From c80ee5b7014d12675e2c615574c9f3f3354ffd1b Mon Sep 17 00:00:00 2001 From: Chris Robinson Date: Tue, 28 May 2019 16:22:36 -0700 Subject: Use std::array for most mixing buffer arrays --- Alc/alc.cpp | 2 +- Alc/alu.cpp | 58 +++++++++++++++++++++++++++++------------------------- Alc/bformatdec.cpp | 19 +++++++++--------- Alc/bformatdec.h | 4 ++-- Alc/effects/base.h | 2 +- Alc/mastering.cpp | 20 +++++++++---------- Alc/mastering.h | 4 ++-- Alc/mixvoice.cpp | 29 ++++++++++++++------------- Alc/panning.cpp | 2 +- Alc/uhjfilter.cpp | 12 +++++------ Alc/uhjfilter.h | 3 ++- 11 files changed, 81 insertions(+), 74 deletions(-) (limited to 'Alc') diff --git a/Alc/alc.cpp b/Alc/alc.cpp index eb6269ec..6d09b118 100644 --- a/Alc/alc.cpp +++ b/Alc/alc.cpp @@ -1937,7 +1937,7 @@ static ALCenum UpdateDeviceParams(ALCdevice *device, const ALCint *attrList) num_chans*sizeof(device->MixBuffer[0])); device->MixBuffer.resize(num_chans); - device->Dry.Buffer = &reinterpret_cast(device->MixBuffer[0]); + device->Dry.Buffer = device->MixBuffer.data(); if(device->RealOut.NumChannels != 0) device->RealOut.Buffer = device->Dry.Buffer + device->Dry.NumChannels; else diff --git a/Alc/alu.cpp b/Alc/alu.cpp index 94ba3aa8..2ce74537 100644 --- a/Alc/alu.cpp +++ b/Alc/alu.cpp @@ -133,8 +133,9 @@ void ProcessHrtf(ALCdevice *device, const ALsizei SamplesToDo) ASSUME(lidx >= 0 && ridx >= 0); DirectHrtfState *state{device->mHrtfState.get()}; - MixDirectHrtf(device->RealOut.Buffer[lidx], device->RealOut.Buffer[ridx], device->Dry.Buffer, - device->HrtfAccumData, state, device->Dry.NumChannels, SamplesToDo); + MixDirectHrtf(device->RealOut.Buffer[lidx].data(), device->RealOut.Buffer[ridx].data(), + &reinterpret_cast(device->Dry.Buffer[0]), device->HrtfAccumData, + state, device->Dry.NumChannels, SamplesToDo); } void ProcessAmbiDec(ALCdevice *device, const ALsizei SamplesToDo) @@ -165,8 +166,8 @@ void ProcessBs2b(ALCdevice *device, const ALsizei SamplesToDo) ASSUME(lidx >= 0 && ridx >= 0); /* Apply binaural/crossfeed filter */ - bs2b_cross_feed(device->Bs2b.get(), device->RealOut.Buffer[lidx], - device->RealOut.Buffer[ridx], SamplesToDo); + bs2b_cross_feed(device->Bs2b.get(), device->RealOut.Buffer[lidx].data(), + device->RealOut.Buffer[ridx].data(), SamplesToDo); } } // namespace @@ -1465,14 +1466,17 @@ void ProcessContext(ALCcontext *ctx, const ALsizei SamplesToDo) [SamplesToDo](const ALeffectslot *slot) -> void { EffectState *state{slot->Params.mEffectState}; - state->process(SamplesToDo, slot->Wet.Buffer, slot->Wet.NumChannels, - state->mOutBuffer, state->mOutChannels); + state->process(SamplesToDo, + &reinterpret_cast(slot->Wet.Buffer[0]), + slot->Wet.NumChannels, + &reinterpret_cast(state->mOutBuffer[0]), + state->mOutChannels); } ); } -void ApplyStablizer(FrontStablizer *Stablizer, ALfloat (*Buffer)[BUFFERSIZE], const int lidx, +void ApplyStablizer(FrontStablizer *Stablizer, FloatBufferLine *Buffer, const int lidx, const int ridx, const int cidx, const ALsizei SamplesToDo, const ALsizei NumChannels) { ASSUME(SamplesToDo > 0); @@ -1487,16 +1491,17 @@ void ApplyStablizer(FrontStablizer *Stablizer, ALfloat (*Buffer)[BUFFERSIZE], co continue; auto &DelayBuf = Stablizer->DelayBuf[i]; - auto buffer_end = Buffer[i] + SamplesToDo; + auto buffer_end = Buffer[i].begin() + SamplesToDo; if(LIKELY(SamplesToDo >= ALsizei{FrontStablizer::DelayLength})) { - auto delay_end = std::rotate(Buffer[i], buffer_end - FrontStablizer::DelayLength, - buffer_end); - std::swap_ranges(Buffer[i], delay_end, std::begin(DelayBuf)); + auto delay_end = std::rotate(Buffer[i].begin(), + buffer_end - FrontStablizer::DelayLength, buffer_end); + std::swap_ranges(Buffer[i].begin(), delay_end, std::begin(DelayBuf)); } else { - auto delay_start = std::swap_ranges(Buffer[i], buffer_end, std::begin(DelayBuf)); + auto delay_start = std::swap_ranges(Buffer[i].begin(), buffer_end, + std::begin(DelayBuf)); std::rotate(std::begin(DelayBuf), delay_start, std::end(DelayBuf)); } } @@ -1509,7 +1514,7 @@ void ApplyStablizer(FrontStablizer *Stablizer, ALfloat (*Buffer)[BUFFERSIZE], co /* This applies the band-splitter, preserving phase at the cost of some * delay. The shorter the delay, the more error seeps into the result. */ - auto apply_splitter = [&APFilter,&tmpbuf,SamplesToDo](const ALfloat *Buffer, + auto apply_splitter = [&APFilter,&tmpbuf,SamplesToDo](const FloatBufferLine &Buffer, ALfloat (&DelayBuf)[FrontStablizer::DelayLength], BandSplitter &Filter, ALfloat (&splitbuf)[2][BUFFERSIZE]) -> void { @@ -1520,7 +1525,7 @@ void ApplyStablizer(FrontStablizer *Stablizer, ALfloat (*Buffer)[BUFFERSIZE], co */ auto tmpbuf_end = std::begin(tmpbuf) + SamplesToDo; std::copy_n(std::begin(DelayBuf), FrontStablizer::DelayLength, tmpbuf_end); - std::reverse_copy(Buffer, Buffer+SamplesToDo, std::begin(tmpbuf)); + std::reverse_copy(Buffer.begin(), Buffer.begin()+SamplesToDo, std::begin(tmpbuf)); std::copy_n(std::begin(tmpbuf), FrontStablizer::DelayLength, std::begin(DelayBuf)); /* Apply an all-pass on the reversed signal, then reverse the samples @@ -1568,8 +1573,8 @@ void ApplyStablizer(FrontStablizer *Stablizer, ALfloat (*Buffer)[BUFFERSIZE], co } } -void ApplyDistanceComp(ALfloat (*Samples)[BUFFERSIZE], const DistanceComp &distcomp, - const ALsizei SamplesToDo, const ALsizei numchans) +void ApplyDistanceComp(FloatBufferLine *Samples, const DistanceComp &distcomp, + const ALsizei SamplesToDo, const ALsizei numchans) { ASSUME(SamplesToDo > 0); ASSUME(numchans > 0); @@ -1583,7 +1588,7 @@ void ApplyDistanceComp(ALfloat (*Samples)[BUFFERSIZE], const DistanceComp &distc if(base < 1) continue; - ALfloat *inout{al::assume_aligned<16>(Samples[c])}; + ALfloat *inout{al::assume_aligned<16>(Samples[c].data())}; auto inout_end = inout + SamplesToDo; if(LIKELY(SamplesToDo >= base)) { @@ -1599,8 +1604,8 @@ void ApplyDistanceComp(ALfloat (*Samples)[BUFFERSIZE], const DistanceComp &distc } } -void ApplyDither(ALfloat (*Samples)[BUFFERSIZE], ALuint *dither_seed, const ALfloat quant_scale, - const ALsizei SamplesToDo, const ALsizei numchans) +void ApplyDither(FloatBufferLine *Samples, ALuint *dither_seed, const ALfloat quant_scale, + const ALsizei SamplesToDo, const ALsizei numchans) { ASSUME(numchans > 0); @@ -1610,11 +1615,10 @@ void ApplyDither(ALfloat (*Samples)[BUFFERSIZE], ALuint *dither_seed, const ALfl */ const ALfloat invscale{1.0f / quant_scale}; ALuint