();
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return AmbiIndex::FromACN();
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}
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inline auto& GetAmbi2DLayout(AmbiLayout layouttype) noexcept
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{
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if(layouttype == AmbiLayout::FuMa) return AmbiIndex::FromFuMa2D();
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return AmbiIndex::FromACN2D();
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}
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bool CalcContextParams(ContextBase *ctx)
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{
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ContextProps *props{ctx->mParams.ContextUpdate.exchange(nullptr, std::memory_order_acq_rel)};
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if(!props) return false;
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ctx->mParams.DopplerFactor = props->DopplerFactor;
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ctx->mParams.SpeedOfSound = props->SpeedOfSound * props->DopplerVelocity;
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ctx->mParams.SourceDistanceModel = props->SourceDistanceModel;
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ctx->mParams.mDistanceModel = props->mDistanceModel;
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AtomicReplaceHead(ctx->mFreeContextProps, props);
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return true;
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}
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bool CalcListenerParams(ContextBase *ctx)
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{
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ListenerProps *props{ctx->mParams.ListenerUpdate.exchange(nullptr,
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std::memory_order_acq_rel)};
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if(!props) return false;
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/* AT then UP */
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alu::Vector N{props->OrientAt[0], props->OrientAt[1], props->OrientAt[2], 0.0f};
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N.normalize();
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alu::Vector V{props->OrientUp[0], props->OrientUp[1], props->OrientUp[2], 0.0f};
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V.normalize();
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/* Build and normalize right-vector */
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alu::Vector U{N.cross_product(V)};
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U.normalize();
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const alu::MatrixR<double> rot{
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U[0], V[0], -N[0], 0.0,
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U[1], V[1], -N[1], 0.0,
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U[2], V[2], -N[2], 0.0,
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0.0, 0.0, 0.0, 1.0};
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const alu::VectorR<double> pos{props->Position[0],props->Position[1],props->Position[2],1.0};
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const alu::VectorR<double> vel{props->Velocity[0],props->Velocity[1],props->Velocity[2],0.0};
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const alu::Vector P{alu::cast_to<float>(rot * pos)};
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ctx->mParams.Matrix = alu::Matrix{
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U[0], V[0], -N[0], 0.0f,
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U[1], V[1], -N[1], 0.0f,
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U[2], V[2], -N[2], 0.0f,
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-P[0], -P[1], -P[2], 1.0f};
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ctx->mParams.Velocity = alu::cast_to<float>(rot * vel);
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ctx->mParams.Gain = props->Gain * ctx->mGainBoost;
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ctx->mParams.MetersPerUnit = props->MetersPerUnit;
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AtomicReplaceHead(ctx->mFreeListenerProps, props);
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return true;
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}
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bool CalcEffectSlotParams(EffectSlot *slot, EffectSlot **sorted_slots, ContextBase *context)
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{
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EffectSlotProps *props{slot->Update.exchange(nullptr, std::memory_order_acq_rel)};
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if(!props) return false;
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/* If the effect slot target changed, clear the first sorted entry to force
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* a re-sort.
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*/
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if(slot->Target != props->Target)
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*sorted_slots = nullptr;
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slot->Gain = props->Gain;
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slot->AuxSendAuto = props->AuxSendAuto;
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slot->Target = props->Target;
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slot->EffectType = props->Type;
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slot->mEffectProps = props->Props;
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if(props->Type == EffectSlotType::Reverb || props->Type == EffectSlotType::EAXReverb)
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{
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slot->RoomRolloff = props->Props.Reverb.RoomRolloffFactor;
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slot->DecayTime = props->Props.Reverb.DecayTime;
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slot->DecayLFRatio = props->Props.Reverb.DecayLFRatio;
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slot->DecayHFRatio = props->Props.Reverb.DecayHFRatio;
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slot->DecayHFLimit = props->Props.Reverb.DecayHFLimit;
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slot->AirAbsorptionGainHF = props->Props.Reverb.AirAbsorptionGainHF;
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}
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else
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{
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slot->RoomRolloff = 0.0f;
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slot->DecayTime = 0.0f;
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slot->DecayLFRatio = 0.0f;
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slot->DecayHFRatio = 0.0f;
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slot->DecayHFLimit = false;
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slot->AirAbsorptionGainHF = 1.0f;
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}
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EffectState *state{props->State.release()};
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EffectState *oldstate{slot->mEffectState};
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slot->mEffectState = state;
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/* Only release the old state if it won't get deleted, since we can't be
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* deleting/freeing anything in the mixer.
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*/
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if(!oldstate->releaseIfNoDelete())
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{
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/* Otherwise, if it would be deleted send it off with a release event. */
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RingBuffer *ring{context->mAsyncEvents.get()};
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auto evt_vec = ring->getWriteVector();
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if LIKELY(evt_vec.first.len > 0)
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{
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AsyncEvent *evt{::new(evt_vec.first.buf) AsyncEvent{EventType_ReleaseEffectState}};
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evt->u.mEffectState = oldstate;
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ring->writeAdvance(1);
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}
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else
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{
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/* If writing the event failed, the queue was probably full. Store
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* the old state in the property object where it can eventually be
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* cleaned up sometime later (not ideal, but better than blocking
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* or leaking).
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*/
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props->State.reset(oldstate);
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}
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}
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AtomicReplaceHead(context->mFreeEffectslotProps, props);
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EffectTarget output;
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if(EffectSlot *target{slot->Target})
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output = EffectTarget{&target->Wet, nullptr};
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else
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{
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DeviceBase *device{context->mDevice};
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output = EffectTarget{&device->Dry, &device->RealOut};
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}
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state->update(context, slot, &slot->mEffectProps, output);
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return true;
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}
|
|
|
|
|
|
/* Scales the given azimuth toward the side (+/- pi/2 radians) for positions in
|
|
* front.
|
|
*/
|
|
inline float ScaleAzimuthFront(float azimuth, float scale)
|
|
{
|
|
const float abs_azi{std::fabs(azimuth)};
|
|
if(!(abs_azi >= al::MathDefs<float>::Pi()*0.5f))
|
|
return std::copysign(minf(abs_azi*scale, al::MathDefs<float>::Pi()*0.5f), azimuth);
|
|
return azimuth;
|
|
}
|
|
|
|
/* Wraps the given value in radians to stay between [-pi,+pi] */
|
|
inline float WrapRadians(float r)
|
|
{
|
|
constexpr float Pi{al::MathDefs<float>::Pi()};
|
|
constexpr float Pi2{al::MathDefs<float>::Tau()};
|
|
if(r > Pi) return std::fmod(Pi+r, Pi2) - Pi;
|
|
if(r < -Pi) return Pi - std::fmod(Pi-r, Pi2);
|
|
return r;
|
|
}
|
|
|
|
/* Begin ambisonic rotation helpers.
|
|
*
|
|
* Rotating first-order B-Format just needs a straight-forward X/Y/Z rotation
|
|
* matrix. Higher orders, however, are more complicated. The method implemented
|
|
* here is a recursive algorithm (the rotation for first-order is used to help
|
|
* generate the second-order rotation, which helps generate the third-order
|
|
* rotation, etc).
|
|
*
|
|
* Adapted from
|
|
* <https://github.com/polarch/Spherical-Harmonic-Transform/blob/master/getSHrotMtx.m>,
|
|
* provided under the BSD 3-Clause license.
|
|
*
|
|
* Copyright (c) 2015, Archontis Politis
|
|
* Copyright (c) 2019, Christopher Robinson
|
|
*
|
|
* The u, v, and w coefficients used for generating higher-order rotations are
|
|
* precomputed since they're constant. The second-order coefficients are
|
|
* followed by the third-order coefficients, etc.
|
|
*/
|
|
struct RotatorCoeffs {
|
|
float u, v, w;
|
|
|
|
template<size_t N0, size_t N1>
|
|
static std::array<RotatorCoeffs,N0+N1> ConcatArrays(const std::array<RotatorCoeffs,N0> &lhs,
|
|
const std::array<RotatorCoeffs,N1> &rhs)
|
|
{
|
|
std::array<RotatorCoeffs,N0+N1> ret;
|
|
auto iter = std::copy(lhs.cbegin(), lhs.cend(), ret.begin());
|
|
std::copy(rhs.cbegin(), rhs.cend(), iter);
|
|
return ret;
|
|
}
|
|
|
|
template<int l, int num_elems=l*2+1>
|
|
static std::array<RotatorCoeffs,num_elems*num_elems> GenCoeffs()
|
|
{
|
|
std::array<RotatorCoeffs,num_elems*num_elems> ret{};
|
|
auto coeffs = ret.begin();
|
|
|
|
for(int m{-l};m <= l;++m)
|
|
{
|
|
for(int n{-l};n <= l;++n)
|
|
{
|
|
// compute u,v,w terms of Eq.8.1 (Table I)
|
|
const bool d{m == 0}; // the delta function d_m0
|
|
const float denom{static_cast<float>((std::abs(n) == l) ?
|
|
(2*l) * (2*l - 1) : (l*l - n*n))};
|
|
|
|
const int abs_m{std::abs(m)};
|
|
coeffs->u = std::sqrt(static_cast<float>(l*l - m*m)/denom);
|
|
coeffs->v = std::sqrt(static_cast<float>(l+abs_m-1) * static_cast<float>(l+abs_m) /
|
|
denom) * (1.0f+d) * (1.0f - 2.0f*d) * 0.5f;
|
|
coeffs->w = std::sqrt(static_cast<float>(l-abs_m-1) * static_cast<float>(l-abs_m) /
|
|
denom) * (1.0f-d) * -0.5f;
|
|
++coeffs;
|
|
}
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
};
|
|
const auto RotatorCoeffArray = RotatorCoeffs::ConcatArrays(RotatorCoeffs::GenCoeffs<2>(),
|
|
RotatorCoeffs::GenCoeffs<3>());
|
|
|
|
/**
|
|
* Given the matrix, pre-filled with the (zeroth- and) first-order rotation
|
|
* coefficients, this fills in the coefficients for the higher orders up to and
|
|
* including the given order. The matrix is in ACN layout.
