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-rw-r--r--alc/alu.cpp17
-rw-r--r--alc/voice.cpp140
-rw-r--r--alc/voice.h15
3 files changed, 103 insertions, 69 deletions
diff --git a/alc/alu.cpp b/alc/alu.cpp
index 67bb8ebc..86cbefa5 100644
--- a/alc/alu.cpp
+++ b/alc/alu.cpp
@@ -789,23 +789,18 @@ void CalcPanningAndFilters(Voice *voice, const float xpos, const float ypos, con
case FmtBFormat2D:
case FmtBFormat3D:
- DirectChannels = DirectMode::Off;
- break;
-
- /* TODO: UHJ2 should be treated as BFormat2D for panning. */
case FmtUHJ2:
DirectChannels = DirectMode::Off;
- chans = StereoMap;
- downmix_gain = 1.0f / 2.0f;
break;
}
voice->mFlags &= ~(VoiceHasHrtf | VoiceHasNfc);
- if(voice->mFmtChannels == FmtBFormat2D || voice->mFmtChannels == FmtBFormat3D)
+ if(voice->mFmtChannels == FmtBFormat2D || voice->mFmtChannels == FmtBFormat3D
+ || voice->mFmtChannels == FmtUHJ2)
{
/* Special handling for B-Format sources. */
- if(Device->AvgSpeakerDist > 0.0f)
+ if(Device->AvgSpeakerDist > 0.0f && voice->mFmtChannels != FmtUHJ2)
{
if(!(Distance > std::numeric_limits<float>::epsilon()))
{
@@ -904,7 +899,8 @@ void CalcPanningAndFilters(Voice *voice, const float xpos, const float ypos, con
/* Convert the rotation matrix for input ordering and scaling, and
* whether input is 2D or 3D.
*/
- const uint8_t *index_map{(voice->mFmtChannels == FmtBFormat2D) ?
+ const uint8_t *index_map{
+ (voice->mFmtChannels == FmtBFormat2D || voice->mFmtChannels == FmtUHJ2) ?
GetAmbi2DLayout(voice->mAmbiLayout).data() :
GetAmbiLayout(voice->mAmbiLayout).data()};
@@ -1561,7 +1557,8 @@ void CalcSourceParams(Voice *voice, ALCcontext *context, bool force)
}
if((voice->mProps.DirectChannels != DirectMode::Off && voice->mFmtChannels != FmtMono
- && voice->mFmtChannels != FmtBFormat2D && voice->mFmtChannels != FmtBFormat3D)
+ && voice->mFmtChannels != FmtBFormat2D && voice->mFmtChannels != FmtBFormat3D
+ && voice->mFmtChannels != FmtUHJ2)
|| voice->mProps.mSpatializeMode==SpatializeMode::Off
|| (voice->mProps.mSpatializeMode==SpatializeMode::Auto && voice->mFmtChannels != FmtMono))
CalcNonAttnSourceParams(voice, &voice->mProps, context);
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.
*/
diff --git a/alc/voice.h b/alc/voice.h
index 96975efa..8f3476f1 100644
--- a/alc/voice.h
+++ b/alc/voice.h
@@ -15,6 +15,7 @@
#include "core/filters/splitter.h"
#include "core/mixer/defs.h"
#include "core/mixer/hrtfdefs.h"
+#include "core/uhjfilter.h"
#include "vector.h"
struct ALCcontext;
@@ -37,6 +38,12 @@ enum class DirectMode : unsigned char {
};
+/* Maximum number of extra source samples that may need to be loaded, for
+ * resampling or conversion purposes.
+ */
+constexpr uint MaxPostVoiceLoad{MaxResamplerEdge + UhjDecoder::sFilterDelay};
+
+
enum {
AF_None = 0,
AF_LowPass = 1,
@@ -191,11 +198,13 @@ struct Voice {
FmtChannels mFmtChannels;
FmtType mFmtType;
uint mFrequency;
- uint mSampleSize;
+ uint mFrameSize;
AmbiLayout mAmbiLayout;
AmbiScaling mAmbiScaling;
uint mAmbiOrder;
+ std::unique_ptr<UhjDecoder> mDecoder;
+
/** Current target parameters used for mixing. */
uint mStep{0};
@@ -218,7 +227,9 @@ struct Voice {
* now current (which may be overwritten if the buffer data is still
* available).
*/
- using BufferLine = std::array<float,BufferLineSize+MaxResamplerPadding>;
+ static constexpr size_t LineSize{BufferLineSize + MaxResamplerPadding +
+ UhjDecoder::sFilterDelay};
+ using BufferLine = std::array<float,LineSize>;
al::vector<BufferLine,16> mVoiceSamples{2};
struct ChannelData {
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-28 22:49:41 +0000