seed{*dither_seed}; - auto dither_channel = [&seed,invscale,quant_scale,SamplesToDo](ALfloat *input) -> void + auto dither_channel = [&seed,invscale,quant_scale,SamplesToDo](FloatBufferLine &input) -> void { ASSUME(SamplesToDo > 0); - ALfloat *buffer{al::assume_aligned<16>(input)}; - auto dither_sample = [&seed,invscale,quant_scale](ALfloat sample) noexcept -> ALfloat + auto dither_sample = [&seed,invscale,quant_scale](const ALfloat sample) noexcept -> ALfloat { ALfloat val{sample * quant_scale}; ALuint rng0{dither_rng(&seed)}; @@ -1622,7 +1626,7 @@ void ApplyDither(ALfloat (*Samples)[BUFFERSIZE], ALuint *dither_seed, const ALfl val += static_cast(rng0*(1.0/UINT_MAX) - rng1*(1.0/UINT_MAX)); return fast_roundf(val) * invscale; }; - std::transform(buffer, buffer+SamplesToDo, buffer, dither_sample); + std::transform(input.begin(), input.begin()+SamplesToDo, input.begin(), dither_sample); }; std::for_each(Samples, Samples+numchans, dither_channel); *dither_seed = seed; @@ -1660,7 +1664,7 @@ template<> inline ALubyte SampleConv(ALfloat val) noexcept { return SampleConv(val) + 128; } template -void Write(const ALfloat (*InBuffer)[BUFFERSIZE], ALvoid *OutBuffer, const ALsizei Offset, +void Write(const FloatBufferLine *InBuffer, ALvoid *OutBuffer, const ALsizei Offset, const ALsizei SamplesToDo, const ALsizei numchans) { using SampleType = typename DevFmtTypeTraits::Type; @@ -1668,7 +1672,7 @@ void Write(const ALfloat (*InBuffer)[BUFFERSIZE], ALvoid *OutBuffer, const ALsiz ASSUME(Offset >= 0); ASSUME(numchans > 0); SampleType *outbase = static_cast(OutBuffer) + Offset*numchans; - auto conv_channel = [&outbase,SamplesToDo,numchans](const ALfloat *inbuf) -> void + auto conv_channel = [&outbase,SamplesToDo,numchans](const FloatBufferLine &inbuf) -> void { ASSUME(SamplesToDo > 0); SampleType *out{outbase++}; @@ -1677,7 +1681,7 @@ void Write(const ALfloat (*InBuffer)[BUFFERSIZE], ALvoid *OutBuffer, const ALsiz *out = SampleConv(s); out += numchans; }; - std::for_each(inbuf, inbuf+SamplesToDo, conv_sample); + std::for_each(inbuf.begin(), inbuf.begin()+SamplesToDo, conv_sample); }; std::for_each(InBuffer, InBuffer+numchans, conv_channel); } @@ -1758,7 +1762,7 @@ void aluMixData(ALCdevice *device, ALvoid *OutBuffer, ALsizei NumSamples) if(LIKELY(OutBuffer)) { - ALfloat (*Buffer)[BUFFERSIZE]{device->RealOut.Buffer}; + FloatBufferLine *Buffer{device->RealOut.Buffer}; ALsizei Channels{device->RealOut.NumChannels}; /* Finally, interleave and convert samples, writing to the device's diff --git a/Alc/bformatdec.cpp b/Alc/bformatdec.cpp index 563282a7..33659e04 100644 --- a/Alc/bformatdec.cpp +++ b/Alc/bformatdec.cpp @@ -146,7 +146,7 @@ BFormatDec::BFormatDec(const ALsizei inchans, const ALsizei chancount, } -void BFormatDec::process(ALfloat (*OutBuffer)[BUFFERSIZE], const ALsizei OutChannels, const ALfloat (*InSamples)[BUFFERSIZE], const ALsizei SamplesToDo) +void BFormatDec::process(FloatBufferLine *OutBuffer, const ALsizei OutChannels, const FloatBufferLine *InSamples, const ALsizei SamplesToDo) { ASSUME(OutChannels > 0); ASSUME(mNumChannels > 0); @@ -154,19 +154,19 @@ void BFormatDec::process(ALfloat (*OutBuffer)[BUFFERSIZE], const ALsizei OutChan if(mDualBand) { for(ALsizei i{0};i < mNumChannels;i++) - mXOver[i].process(mSamplesHF[i].data(), mSamplesLF[i].data(), InSamples[i], - SamplesToDo); + mXOver[i].process(mSamplesHF[i].data(), mSamplesLF[i].data(), InSamples[i].data(), + SamplesToDo); for(ALsizei chan{0};chan < OutChannels;chan++) { if(UNLIKELY(!