|
|
*/
|
|
void AmbiRotator(std::array<std::array<float,MaxAmbiChannels>,MaxAmbiChannels> &matrix,
|
|
const int order)
|
|
{
|
|
/* Don't do anything for < 2nd order. */
|
|
if(order < 2) return;
|
|
|
|
auto P = [](const int i, const int l, const int a, const int n, const size_t last_band,
|
|
const std::array<std::array<float,MaxAmbiChannels>,MaxAmbiChannels> &R)
|
|
{
|
|
const float ri1{ R[static_cast<uint>(i+2)][ 1+2]};
|
|
const float rim1{R[static_cast<uint>(i+2)][-1+2]};
|
|
const float ri0{ R[static_cast<uint>(i+2)][ 0+2]};
|
|
|
|
auto vec = R[static_cast<uint>(a+l-1) + last_band].cbegin() + last_band;
|
|
if(n == -l)
|
|
return ri1*vec[0] + rim1*vec[static_cast<uint>(l-1)*size_t{2}];
|
|
if(n == l)
|
|
return ri1*vec[static_cast<uint>(l-1)*size_t{2}] - rim1*vec[0];
|
|
return ri0*vec[static_cast<uint>(n+l-1)];
|
|
};
|
|
|
|
auto U = [P](const int l, const int m, const int n, const size_t last_band,
|
|
const std::array<std::array<float,MaxAmbiChannels>,MaxAmbiChannels> &R)
|
|
{
|
|
return P(0, l, m, n, last_band, R);
|
|
};
|
|
auto V = [P](const int l, const int m, const int n, const size_t last_band,
|
|
const std::array<std::array<float,MaxAmbiChannels>,MaxAmbiChannels> &R)
|
|
{
|
|
if(m > 0)
|
|
{
|
|
const bool d{m == 1};
|
|
const float p0{P( 1, l, m-1, n, last_band, R)};
|
|
const float p1{P(-1, l, -m+1, n, last_band, R)};
|
|
return d ? p0*std::sqrt(2.0f) : (p0 - p1);
|
|
}
|
|
const bool d{m == -1};
|
|
const float p0{P( 1, l, m+1, n, last_band, R)};
|
|
const float p1{P(-1, l, -m-1, n, last_band, R)};
|
|
return d ? p1*std::sqrt(2.0f) : (p0 + p1);
|
|
};
|
|
auto W = [P](const int l, const int m, const int n, const size_t last_band,
|
|
const std::array<std::array<float,MaxAmbiChannels>,MaxAmbiChannels> &R)
|
|
{
|
|
assert(m != 0);
|
|
if(m > 0)
|
|
{
|
|
const float p0{P( 1, l, m+1, n, last_band, R)};
|
|
const float p1{P(-1, l, -m-1, n, last_band, R)};
|
|
return p0 + p1;
|
|
}
|
|
const float p0{P( 1, l, m-1, n, last_band, R)};
|
|
const float p1{P(-1, l, -m+1, n, last_band, R)};
|
|
return p0 - p1;
|
|
};
|
|
|
|
// compute rotation matrix of each subsequent band recursively
|
|
auto coeffs = RotatorCoeffArray.cbegin();
|
|
size_t band_idx{4}, last_band{1};
|
|
for(int l{2};l <= order;++l)
|
|
{
|
|
size_t y{band_idx};
|
|
for(int m{-l};m <= l;++m,++y)
|
|
{
|
|
size_t x{band_idx};
|
|
for(int n{-l};n <= l;++n,++x)
|
|
{
|
|
float r{0.0f};
|
|
|
|
// computes Eq.8.1
|
|
const float u{coeffs->u};
|
|
if(u != 0.0f) r += u * U(l, m, n, last_band, matrix);
|
|
const float v{coeffs->v};
|
|
if(v != 0.0f) r += v * V(l, m, n, last_band, matrix);
|
|
const float w{coeffs->w};
|
|
if(w != 0.0f) r += w * W(l, m, n, last_band, matrix);
|
|
|
|
matrix[y][x] = r;
|
|
++coeffs;
|
|
}
|
|
}
|
|
last_band = band_idx;
|
|
band_idx += static_cast<uint>(l)*size_t{2} + 1;
|
|
}
|
|
}
|
|
/* End ambisonic rotation helpers. */
|
|
|
|
|
|
struct GainTriplet { float Base, HF, LF; };
|
|
|
|
void CalcPanningAndFilters(Voice *voice, const float xpos, const float ypos, const float zpos,
|
|
const float Distance, const float Spread, const GainTriplet &DryGain,
|
|
const al::span<const GainTriplet,MAX_SENDS> WetGain, EffectSlot *(&SendSlots)[MAX_SENDS],
|
|
const VoiceProps *props, const ContextParams &Context, const DeviceBase *Device)
|
|
{
|
|
static const ChanMap MonoMap[1]{
|
|
{ FrontCenter, 0.0f, 0.0f }
|
|
}, RearMap[2]{
|
|
{ BackLeft, Deg2Rad(-150.0f), Deg2Rad(0.0f) },
|
|
{ BackRight, Deg2Rad( 150.0f), Deg2Rad(0.0f) }
|
|
}, QuadMap[4]{
|
|
{ FrontLeft, Deg2Rad( -45.0f), Deg2Rad(0.0f) },
|
|
{ FrontRight, Deg2Rad( 45.0f), Deg2Rad(0.0f) },
|
|
{ BackLeft, Deg2Rad(-135.0f), Deg2Rad(0.0f) },
|
|
{ BackRight, Deg2Rad( 135.0f), Deg2Rad(0.0f) }
|
|
}, X51Map[6]{
|
|
{ FrontLeft, Deg2Rad( -30.0f), Deg2Rad(0.0f) },
|
|
{ FrontRight, Deg2Rad( 30.0f), Deg2Rad(0.0f) },
|
|
{ FrontCenter, Deg2Rad( 0.0f), Deg2Rad(0.0f) },
|
|
{ LFE, 0.0f, 0.0f },
|
|
{ SideLeft, Deg2Rad(-110.0f), Deg2Rad(0.0f) },
|
|
{ SideRight, Deg2Rad( 110.0f), Deg2Rad(0.0f) }
|
|
}, X61Map[7]{
|
|
{ FrontLeft, Deg2Rad(-30.0f), Deg2Rad(0.0f) },
|
|
{ FrontRight, Deg2Rad( 30.0f), Deg2Rad(0.0f) },
|
|
{ FrontCenter, Deg2Rad( 0.0f), Deg2Rad(0.0f) },
|
|
{ LFE, 0.0f, 0.0f },
|
|
{ BackCenter, Deg2Rad(180.0f), Deg2Rad(0.0f) },
|
|
{ SideLeft, Deg2Rad(-90.0f), Deg2Rad(0.0f) },
|
|
{ SideRight, Deg2Rad( 90.0f), Deg2Rad(0.0f) }
|
|
}, X71Map[8]{
|
|
{ FrontLeft, Deg2Rad( -30.0f), Deg2Rad(0.0f) },
|
|
{ FrontRight, Deg2Rad( 30.0f), Deg2Rad(0.0f) },
|
|
{ FrontCenter, Deg2Rad( 0.0f), Deg2Rad(0.0f) },
|
|
{ LFE, 0.0f, 0.0f },
|
|
{ BackLeft, Deg2Rad(-150.0f), Deg2Rad(0.0f) },
|
|
{ BackRight, Deg2Rad( 150.0f), Deg2Rad(0.0f) },
|
|
{ SideLeft, Deg2Rad( -90.0f), Deg2Rad(0.0f) },
|
|
{ SideRight, Deg2Rad( 90.0f), Deg2Rad(0.0f) }
|
|
};
|
|
|
|
ChanMap StereoMap[2]{
|
|
{ FrontLeft, Deg2Rad(-30.0f), Deg2Rad(0.0f) },
|
|
{ FrontRight, Deg2Rad( 30.0f), Deg2Rad(0.0f) }
|
|
};
|
|
|
|
const auto Frequency = static_cast<float>(Device->Frequency);
|
|
const uint NumSends{Device->NumAuxSends};
|
|
|
|
const size_t num_channels{voice->mChans.size()};
|
|
ASSUME(num_channels > 0);
|
|
|
|
for(auto &chandata : voice->mChans)
|
|
{
|
|
chandata.mDryParams.Hrtf.Target = HrtfFilter{};
|
|
chandata.mDryParams.Gains.Target.fill(0.0f);
|
|
std::for_each(chandata.mWetParams.begin(), chandata.mWetParams.begin()+NumSends,
|
|
[](SendParams ¶ms) -> void { params.Gains.Target.fill(0.0f); });
|
|
}
|
|
|
|
DirectMode DirectChannels{props->DirectChannels};
|
|
const ChanMap *chans{nullptr};
|
|
float downmix_gain{1.0f};
|
|
switch(voice->mFmtChannels)
|
|
{
|
|
case FmtMono:
|
|
chans = MonoMap;
|
|
/* Mono buffers are never played direct. */
|
|
DirectChannels = DirectMode::Off;
|
|
break;
|
|
|
|
case FmtStereo:
|
|
if(DirectChannels == DirectMode::Off)
|
|
{
|
|
/* Convert counter-clockwise to clock-wise, and wrap between
|
|
* [-pi,+pi].
