mirror of https://github.com/AxioDL/amuse.git
MusyX 1 compatibility fixes, Rogue Squadron N64 decoding confirmed
This commit is contained in:
Jack Andersen
parent
3ba25fd06a
commit
80cc9f7c94
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@ -10,7 +10,6 @@ namespace amuse
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class AudioGroupData
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{
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friend class Engine;
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friend class AudioGroupProject;
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protected:
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unsigned char* m_proj;
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unsigned char* m_pool;
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@ -41,6 +40,8 @@ public:
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{
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return m_proj != nullptr && m_pool != nullptr && m_sdir != nullptr && m_samp != nullptr;
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}
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DataFormat getDataFormat() const {return m_fmt;}
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};
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/** A buffer-owning version of AudioGroupData */
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@ -26,7 +26,9 @@ public:
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uint32_t m_adpcmParmOffset;
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void swapBig();
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};
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struct ADPCMParms
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union ADPCMParms
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{
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struct DSPParms
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{
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uint16_t m_bytesPerFrame;
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uint8_t m_ps;
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@ -34,13 +36,19 @@ public:
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int16_t m_hist2;
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int16_t m_hist1;
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int16_t m_coefs[8][2];
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void swapBig();
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} dsp;
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struct VADPCMParms
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{
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int16_t m_coefs[8][2][8];
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} vadpcm;
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void swapBigDSP();
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void swapBigVADPCM();
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};
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private:
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std::unordered_map<uint16_t, std::pair<Entry, ADPCMParms>> m_entries;
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public:
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AudioGroupSampleDirectory(const unsigned char* data, GCNDataTag);
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AudioGroupSampleDirectory(const unsigned char* data, N64DataTag);
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AudioGroupSampleDirectory(const unsigned char* data, const unsigned char* sampData, N64DataTag);
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AudioGroupSampleDirectory(const unsigned char* data, PCDataTag);
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};
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@ -48,10 +48,11 @@ class Voice : public Entity
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enum class SampleFormat : uint8_t
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{
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DSP,
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IMA,
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PCM,
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N64
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DSP, /**< GCN DSP-ucode ADPCM (very common for GameCube games) */
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IMA, /**< IMA-ADPCM (rarely used within MusyX itself) */
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PCM, /**< Big-endian PCM found in MusyX2 demo GM instruments */
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N64, /**< 2-stage VADPCM coding with SAMP-embedded codebooks */
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PCM_PC /**< Little-endian PCM found in PC Rogue Squadron (actually enum 0 which conflicts with DSP-ADPCM) */
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};
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const Sample* m_curSample = nullptr; /**< Current sample entry playing */
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const unsigned char* m_curSampleData = nullptr; /**< Current sample data playing */
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@ -15,7 +15,7 @@ AudioGroup::AudioGroup(const AudioGroupData& data, GCNDataTag)
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AudioGroup::AudioGroup(const AudioGroupData& data, N64DataTag)
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: m_proj(data.getProj(), N64DataTag{}),
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m_pool(data.getPool()),
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m_sdir(data.getSdir(), N64DataTag{}),
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m_sdir(data.getSdir(), data.getSamp(), N64DataTag{}),
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m_samp(data.getSamp()),