(mEnabled&(1<(mSamplesHF[0]), + MixRowSamples(OutBuffer[chan].data(), mMatrix.Dual[chan][sHFBand], + &reinterpret_cast(mSamplesHF[0]), mNumChannels, 0, SamplesToDo); - MixRowSamples(OutBuffer[chan], mMatrix.Dual[chan][sLFBand], - &reinterpret_cast(mSamplesLF[0]), + MixRowSamples(OutBuffer[chan].data(), mMatrix.Dual[chan][sLFBand], + &reinterpret_cast(mSamplesLF[0]), mNumChannels, 0, SamplesToDo); } } @@ -177,8 +177,9 @@ void BFormatDec::process(ALfloat (*OutBuffer)[BUFFERSIZE], const ALsizei OutChan if(UNLIKELY(!(mEnabled&(1<(InSamples[0]), mNumChannels, 0, + SamplesToDo); } } } diff --git a/Alc/bformatdec.h b/Alc/bformatdec.h index d82f08ac..ef81b41d 100644 --- a/Alc/bformatdec.h +++ b/Alc/bformatdec.h @@ -43,8 +43,8 @@ public: const ALsizei (&chanmap)[MAX_OUTPUT_CHANNELS]); /* Decodes the ambisonic input to the given output channels. */ - void process(ALfloat (*OutBuffer)[BUFFERSIZE], const ALsizei OutChannels, - const ALfloat (*InSamples)[BUFFERSIZE], const ALsizei SamplesToDo); + void process(FloatBufferLine *OutBuffer, const ALsizei OutChannels, + const FloatBufferLine *InSamples, const ALsizei SamplesToDo); /* Retrieves per-order HF scaling factors for "upsampling" ambisonic data. */ static std::array GetHFOrderScales(const ALsizei in_order, diff --git a/Alc/effects/base.h b/Alc/effects/base.h index ba1fbf96..467fb5af 100644 --- a/Alc/effects/base.h +++ b/Alc/effects/base.h @@ -143,7 +143,7 @@ struct EffectTarget { struct EffectState { RefCount mRef{1u}; - ALfloat (*mOutBuffer)[BUFFERSIZE]{nullptr}; + FloatBufferLine *mOutBuffer{nullptr}; ALsizei mOutChannels{0}; diff --git a/Alc/mastering.cpp b/Alc/mastering.cpp index 23386cf4..2cf6acf5 100644 --- a/Alc/mastering.cpp +++ b/Alc/mastering.cpp @@ -97,7 +97,7 @@ void ShiftSlidingHold(SlidingHold *Hold, const ALsizei n) /* Multichannel compression is linked via the absolute maximum of all * channels. */ -void LinkChannels(Compressor *Comp, const ALsizei SamplesToDo, const ALfloat (*RESTRICT OutBuffer)[BUFFERSIZE]) +void LinkChannels(Compressor *Comp, const ALsizei SamplesToDo, const FloatBufferLine *OutBuffer) { const ALsizei index{Comp->mLookAhead}; const ALsizei numChans{Comp->mNumChans}; @@ -109,9 +109,9 @@ void LinkChannels(Compressor *Comp, const ALsizei SamplesToDo, const ALfloat (*R auto side_begin = std::begin(Comp->mSideChain) + index; std::fill(side_begin, side_begin+SamplesToDo, 0.0f); - auto fill_max = [SamplesToDo,side_begin](const ALfloat *input) -> void + auto fill_max = [SamplesToDo,side_begin](const FloatBufferLine &input) -> void { - const ALfloat *RESTRICT buffer{al::assume_aligned<16>(input)}; + const ALfloat *RESTRICT buffer{al::assume_aligned<16>(input.data())}; auto max_abs = std::bind(maxf, _1, std::bind(static_cast(std::fabs), _2)); std::transform(side_begin, side_begin+SamplesToDo, buffer, side_begin, max_abs); }; @@ -293,7 +293,7 @@ void GainCompressor(Compressor *Comp, const ALsizei SamplesToDo) * reaching the offending impulse. This is best used when operating as a * limiter. */ -void SignalDelay(Compressor *Comp, const ALsizei SamplesToDo, ALfloat (*RESTRICT OutBuffer)[BUFFERSIZE]) +void SignalDelay(Compressor *Comp, const ALsizei SamplesToDo, FloatBufferLine *OutBuffer) { static constexpr ALsizei mask{BUFFERSIZE - 1}; const ALsizei numChans{Comp->mNumChans}; @@ -305,7 +305,7 @@ void SignalDelay(Compressor *Comp, const ALsizei SamplesToDo, ALfloat (*RESTRICT for(ALsizei c{0};c < numChans;c++) { - ALfloat *RESTRICT inout{al::assume_aligned<16>(OutBuffer[c])}; + ALfloat *RESTRICT inout{al::assume_aligned<16>(OutBuffer[c].data())}; ALfloat *RESTRICT delay{al::assume_aligned<16>(Comp->mDelay[c])}; for(ALsizei i{0};i < SamplesToDo;i++) { @@ -425,7 +425,7 @@ Compressor::~Compressor() } -void Compressor::process(const ALsizei SamplesToDo, ALfloat (*OutBuffer)[BUFFERSIZE]) +void Compressor::process(const ALsizei SamplesToDo, FloatBufferLine *OutBuffer) { const ALsizei numChans{mNumChans}; @@ -435,9 +435,9 @@ void Compressor::process(const ALsizei SamplesToDo, ALfloat (*OutBuffer)[BUFFERS const ALfloat preGain{mPreGain}; if(preGain != 1.0f) { - auto apply_gain = [SamplesToDo,preGain](ALfloat *input) noexcept -> void + auto apply_gain = [SamplesToDo,preGain](FloatBufferLine &input) noexcept -> void { - ALfloat *buffer{al::assume_aligned<16>(input)}; + ALfloat *buffer{al::assume_aligned<16>(input.data())}; std::transform(buffer, buffer+SamplesToDo, buffer, std::bind(std::multiplies{}, _1, preGain)); }; @@ -460,9 +460,9 @@ void Compressor::process(const ALsizei SamplesToDo, ALfloat (*OutBuffer)[BUFFERS SignalDelay(this, SamplesToDo, OutBuffer); const ALfloat (&sideChain)[BUFFERSIZE*2] = mSideChain; - auto apply_comp = [SamplesToDo,&sideChain](ALfloat *input) noexcept -> void + auto apply_comp = [SamplesToDo,&sideChain](FloatBufferLine &input) noexcept -> void { - ALfloat *buffer{al::assume_aligned<16>(input)}; + ALfloat *buffer{al::assume_aligned<16>(input.data())}; const ALfloat *gains{al::assume_aligned<16>(&sideChain[0])}; std::transform(gains, gains+SamplesToDo, buffer, buffer, std::bind(std::multiplies{}, _1, _2)); diff --git a/Alc/mastering.h b/Alc/mastering.h index a9411bd0..6cc40ba3 100644 --- a/Alc/mastering.h +++ b/Alc/mastering.h @@ -6,7 +6,7 @@ #include "AL/al.h" #include "almalloc.h" -/* For BUFFERSIZE. */ +/* For FloatBufferLine/BUFFERSIZE. */ #include "alMain.h" @@ -65,7 +65,7 @@ struct Compressor { ~Compressor(); - void process(const ALsizei SamplesToDo, ALfloat (*OutBuffer)[BUFFERSIZE]); + void process(const ALsizei SamplesToDo, FloatBufferLine *OutBuffer); ALsizei getLookAhead() const noexcept { return mLookAhead; } DEF_PLACE_NEWDEL() diff --git a/Alc/mixvoice.cpp b/Alc/mixvoice.cpp index 284621ab..6df9f430 100644 --- a/Alc/mixvoice.cpp +++ b/Alc/mixvoice.cpp @@ -744,10 +744,9 @@ void MixVoice(ALvoice *voice, ALvoice::State vstate, const ALuint SourceID, ALCc hrtfparams.Gain = 0.0f; hrtfparams.GainStep = gain / static_cast(fademix); - MixHrtfBlendSamples( - voice->mDirect.Buffer[OutLIdx], voice->mDirect.Buffer[OutRIdx], - HrtfSamples, AccumSamples, OutPos, IrSize, &parms.Hrtf.Old, - &hrtfparams, fademix); + MixHrtfBlendSamples(voice->mDirect.Buffer[OutLIdx].data(), + voice->mDirect.Buffer[OutRIdx].data(), HrtfSamples, AccumSamples, + OutPos, IrSize, &parms.Hrtf.Old, &hrtfparams, fademix); /* Update the old parameters with the result. */ parms.Hrtf.Old = parms.Hrtf.Target; if(fademix < Counter) @@ -778,10 +777,9 @@ void MixVoice(ALvoice *voice, ALvoice::State vstate, const ALuint SourceID, ALCc hrtfparams.Gain = parms.Hrtf.Old.Gain; hrtfparams.GainStep = (gain - parms.Hrtf.Old.Gain) / static_cast(todo); - MixHrtfSamples( - voice->mDirect.Buffer[OutLIdx], voice->mDirect.Buffer[OutRIdx], - HrtfSamples+fademix, AccumSamples+fademix, OutPos+fademix, IrSize, - &hrtfparams, todo); + MixHrtfSamples(voice->mDirect.Buffer[OutLIdx].data(), + voice->mDirect.Buffer[OutRIdx].data(), HrtfSamples+fademix, + AccumSamples+fademix, OutPos+fademix, IrSize, &hrtfparams, todo); /* Store the interpolated gain or the final target gain * depending if the fade is done. */ @@ -803,8 +801,8 @@ void MixVoice(ALvoice *voice, ALvoice::State vstate, const ALuint SourceID, ALCc SilentTarget : parms.Gains.Target}; MixSamples(samples, voice->mDirect.ChannelsPerOrder[0], - voice->mDirect.Buffer, parms.Gains.Current, TargetGains, Counter, - OutPos, DstBufferSize); + &reinterpret_cast(voice->mDirect.Buffer[0]), + parms.Gains.Current, TargetGains, Counter, OutPos, DstBufferSize); ALfloat (&nfcsamples)[BUFFERSIZE] = Device->NfcSampleData; ALsizei chanoffset{voice->mDirect.ChannelsPerOrder[0]}; @@ -815,8 +813,9 @@ void MixVoice(ALvoice *voice, ALvoice::State vstate, const ALuint SourceID, ALCc return; (parms.NFCtrlFilter.*process)(nfcsamples, samples, DstBufferSize); MixSamples(nfcsamples, voice->mDirect.ChannelsPerOrder[order], - voice->mDirect.Buffer+chanoffset, parms.Gains.Current+chanoffset, - TargetGains+chanoffset, Counter, OutPos, DstBufferSize); + &reinterpret_cast(voice->mDirect.Buffer[chanoffset]), + parms.Gains.Current+chanoffset, TargetGains+chanoffset, Counter, + OutPos, DstBufferSize); chanoffset += voice->mDirect.ChannelsPerOrder[order]; }; apply_nfc(&NfcFilter::process1, 1); @@ -827,7 +826,8 @@ void MixVoice(ALvoice *voice, ALvoice::State vstate, const ALuint SourceID, ALCc { const ALfloat *TargetGains{UNLIKELY(vstate == ALvoice::Stopping) ? SilentTarget : parms.Gains.Target}; - MixSamples(samples, voice->mDirect.Channels, voice->mDirect.Buffer, + MixSamples(samples, voice->mDirect.Channels, + &reinterpret_cast(voice->mDirect.Buffer[0]), parms.Gains.Current, TargetGains, Counter, OutPos, DstBufferSize); } } @@ -844,7 +844,8 @@ void MixVoice(ALvoice *voice, ALvoice::State vstate, const ALuint SourceID, ALCc const ALfloat *TargetGains{UNLIKELY(vstate==ALvoice::Stopping) ? SilentTarget : parms.Gains.Target}; - MixSamples(samples, send.Channels, send.Buffer, parms.Gains.Current, + MixSamples(samples, send.Channels, + &reinterpret_cast(send.Buffer[0]), parms.Gains.Current, TargetGains, Counter, OutPos, DstBufferSize); }; std::for_each(voice->mSend.begin(), voice->mSend.end(), mix_send); diff --git a/Alc/panning.cpp b/Alc/panning.cpp index a209c6cf..d1d40927 100644 --- a/Alc/panning.cpp +++ b/Alc/panning.cpp @@ -976,6 +976,6 @@ void