diff options
Diffstat (limited to 'alc/voice.cpp')
| -rw-r--r-- | alc/voice.cpp | 140 |
1 files changed, 83 insertions, 57 deletions
diff --git a/alc/voice.cpp b/alc/voice.cpp index f9eca51c..c3e3dca2 100644 --- a/alc/voice.cpp +++ b/alc/voice.cpp @@ -55,6 +55,7 @@ #include "core/logging.h" #include "core/mixer/defs.h" #include "core/mixer/hrtfdefs.h" +#include "core/resampler_limits.h" #include "hrtf.h" #include "inprogext.h" #include "opthelpers.h" @@ -81,8 +82,6 @@ MixerFunc MixSamples{Mix_<CTag>}; namespace { -constexpr uint ResamplerPrePadding{MaxResamplerPadding / 2}; - using HrtfMixerFunc = void(*)(const float *InSamples, float2 *AccumSamples, const uint IrSize, const MixHrtfFilter *hrtfparams, const size_t BufferSize); using HrtfMixerBlendFunc = void(*)(const float *InSamples, float2 *AccumSamples, @@ -224,17 +223,32 @@ const float *DoFilters(BiquadFilter &lpfilter, BiquadFilter &hpfilter, float *ds void LoadSamples(const al::span<Voice::BufferLine> dstSamples, const size_t dstOffset, - const al::byte *src, const size_t srcOffset, const size_t srcstep, FmtType srctype, + const al::byte *src, const size_t srcOffset, const FmtType srctype, const FmtChannels srcchans, const size_t samples) noexcept { #define HANDLE_FMT(T) case T: \ { \ constexpr size_t sampleSize{sizeof(al::FmtTypeTraits<T>::Type)}; \ - src += srcOffset*srcstep*sampleSize; \ - for(auto &dst : dstSamples) \ + if(srcchans == FmtUHJ2) \ + { \ + constexpr size_t srcstep{2u}; \ + src += srcOffset*srcstep*sampleSize; \ + al::LoadSampleArray<T>(dstSamples[0].data() + dstOffset, src, \ + srcstep, samples); \ + al::LoadSampleArray<T>(dstSamples[1].data() + dstOffset, \ + src + sampleSize, srcstep, samples); \ + std::fill_n(dstSamples[2].data() + dstOffset, samples, 0.0f); \ + } \ + else \ { \ - al::LoadSampleArray<T>(dst.data() + dstOffset, src, srcstep, samples); \ - src += sampleSize; \ + const size_t srcstep{dstSamples.size()}; \ + src += srcOffset*srcstep*sampleSize; \ + for(auto &dst : dstSamples) \ + { \ + al::LoadSampleArray<T>(dst.data() + dstOffset, src, srcstep, \ + samples); \ + src += sampleSize; \ + } \ } \ } \ break @@ -252,10 +266,9 @@ void LoadSamples(const al::span<Voice::BufferLine> dstSamples, const size_t dstO } void LoadBufferStatic(VoiceBufferItem *buffer, VoiceBufferItem *bufferLoopItem, - const size_t dataPosInt, const FmtType sampleType, const size_t samplesToLoad, - const al::span<Voice::BufferLine> voiceSamples) + const size_t dataPosInt, const FmtType sampleType, const FmtChannels sampleChannels, + const size_t samplesToLoad, const al::span<Voice::BufferLine> voiceSamples) { - const size_t numChannels{voiceSamples.size()}; const uint loopStart{buffer->mLoopStart}; const uint loopEnd{buffer->mLoopEnd}; ASSUME(loopEnd > loopStart); @@ -265,14 +278,14 @@ void LoadBufferStatic(VoiceBufferItem *buffer, VoiceBufferItem *bufferLoopItem, { /* Load what's left to play from the buffer */ const size_t remaining{minz(samplesToLoad, buffer->mSampleLen-dataPosInt)}; - LoadSamples(voiceSamples, ResamplerPrePadding, buffer->mSamples, dataPosInt, numChannels, - sampleType, remaining); + LoadSamples(voiceSamples, MaxResamplerEdge, buffer->mSamples, dataPosInt, sampleType, + sampleChannels, remaining); if(const size_t toFill{samplesToLoad - remaining}) { for(auto &chanbuffer : voiceSamples) { - auto srcsamples = chanbuffer.data() + ResamplerPrePadding - 1 + remaining; + auto srcsamples = chanbuffer.data() + MaxResamplerEdge - 1 + remaining; std::fill_n(srcsamples + 1, toFill, *srcsamples); } } @@ -281,46 +294,44 @@ void LoadBufferStatic(VoiceBufferItem *buffer, VoiceBufferItem *bufferLoopItem, { /* Load what's left of this loop iteration */ const size_t remaining{minz(samplesToLoad, loopEnd-dataPosInt)}; - LoadSamples(voiceSamples, ResamplerPrePadding, buffer->mSamples, dataPosInt, numChannels, - sampleType, remaining); + LoadSamples(voiceSamples, MaxResamplerEdge, buffer->mSamples, dataPosInt, sampleType, + sampleChannels, remaining); /* Load repeats of the loop to fill the buffer. */ const auto loopSize = static_cast<size_t>(loopEnd - loopStart); size_t samplesLoaded{remaining}; while(const size_t toFill{minz(samplesToLoad - samplesLoaded, loopSize)}) { - LoadSamples(voiceSamples, ResamplerPrePadding + samplesLoaded, buffer->mSamples, - loopStart, numChannels, sampleType, toFill); + LoadSamples(voiceSamples, MaxResamplerEdge + samplesLoaded, buffer->mSamples, + loopStart, sampleType, sampleChannels, toFill); samplesLoaded += toFill; } } } void LoadBufferCallback(VoiceBufferItem *buffer, const size_t numCallbackSamples, - const FmtType sampleType, const size_t samplesToLoad, + const FmtType sampleType, const FmtChannels sampleChannels, const size_t samplesToLoad, const al::span<Voice::BufferLine> voiceSamples) { - const size_t numChannels{voiceSamples.size()}; /* Load what's left to play from the buffer */ const size_t remaining{minz(samplesToLoad, numCallbackSamples)}; - LoadSamples(voiceSamples, ResamplerPrePadding, buffer->mSamples, 0, numChannels, sampleType, + LoadSamples(voiceSamples, MaxResamplerEdge, buffer->mSamples, 0, sampleType, sampleChannels, remaining); if(const size_t toFill{samplesToLoad - remaining}) { for(auto &chanbuffer : voiceSamples) { - auto srcsamples = chanbuffer.data() + ResamplerPrePadding - 1 + remaining; + auto srcsamples = chanbuffer.data() + MaxResamplerEdge - 1 + remaining; std::fill_n(srcsamples + 1, toFill, *srcsamples); } } } void LoadBufferQueue(VoiceBufferItem *buffer, VoiceBufferItem *bufferLoopItem, - size_t dataPosInt, const FmtType sampleType, const size_t samplesToLoad, - const al::span<Voice::BufferLine> voiceSamples) + size_t dataPosInt, const FmtType sampleType, const FmtChannels sampleChannels, + const size_t samplesToLoad, const al::span<Voice::BufferLine> voiceSamples) { - const size_t numChannels{voiceSamples.size()}; /* Crawl the buffer queue to fill in the temp buffer */ size_t samplesLoaded{0}; while(buffer && samplesLoaded != samplesToLoad) @@ -334,8 +345,8 @@ void LoadBufferQueue(VoiceBufferItem *buffer, VoiceBufferItem *bufferLoopItem, } const size_t remaining{minz(samplesToLoad-samplesLoaded, buffer->mSampleLen-dataPosInt)}; - LoadSamples(voiceSamples, ResamplerPrePadding+samplesLoaded, buffer->mSamples, dataPosInt, - numChannels, sampleType, remaining); + LoadSamples(voiceSamples, MaxResamplerEdge+samplesLoaded, buffer->mSamples, dataPosInt, + sampleType, sampleChannels, remaining); samplesLoaded += remaining; if(samplesLoaded == samplesToLoad) @@ -350,7 +361,7 @@ void LoadBufferQueue(VoiceBufferItem *buffer, VoiceBufferItem *bufferLoopItem, size_t chanidx{0}; for(auto &chanbuffer : voiceSamples) { - auto srcsamples = chanbuffer.data() + ResamplerPrePadding - 1 + samplesLoaded; + auto srcsamples = chanbuffer.data() + MaxResamplerEdge - 1 + samplesLoaded; std::fill_n(srcsamples + 1, toFill, *srcsamples); ++chanidx; } @@ -517,6 +528,8 @@ void Voice::mix(const State vstate, ALCcontext *Context, const uint SamplesToDo) else if UNLIKELY(!BufferListItem) Counter = std::min(Counter, 64u); + const uint PostPadding{MaxResamplerEdge + + ((mFmtChannels==FmtUHJ2) ? uint{UhjDecoder::sFilterDelay} : 0u)}; uint buffers_done{0u}; uint OutPos{0u}; do { @@ -531,7 +544,7 @@ void Voice::mix(const State vstate, ALCcontext *Context, const uint SamplesToDo) /* Calculate the last read src sample pos. */ DataSize64 = (DataSize64*increment + DataPosFrac) >> MixerFracBits; /* +1 to get the src sample count, include padding. */ - DataSize64 += 1 + ResamplerPrePadding; + DataSize64 += 1 + PostPadding; /* Result is guaranteed to be <= BufferLineSize+ResamplerPrePadding * since we won't use more src samples than dst samples+padding. @@ -543,18 +556,18 @@ void Voice::mix(const State vstate, ALCcontext *Context, const uint SamplesToDo) uint64_t DataSize64{DstBufferSize}; /* Calculate the end src sample pos, include padding. */ DataSize64 = (DataSize64*increment + DataPosFrac) >> MixerFracBits; - DataSize64 += ResamplerPrePadding; + DataSize64 += PostPadding; - if(DataSize64 <= BufferLineSize + ResamplerPrePadding) + if(DataSize64 <= LineSize - MaxResamplerEdge) SrcBufferSize = static_cast<uint>(DataSize64); else { /* If the source size got saturated, we can't fill the desired * dst size. Figure out how many samples we can actually mix. */ - SrcBufferSize = BufferLineSize + ResamplerPrePadding; + SrcBufferSize = LineSize - MaxResamplerEdge; - DataSize64 = SrcBufferSize - ResamplerPrePadding; + DataSize64 = SrcBufferSize - PostPadding; DataSize64 = ((DataSize64<<MixerFracBits) - DataPosFrac) / increment; if(DataSize64 < DstBufferSize) { @@ -563,6 +576,7 @@ void Voice::mix(const State vstate, ALCcontext *Context, const uint SamplesToDo) */ DstBufferSize = static_cast<uint>(DataSize64) & ~3u; } + ASSUME(DstBufferSize > 0); } } @@ -570,11 +584,8 @@ void Voice::mix(const State vstate, ALCcontext *Context, const uint SamplesToDo) { if(SrcBufferSize > mNumCallbackSamples) { - const size_t FrameSize{mChans.size() * mSampleSize}; - ASSUME(FrameSize > 0); - - const size_t byteOffset{mNumCallbackSamples*FrameSize}; - const size_t needBytes{SrcBufferSize*FrameSize - byteOffset}; + const size_t byteOffset{mNumCallbackSamples*mFrameSize}; + const size_t needBytes{SrcBufferSize*mFrameSize - byteOffset}; const int gotBytes{BufferListItem->mCallback(BufferListItem->mUserData, &BufferListItem->mSamples[byteOffset], static_cast<int>(needBytes))}; @@ -584,7 +595,7 @@ void Voice::mix(const State vstate, ALCcontext *Context, const uint SamplesToDo) { mFlags |= VoiceCallbackStopped; mNumCallbackSamples += static_cast<uint>(static_cast<uint>(gotBytes) / - FrameSize); + mFrameSize); } else mNumCallbackSamples = SrcBufferSize; @@ -595,7 +606,8 @@ void Voice::mix(const State vstate, ALCcontext *Context, const uint SamplesToDo) { for(auto &chanbuffer : mVoiceSamples) { - auto srciter = chanbuffer.data() + ResamplerPrePadding; + auto srciter = chanbuffer.data() + MaxResamplerEdge; + auto srcend = chanbuffer.data() + MaxResamplerPadding; /* When loading from a voice that ended