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m_fmt(DataFormat::N64)
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{}
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@ -187,8 +187,7 @@ void AudioGroupProject::_allocateConvBuffers(const unsigned char* data, N64DataT
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/* MIDI setups */
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const uint8_t* setupData = subData + header.midiSetupsOff;
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const uint8_t* setupEnd = subData + header.groupEndOff;
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while (setupData < setupEnd)
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while (*reinterpret_cast<const uint32_t*>(setupData) != 0xffffffff)
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{
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++midiSetupCount;
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setupData += 8 * 16 + 4;
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@ -252,8 +251,7 @@ AudioGroupProject::AudioGroupProject(const unsigned char* data, N64DataTag)
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/* MIDI setups */
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const uint8_t* setupData = subData + header.midiSetupsOff;
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const uint8_t* setupEnd = subData + header.groupEndOff;
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while (setupData < setupEnd)
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while (*reinterpret_cast<const uint32_t*>(setupData) != 0xffffffff)
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{
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uint16_t songId = SBig(*reinterpret_cast<const uint16_t*>(setupData));
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const std::array<MusyX1MIDISetup, 16>* midiSetups =
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@ -329,8 +327,7 @@ void AudioGroupProject::_allocateConvBuffers(const unsigned char* data, PCDataTa
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/* MIDI setups */
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const uint8_t* setupData = subData + group->midiSetupsOff;
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const uint8_t* setupEnd = subData + group->groupEndOff;
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while (setupData < setupEnd)
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while (*reinterpret_cast<const uint32_t*>(setupData) != 0xffffffff)
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{
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++midiSetupCount;
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setupData += 8 * 16 + 4;
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@ -392,10 +389,9 @@ AudioGroupProject::AudioGroupProject(const unsigned char* data, PCDataTag)
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/* MIDI setups */
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const uint8_t* setupData = subData + group->midiSetupsOff;
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const uint8_t* setupEnd = subData + group->groupEndOff;
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while (setupData < setupEnd)
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while (*reinterpret_cast<const uint32_t*>(setupData) != 0xffffffff)
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{
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uint16_t songId = SBig(*reinterpret_cast<const uint16_t*>(setupData));
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uint16_t songId = *reinterpret_cast<const uint16_t*>(setupData);
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const std::array<MusyX1MIDISetup, 16>* midiSetups =
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reinterpret_cast<const std::array<MusyX1MIDISetup, 16>*>(setupData + 4);
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@ -438,7 +434,7 @@ AudioGroupProject::AudioGroupProject(const unsigned char* data, PCDataTag)
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AudioGroupProject AudioGroupProject::CreateAudioGroupProject(const AudioGroupData& data)
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{
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switch (data.m_fmt)
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switch (data.getDataFormat())
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{
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case DataFormat::GCN:
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return AudioGroupProject(data.getProj(), GCNDataTag{});
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@ -1,5 +1,6 @@
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#include "amuse/AudioGroupSampleDirectory.hpp"
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#include "amuse/Common.hpp"
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#include <string.h>
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namespace amuse
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{
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@ -16,18 +17,25 @@ void AudioGroupSampleDirectory::Entry::swapBig()
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m_adpcmParmOffset = SBig(m_adpcmParmOffset);
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}
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void AudioGroupSampleDirectory::ADPCMParms::swapBig()
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void AudioGroupSampleDirectory::ADPCMParms::swapBigDSP()
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{
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m_bytesPerFrame = SBig(m_bytesPerFrame);
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m_hist2 = SBig(m_hist2);
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m_hist1 = SBig(m_hist1);
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dsp.m_bytesPerFrame = SBig(dsp.m_bytesPerFrame);