|
|
*/
|
|
StereoMap[0].angle = WrapRadians(-props->StereoPan[0]);
|
|
StereoMap[1].angle = WrapRadians(-props->StereoPan[1]);
|
|
}
|
|
|
|
chans = StereoMap;
|
|
downmix_gain = 1.0f / 2.0f;
|
|
break;
|
|
|
|
case FmtRear:
|
|
chans = RearMap;
|
|
downmix_gain = 1.0f / 2.0f;
|
|
break;
|
|
|
|
case FmtQuad:
|
|
chans = QuadMap;
|
|
downmix_gain = 1.0f / 4.0f;
|
|
break;
|
|
|
|
case FmtX51:
|
|
chans = X51Map;
|
|
/* NOTE: Excludes LFE. */
|
|
downmix_gain = 1.0f / 5.0f;
|
|
break;
|
|
|
|
case FmtX61:
|
|
chans = X61Map;
|
|
/* NOTE: Excludes LFE. */
|
|
downmix_gain = 1.0f / 6.0f;
|
|
break;
|
|
|
|
case FmtX71:
|
|
chans = X71Map;
|
|
/* NOTE: Excludes LFE. */
|
|
downmix_gain = 1.0f / 7.0f;
|
|
break;
|
|
|
|
case FmtBFormat2D:
|
|
case FmtBFormat3D:
|
|
case FmtUHJ2:
|
|
case FmtUHJ3:
|
|
case FmtUHJ4:
|
|
DirectChannels = DirectMode::Off;
|
|
break;
|
|
}
|
|
|
|
voice->mFlags &= ~(VoiceHasHrtf | VoiceHasNfc);
|
|
if(voice->mFmtChannels == FmtBFormat2D || voice->mFmtChannels == FmtBFormat3D
|
|
|| voice->mFmtChannels == FmtUHJ2 || voice->mFmtChannels == FmtUHJ3
|
|
|| voice->mFmtChannels == FmtUHJ4)
|
|
{
|
|
/* Special handling for B-Format sources. */
|
|
|
|
if(Device->AvgSpeakerDist > 0.0f && voice->mFmtChannels != FmtUHJ2)
|
|
{
|
|
if(!(Distance > std::numeric_limits<float>::epsilon()))
|
|
{
|
|
/* NOTE: The NFCtrlFilters were created with a w0 of 0, which
|
|
* is what we want for FOA input. The first channel may have
|
|
* been previously re-adjusted if panned, so reset it.
|
|
*/
|
|
voice->mChans[0].mDryParams.NFCtrlFilter.adjust(0.0f);
|
|
}
|
|
else
|
|
{
|
|
/* Clamp the distance for really close sources, to prevent
|
|
* excessive bass.
|
|
*/
|
|
const float mdist{maxf(Distance, Device->AvgSpeakerDist/4.0f)};
|
|
const float w0{SpeedOfSoundMetersPerSec / (mdist * Frequency)};
|
|
|
|
/* Only need to adjust the first channel of a B-Format source. */
|
|
voice->mChans[0].mDryParams.NFCtrlFilter.adjust(w0);
|
|
}
|
|
|
|
voice->mFlags |= VoiceHasNfc;
|
|
}
|
|
|
|
/* Panning a B-Format sound toward some direction is easy. Just pan the
|
|
* first (W) channel as a normal mono sound. The angular spread is used
|
|
* as a directional scalar to blend between full coverage and full
|
|
* panning.
|
|
*/
|
|
const float coverage{!(Distance > std::numeric_limits<float>::epsilon()) ? 1.0f :
|
|
(Spread * (1.0f/al::MathDefs<float>::Tau()))};
|
|
|
|
auto calc_coeffs = [xpos,ypos,zpos](RenderMode mode)
|
|
{
|
|
if(mode != RenderMode::Pairwise)
|
|
return CalcDirectionCoeffs({xpos, ypos, zpos}, 0.0f);
|
|
|
|
/* Clamp Y, in case rounding errors caused it to end up outside
|
|
* of -1...+1.
|
|
*/
|
|
const float ev{std::asin(clampf(ypos, -1.0f, 1.0f))};
|
|
/* Negate Z for right-handed coords with -Z in front. */
|
|
const float az{std::atan2(xpos, -zpos)};
|
|
|
|
/* A scalar of 1.5 for plain stereo results in +/-60 degrees
|
|
* being moved to +/-90 degrees for direct right and left
|
|
* speaker responses.
|
|
*/
|
|
return CalcAngleCoeffs(ScaleAzimuthFront(az, 1.5f), ev, 0.0f);
|
|
};
|
|
auto coeffs = calc_coeffs(Device->mRenderMode);
|
|
std::transform(coeffs.begin()+1, coeffs.end(), coeffs.begin()+1,
|
|
std::bind(std::multiplies<float>{}, _1, 1.0f-coverage));
|
|
|
|
/* NOTE: W needs to be scaled according to channel scaling. */
|
|
auto&& scales = GetAmbiScales(voice->mAmbiScaling);
|
|
ComputePanGains(&Device->Dry, coeffs.data(), DryGain.Base*scales[0],
|
|
voice->mChans[0].mDryParams.Gains.Target);
|
|
for(uint i{0};i < NumSends;i++)
|
|
{
|
|
if(const EffectSlot *Slot{SendSlots[i]})
|
|
ComputePanGains(&Slot->Wet, coeffs.data(), WetGain[i].Base*scales[0],
|
|
voice->mChans[0].mWetParams[i].Gains.Target);
|
|
}
|
|
|
|
if(coverage > 0.0f)
|
|
{
|
|
/* Local B-Format sources have their XYZ channels rotated according
|
|
* to the orientation.
|
|
*/
|
|
/* AT then UP */
|
|
alu::Vector N{props->OrientAt[0], props->OrientAt[1], props->OrientAt[2], 0.0f};
|
|
N.normalize();
|
|
alu::Vector V{props->OrientUp[0], props->OrientUp[1], props->OrientUp[2], 0.0f};
|
|
V.normalize();
|
|
if(!props->HeadRelative)
|
|
{
|
|
N = Context.Matrix * N;
|
|
V = Context.Matrix * V;
|
|
}
|
|
/* Build and normalize right-vector */
|
|
alu::Vector U{N.cross_product(V)};
|
|
U.normalize();
|
|
|
|
/* Build a rotation matrix. Manually fill the zeroth- and first-
|
|
* order elements, then construct the rotation for the higher
|
|
* orders.
|
|
*/
|
|
std::array<std::array<float,MaxAmbiChannels>,MaxAmbiChannels> shrot{};
|
|
shrot[0][0] = 1.0f;
|
|
shrot[1][1] = U[0]; shrot[1][2] = -V[0]; shrot[1][3] = -N[0];
|
|
shrot[2][1] = -U[1]; shrot[2][2] = V[1]; shrot[2][3] = N[1];
|
|
shrot[3][1] = U[2]; shrot[3][2] = -V[2]; shrot[3][3] = -N[2];
|
|
AmbiRotator(shrot, static_cast<int>(minu(voice->mAmbiOrder, Device->mAmbiOrder)));
|
|
|
|
/* Convert the rotation matrix for input ordering and scaling, and
|
|
* whether input is 2D or 3D.
|
|
*/
|
|
const uint8_t *index_map{
|
|
(voice->mFmtChannels == FmtBFormat2D || voice->mFmtChannels == FmtUHJ2
|
|
|| voice->mFmtChannels == FmtUHJ3) ?
|
|
GetAmbi2DLayout(voice->mAmbiLayout).data() :
|
|
GetAmbiLayout(voice->mAmbiLayout).data()};
|
|
|
|
static const uint8_t ChansPerOrder[MaxAmbiOrder+1]{1, 3, 5, 7,};
|
|
static const uint8_t OrderOffset[MaxAmbiOrder+1]{0, 1, 4, 9,};
|
|
for(size_t c{1};c < num_channels;c++)
|
|
{
|
|
const size_t acn{index_map[c]};
|
|
const size_t order{AmbiIndex::OrderFromChannel()[acn]};
|
|
const size_t tocopy{ChansPerOrder[order]};
|
|
const size_t offset{OrderOffset[order]};
|
|
const float scale{scales[acn] * coverage};
|
|
auto in = shrot.cbegin() + offset;
|
|
|
|
coeffs = std::array<float,MaxAmbiChannels>{};
|
|
for(size_t x{0};x < tocopy;++x)
|
|
coeffs[offset+x] = in[x][acn] * scale;
|
|
|
|
ComputePanGains(&Device->Dry, coeffs.data(), DryGain.Base,
|
|
voice->mChans[c].mDryParams.Gains.Target);
|
|
|
|
for(uint i{0};i < NumSends;i++)
|
|
{
|
|
if(const EffectSlot *Slot{SendSlots[i]})
|
|
ComputePanGains(&Slot->Wet, coeffs.data(), WetGain[i].Base,
|
|
voice->mChans[c].mWetParams[i].Gains.Target);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
else if(DirectChannels != DirectMode::Off && Device->FmtChans != DevFmtAmbi3D)
|
|
{
|
|
/* Direct source channels always play local. Skip the virtual channels
|
|
* and write inputs to the matching real outputs.
|
|
*/
|
|
voice->mDirect.Buffer = Device->RealOut.Buffer;
|
|
|
|
for(size_t c{0};c < num_channels;c++)
|
|
{
|
|
uint idx{GetChannelIdxByName(Device->RealOut, chans[c].channel)};
|
|
if(idx != INVALID_CHANNEL_INDEX)
|
|
voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base;
|
|
else if(DirectChannels == DirectMode::RemixMismatch)
|
|
{
|
|
auto match_channel = [chans,c](const InputRemixMap &map) noexcept -> bool
|
|
{ return chans[c].channel == map.channel; };
|
|
auto remap = std::find_if(Device->RealOut.RemixMap.cbegin(),
|
|
Device->RealOut.RemixMap.cend(), match_channel);
|
|
if(remap != Device->RealOut.RemixMap.cend())
|
|
for(const auto &target : remap->targets)
|
|
{
|
|
idx = GetChannelIdxByName(Device->RealOut, target.channel);
|
|
if(idx != INVALID_CHANNEL_INDEX)
|
|
voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base *
|
|
target.mix;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Auxiliary sends still use normal channel panning since they mix to
|
|
* B-Format, which can't channel-match.
|
|
*/
|
|
for(size_t c{0};c < num_channels;c++)
|
|
{
|
|
const auto coeffs = CalcAngleCoeffs(chans[c].angle, chans[c].elevation, 0.0f);
|
|
|
|
for(uint i{0};i < NumSends;i++)
|
|
{
|
|
if(const EffectSlot *Slot{SendSlots[i]})
|
|
ComputePanGains(&Slot->Wet, coeffs.data(), WetGain[i].Base,
|
|
voice->mChans[c].mWetParams[i].Gains.Target);
|
|
}
|
|
}
|
|
}
|
|
else if(Device->mRenderMode == RenderMode::Hrtf)
|
|
{
|
|
/* Full HRTF rendering. Skip the virtual channels and render to the
|
|
* real outputs.
