axmol/thirdparty/openal/alc/alcmain.h

372 lines
10 KiB
C++

#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