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dsp.m_hist2 = SBig(dsp.m_hist2);
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dsp.m_hist1 = SBig(dsp.m_hist1);
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for (int i=0 ; i<8 ; ++i)
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{
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m_coefs[i][0] = SBig(m_coefs[i][0]);
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m_coefs[i][1] = SBig(m_coefs[i][1]);
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dsp.m_coefs[i][0] = SBig(dsp.m_coefs[i][0]);
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dsp.m_coefs[i][1] = SBig(dsp.m_coefs[i][1]);
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}
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}
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void AudioGroupSampleDirectory::ADPCMParms::swapBigVADPCM()
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{
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int16_t* allCoefs = reinterpret_cast<int16_t*>(vadpcm.m_coefs[0][0]);
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for (int i=0 ; i<128 ; ++i)
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allCoefs[i] = SBig(allCoefs[i]);
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}
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AudioGroupSampleDirectory::AudioGroupSampleDirectory(const unsigned char* data, GCNDataTag)
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{
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const unsigned char* cur = data;
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@ -46,7 +54,7 @@ AudioGroupSampleDirectory::AudioGroupSampleDirectory(const unsigned char* data,
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reinterpret_cast<const AudioGroupSampleDirectory::ADPCMParms*>(data +
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store.first.m_adpcmParmOffset);
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store.second = *adpcm;
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store.second.swapBig();
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store.second.swapBigDSP();
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}
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cur += 32;
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@ -86,7 +94,8 @@ struct MusyX1SdirEntry
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}
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};
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AudioGroupSampleDirectory::AudioGroupSampleDirectory(const unsigned char* data, N64DataTag)
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AudioGroupSampleDirectory::AudioGroupSampleDirectory(const unsigned char* data,
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const unsigned char* sampData, N64DataTag)
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{
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const unsigned char* cur = data;
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while (*reinterpret_cast<const uint32_t*>(cur) != 0xffffffff)
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@ -97,6 +106,9 @@ AudioGroupSampleDirectory::AudioGroupSampleDirectory(const unsigned char* data,
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std::pair<Entry, ADPCMParms>& store = m_entries[ent.m_sfxId];
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ent.setIntoMusyX2(store.first);
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memcpy(&store.second.vadpcm.m_coefs, sampData + ent.m_sampleOff, 256);
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store.second.swapBigVADPCM();
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cur += 24;
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}
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}
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@ -518,19 +518,16 @@ static bool ValidateRS1N64(FILE* fp)
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if (endPos > 32 * 1024 * 1024)
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return false; /* N64 ROM definitely won't exceed 32MB */
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FSeek(fp, 59, SEEK_SET);
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uint32_t gameId;
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fread(&gameId, 1, 4, fp);
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if (gameId != 0x4e525345 && gameId != 0x4553524e)
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return false; /* GameId not 'NRSE' */
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FSeek(fp, 0, SEEK_SET);
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std::unique_ptr<uint8_t[]> data(new uint8_t[endPos]);
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fread(data.get(), 1, endPos, fp);
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if ((data[0] & 0x80) != 0x80 && (data[3] & 0x80) == 0x80)
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SwapN64Rom(data.get(), endPos);
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const uint32_t* gameId = reinterpret_cast<const uint32_t*>(&data[59]);
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if (*gameId != 0x4e525345 && *gameId != 0x4553524e)
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return false; /* GameId not 'NRSE' */
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const uint8_t* dataSeg = reinterpret_cast<const uint8_t*>(memmem(data.get(), endPos,
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"dbg_data\0\0\0\0\0\0\0\0", 16));
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if (dataSeg)
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@ -1,6 +1,7 @@
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#include "amuse/N64MusyXCodec.h"
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#include <stdint.h>
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#include <string.h>