|
|
*/
|
|
voice->mDirect.Buffer = Device->RealOut.Buffer;
|
|
|
|
if(Distance > std::numeric_limits<float>::epsilon())
|
|
{
|
|
const float ev{std::asin(clampf(ypos, -1.0f, 1.0f))};
|
|
const float az{std::atan2(xpos, -zpos)};
|
|
|
|
/* Get the HRIR coefficients and delays just once, for the given
|
|
* source direction.
|
|
*/
|
|
GetHrtfCoeffs(Device->mHrtf.get(), ev, az, Distance, Spread,
|
|
voice->mChans[0].mDryParams.Hrtf.Target.Coeffs,
|
|
voice->mChans[0].mDryParams.Hrtf.Target.Delay);
|
|
voice->mChans[0].mDryParams.Hrtf.Target.Gain = DryGain.Base * downmix_gain;
|
|
|
|
/* Remaining channels use the same results as the first. */
|
|
for(size_t c{1};c < num_channels;c++)
|
|
{
|
|
/* Skip LFE */
|
|
if(chans[c].channel == LFE) continue;
|
|
voice->mChans[c].mDryParams.Hrtf.Target = voice->mChans[0].mDryParams.Hrtf.Target;
|
|
}
|
|
|
|
/* Calculate the directional coefficients once, which apply to all
|
|
* input channels of the source sends.
|
|
*/
|
|
const auto coeffs = CalcDirectionCoeffs({xpos, ypos, zpos}, Spread);
|
|
|
|
for(size_t c{0};c < num_channels;c++)
|
|
{
|
|
/* Skip LFE */
|
|
if(chans[c].channel == LFE)
|
|
continue;
|
|
for(uint i{0};i < NumSends;i++)
|
|
{
|
|
if(const EffectSlot *Slot{SendSlots[i]})
|
|
ComputePanGains(&Slot->Wet, coeffs.data(), WetGain[i].Base * downmix_gain,
|
|
voice->mChans[c].mWetParams[i].Gains.Target);
|
|
}
|
|
}
|
|
}
|
|
else
|
|
{
|
|
/* Local sources on HRTF play with each channel panned to its
|
|
* relative location around the listener, providing "virtual
|
|
* speaker" responses.
|
|
*/
|
|
for(size_t c{0};c < num_channels;c++)
|
|
{
|
|
/* Skip LFE */
|
|
if(chans[c].channel == LFE)
|
|
continue;
|
|
|
|
/* Get the HRIR coefficients and delays for this channel
|
|
* position.
|
|
*/
|
|
GetHrtfCoeffs(Device->mHrtf.get(), chans[c].elevation, chans[c].angle,
|
|
std::numeric_limits<float>::infinity(), Spread,
|
|
voice->mChans[c].mDryParams.Hrtf.Target.Coeffs,
|
|
voice->mChans[c].mDryParams.Hrtf.Target.Delay);
|
|
voice->mChans[c].mDryParams.Hrtf.Target.Gain = DryGain.Base;
|
|
|
|
/* Normal panning for auxiliary sends. */
|
|
const auto coeffs = CalcAngleCoeffs(chans[c].angle, chans[c].elevation, Spread);
|
|
|
|
for(uint i{0};i < NumSends;i++)
|
|
{
|
|
if(const EffectSlot *Slot{SendSlots[i]})
|
|
ComputePanGains(&Slot->Wet, coeffs.data(), WetGain[i].Base,
|
|
voice->mChans[c].mWetParams[i].Gains.Target);
|
|
}
|
|
}
|
|
}
|
|
|
|
voice->mFlags |= VoiceHasHrtf;
|
|
}
|
|
else
|
|
{
|
|
/* Non-HRTF rendering. Use normal panning to the output. */
|
|
|
|
if(Distance > std::numeric_limits<float>::epsilon())
|
|
{
|
|
/* Calculate NFC filter coefficient if needed. */
|
|
if(Device->AvgSpeakerDist > 0.0f)
|
|
{
|
|
/* Clamp the distance for really close sources, to prevent
|
|
* excessive bass.
|
|
*/
|
|
const float mdist{maxf(Distance, Device->AvgSpeakerDist/4.0f)};
|
|
const float w0{SpeedOfSoundMetersPerSec / (mdist * Frequency)};
|
|
|
|
/* Adjust NFC filters. */
|
|
for(size_t c{0};c < num_channels;c++)
|
|
voice->mChans[c].mDryParams.NFCtrlFilter.adjust(w0);
|
|
|
|
voice->mFlags |= VoiceHasNfc;
|
|
}
|
|
|
|
/* Calculate the directional coefficients once, which apply to all
|
|
* input channels.
|
|
*/
|
|
auto calc_coeffs = [xpos,ypos,zpos,Spread](RenderMode mode)
|
|
{
|
|
if(mode != RenderMode::Pairwise)
|
|
return CalcDirectionCoeffs({xpos, ypos, zpos}, Spread);
|
|
const float ev{std::asin(clampf(ypos, -1.0f, 1.0f))};
|
|
const float az{std::atan2(xpos, -zpos)};
|
|
return CalcAngleCoeffs(ScaleAzimuthFront(az, 1.5f), ev, Spread);
|
|
};
|
|
const auto coeffs = calc_coeffs(Device->mRenderMode);
|
|
|
|
for(size_t c{0};c < num_channels;c++)
|
|
{
|
|
/* Special-case LFE */
|
|
if(chans[c].channel == LFE)
|
|
{
|
|
if(Device->Dry.Buffer.data() == Device->RealOut.Buffer.data())
|
|
{
|
|
const uint idx{GetChannelIdxByName(Device->RealOut, chans[c].channel)};
|
|
if(idx != INVALID_CHANNEL_INDEX)
|
|
voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base;
|
|
}
|
|
continue;
|
|
}
|
|
|
|
ComputePanGains(&Device->Dry, coeffs.data(), DryGain.Base * downmix_gain,
|
|
voice->mChans[c].mDryParams.Gains.Target);
|
|
for(uint i{0};i < NumSends;i++)
|
|
{
|
|
if(const EffectSlot *Slot{SendSlots[i]})
|
|
ComputePanGains(&Slot->Wet, coeffs.data(), WetGain[i].Base * downmix_gain,
|
|
voice->mChans[c].mWetParams[i].Gains.Target);
|
|
}
|
|
}
|
|
}
|
|
else
|
|
{
|
|
if(Device->AvgSpeakerDist > 0.0f)
|
|
{
|
|
/* If the source distance is 0, simulate a plane-wave by using
|
|
* infinite distance, which results in a w0 of 0.
|
|
*/
|
|
constexpr float w0{0.0f};
|
|
for(size_t c{0};c < num_channels;c++)
|
|
voice->mChans[c].mDryParams.NFCtrlFilter.adjust(w0);
|
|
|
|
voice->mFlags |= VoiceHasNfc;
|
|
}
|
|
|
|
for(size_t c{0};c < num_channels;c++)
|
|
{
|
|
/* Special-case LFE */
|
|
if(chans[c].channel == LFE)
|
|
{
|
|
if(Device->Dry.Buffer.data() == Device->RealOut.Buffer.data())
|
|
{
|
|
const uint idx{GetChannelIdxByName(Device->RealOut, chans[c].channel)};
|
|
if(idx != INVALID_CHANNEL_INDEX)
|
|
voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base;
|
|
}
|
|
continue;
|
|
}
|
|
|
|
const auto coeffs = CalcAngleCoeffs((Device->mRenderMode == RenderMode::Pairwise)
|
|
? ScaleAzimuthFront(chans[c].angle, 3.0f) : chans[c].angle,
|
|
chans[c].elevation, Spread);
|
|
|
|
ComputePanGains(&Device->Dry, coeffs.data(), DryGain.Base,
|
|
voice->mChans[c].mDryParams.Gains.Target);
|
|
for(uint i{0};i < NumSends;i++)
|
|
{
|
|
if(const EffectSlot *Slot{SendSlots[i]})
|
|
ComputePanGains(&Slot->Wet, coeffs.data(), WetGain[i].Base,
|
|
voice->mChans[c].mWetParams[i].Gains.Target);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
{
|
|
const float hfNorm{props->Direct.HFReference / Frequency};
|
|
const float lfNorm{props->Direct.LFReference / Frequency};
|
|
|
|
voice->mDirect.FilterType = AF_None;
|
|
if(DryGain.HF != 1.0f) voice->mDirect.FilterType |= AF_LowPass;
|
|
if(DryGain.LF != 1.0f) voice->mDirect.FilterType |= AF_HighPass;
|
|
|
|
auto &lowpass = voice->mChans[0].mDryParams.LowPass;
|
|
auto &highpass = voice->mChans[0].mDryParams.HighPass;
|
|
lowpass.setParamsFromSlope(BiquadType::HighShelf, hfNorm, DryGain.HF, 1.0f);
|
|
highpass.setParamsFromSlope(BiquadType::LowShelf, lfNorm, DryGain.LF, 1.0f);
|
|
for(size_t c{1};c < num_channels;c++)
|
|
{
|
|
voice->mChans[c].mDryParams.LowPass.copyParamsFrom(lowpass);
|
|
voice->mChans[c].mDryParams.HighPass.copyParamsFrom(highpass);
|
|
}
|
|
}
|
|
for(uint i{0};i < NumSends;i++)
|
|
{
|
|
const float hfNorm{props->Send[i].HFReference / Frequency};
|
|
const float lfNorm{props->Send[i].LFReference / Frequency};
|
|
|
|
voice->mSend[i].FilterType = AF_None;
|
|
if(WetGain[i].HF != 1.0f) voice->mSend[i].FilterType |= AF_LowPass;
|
|
if(WetGain[i].LF != 1.0f) voice->mSend[i].FilterType |= AF_HighPass;
|
|
|
|
auto &lowpass = voice->mChans[0].mWetParams[i].LowPass;
|
|
auto &highpass = voice->mChans[0].mWetParams[i].HighPass;
|
|
lowpass.setParamsFromSlope(BiquadType::HighShelf, hfNorm, WetGain[i].HF, 1.0f);
|
|
highpass.setParamsFromSlope(BiquadType::LowShelf, lfNorm, WetGain[i].LF, 1.0f);
|
|
for(size_t c{1};c < num_channels;c++)
|
|
{
|
|
voice->mChans[c].mWetParams[i].LowPass.copyParamsFrom(lowpass);
|
|
voice->mChans[c].mWetParams[i].HighPass.copyParamsFrom(highpass);
|
|
}
|
|
}
|
|
}
|
|
|
|
