update openal-soft

sync point: master-ac5d40e40a0155351fe1be4aab30017b6a13a859

Engine/lib/openal-soft/Alc/alcmain.h

@@ -0,0 +1,371 @@
+#ifndef ALC_MAIN_H
+#define ALC_MAIN_H
+
+#include <algorithm>
+#include <array>
+#include <atomic>
+#include <bitset>
+#include <chrono>
+#include <cstdint>
+#include <cstddef>
+#include <memory>
+#include <mutex>
+#include <string>
+#include <thread>
+#include <utility>
+
+#include "AL/al.h"
+#include "AL/alc.h"
+#include "AL/alext.h"
+
+#include "almalloc.h"
+#include "alnumeric.h"
+#include "alspan.h"
+#include "atomic.h"
+#include "core/ambidefs.h"
+#include "core/bufferline.h"
+#include "core/devformat.h"
+#include "core/filters/splitter.h"
+#include "core/mixer/defs.h"
+#include "hrtf.h"
+#include "inprogext.h"
+#include "intrusive_ptr.h"
+#include "vector.h"
+
+class BFormatDec;
+struct ALbuffer;
+struct ALeffect;
+struct ALfilter;
+struct BackendBase;
+struct Compressor;
+struct EffectState;
+struct Uhj2Encoder;
+struct bs2b;
+
+using uint = unsigned int;
+
+
+#define MIN_OUTPUT_RATE      8000
+#define MAX_OUTPUT_RATE      192000
+#define DEFAULT_OUTPUT_RATE  44100
+
+#define DEFAULT_UPDATE_SIZE  882 /* 20ms */
+#define DEFAULT_NUM_UPDATES  3
+
+
+enum class DeviceType : unsigned char {
+    Playback,
+    Capture,
+    Loopback
+};
+
+
+enum class RenderMode : unsigned char {
+    Normal,
+    Pairwise,
+    Hrtf
+};
+
+
+struct InputRemixMap {
+    struct TargetMix { Channel channel; float mix; };
+
+    Channel channel;
+    std::array<TargetMix,2> targets;
+};
+
+
+struct BufferSubList {
+    uint64_t FreeMask{~0_u64};
+    ALbuffer *Buffers{nullptr}; /* 64 */
+
+    BufferSubList() noexcept = default;
+    BufferSubList(const BufferSubList&) = delete;
+    BufferSubList(BufferSubList&& rhs) noexcept : FreeMask{rhs.FreeMask}, Buffers{rhs.Buffers}
+    { rhs.FreeMask = ~0_u64; rhs.Buffers = nullptr; }
+    ~BufferSubList();
+
+    BufferSubList& operator=(const BufferSubList&) = delete;
+    BufferSubList& operator=(BufferSubList&& rhs) noexcept
+    { std::swap(FreeMask, rhs.FreeMask); std::swap(Buffers, rhs.Buffers); return *this; }
+};
+
+struct EffectSubList {
+    uint64_t FreeMask{~0_u64};
+    ALeffect *Effects{nullptr}; /* 64 */
+
+    EffectSubList() noexcept = default;
+    EffectSubList(const EffectSubList&) = delete;
+    EffectSubList(EffectSubList&& rhs) noexcept : FreeMask{rhs.FreeMask}, Effects{rhs.Effects}
+    { rhs.FreeMask = ~0_u64; rhs.Effects = nullptr; }
+    ~EffectSubList();
+
+    EffectSubList& operator=(const EffectSubList&) = delete;
+    EffectSubList& operator=(EffectSubList&& rhs) noexcept
+    { std::swap(FreeMask, rhs.FreeMask); std::swap(Effects, rhs.Effects); return *this; }
+};
+
+struct FilterSubList {
+    uint64_t FreeMask{~0_u64};
+    ALfilter *Filters{nullptr}; /* 64 */
+
+    FilterSubList() noexcept = default;
+    FilterSubList(const FilterSubList&) = delete;
+    FilterSubList(FilterSubList&& rhs) noexcept : FreeMask{rhs.FreeMask}, Filters{rhs.Filters}
+    { rhs.FreeMask = ~0_u64; rhs.Filters = nullptr; }
+    ~FilterSubList();
+
+    FilterSubList& operator=(const FilterSubList&) = delete;
+    FilterSubList& operator=(FilterSubList&& rhs) noexcept
+    { std::swap(FreeMask, rhs.FreeMask); std::swap(Filters, rhs.Filters); return *this; }
+};
+
+
+/* Maximum delay in samples for speaker distance compensation. */
+#define MAX_DELAY_LENGTH 1024
+
+struct DistanceComp {
+    struct ChanData {