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#include <stdio.h>
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/* Acknowledgements:
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* SubDrag for N64 Sound Tool (http://www.goldeneyevault.com/viewfile.php?id=212)
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@ -72,68 +73,44 @@ static unsigned adpcm_decode_upto_8_samples(int16_t* dst, const int16_t* src,
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return size;
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}
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static unsigned adpcm_decode_upto_8_samples_ranged(int16_t* dst, const int16_t* src,
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const int16_t* cb_entry,
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const int16_t* last_samples,
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unsigned firstSample,
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unsigned size)
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{
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if (firstSample >= size)
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return 0;
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const int16_t* book1 = cb_entry;
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const int16_t* book2 = cb_entry + 8;
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int16_t l1 = last_samples[0];
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int16_t l2 = last_samples[1];
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int accu;
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unsigned ret = 0;
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for (unsigned i=firstSample ; i<size ; ++i)
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{
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accu = (int)src[i] << 11;
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accu += book1[i] * l1 + book2[i] * l2 + rdot(i, book2, src);
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*dst++ = N64MusyXSampClamp(accu >> 11);
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++ret;
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}
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return ret;
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}
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unsigned N64MusyXDecompressFrame(int16_t* out, const uint8_t* in,
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const int16_t coefs[8][2][8],
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unsigned lastSample)
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{
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int16_t frame[32];
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unsigned remSamples = lastSample;
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unsigned samples = 0;
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unsigned procSamples;
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unsigned procSamples, procRes;
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unsigned char c2 = in[0x8];
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c2 = c2 % 0x80;
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const int16_t* bookPredictors = coefs[(c2 & 0xf0) >> 8][0];
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const int16_t* bookPredictors = coefs[(c2 & 0xf0) >> 4][0];
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unsigned int rshift = (c2 & 0x0f);
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adpcm_get_predicted_frame(frame, &in[0x0], &in[0x8], rshift);
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procSamples = (lastSample < 2) ? lastSample : 2;
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procSamples = (remSamples < 2) ? remSamples : 2;
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memcpy(out, frame, 2 * procSamples);
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samples += procSamples;
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remSamples -= procSamples;
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if (samples == lastSample)
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return samples;
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#define adpcm_decode_upto_end_samples(inOffset, bookPredictors, outOffset, size) \
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procSamples = (lastSample < size) ? lastSample : size; \
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samples += adpcm_decode_upto_8_samples(out + inOffset, frame + inOffset, bookPredictors, out + outOffset, procSamples); \
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#define adpcm_decode_upto_end_samples(inOffset, outOffset, size) \
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procSamples = (remSamples < size) ? remSamples : size; \
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procRes = adpcm_decode_upto_8_samples(out + inOffset, frame + inOffset, bookPredictors, \
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out + outOffset, procSamples); \
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samples += procRes; \
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remSamples -= procRes; \
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if (samples == lastSample) \
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return samples;
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adpcm_decode_upto_end_samples(2, bookPredictors, 0, 6);
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adpcm_decode_upto_end_samples(8, bookPredictors, 6, 8);
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adpcm_decode_upto_end_samples(16, bookPredictors, 14, 8);
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adpcm_decode_upto_end_samples(24, bookPredictors, 22, 8);
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adpcm_decode_upto_end_samples(2, 0, 6);