void CalcNonAttnSourceParams(Voice *voice, const VoiceProps *props, const ContextBase *context)
|
|
{
|
|
const DeviceBase *Device{context->mDevice};
|
|
EffectSlot *SendSlots[MAX_SENDS];
|
|
|
|
voice->mDirect.Buffer = Device->Dry.Buffer;
|
|
for(uint i{0};i < Device->NumAuxSends;i++)
|
|
{
|
|
SendSlots[i] = props->Send[i].Slot;
|
|
if(!SendSlots[i] || SendSlots[i]->EffectType == EffectSlotType::None)
|
|
{
|
|
SendSlots[i] = nullptr;
|
|
voice->mSend[i].Buffer = {};
|
|
}
|
|
else
|
|
voice->mSend[i].Buffer = SendSlots[i]->Wet.Buffer;
|
|
}
|
|
|
|
/* Calculate the stepping value */
|
|
const auto Pitch = static_cast<float>(voice->mFrequency) /
|
|
static_cast<float>(Device->Frequency) * props->Pitch;
|
|
if(Pitch > float{MaxPitch})
|
|
voice->mStep = MaxPitch<<MixerFracBits;
|
|
else
|
|
voice->mStep = maxu(fastf2u(Pitch * MixerFracOne), 1);
|
|
voice->mResampler = PrepareResampler(props->mResampler, voice->mStep, &voice->mResampleState);
|
|
|
|
/* Calculate gains */
|
|
GainTriplet DryGain;
|
|
DryGain.Base = minf(clampf(props->Gain, props->MinGain, props->MaxGain) * props->Direct.Gain *
|
|
context->mParams.Gain, GainMixMax);
|
|
DryGain.HF = props->Direct.GainHF;
|
|
DryGain.LF = props->Direct.GainLF;
|
|
GainTriplet WetGain[MAX_SENDS];
|
|
for(uint i{0};i < Device->NumAuxSends;i++)
|
|
{
|
|
WetGain[i].Base = minf(clampf(props->Gain, props->MinGain, props->MaxGain) *
|
|
props->Send[i].Gain * context->mParams.Gain, GainMixMax);
|
|
WetGain[i].HF = props->Send[i].GainHF;
|
|
WetGain[i].LF = props->Send[i].GainLF;
|
|
}
|
|
|
|
CalcPanningAndFilters(voice, 0.0f, 0.0f, -1.0f, 0.0f, 0.0f, DryGain, WetGain, SendSlots, props,
|
|
context->mParams, Device);
|
|
}
|
|
|
|
void CalcAttnSourceParams(Voice *voice, const VoiceProps *props, const ContextBase *context)
|
|
{
|
|
const DeviceBase *Device{context->mDevice};
|
|
const uint NumSends{Device->NumAuxSends};
|
|
|
|
/* Set mixing buffers and get send parameters. */
|
|
voice->mDirect.Buffer = Device->Dry.Buffer;
|
|
EffectSlot *SendSlots[MAX_SENDS];
|
|
float RoomRolloff[MAX_SENDS];
|
|
GainTriplet DecayDistance[MAX_SENDS];
|
|
for(uint i{0};i < NumSends;i++)
|
|
{
|
|
SendSlots[i] = props->Send[i].Slot;
|
|
if(!SendSlots[i] || SendSlots[i]->EffectType == EffectSlotType::None)
|
|
{
|
|
SendSlots[i] = nullptr;
|
|
RoomRolloff[i] = 0.0f;
|
|
DecayDistance[i].Base = 0.0f;
|
|
DecayDistance[i].LF = 0.0f;
|
|
DecayDistance[i].HF = 0.0f;
|
|
}
|
|
else if(SendSlots[i]->AuxSendAuto)
|
|
{
|
|
RoomRolloff[i] = SendSlots[i]->RoomRolloff + props->RoomRolloffFactor;
|
|
/* Calculate the distances to where this effect's decay reaches
|
|
* -60dB.
|
|
*/
|
|
DecayDistance[i].Base = SendSlots[i]->DecayTime * SpeedOfSoundMetersPerSec;
|
|
DecayDistance[i].LF = DecayDistance[i].Base * SendSlots[i]->DecayLFRatio;
|
|
DecayDistance[i].HF = DecayDistance[i].Base * SendSlots[i]->DecayHFRatio;
|
|
if(SendSlots[i]->DecayHFLimit)
|
|
{
|
|
const float airAbsorption{SendSlots[i]->AirAbsorptionGainHF};
|
|
if(airAbsorption < 1.0f)
|
|
{
|
|
/* Calculate the distance to where this effect's air
|
|
* absorption reaches -60dB, and limit the effect's HF
|
|
* decay distance (so it doesn't take any longer to decay
|
|
* than the air would allow).
|
|
*/
|
|
constexpr float log10_decaygain{-3.0f/*std::log10(ReverbDecayGain)*/};
|
|
const float absorb_dist{log10_decaygain / std::log10(airAbsorption)};
|
|
DecayDistance[i].HF = minf(absorb_dist, DecayDistance[i].HF);
|
|
}
|
|
}
|
|
}
|
|
else
|
|
{
|
|
/* If the slot's auxiliary send auto is off, the data sent to the
|
|
* effect slot is the same as the dry path, sans filter effects */
|
|
RoomRolloff[i] = props->RolloffFactor;
|
|
DecayDistance[i].Base = 0.0f;
|
|
DecayDistance[i].LF = 0.0f;
|
|
DecayDistance[i].HF = 0.0f;
|
|
}
|
|
|
|
if(!SendSlots[i])
|
|
voice->mSend[i].Buffer = {};
|
|
else
|
|
voice->mSend[i].Buffer = SendSlots[i]->Wet.Buffer;
|
|
}
|
|
|
|
/* Transform source to listener space (convert to head relative) */
|
|
alu::Vector Position{props->Position[0], props->Position[1], props->Position[2], 1.0f};
|
|
alu::Vector Velocity{props->Velocity[0], props->Velocity[1], props->Velocity[2], 0.0f};
|
|
alu::Vector Direction{props->Direction[0], props->Direction[1], props->Direction[2], 0.0f};
|
|
if(!props->HeadRelative)
|
|
{
|
|
/* Transform source vectors */
|
|
Position = context->mParams.Matrix * Position;
|
|
Velocity = context->mParams.Matrix * Velocity;
|
|
Direction = context->mParams.Matrix * Direction;
|
|
}
|
|
else
|
|
{
|
|
/* Offset the source velocity to be relative of the listener velocity */
|
|
Velocity += context->mParams.Velocity;
|
|
}
|
|
|
|
const bool directional{Direction.normalize() > 0.0f};
|
|
alu::Vector ToSource{Position[0], Position[1], Position[2], 0.0f};
|
|
const float Distance{ToSource.normalize(props->RefDistance / 1024.0f)};
|
|
|
|
/* Initial source gain */
|
|
GainTriplet DryGain{props->Gain, 1.0f, 1.0f};
|
|
GainTriplet WetGain[MAX_SENDS];
|
|
for(uint i{0};i < NumSends;i++)
|
|
WetGain[i] = DryGain;
|
|
|
|
/* Calculate distance attenuation */
|
|
float ClampedDist{Distance};
|
|
|
|
switch(context->mParams.SourceDistanceModel ? props->mDistanceModel
|
|
: context->mParams.mDistanceModel)
|
|
{
|
|
case DistanceModel::InverseClamped:
|
|
ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance);
|
|
if(props->MaxDistance < props->RefDistance) break;
|
|
/*fall-through*/
|
|
case DistanceModel::Inverse:
|
|
if(!(props->RefDistance > 0.0f))
|
|
ClampedDist = props->RefDistance;
|
|
else
|
|
{
|
|
float dist{lerp(props->RefDistance, ClampedDist, props->RolloffFactor)};
|
|
if(dist > 0.0f) DryGain.Base *= props->RefDistance / dist;
|
|
for(uint i{0};i < NumSends;i++)
|
|
{
|
|
dist = lerp(props->RefDistance, ClampedDist, RoomRolloff[i]);
|
|
if(dist > 0.0f) WetGain[i].Base *= props->RefDistance / dist;
|
|
}
|
|
}
|
|
break;
|
|
|
|
case DistanceModel::LinearClamped:
|
|
ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance);
|
|
if(props->MaxDistance < props->RefDistance) break;
|
|
/*fall-through*/
|
|
case DistanceModel::Linear:
|
|
if(!(props->MaxDistance != props->RefDistance))
|
|
ClampedDist = props->RefDistance;
|
|
else
|
|
{
|
|
float attn{props->RolloffFactor * (ClampedDist-props->RefDistance) /
|
|
(props->MaxDistance-props->RefDistance)};
|
|
DryGain.Base *= maxf(1.0f - attn, 0.0f);
|
|
for(uint i{0};i < NumSends;i++)
|
|
{
|
|
attn = RoomRolloff[i] * (ClampedDist-props->RefDistance) /
|
|
(props->MaxDistance-props->RefDistance);
|
|
WetGain[i].Base *= maxf(1.0f - attn, 0.0f);
|
|
}
|
|
}
|
|
break;
|
|
|
|
case DistanceModel::ExponentClamped:
|
|
ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance);
|
|
if(props->MaxDistance < props->RefDistance) break;
|
|
/*fall-through*/
|
|
case DistanceModel::Exponent:
|
|
if(!(ClampedDist > 0.0f && props->RefDistance > 0.0f))
|
|
ClampedDist = props->RefDistance;
|
|
else
|
|
{
|
|
const float dist_ratio{ClampedDist/props->RefDistance};
|
|
DryGain.Base *= std::pow(dist_ratio, -props->RolloffFactor);
|
|
for(uint i{0};i < NumSends;i++)
|
|
WetGain[i].Base *= std::pow(dist_ratio, -RoomRolloff[i]);
|
|
}
|
|
break;
|
|
|
|
case DistanceModel::Disable:
|
|
ClampedDist = props->RefDistance;
|
|
break;
|
|
}
|
|
|
|
/* Calculate directional soundcones */
|
|
if(directional && props->InnerAngle < 360.0f)
|
|
{
|
|
const float Angle{Rad2Deg(std::acos(Direction.dot_product(ToSource)) * ConeScale * -2.0f)};