+        float Gain{1.0f};
+        uint Length{0u}; /* Valid range is [0...MAX_DELAY_LENGTH). */
+        float *Buffer{nullptr};
+    };
+
+    std::array<ChanData,MAX_OUTPUT_CHANNELS> mChannels;
+    al::FlexArray<float,16> mSamples;
+
+    DistanceComp(size_t count) : mSamples{count} { }
+
+    static std::unique_ptr<DistanceComp> Create(size_t numsamples)
+    { return std::unique_ptr<DistanceComp>{new(FamCount(numsamples)) DistanceComp{numsamples}}; }
+
+    DEF_FAM_NEWDEL(DistanceComp, mSamples)
+};
+
+
+struct BFChannelConfig {
+    float Scale;
+    uint Index;
+};
+
+
+struct MixParams {
+    /* Coefficient channel mapping for mixing to the buffer. */
+    std::array<BFChannelConfig,MAX_OUTPUT_CHANNELS> AmbiMap{};
+
+    al::span<FloatBufferLine> Buffer;
+};
+
+struct RealMixParams {
+    al::span<const InputRemixMap> RemixMap;
+    std::array<uint,MaxChannels> ChannelIndex{};
+
+    al::span<FloatBufferLine> Buffer;
+};
+
+enum {
+    // Frequency was requested by the app or config file
+    FrequencyRequest,
+    // Channel configuration was requested by the config file
+    ChannelsRequest,
+    // Sample type was requested by the config file
+    SampleTypeRequest,
+
+    // Specifies if the DSP is paused at user request
+    DevicePaused,
+    // Specifies if the device is currently running
+    DeviceRunning,
+
+    DeviceFlagsCount
+};
+
+struct ALCdevice : public al::intrusive_ref<ALCdevice> {
+    std::atomic<bool> Connected{true};
+    const DeviceType Type{};
+
+    uint Frequency{};
+    uint UpdateSize{};
+    uint BufferSize{};
+
+    DevFmtChannels FmtChans{};
+    DevFmtType FmtType{};
+    bool IsHeadphones{false};
+    uint mAmbiOrder{0};
+    float mXOverFreq{400.0f};
+    /* For DevFmtAmbi* output only, specifies the channel order and
+     * normalization.
+     */
+    DevAmbiLayout mAmbiLayout{DevAmbiLayout::Default};
+    DevAmbiScaling mAmbiScale{DevAmbiScaling::Default};
+
+    std::string DeviceName;
+
+    // Device flags
+    std::bitset<DeviceFlagsCount> Flags{};
+
+    // Maximum number of sources that can be created
+    uint SourcesMax{};
+    // Maximum number of slots that can be created
+    uint AuxiliaryEffectSlotMax{};
+
+    /* Rendering mode. */
+    RenderMode mRenderMode{RenderMode::Normal};
+
+    /* The average speaker distance as determined by the ambdec configuration,
+     * HRTF data set, or the NFC-HOA reference delay. Only used for NFC.
+     */
+    float AvgSpeakerDist{0.0f};
+
+    uint SamplesDone{0u};
+    std::chrono::nanoseconds ClockBase{0};
+    std::chrono::nanoseconds FixedLatency{0};
+
+    /* Temp storage used for mixer processing. */
+    alignas(16) float SourceData[BufferLineSize + MaxResamplerPadding];
+    alignas(16) float ResampledData[BufferLineSize];
+    alignas(16) float FilteredData[BufferLineSize];
+    union {
+        alignas(16) float HrtfSourceData[BufferLineSize + HrtfHistoryLength];
+        alignas(16) float NfcSampleData[BufferLineSize];
+    };
+
+    /* Persistent storage for HRTF mixing. */
+    alignas(16) float2 HrtfAccumData[BufferLineSize + HrirLength + HrtfDirectDelay];
+
+    /* Mixing buffer used by the Dry mix and Real output. */
+    al::vector<FloatBufferLine, 16> MixBuffer;
+
+    /* The "dry" path corresponds to the main output. */
+    MixParams Dry;
+    uint NumChannelsPerOrder[MaxAmbiOrder+1]{};
+
+    /* "Real" output, which will be written to the device buffer. May alias the
+     * dry buffer.