prematurely, only take * the samples that get closest to 0 amplitude. This helps @@ -603,29 +615,41 @@ void Voice::mix(const State vstate, ALCcontext *Context, const uint SamplesToDo) */ auto abs_lt = [](const float lhs, const float rhs) noexcept -> bool { return std::abs(lhs) < std::abs(rhs); }; - srciter = std::min_element(srciter, srciter+(MaxResamplerPadding>>1), abs_lt); + srciter = std::min_element(srciter, srcend, abs_lt); - std::fill(srciter+1, chanbuffer.data() + ResamplerPrePadding + SrcBufferSize, - *srciter); + SrcBufferSize = SrcBufferSize - PostPadding + MaxResamplerPadding; + std::fill(srciter+1, chanbuffer.data() + SrcBufferSize, *srciter); } } - else if((mFlags&VoiceIsStatic)) - LoadBufferStatic(BufferListItem, BufferLoopItem, DataPosInt, mFmtType, SrcBufferSize, - mVoiceSamples); - else if((mFlags&VoiceIsCallback)) - LoadBufferCallback(BufferListItem, mNumCallbackSamples, mFmtType, SrcBufferSize, - mVoiceSamples); else - LoadBufferQueue(BufferListItem, BufferLoopItem, DataPosInt, mFmtType, SrcBufferSize, - mVoiceSamples); + { + if((mFlags&VoiceIsStatic)) + LoadBufferStatic(BufferListItem, BufferLoopItem, DataPosInt, mFmtType, mFmtChannels, + SrcBufferSize, mVoiceSamples); + else if((mFlags&VoiceIsCallback)) + LoadBufferCallback(BufferListItem, mNumCallbackSamples, mFmtType, mFmtChannels, + SrcBufferSize, mVoiceSamples); + else + LoadBufferQueue(BufferListItem, BufferLoopItem, DataPosInt, mFmtType, mFmtChannels, + SrcBufferSize, mVoiceSamples); + + if(mDecoder) + { + std::array<float*,3> samples{{mVoiceSamples[0].data() + MaxResamplerEdge, + mVoiceSamples[1].data() + MaxResamplerEdge, + mVoiceSamples[2].data() + MaxResamplerEdge}}; + const size_t srcOffset{(increment*DstBufferSize + DataPosFrac)>>MixerFracBits}; + SrcBufferSize = SrcBufferSize - PostPadding + MaxResamplerEdge; + mDecoder->decode(samples, SrcBufferSize, srcOffset); + } + } - ASSUME(DstBufferSize > 0); auto voiceSamples = mVoiceSamples.begin(); for(auto &chandata : mChans) { /* Resample, then apply ambisonic upsampling as needed. */ float *ResampledData{Resample(&mResampleState, - voiceSamples->data() + ResamplerPrePadding, DataPosFrac, increment, + voiceSamples->data() + MaxResamplerEdge, DataPosFrac, increment, {Device->ResampledData, DstBufferSize})}; if((mFlags&VoiceIsAmbisonic)) chandata.mAmbiSplitter.processHfScale({ResampledData, DstBufferSize}, @@ -720,11 +744,8 @@ void Voice::mix(const State vstate, ALCcontext *Context, const uint SamplesToDo) { if(SrcSamplesDone < mNumCallbackSamples) { - const size_t FrameSize{mChans.size() * mSampleSize}; - ASSUME(FrameSize > 0); - - const size_t byteOffset{SrcSamplesDone*FrameSize}; - const size_t byteEnd{mNumCallbackSamples*FrameSize}; + const size_t byteOffset{SrcSamplesDone*mFrameSize}; + const size_t byteEnd{mNumCallbackSamples*mFrameSize}; al::byte *data{BufferListItem->mSamples}; std::copy(data+byteOffset, data+byteEnd, data); mNumCallbackSamples -= SrcSamplesDone; @@ -802,6 +823,11 @@ void Voice::mix(const State vstate, ALCcontext *Context, const uint SamplesToDo) void Voice::prepare(ALCdevice *device) { + if(mFmtChannels == FmtUHJ2 && !mDecoder) + mDecoder = std::make_unique<UhjDecoder>(); + else if(mFmtChannels != FmtUHJ2) + mDecoder = nullptr; + /* Clear the stepping value explicitly so the mixer knows not to mix this * until the update gets applied. */ |