|
|
|
|
float ConeGain, ConeHF;
|
|
if(!(Angle > props->InnerAngle))
|
|
{
|
|
ConeGain = 1.0f;
|
|
ConeHF = 1.0f;
|
|
}
|
|
else if(Angle < props->OuterAngle)
|
|
{
|
|
const float scale{(Angle-props->InnerAngle) / (props->OuterAngle-props->InnerAngle)};
|
|
ConeGain = lerp(1.0f, props->OuterGain, scale);
|
|
ConeHF = lerp(1.0f, props->OuterGainHF, scale);
|
|
}
|
|
else
|
|
{
|
|
ConeGain = props->OuterGain;
|
|
ConeHF = props->OuterGainHF;
|
|
}
|
|
|
|
DryGain.Base *= ConeGain;
|
|
if(props->DryGainHFAuto)
|
|
DryGain.HF *= ConeHF;
|
|
if(props->WetGainAuto)
|
|
std::for_each(std::begin(WetGain), std::begin(WetGain)+NumSends,
|
|
[ConeGain](GainTriplet &gain) noexcept -> void { gain.Base *= ConeGain; });
|
|
if(props->WetGainHFAuto)
|
|
std::for_each(std::begin(WetGain), std::begin(WetGain)+NumSends,
|
|
[ConeHF](GainTriplet &gain) noexcept -> void { gain.HF *= ConeHF; });
|
|
}
|
|
|
|
/* Apply gain and frequency filters */
|
|
DryGain.Base = minf(clampf(DryGain.Base, props->MinGain, props->MaxGain) * props->Direct.Gain *
|
|
context->mParams.Gain, GainMixMax);
|
|
DryGain.HF *= props->Direct.GainHF;
|
|
DryGain.LF *= props->Direct.GainLF;
|
|
for(uint i{0};i < NumSends;i++)
|
|
{
|
|
WetGain[i].Base = minf(clampf(WetGain[i].Base, props->MinGain, props->MaxGain) *
|
|
props->Send[i].Gain * context->mParams.Gain, GainMixMax);
|
|
WetGain[i].HF *= props->Send[i].GainHF;
|
|
WetGain[i].LF *= props->Send[i].GainLF;
|
|
}
|
|
|
|
/* Distance-based air absorption and initial send decay. */
|
|
if(ClampedDist > props->RefDistance && props->RolloffFactor > 0.0f)
|
|
{
|
|
const float meters_base{(ClampedDist-props->RefDistance) * props->RolloffFactor *
|
|
context->mParams.MetersPerUnit};
|
|
if(props->AirAbsorptionFactor > 0.0f)
|
|
{
|
|
const float hfattn{std::pow(AirAbsorbGainHF, meters_base*props->AirAbsorptionFactor)};
|
|
DryGain.HF *= hfattn;
|
|
std::for_each(std::begin(WetGain), std::begin(WetGain)+NumSends,
|
|
[hfattn](GainTriplet &gain) noexcept -> void { gain.HF *= hfattn; });
|
|
}
|
|
|
|
if(props->WetGainAuto)
|
|
{
|
|
/* Apply a decay-time transformation to the wet path, based on the
|
|
* source distance in meters. The initial decay of the reverb
|
|
* effect is calculated and applied to the wet path.
|
|
*/
|
|
for(uint i{0};i < NumSends;i++)
|
|
{
|
|
if(!(DecayDistance[i].Base > 0.0f))
|
|
continue;
|
|
|
|
const float gain{std::pow(ReverbDecayGain, meters_base/DecayDistance[i].Base)};
|
|
WetGain[i].Base *= gain;
|
|
/* Yes, the wet path's air absorption is applied with
|
|
* WetGainAuto on, rather than WetGainHFAuto.
|
|
*/
|
|
if(gain > 0.0f)
|
|
{
|
|
float gainhf{std::pow(ReverbDecayGain, meters_base/DecayDistance[i].HF)};
|
|
WetGain[i].HF *= minf(gainhf / gain, 1.0f);
|
|
float gainlf{std::pow(ReverbDecayGain, meters_base/DecayDistance[i].LF)};
|
|
WetGain[i].LF *= minf(gainlf / gain, 1.0f);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
/* Initial source pitch */
|
|
float Pitch{props->Pitch};
|
|
|
|
/* Calculate velocity-based doppler effect */
|
|
float DopplerFactor{props->DopplerFactor * context->mParams.DopplerFactor};
|
|
if(DopplerFactor > 0.0f)
|
|
{
|
|
const alu::Vector &lvelocity = context->mParams.Velocity;
|
|
float vss{Velocity.dot_product(ToSource) * -DopplerFactor};
|
|
float vls{lvelocity.dot_product(ToSource) * -DopplerFactor};
|
|
|
|
const float SpeedOfSound{context->mParams.SpeedOfSound};
|
|
if(!(vls < SpeedOfSound))
|
|
{
|
|
/* Listener moving away from the source at the speed of sound.
|
|
* Sound waves can't catch it.
|
|
*/
|
|
Pitch = 0.0f;
|
|
}
|
|
else if(!(vss < SpeedOfSound))
|
|
{
|
|
/* Source moving toward the listener at the speed of sound. Sound
|
|
* waves bunch up to extreme frequencies.
|
|
*/
|
|
Pitch = std::numeric_limits<float>::infinity();
|
|
}
|
|
else
|
|
{
|
|
/* Source and listener movement is nominal. Calculate the proper
|
|
* doppler shift.
|
|
*/
|
|
Pitch *= (SpeedOfSound-vls) / (SpeedOfSound-vss);
|
|
}
|
|
}
|
|
|
|
/* Adjust pitch based on the buffer and output frequencies, and calculate
|
|
* fixed-point stepping value.
|
|
*/
|
|
Pitch *= static_cast<float>(voice->mFrequency) / static_cast<float>(Device->Frequency);
|
|
if(Pitch > float{MaxPitch})
|
|
voice->mStep = MaxPitch<<MixerFracBits;
|
|
else
|
|
voice->mStep = maxu(fastf2u(Pitch * MixerFracOne), 1);
|
|
voice->mResampler = PrepareResampler(props->mResampler, voice->mStep, &voice->mResampleState);
|
|
|
|
float spread{0.0f};
|
|
if(props->Radius > Distance)
|
|
spread = al::MathDefs<float>::Tau() - Distance/props->Radius*al::MathDefs<float>::Pi();
|
|
else if(Distance > 0.0f)
|
|
spread = std::asin(props->Radius/Distance) * 2.0f;
|
|
|
|
CalcPanningAndFilters(voice, ToSource[0], ToSource[1], ToSource[2]*ZScale,
|
|
Distance*context->mParams.MetersPerUnit, spread, DryGain, WetGain, SendSlots, props,
|
|
context->mParams, Device);
|
|
}
|
|
|
|
void CalcSourceParams(Voice *voice, ContextBase *context, bool force)
|
|
{
|
|
VoicePropsItem *props{voice->mUpdate.exchange(nullptr, std::memory_order_acq_rel)};
|
|
if(!props && !force) return;
|
|
|
|
if(props)
|
|
{
|
|
voice->mProps = *props;
|
|
|
|
AtomicReplaceHead(context->mFreeVoiceProps, props);
|
|
}
|
|
|
|
if((voice->mProps.DirectChannels != DirectMode::Off && voice->mFmtChannels != FmtMono
|
|
&& voice->mFmtChannels != FmtBFormat2D && voice->mFmtChannels != FmtBFormat3D
|
|
&& voice->mFmtChannels != FmtUHJ2 && voice->mFmtChannels != FmtUHJ3
|
|
&& voice->mFmtChannels != FmtUHJ3)
|
|
|| voice->mProps.mSpatializeMode==SpatializeMode::Off
|
|
|| (voice->mProps.mSpatializeMode==SpatializeMode::Auto && voice->mFmtChannels != FmtMono))
|
|
CalcNonAttnSourceParams(voice, &voice->mProps, context);
|
|
else
|
|
CalcAttnSourceParams(voice, &voice->mProps, context);
|
|
}
|
|
|
|
|
|
void SendSourceStateEvent(ContextBase *context, uint id, VChangeState state)
|
|
{
|
|
RingBuffer *ring{context->mAsyncEvents.get()};
|
|
auto evt_vec = ring->getWriteVector();
|
|
if(evt_vec.first.len < 1) return;
|
|
|
|
AsyncEvent *evt{::new(evt_vec.first.buf) AsyncEvent{EventType_SourceStateChange}};
|
|
evt->u.srcstate.id = id;
|
|
switch(state)
|
|
{
|
|
case VChangeState::Reset:
|
|
evt->u.srcstate.state = AsyncEvent::SrcState::Reset;
|
|
break;
|
|
case VChangeState::Stop:
|
|
evt->u.srcstate.state = AsyncEvent::SrcState::Stop;
|
|
break;
|
|
case VChangeState::Play:
|
|
evt->u.srcstate.state = AsyncEvent::SrcState::Play;
|
|
break;
|
|
case VChangeState::Pause:
|
|
evt->u.srcstate.state = AsyncEvent::SrcState::Pause;
|
|
break;
|
|
/* Shouldn't happen. */
|
|
case VChangeState::Restart:
|
|
ASSUME(0);
|
|
}
|
|
|
|
ring->writeAdvance(1);
|
|
}
|
|
|
|
void ProcessVoiceChanges(ContextBase *ctx)
|
|
{
|
|
VoiceChange *cur{ctx->mCurrentVoiceChange.load(std::memory_order_acquire)};
|
|
VoiceChange *next{cur->mNext.load(std::memory_order_acquire)};
|
|
if(!next) return;
|
|
|
|
const uint enabledevt{ctx->mEnabledEvts.load(std::memory_order_acquire)};
|
|
do {
|
|
cur = next;
|
|
|
|
bool sendevt{false};
|
|
if(cur->mState == VChangeState::Reset || cur->mState == VChangeState::Stop)
|
|
{
|
|
if(Voice *voice{cur->mVoice})
|
|
{
|
|
voice->mCurrentBuffer.store(nullptr, std::memory_order_relaxed);
|
|
voice->mLoopBuffer.store(nullptr, std::memory_order_relaxed);
|
|
/* A source ID indicates the voice was playing or paused, which
|
|
* gets a reset/stop event.