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adpcm_decode_upto_end_samples(8, 6, 8);
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adpcm_decode_upto_end_samples(16, 14, 8);
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adpcm_decode_upto_end_samples(24, 22, 8);
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out += 32;
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@ -141,21 +118,22 @@ unsigned N64MusyXDecompressFrame(int16_t* out, const uint8_t* in,
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c2 = in[0x18];
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c2 = c2 % 0x80;
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bookPredictors = coefs[(c2 & 0xf0) >> 8][0];
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bookPredictors = coefs[(c2 & 0xf0) >> 4][0];
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rshift = (c2 & 0x0f);
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adpcm_get_predicted_frame(frame, &in[0x4], &in[0x18], rshift);
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procSamples = (lastSample < 2) ? lastSample : 2;
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procSamples = (remSamples < 2) ? remSamples : 2;
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memcpy(out, frame, 2 * procSamples);
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samples += procSamples;
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remSamples -= procSamples;
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if (samples == lastSample)
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return samples;
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adpcm_decode_upto_end_samples(2, bookPredictors, 0, 6);
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adpcm_decode_upto_end_samples(8, bookPredictors, 6, 8);
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adpcm_decode_upto_end_samples(16, bookPredictors, 14, 8);
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adpcm_decode_upto_end_samples(24, bookPredictors, 22, 8);
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adpcm_decode_upto_end_samples(2, 0, 6);
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adpcm_decode_upto_end_samples(8, 6, 8);
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adpcm_decode_upto_end_samples(16, 14, 8);
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adpcm_decode_upto_end_samples(24, 22, 8);
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return samples;
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}
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@ -164,61 +142,10 @@ unsigned N64MusyXDecompressFrameRanged(int16_t* out, const uint8_t* in,
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const int16_t coefs[8][2][8],
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unsigned firstSample, unsigned lastSample)
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{
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int16_t frame[32];
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unsigned samples = 0;
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unsigned procSamples;
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unsigned char c2 = in[0x8];
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c2 = c2 % 0x80;
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const int16_t* bookPredictors = coefs[(c2 & 0xf0) >> 8][0];
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unsigned int rshift = (c2 & 0x0f);
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adpcm_get_predicted_frame(frame, &in[0x0], &in[0x8], rshift);
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procSamples = (lastSample < 2) ? lastSample : 2;
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memcpy(out, frame, 2 * procSamples);
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samples += procSamples;
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firstSample = (2 > firstSample) ? 0 : (firstSample - 2);
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if (samples == lastSample)
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return samples;
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#define adpcm_decode_upto_end_samples_ranged(inOffset, bookPredictors, outOffset, size) \
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procSamples = (lastSample < size) ? lastSample : size; \
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samples += adpcm_decode_upto_8_samples_ranged(out + inOffset, frame + inOffset, bookPredictors, out + outOffset, firstSample, procSamples); \
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firstSample = (size > firstSample) ? 0 : (firstSample - size); \
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if (samples == lastSample) \
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return samples;\
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adpcm_decode_upto_end_samples_ranged(2, bookPredictors, 0, 6);
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adpcm_decode_upto_end_samples_ranged(8, bookPredictors, 6, 8);
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adpcm_decode_upto_end_samples_ranged(16, bookPredictors, 14, 8);
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adpcm_decode_upto_end_samples_ranged(24, bookPredictors, 22, 8);
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out += 32;
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c2 = in[0x18];
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c2 = c2 % 0x80;
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bookPredictors = coefs[(c2 & 0xf0) >> 8][0];
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rshift = (c2 & 0x0f);
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adpcm_get_predicted_frame(frame, &in[0x4], &in[0x18], rshift);
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procSamples = (lastSample < 2) ? lastSample : 2;
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memcpy(out, frame, 2 * procSamples);