aluInitEffectPanning(ALeffectslot *slot, ALCdevice *device) { return BFChannelConfig{1.0f, acn}; } ); std::fill(iter, slot->Wet.AmbiMap.end(), BFChannelConfig{}); - slot->Wet.Buffer = &reinterpret_cast(slot->MixBuffer[0]); + slot->Wet.Buffer = slot->MixBuffer.data(); slot->Wet.NumChannels = static_cast(count); } diff --git a/Alc/uhjfilter.cpp b/Alc/uhjfilter.cpp index 64d5f76c..acdf6f0b 100644 --- a/Alc/uhjfilter.cpp +++ b/Alc/uhjfilter.cpp @@ -59,7 +59,7 @@ void allpass_process(AllPassState *state, ALfloat *dst, const ALfloat *src, cons * know which is the intended result. */ -void Uhj2Encoder::encode(ALfloat *LeftOut, ALfloat *RightOut, ALfloat (*InSamples)[BUFFERSIZE], const ALsizei SamplesToDo) +void Uhj2Encoder::encode(FloatBufferLine &LeftOut, FloatBufferLine &RightOut, FloatBufferLine *InSamples, const ALsizei SamplesToDo) { alignas(16) ALfloat D[MAX_UPDATE_SAMPLES], S[MAX_UPDATE_SAMPLES]; alignas(16) ALfloat temp[MAX_UPDATE_SAMPLES]; @@ -72,7 +72,7 @@ void Uhj2Encoder::encode(ALfloat *LeftOut, ALfloat *RightOut, ALfloat (*InSample ASSUME(todo > 0); /* D = 0.6554516*Y */ - const ALfloat *RESTRICT input{al::assume_aligned<16>(InSamples[2]+base)}; + const ALfloat *RESTRICT input{al::assume_aligned<16>(InSamples[2].data()+base)}; for(ALsizei i{0};i < todo;i++) temp[i] = 0.6554516f*input[i]; allpass_process(&mFilter1_Y[0], temp, temp, Filter1CoeffSqr[0], todo); @@ -89,8 +89,8 @@ void Uhj2Encoder::encode(ALfloat *LeftOut, ALfloat *RightOut, ALfloat (*InSample mLastY = temp[todo-1]; /* D += j(-0.3420201*W + 0.5098604*X) */ - const ALfloat *RESTRICT input0{al::assume_aligned<16>(InSamples[0]+base)}; - const ALfloat *RESTRICT input1{al::assume_aligned<16>(InSamples[1]+base)}; + const ALfloat *RESTRICT input0{al::assume_aligned<16>(InSamples[0].data()+base)}; + const ALfloat *RESTRICT input1{al::assume_aligned<16>(InSamples[1].data()+base)}; for(ALsizei i{0};i < todo;i++) temp[i] = -0.3420201f*input0[i] + 0.5098604f*input1[i]; allpass_process(&mFilter2_WX[0], temp, temp, Filter2CoeffSqr[0], todo); @@ -113,11 +113,11 @@ void Uhj2Encoder::encode(ALfloat *LeftOut, ALfloat *RightOut, ALfloat (*InSample mLastWX = temp[todo-1]; /* Left = (S + D)/2.0 */ - ALfloat *RESTRICT left = al::assume_aligned<16>(LeftOut+base); + ALfloat *RESTRICT left = al::assume_aligned<16>(LeftOut.data()+base); for(ALsizei i{0};i < todo;i++) left[i] += (S[i] + D[i]) * 0.5f; /* Right = (S - D)/2.0 */ - ALfloat *RESTRICT right = al::assume_aligned<16>(RightOut+base); + ALfloat *RESTRICT right = al::assume_aligned<16>(RightOut.data()+base); for(ALsizei i{0};i < todo;i++) right[i] += (S[i] - D[i]) * 0.5f; diff --git a/Alc/uhjfilter.h b/Alc/uhjfilter.h index 1351491b..181e036a 100644 --- a/Alc/uhjfilter.h +++ b/Alc/uhjfilter.h @@ -45,7 +45,8 @@ struct Uhj2Encoder { /* Encodes a 2-channel UHJ (stereo-compatible) signal from a B-Format input * signal. The input must use FuMa channel ordering and scaling. */ - void encode(ALfloat *LeftOut, ALfloat *RightOut, ALfloat (*InSamples)[BUFFERSIZE], const ALsizei SamplesToDo); + void encode(FloatBufferLine &LeftOut, FloatBufferLine &RightOut, FloatBufferLine *InSamples, + const ALsizei SamplesToDo); DEF_NEWDEL(Uhj2Encoder) }; -- cgit v1.2.3