|
|
*/
|
|
sendevt = voice->mSourceID.exchange(0u, std::memory_order_relaxed) != 0u;
|
|
Voice::State oldvstate{Voice::Playing};
|
|
voice->mPlayState.compare_exchange_strong(oldvstate, Voice::Stopping,
|
|
std::memory_order_relaxed, std::memory_order_acquire);
|
|
voice->mPendingChange.store(false, std::memory_order_release);
|
|
}
|
|
/* Reset state change events are always sent, even if the voice is
|
|
* already stopped or even if there is no voice.
|
|
*/
|
|
sendevt |= (cur->mState == VChangeState::Reset);
|
|
}
|
|
else if(cur->mState == VChangeState::Pause)
|
|
{
|
|
Voice *voice{cur->mVoice};
|
|
Voice::State oldvstate{Voice::Playing};
|
|
sendevt = voice->mPlayState.compare_exchange_strong(oldvstate, Voice::Stopping,
|
|
std::memory_order_release, std::memory_order_acquire);
|
|
}
|
|
else if(cur->mState == VChangeState::Play)
|
|
{
|
|
/* NOTE: When playing a voice, sending a source state change event
|
|
* depends if there's an old voice to stop and if that stop is
|
|
* successful. If there is no old voice, a playing event is always
|
|
* sent. If there is an old voice, an event is sent only if the
|
|
* voice is already stopped.
|
|
*/
|
|
if(Voice *oldvoice{cur->mOldVoice})
|
|
{
|
|
oldvoice->mCurrentBuffer.store(nullptr, std::memory_order_relaxed);
|
|
oldvoice->mLoopBuffer.store(nullptr, std::memory_order_relaxed);
|
|
oldvoice->mSourceID.store(0u, std::memory_order_relaxed);
|
|
Voice::State oldvstate{Voice::Playing};
|
|
sendevt = !oldvoice->mPlayState.compare_exchange_strong(oldvstate, Voice::Stopping,
|
|
std::memory_order_relaxed, std::memory_order_acquire);
|
|
oldvoice->mPendingChange.store(false, std::memory_order_release);
|
|
}
|
|
else
|
|
sendevt = true;
|
|
|
|
Voice *voice{cur->mVoice};
|
|
voice->mPlayState.store(Voice::Playing, std::memory_order_release);
|
|
}
|
|
else if(cur->mState == VChangeState::Restart)
|
|
{
|
|
/* Restarting a voice never sends a source change event. */
|
|
Voice *oldvoice{cur->mOldVoice};
|
|
oldvoice->mCurrentBuffer.store(nullptr, std::memory_order_relaxed);
|
|
oldvoice->mLoopBuffer.store(nullptr, std::memory_order_relaxed);
|
|
/* If there's no sourceID, the old voice finished so don't start
|
|
* the new one at its new offset.
|
|
*/
|
|
if(oldvoice->mSourceID.exchange(0u, std::memory_order_relaxed) != 0u)
|
|
{
|
|
/* Otherwise, set the voice to stopping if it's not already (it
|
|
* might already be, if paused), and play the new voice as
|
|
* appropriate.
|
|
*/
|
|
Voice::State oldvstate{Voice::Playing};
|
|
oldvoice->mPlayState.compare_exchange_strong(oldvstate, Voice::Stopping,
|
|
std::memory_order_relaxed, std::memory_order_acquire);
|
|
|
|
Voice *voice{cur->mVoice};
|
|
voice->mPlayState.store((oldvstate == Voice::Playing) ? Voice::Playing
|
|
: Voice::Stopped, std::memory_order_release);
|
|
}
|
|
oldvoice->mPendingChange.store(false, std::memory_order_release);
|
|
}
|
|
if(sendevt && (enabledevt&EventType_SourceStateChange))
|
|
SendSourceStateEvent(ctx, cur->mSourceID, cur->mState);
|
|
|
|
next = cur->mNext.load(std::memory_order_acquire);
|
|
} while(next);
|
|
ctx->mCurrentVoiceChange.store(cur, std::memory_order_release);
|
|
}
|
|
|
|
void ProcessParamUpdates(ContextBase *ctx, const EffectSlotArray &slots,
|
|
const al::span<Voice*> voices)
|
|
{
|
|
ProcessVoiceChanges(ctx);
|
|
|
|
IncrementRef(ctx->mUpdateCount);
|
|
if LIKELY(!ctx->mHoldUpdates.load(std::memory_order_acquire))
|
|
{
|
|
bool force{CalcContextParams(ctx)};
|
|
force |= CalcListenerParams(ctx);
|
|
auto sorted_slots = const_cast<EffectSlot**>(slots.data() + slots.size());
|
|
for(EffectSlot *slot : slots)
|
|
force |= CalcEffectSlotParams(slot, sorted_slots, ctx);
|
|
|
|
for(Voice *voice : voices)
|
|
{
|
|
/* Only update voices that have a source. */
|
|
if(voice->mSourceID.load(std::memory_order_relaxed) != 0)
|
|
CalcSourceParams(voice, ctx, force);
|
|
}
|
|
}
|
|
IncrementRef(ctx->mUpdateCount);
|
|
}
|
|
|
|
void ProcessContexts(DeviceBase *device, const uint SamplesToDo)
|
|
{
|
|
ASSUME(SamplesToDo > 0);
|
|
|
|
for(ContextBase *ctx : *device->mContexts.load(std::memory_order_acquire))
|
|
{
|
|
const EffectSlotArray &auxslots = *ctx->mActiveAuxSlots.load(std::memory_order_acquire);
|
|
const al::span<Voice*> voices{ctx->getVoicesSpanAcquired()};
|
|
|
|
/* Process pending propery updates for objects on the context. */
|
|
ProcessParamUpdates(ctx, auxslots, voices);
|
|
|
|
/* Clear auxiliary effect slot mixing buffers. */
|
|
for(EffectSlot *slot : auxslots)
|
|
{
|
|
for(auto &buffer : slot->Wet.Buffer)
|
|
buffer.fill(0.0f);
|
|
}
|
|
|
|
/* Process voices that have a playing source. */
|
|
for(Voice *voice : voices)
|
|
{
|
|
const Voice::State vstate{voice->mPlayState.load(std::memory_order_acquire)};
|
|
if(vstate != Voice::Stopped && vstate != Voice::Pending)
|
|
voice->mix(vstate, ctx, SamplesToDo);
|
|
}
|
|
|
|
/* Process effects. */
|
|
if(const size_t num_slots{auxslots.size()})
|
|
{
|
|
auto slots = auxslots.data();
|
|
auto slots_end = slots + num_slots;
|
|
|
|
/* Sort the slots into extra storage, so that effect slots come
|
|
* before their effect slot target (or their targets' target).
|
|
*/
|
|
const al::span<EffectSlot*> sorted_slots{const_cast<EffectSlot**>(slots_end),
|
|
num_slots};
|
|
/* Skip sorting if it has already been done. */
|
|
if(!sorted_slots[0])
|
|
{
|
|
/* First, copy the slots to the sorted list, then partition the
|
|
* sorted list so that all slots without a target slot go to
|
|
* the end.
|
|
*/
|
|
std::copy(slots, slots_end, sorted_slots.begin());
|
|
auto split_point = std::partition(sorted_slots.begin(), sorted_slots.end(),
|
|
[](const EffectSlot *slot) noexcept -> bool
|
|
{ return slot->Target != nullptr; });
|
|
/* There must be at least one slot without a slot target. */
|
|
assert(split_point != sorted_slots.end());
|
|
|
|
/* Simple case: no more than 1 slot has a target slot. Either
|
|
* all slots go right to the output, or the remaining one must
|
|
* target an already-partitioned slot.
|
|
*/
|
|
if(split_point - sorted_slots.begin() > 1)
|
|
{
|
|
/* At least two slots target other slots. Starting from the
|
|
* back of the sorted list, continue partitioning the front
|
|
* of the list given each target until all targets are
|
|
* accounted for. This ensures all slots without a target
|
|
* go last, all slots directly targeting those last slots
|
|
* go second-to-last, all slots directly targeting those
|
|
* second-last slots go third-to-last, etc.
|
|
*/
|
|
auto next_target = sorted_slots.end();
|
|
do {
|
|
/* This shouldn't happen, but if there's unsorted slots
|
|
* left that don't target any sorted slots, they can't
|
|
* contribute to the output, so leave them.
|
|
*/
|
|
if UNLIKELY(next_target == split_point)
|
|
break;
|
|
|
|
--next_target;
|
|
split_point = std::partition(sorted_slots.begin(), split_point,
|
|
[next_target](const EffectSlot *slot) noexcept -> bool
|
|
{ return slot->Target != *next_target; });
|
|
} while(split_point - sorted_slots.begin() > 1);
|
|
}
|
|
}
|
|
|
|
for(const EffectSlot *slot : sorted_slots)
|
|
{
|
|
EffectState *state{slot->mEffectState};
|
|
state->process(SamplesToDo, slot->Wet.Buffer, state->mOutTarget);
|
|
}
|
|
}
|
|
|
|
/* Signal the event handler if there are any events to read. */
|
|
RingBuffer *ring{ctx->mAsyncEvents.get()};
|
|
if(ring->readSpace() > 0)
|
|
ctx->mEventSem.post();
|
|
}
|
|
}
|
|
|
|
|
|
void ApplyDistanceComp(const al::span<FloatBufferLine> Samples, const size_t SamplesToDo,
|
|
const DistanceComp::ChanData *distcomp)
|
|
{
|
|
ASSUME(SamplesToDo > 0);
|
|
|
|
for(auto &chanbuffer : Samples)
|
|
{
|
|
const float gain{distcomp->Gain};
|
|
const size_t base{distcomp->Length};
|
|
float *distbuf{al::assume_aligned<16>(distcomp->Buffer)};
|
|
++distcomp;
|
|
|
|
if(base < 1)
|
|
continue;
|
|
|
|
float *inout{al::assume_aligned<16>(chanbuffer.data())};
|
|
auto inout_end = inout + SamplesToDo;
|
|
if LIKELY(SamplesToDo >= base)
|
|
{
|
|
auto delay_end = std::rotate(inout, inout_end - base, inout_end);
|
|
std::swap_ranges(inout, delay_end, distbuf);
|
|
}
|
|
else
|
|
{
|
|
auto delay_start = std::swap_ranges(inout, inout_end, distbuf);
|
|
std::rotate(distbuf, delay_start, distbuf + base);
|
|
}
|
|
std::transform(inout, inout_end, inout, std::bind(std::multiplies<float>{}, _1, gain));
|
|
}
|
|
}
|
|
|
|
void ApplyDither(const al::span<FloatBufferLine> Samples, uint *dither_seed,
|
|
const float quant_scale, const size_t SamplesToDo)
|
|
{
|
|
ASSUME(SamplesToDo > 0);
|
|
|
|
/* Dithering. Generate whitenoise (uniform distribution of random values
|
|
* between -1 and +1) and add it to the sample values, after scaling up to
|
|
* the desired quantization depth amd before rounding.