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samples += procSamples;
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firstSample = (2 > firstSample) ? 0 : (firstSample - 2);
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if (samples == lastSample)
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return samples;
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adpcm_decode_upto_end_samples_ranged(2, bookPredictors, 0, 6);
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adpcm_decode_upto_end_samples_ranged(8, bookPredictors, 6, 8);
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adpcm_decode_upto_end_samples_ranged(16, bookPredictors, 14, 8);
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adpcm_decode_upto_end_samples_ranged(24, bookPredictors, 22, 8);
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||||
int16_t final[64];
|
||||
unsigned procSamples = N64MusyXDecompressFrame(final, in, coefs, firstSample + lastSample);
|
||||
unsigned samples = procSamples - firstSample;
|
||||
memcpy(out, final + firstSample, samples * 2);
|
||||
return samples;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -94,8 +94,11 @@ bool Voice::_checkSamplePos(bool& looped)
|
|||
{
|
||||
/* Turn over looped sample */
|
||||
m_curSamplePos = m_curSample->first.m_loopStartSample;
|
||||
m_prev1 = m_curSample->second.m_hist1;
|
||||
m_prev2 = m_curSample->second.m_hist2;
|
||||
if (m_curFormat == SampleFormat::DSP)
|
||||
{
|
||||
m_prev1 = m_curSample->second.dsp.m_hist1;
|
||||
m_prev2 = m_curSample->second.dsp.m_hist2;
|
||||
}
|
||||
looped = true;
|
||||
}
|
||||
else
|
||||
|
|
@ -408,10 +411,17 @@ size_t Voice::supplyAudio(size_t samples, int16_t* data)
|
|||
case SampleFormat::DSP:
|
||||
{
|
||||
decSamples = DSPDecompressFrameRanged(data, m_curSampleData + 8 * block,
|
||||
m_curSample->second.m_coefs,
|
||||
m_curSample->second.dsp.m_coefs,
|
||||
&m_prev1, &m_prev2, rem, remCount);
|
||||
break;
|
||||
}
|
||||
case SampleFormat::N64:
|
||||
{
|
||||
decSamples = N64MusyXDecompressFrameRanged(data, m_curSampleData + 256 + 40 * block,
|
||||
m_curSample->second.vadpcm.m_coefs,
|
||||
rem, remCount);
|
||||
break;
|
||||
}
|
||||
case SampleFormat::PCM:
|
||||
{
|
||||
const int16_t* pcm = reinterpret_cast<const int16_t*>(m_curSampleData);
|
||||
|
|
@ -421,11 +431,12 @@ size_t Voice::supplyAudio(size_t samples, int16_t* data)
|
|||
decSamples = remCount;
|
||||
break;
|
||||
}
|
||||
case SampleFormat::N64:
|
||||
case SampleFormat::PCM_PC:
|
||||
{
|
||||
decSamples = N64MusyXDecompressFrameRanged(data, m_curSampleData + 256 + 40 * block,
|
||||
reinterpret_cast<const int16_t(*)[2][8]>(m_curSampleData),
|
||||
rem, remCount);
|
||||
const int16_t* pcm = reinterpret_cast<const int16_t*>(m_curSampleData);
|
||||
remCount = std::min(samplesRem, m_lastSamplePos - m_curSamplePos);
|
||||
memcpy(data, pcm + m_curSamplePos, remCount * sizeof(int16_t));
|
||||
decSamples = remCount;
|
||||
break;
|
||||
}
|
||||
default:
|
||||
|
|
@ -465,10 +476,17 @@ size_t Voice::supplyAudio(size_t samples, int16_t* data)
|
|||
case SampleFormat::DSP:
|
||||
{
|
||||
decSamples = DSPDecompressFrame(data, m_curSampleData + 8 * block,
|
||||
m_curSample->second.m_coefs,
|
||||
m_curSample->second.dsp.m_coefs,
|
||||
&m_prev1, &m_prev2, remCount);
|
||||
break;
|
||||
}
|
||||
case SampleFormat::N64:
|
||||
{
|
||||
decSamples = N64MusyXDecompressFrame(data, m_curSampleData + 256 + 40 * block,
|
||||
m_curSample->second.vadpcm.m_coefs,
|
||||
remCount);
|
||||
break;
|
||||
}
|
||||
case SampleFormat::PCM:
|
||||
{
|
||||
const int16_t* pcm = reinterpret_cast<const int16_t*>(m_curSampleData);
|
||||
|
|
@ -478,11 +496,12 @@ size_t Voice::supplyAudio(size_t samples, int16_t* data)
|
|||
decSamples = remCount;
|
||||
break;
|
||||
}
|
||||
case SampleFormat::N64:
|
||||
case SampleFormat::PCM_PC:
|
||||
{
|
||||
decSamples = N64MusyXDecompressFrame(data, m_curSampleData + 256 + 40 * block,
|
||||
reinterpret_cast<const int16_t(*)[2][8]>(m_curSampleData),
|
||||
remCount);
|
||||
const int16_t* pcm = reinterpret_cast<const int16_t*>(m_curSampleData);
|
||||
remCount = std::min(samplesRem, m_lastSamplePos - m_curSamplePos);
|
||||
memcpy(data, pcm + m_curSamplePos, remCount * sizeof(int16_t));
|
||||
decSamples = remCount;
|
||||
break;
|
||||
}
|
||||
default:
|
||||
|
|
@ -725,13 +744,17 @@ void Voice::startSample(int16_t sampId, int32_t offset)
|
|||
m_curSampleData = m_audioGroup.getSampleData(m_curSample->first.m_sampleOff);
|
||||
m_prev1 = 0;
|
||||
m_prev2 = 0;
|
||||
if (m_audioGroup.getDataFormat() == DataFormat::PC)
|
||||
m_curFormat = SampleFormat::PCM_PC;
|
||||
else
|
||||
m_curFormat = SampleFormat(m_curSample->first.m_numSamples >> 24);
|
||||
m_lastSamplePos = m_curSample->first.m_loopLengthSamples ?
|
||||
(m_curSample->first.m_loopStartSample + m_curSample->first.m_loopLengthSamples) : numSamples;
|
||||
|
||||
if (m_curFormat != SampleFormat::DSP &&
|
||||
m_curFormat != SampleFormat::PCM &&
|
||||
m_curFormat != SampleFormat::N64)
|
||||
m_curFormat != SampleFormat::N64 &&
|
||||
m_curFormat != SampleFormat::PCM_PC)
|
||||
{
|
||||
m_curSample = nullptr;
|
||||
return;
|
||||
|
|
@ -746,11 +769,11 @@ void Voice::startSample(int16_t sampId, int32_t offset)
|
|||
uint32_t block = m_curSamplePos / 14;
|
||||
uint32_t rem = m_curSamplePos % 14;
|
||||
for (uint32_t b = 0 ; b < block ; ++b)
|
||||
DSPDecompressFrameStateOnly(m_curSampleData + 8 * b, m_curSample->second.m_coefs,
|
||||
DSPDecompressFrameStateOnly(m_curSampleData + 8 * b, m_curSample->second.dsp.m_coefs,
|
||||
&m_prev1, &m_prev2, 14);
|
||||
|
||||
if (rem)
|
||||
DSPDecompressFrameStateOnly(m_curSampleData + 8 * block, m_curSample->second.m_coefs,
|
||||
DSPDecompressFrameStateOnly(m_curSampleData + 8 * block, m_curSample->second.dsp.m_coefs,
|
||||
&m_prev1, &m_prev2, rem);
|
||||
}
|
||||
}
|
||||
|
|
|
|||
Loading…
Reference in New Issue