+     */
+    RealMixParams RealOut;
+
+    /* HRTF state and info */
+    std::unique_ptr<DirectHrtfState> mHrtfState;
+    al::intrusive_ptr<HrtfStore> mHrtf;
+    uint mIrSize{0};
+
+    /* Ambisonic-to-UHJ encoder */
+    std::unique_ptr<Uhj2Encoder> Uhj_Encoder;
+
+    /* Ambisonic decoder for speakers */
+    std::unique_ptr<BFormatDec> AmbiDecoder;
+
+    /* Stereo-to-binaural filter */
+    std::unique_ptr<bs2b> Bs2b;
+
+    using PostProc = void(ALCdevice::*)(const size_t SamplesToDo);
+    PostProc PostProcess{nullptr};
+
+    std::unique_ptr<Compressor> Limiter;
+
+    /* Delay buffers used to compensate for speaker distances. */
+    std::unique_ptr<DistanceComp> ChannelDelays;
+
+    /* Dithering control. */
+    float DitherDepth{0.0f};
+    uint DitherSeed{0u};
+
+    /* Running count of the mixer invocations, in 31.1 fixed point. This
+     * actually increments *twice* when mixing, first at the start and then at
+     * the end, so the bottom bit indicates if the device is currently mixing
+     * and the upper bits indicates how many mixes have been done.
+     */
+    RefCount MixCount{0u};
+
+    // Contexts created on this device
+    std::atomic<al::FlexArray<ALCcontext*>*> mContexts{nullptr};
+
+    /* This lock protects the device state (format, update size, etc) from
+     * being from being changed in multiple threads, or being accessed while
+     * being changed. It's also used to serialize calls to the backend.
+     */
+    std::mutex StateLock;
+    std::unique_ptr<BackendBase> Backend;
+
+
+    ALCuint NumMonoSources{};
+    ALCuint NumStereoSources{};
+    ALCuint NumAuxSends{};
+
+    std::string HrtfName;
+    al::vector<std::string> HrtfList;
+    ALCenum HrtfStatus{ALC_FALSE};
+
+    ALCenum LimiterState{ALC_DONT_CARE_SOFT};
+
+    std::atomic<ALCenum> LastError{ALC_NO_ERROR};
+
+    // Map of Buffers for this device
+    std::mutex BufferLock;
+    al::vector<BufferSubList> BufferList;
+
+    // Map of Effects for this device
+    std::mutex EffectLock;
+    al::vector<EffectSubList> EffectList;
+
+    // Map of Filters for this device
+    std::mutex FilterLock;
+    al::vector<FilterSubList> FilterList;
+
+
+    ALCdevice(DeviceType type);
+    ALCdevice(const ALCdevice&) = delete;
+    ALCdevice& operator=(const ALCdevice&) = delete;
+    ~ALCdevice();
+
+    uint bytesFromFmt() const noexcept { return BytesFromDevFmt(FmtType); }
+    uint channelsFromFmt() const noexcept { return ChannelsFromDevFmt(FmtChans, mAmbiOrder); }
+    uint frameSizeFromFmt() const noexcept { return bytesFromFmt() * channelsFromFmt(); }
+
+    uint waitForMix() const noexcept
+    {
+        uint refcount;
+        while((refcount=MixCount.load(std::memory_order_acquire))&1) {
+        }
+        return refcount;
+    }
+
+    void ProcessHrtf(const size_t SamplesToDo);
+    void ProcessAmbiDec(const size_t SamplesToDo);
+    void ProcessAmbiDecStablized(const size_t SamplesToDo);
+    void ProcessUhj(const size_t SamplesToDo);
+    void ProcessBs2b(const size_t SamplesToDo);
+
+    inline void postProcess(const size_t SamplesToDo)
+    { if LIKELY(PostProcess) (this->*PostProcess)(SamplesToDo); }
+
+    void renderSamples(void *outBuffer, const uint numSamples, const size_t frameStep);
+
+    /* Caller must lock the device state, and the mixer must not be running. */
+    [[gnu::format(printf,2,3)]] void handleDisconnect(const char *msg, ...);
+
+    DEF_NEWDEL(ALCdevice)
+};
+
+/* Must be less than 15 characters (16 including terminating null) for
+ * compatibility with pthread_setname_np limitations. */
+#define MIXER_THREAD_NAME "alsoft-mixer"
+
+#define RECORD_THREAD_NAME "alsoft-record"
+
+
+extern int RTPrioLevel;
+void SetRTPriority(void);
+
+/**
+ * Returns the index for the given channel name (e.g. FrontCenter), or
+ * INVALID_CHANNEL_INDEX if it doesn't exist.
+ */
+inline uint GetChannelIdxByName(const RealMixParams &real, Channel chan) noexcept
+{ return real.ChannelIndex[chan]; }
+#define INVALID_CHANNEL_INDEX ~0u
+
+
+al::vector<std::string> SearchDataFiles(const char *match, const char *subdir);
+
+#endif