|
|
*/
|
|
const float invscale{1.0f / quant_scale};
|
|
uint seed{*dither_seed};
|
|
auto dither_sample = [&seed,invscale,quant_scale](const float sample) noexcept -> float
|
|
{
|
|
float val{sample * quant_scale};
|
|
uint rng0{dither_rng(&seed)};
|
|
uint rng1{dither_rng(&seed)};
|
|
val += static_cast<float>(rng0*(1.0/UINT_MAX) - rng1*(1.0/UINT_MAX));
|
|
return fast_roundf(val) * invscale;
|
|
};
|
|
for(FloatBufferLine &inout : Samples)
|
|
std::transform(inout.begin(), inout.begin()+SamplesToDo, inout.begin(), dither_sample);
|
|
*dither_seed = seed;
|
|
}
|
|
|
|
|
|
/* Base template left undefined. Should be marked =delete, but Clang 3.8.1
|
|
* chokes on that given the inline specializations.
|
|
*/
|
|
template<typename T>
|
|
inline T SampleConv(float) noexcept;
|
|
|
|
template<> inline float SampleConv(float val) noexcept
|
|
{ return val; }
|
|
template<> inline int32_t SampleConv(float val) noexcept
|
|
{
|
|
/* Floats have a 23-bit mantissa, plus an implied 1 bit and a sign bit.
|
|
* This means a normalized float has at most 25 bits of signed precision.
|
|
* When scaling and clamping for a signed 32-bit integer, these following
|
|
* values are the best a float can give.
|
|
*/
|
|
return fastf2i(clampf(val*2147483648.0f, -2147483648.0f, 2147483520.0f));
|
|
}
|
|
template<> inline int16_t SampleConv(float val) noexcept
|
|
{ return static_cast<int16_t>(fastf2i(clampf(val*32768.0f, -32768.0f, 32767.0f))); }
|
|
template<> inline int8_t SampleConv(float val) noexcept
|
|
{ return static_cast<int8_t>(fastf2i(clampf(val*128.0f, -128.0f, 127.0f))); }
|
|
|
|
/* Define unsigned output variations. */
|
|
template<> inline uint32_t SampleConv(float val) noexcept
|
|
{ return static_cast<uint32_t>(SampleConv<int32_t>(val)) + 2147483648u; }
|
|
template<> inline uint16_t SampleConv(float val) noexcept
|
|
{ return static_cast<uint16_t>(SampleConv<int16_t>(val) + 32768); }
|
|
template<> inline uint8_t SampleConv(float val) noexcept
|
|
{ return static_cast<uint8_t>(SampleConv<int8_t>(val) + 128); }
|
|
|
|
template<DevFmtType T>
|
|
void Write(const al::span<const FloatBufferLine> InBuffer, void *OutBuffer, const size_t Offset,
|
|
const size_t SamplesToDo, const size_t FrameStep)
|
|
{
|
|
ASSUME(FrameStep > 0);
|
|
ASSUME(SamplesToDo > 0);
|
|
|
|
DevFmtType_t<T> *outbase = static_cast<DevFmtType_t<T>*>(OutBuffer) + Offset*FrameStep;
|
|
for(const FloatBufferLine &inbuf : InBuffer)
|
|
{
|
|
DevFmtType_t<T> *out{outbase++};
|
|
auto conv_sample = [FrameStep,&out](const float s) noexcept -> void
|
|
{
|
|
*out = SampleConv<DevFmtType_t<T>>(s);
|
|
out += FrameStep;
|
|
};
|
|
std::for_each(inbuf.begin(), inbuf.begin()+SamplesToDo, conv_sample);
|
|
}
|
|
}
|
|
|
|
} // namespace
|
|
|
|
void DeviceBase::renderSamples(void *outBuffer, const uint numSamples, const size_t frameStep)
|
|
{
|
|
FPUCtl mixer_mode{};
|
|
for(uint written{0u};written < numSamples;)
|
|
{
|
|
const uint samplesToDo{minu(numSamples-written, BufferLineSize)};
|
|
|
|
/* Clear main mixing buffers. */
|
|
for(FloatBufferLine &buffer : MixBuffer)
|
|
buffer.fill(0.0f);
|
|
|
|
/* Increment the mix count at the start (lsb should now be 1). */
|
|
IncrementRef(MixCount);
|
|
|
|
/* Process and mix each context's sources and effects. */
|
|
ProcessContexts(this, samplesToDo);
|
|
|
|
/* Increment the clock time. Every second's worth of samples is
|
|
* converted and added to clock base so that large sample counts don't
|
|
* overflow during conversion. This also guarantees a stable
|
|
* conversion.
|
|
*/
|
|
SamplesDone += samplesToDo;
|
|
ClockBase += std::chrono::seconds{SamplesDone / Frequency};
|
|
SamplesDone %= Frequency;
|
|
|
|
/* Increment the mix count at the end (lsb should now be 0). */
|
|
IncrementRef(MixCount);
|
|
|
|
/* Apply any needed post-process for finalizing the Dry mix to the
|
|
* RealOut (Ambisonic decode, UHJ encode, etc).
|
|
*/
|
|
postProcess(samplesToDo);
|
|
|
|
/* Apply compression, limiting sample amplitude if needed or desired. */
|
|
if(Limiter) Limiter->process(samplesToDo, RealOut.Buffer.data());
|
|
|
|
/* Apply delays and attenuation for mismatched speaker distances. */
|
|
if(ChannelDelays)
|
|
ApplyDistanceComp(RealOut.Buffer, samplesToDo, ChannelDelays->mChannels.data());
|
|
|
|
/* Apply dithering. The compressor should have left enough headroom for
|
|
* the dither noise to not saturate.
|
|
*/
|
|
if(DitherDepth > 0.0f)
|
|
ApplyDither(RealOut.Buffer, &DitherSeed, DitherDepth, samplesToDo);
|
|
|
|
if LIKELY(outBuffer)
|
|
{
|
|
/* Finally, interleave and convert samples, writing to the device's
|
|
* output buffer.
|
|
*/
|
|
switch(FmtType)
|
|
{
|
|
#define HANDLE_WRITE(T) case T: \
|
|
Write<T>(RealOut.Buffer, outBuffer, written, samplesToDo, frameStep); break;
|
|
HANDLE_WRITE(DevFmtByte)
|
|
HANDLE_WRITE(DevFmtUByte)
|
|
HANDLE_WRITE(DevFmtShort)
|
|
HANDLE_WRITE(DevFmtUShort)
|
|
HANDLE_WRITE(DevFmtInt)
|
|
HANDLE_WRITE(DevFmtUInt)
|
|
HANDLE_WRITE(DevFmtFloat)
|
|
#undef HANDLE_WRITE
|
|
}
|
|
}
|
|
|
|
written += samplesToDo;
|
|
}
|
|
}
|
|
|
|
void DeviceBase::handleDisconnect(const char *msg, ...)
|
|
{
|
|
if(!Connected.exchange(false, std::memory_order_acq_rel))
|
|
return;
|
|
|
|
AsyncEvent evt{EventType_Disconnected};
|
|
|
|
va_list args;
|
|
va_start(args, msg);
|
|
int msglen{vsnprintf(evt.u.disconnect.msg, sizeof(evt.u.disconnect.msg), msg, args)};
|
|
va_end(args);
|
|
|
|
if(msglen < 0 || static_cast<size_t>(msglen) >= sizeof(evt.u.disconnect.msg))
|
|
evt.u.disconnect.msg[sizeof(evt.u.disconnect.msg)-1] = 0;
|
|
|
|
IncrementRef(MixCount);
|
|
for(ContextBase *ctx : *mContexts.load())
|
|
{
|
|
const uint enabledevt{ctx->mEnabledEvts.load(std::memory_order_acquire)};
|
|
if((enabledevt&EventType_Disconnected))
|
|
{
|
|
RingBuffer *ring{ctx->mAsyncEvents.get()};
|
|
auto evt_data = ring->getWriteVector().first;
|
|
if(evt_data.len > 0)
|
|
{
|
|
::new(evt_data.buf) AsyncEvent{evt};
|
|
ring->writeAdvance(1);
|
|
ctx->mEventSem.post();
|
|
}
|
|
}
|
|
|
|
if(!ctx->mStopVoicesOnDisconnect)
|
|
{
|
|
ProcessVoiceChanges(ctx);
|
|
continue;
|
|
}
|
|
|
|
auto voicelist = ctx->getVoicesSpanAcquired();
|
|
auto stop_voice = [](Voice *voice) -> void
|
|
{
|
|
voice->mCurrentBuffer.store(nullptr, std::memory_order_relaxed);
|
|
voice->mLoopBuffer.store(nullptr, std::memory_order_relaxed);
|
|
voice->mSourceID.store(0u, std::memory_order_relaxed);
|
|
voice->mPlayState.store(Voice::Stopped, std::memory_order_release);
|
|
};
|
|
std::for_each(voicelist.begin(), voicelist.end(), stop_voice);
|
|
}
|
|
IncrementRef(MixCount);
|
|
}
|