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hrydgard
GitHub Repository: hrydgard/ppsspp
Path: blob/master/Core/HLE/AtracBase.h
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#pragma once
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#include "Common/CommonTypes.h"
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#include "Common/Swap.h"
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#include "Core/MemMap.h"
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#include "Core/HLE/sceAudiocodec.h"
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constexpr u32 ATRAC3_MAX_SAMPLES = 0x400; // 1024
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constexpr u32 ATRAC3PLUS_MAX_SAMPLES = 0x800; // 2048
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// The "state" member of SceAtracIdInfo.
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enum AtracStatus : u8 {
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ATRAC_STATUS_UNINITIALIZED = 0, // bad state
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ATRAC_STATUS_NO_DATA = 1,
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// The entire file is loaded into memory, no further file access needed.
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ATRAC_STATUS_ALL_DATA_LOADED = 2,
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// The buffer is sized to fit the entire file, but it's only partially loaded, so you can start playback before loading the whole file.
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ATRAC_STATUS_HALFWAY_BUFFER = 3,
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// In these ones, the buffer is smaller than the file, and data is streamed into it as needed for playback.
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// These are the most complex modes, both to implement and use.
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ATRAC_STATUS_STREAMED_WITHOUT_LOOP = 4,
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ATRAC_STATUS_STREAMED_LOOP_FROM_END = 5,
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// This means there's additional audio after the loop.
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// i.e. ~~before loop~~ [ ~~this part loops~~ ] ~~after loop~~
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// The "fork in the road" means a second buffer is needed for the second path.
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ATRAC_STATUS_STREAMED_LOOP_WITH_TRAILER = 6,
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// In this mode, the only API to call is sceAtracLowLevelDecode, which decodes a stream packet by packet without any other metadata.
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ATRAC_STATUS_LOW_LEVEL = 8,
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// This mode is for using an Atrac context as the audio source for an sceSas channel. Not used a lot (Sol Trigger).
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ATRAC_STATUS_FOR_SCESAS = 16,
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// Bitwise-and the status with this to check for any of the streaming modes in a single test.
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ATRAC_STATUS_STREAMED_MASK = 4,
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};
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const char *AtracStatusToString(AtracStatus status);
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inline bool AtracStatusIsStreaming(AtracStatus status) {
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return (status & ATRAC_STATUS_STREAMED_MASK) != 0;
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}
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inline bool AtracStatusIsNormal(AtracStatus status) {
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return (int)status >= ATRAC_STATUS_ALL_DATA_LOADED && (int)status <= ATRAC_STATUS_STREAMED_LOOP_WITH_TRAILER;
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}
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struct SceAtracIdInfo {
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s32 decodePos; // Sample position in the song that we'll next be decoding from.
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s32 endSample; // Last sample index of the track.
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s32 loopStart; // Start of the loop (sample index)
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s32 loopEnd; // End of the loop (sample index)
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s32 firstValidSample; // Seems to be the number of skipped samples at the start. After SetID, decodePos will match this. Was previously misnamed 'samplesPerChan'.
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u8 numSkipFrames; // This is 1 for a single frame when a loop is triggered, otherwise seems to stay at 0. Likely mis-named.
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AtracStatus state; // State enum, see AtracStatus.
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u8 curBuffer; // Current buffer (1 == second, 2 == done?) Previously unk
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u8 numChan; // Number of audio channels, usually 2 but 1 is possible.
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u16 sampleSize; // Size in bytes of an encoded audio frame.
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u16 codec; // Codec. 0x1000 is Atrac3+, 0x1001 is Atrac3. See the PSP_CODEC_ enum (only these two are supported).
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s32 dataOff; // File offset in bytes where the Atrac3+ frames start appearing. The first dummy packet starts here.
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s32 curFileOff; // File offset in bytes corresponding to the start of next packet that will be *decoded* (on the next call to sceAtracDecodeData).
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s32 fileDataEnd; // File size in bytes.
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s32 loopNum; // Current loop counter. If 0, will not loop. -1 loops for ever, positive numbers get decremented on the loop end. So to play a song 3 times and then end, set this to 2.
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s32 streamDataByte; // Number of bytes of queued/buffered/uploaded data. In full and half-way modes, this isn't decremented as you decode.
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s32 streamOff; // Streaming modes only: The byte offset inside the RAM buffer where sceAtracDecodeData will read from next. ONLY points to even packet boundaries.
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s32 secondStreamOff; // A kind of stream position in the secondary buffer.
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u32 buffer; // Address in RAM of the main buffer.
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u32 secondBuffer; // Address in RAM of the second buffer, or 0 if not used.
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u32 bufferByte; // Size in bytes of the main buffer.
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u32 secondBufferByte; // Size in bytes of the second buffer.
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// Offset 72 here.
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// make sure the size is 128
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u32 unk[14];
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// Simple helpers. Similar ones are on track_, but we shouldn't need track_ anymore when playing back.
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int SamplesPerFrame() const {
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return codec == 0x1000 ? ATRAC3PLUS_MAX_SAMPLES : ATRAC3_MAX_SAMPLES;
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}
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int SamplesFrameMask() const {
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return SamplesPerFrame() - 1;
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}
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int SkipSamples() const {
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// These first samples are skipped, after first possibly skipping 0-2 full frames, it seems.
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return codec == 0x1000 ? 0x170 : 0x45;
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}
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int BitRate() const {
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int bitrate = (sampleSize * 352800) / 1000;
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if (codec == PSP_CODEC_AT3PLUS) {
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bitrate = ((bitrate >> 11) + 8) & 0xFFFFFFF0;
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} else {
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bitrate = (bitrate + 511) >> 10;
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}
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return bitrate;
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}
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};
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// One of these structs is allocated for each Atrac context.
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// The raw codec state is stored in 'codec'.
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// The internal playback state is stored in 'info', and that is used for all state keeping in the Atrac2 implementation,
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// imitating what happens on hardware as closely as possible.
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struct SceAtracContext {
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// size 128
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SceAudiocodecCodec codec;
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// size 128
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SceAtracIdInfo info;
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};
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struct Atrac3LowLevelParams {
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int encodedChannels;
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int outputChannels;
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int bytesPerFrame;
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};
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struct AtracSingleResetBufferInfo {
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u32_le writePosPtr;
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u32_le writableBytes;
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u32_le minWriteBytes;
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u32_le filePos;
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};
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struct AtracResetBufferInfo {
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AtracSingleResetBufferInfo first;
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AtracSingleResetBufferInfo second;
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};
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struct AtracSasStreamState {
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u32 bufPtr[2]{};
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u32 bufSize[2]{};
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int streamOffset = 0;
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int fileOffset = 0;
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int curBuffer = 0;
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bool isStreaming = false;
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int CurPos() const {
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int retval = fileOffset - bufSize[curBuffer] + streamOffset;
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_dbg_assert_(retval >= 0);
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return retval;
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}
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};
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const int PSP_ATRAC_ALLDATA_IS_ON_MEMORY = -1;
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const int PSP_ATRAC_NONLOOP_STREAM_DATA_IS_ON_MEMORY = -2;
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const int PSP_ATRAC_LOOP_STREAM_DATA_IS_ON_MEMORY = -3;
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// This is not a PSP-native struct.
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// But, it's stored in its entirety in savestates, which makes it awkward to change it.
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// This is used for both first_ and second_, but the latter doesn't use all the fields.
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struct InputBuffer {
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// Address of the buffer.
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u32 addr;
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// Size of data read so far into dataBuf_ (to be removed.)
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u32 size;
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// Offset into addr at which new data is added.
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u32 offset;
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// Last writableBytes number (to be removed.)
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u32 writableBytes;
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// Unused, always 0.
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u32 neededBytes;
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// Total size of the entire file data.
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u32 _filesize_dontuse;
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// Offset into the file at which new data is read.
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u32 fileoffset;
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};
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class AudioDecoder;
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class PointerWrap;
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struct Track;
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class AtracBase {
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public:
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virtual ~AtracBase();
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virtual void DoState(PointerWrap &p) = 0;
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// TODO: Find a way to get rid of this from the base class.
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virtual void UpdateContextFromPSPMem() = 0;
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virtual int Channels() const = 0;
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int GetOutputChannels() const {
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return outputChannels_;
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}
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void SetOutputChannels(int channels) {
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// Only used for sceSas audio. To be refactored away in the future.
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outputChannels_ = channels;
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}
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virtual u32 GetInternalCodecError() const { return 0; }
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PSPPointer<SceAtracContext> context_{};
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virtual AtracStatus BufferState() const = 0;
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virtual int SetLoopNum(int loopNum) = 0;
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virtual int LoopNum() const = 0;
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virtual int LoopStatus() const = 0;
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virtual int CodecType() const = 0;
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AudioDecoder *Decoder() const {
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return decoder_;
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}
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void CreateDecoder(int codecType, int bytesPerFrame, int channels);
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virtual void NotifyGetContextAddress() = 0;
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virtual int GetNextDecodePosition(int *pos) const = 0;
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virtual int RemainingFrames() const = 0;
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virtual bool HasSecondBuffer() const = 0;
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virtual int Bitrate() const = 0;
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virtual int BytesPerFrame() const = 0;
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virtual int SamplesPerFrame() const = 0;
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virtual void GetStreamDataInfo(u32 *writePtr, u32 *writableBytes, u32 *readOffset) = 0; // This should be const, but the legacy impl stops it (it's wrong).
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virtual int AddStreamData(u32 bytesToAdd) = 0;
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virtual int ResetPlayPosition(int sample, int bytesWrittenFirstBuf, int bytesWrittenSecondBuf, bool *delay) = 0;
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virtual int GetBufferInfoForResetting(AtracResetBufferInfo *bufferInfo, int sample, bool *delay) = 0; // NOTE: Not const! This can cause SkipFrames!
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virtual int SetData(const Track &track, u32 buffer, u32 readSize, u32 bufferSize, u32 fileSize, int outputChannels, bool isAA3) = 0;
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virtual int GetSecondBufferInfo(u32 *fileOffset, u32 *desiredSize) const = 0;
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virtual int SetSecondBuffer(u32 secondBuffer, u32 secondBufferSize) = 0;
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virtual u32 DecodeData(u8 *outbuf, u32 outbufPtr, int *SamplesNum, int *finish, int *remains) = 0;
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virtual int DecodeLowLevel(const u8 *srcData, int *bytesConsumed, s16 *dstData, int *bytesWritten) = 0;
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virtual u32 GetNextSamples() = 0; // This should be const, but the legacy impl stops it (it's wrong).
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virtual void InitLowLevel(const Atrac3LowLevelParams &params, int codecType) = 0;
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virtual void CheckForSas() = 0;
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virtual int EnqueueForSas(u32 address, u32 ptr) = 0;
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virtual void DecodeForSas(s16 *dstData, int *bytesWritten, int *finish) = 0;
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virtual const AtracSasStreamState *StreamStateForSas() const { return nullptr; }
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virtual int GetSoundSample(int *endSample, int *loopStartSample, int *loopEndSample) const = 0;
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virtual int GetContextVersion() const = 0;
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protected:
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u16 outputChannels_ = 2;
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// TODO: Save the internal state of this, now technically possible.
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AudioDecoder *decoder_ = nullptr;
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};
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