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-rw-r--r--graphics/video/qt_decoder.cpp2
-rw-r--r--sound/decoders/qdm2.cpp260
-rw-r--r--sound/decoders/qdm2.h267
3 files changed, 273 insertions, 256 deletions
diff --git a/graphics/video/qt_decoder.cpp b/graphics/video/qt_decoder.cpp
index 5351b9b676..05f73526eb 100644
--- a/graphics/video/qt_decoder.cpp
+++ b/graphics/video/qt_decoder.cpp
@@ -1221,7 +1221,7 @@ Audio::AudioStream *QuickTimeDecoder::createAudioStream(Common::SeekableReadStre
#ifdef SOUND_QDM2_H
} else if (_streams[_audioStreamIndex]->codec_tag == MKID_BE('QDM2')) {
// Several Myst ME videos use this codec
- return new Audio::QDM2Stream(stream, _streams[_audioStreamIndex]->extradata);
+ return Audio::makeQDM2Stream(stream, _streams[_audioStreamIndex]->extradata);
#endif
}
diff --git a/sound/decoders/qdm2.cpp b/sound/decoders/qdm2.cpp
index dbfa130be5..aa4eb4b40a 100644
--- a/sound/decoders/qdm2.cpp
+++ b/sound/decoders/qdm2.cpp
@@ -29,11 +29,267 @@
#ifdef SOUND_QDM2_H
+#include "sound/audiostream.h"
#include "sound/decoders/qdm2data.h"
+
+#include "common/array.h"
+#include "common/stream.h"
#include "common/system.h"
namespace Audio {
+enum {
+ SOFTCLIP_THRESHOLD = 27600,
+ HARDCLIP_THRESHOLD = 35716,
+ MPA_MAX_CHANNELS = 2,
+ MPA_FRAME_SIZE = 1152,
+ FF_INPUT_BUFFER_PADDING_SIZE = 8
+};
+
+typedef int8 sb_int8_array[2][30][64];
+
+/* bit input */
+/* buffer, buffer_end and size_in_bits must be present and used by every reader */
+struct GetBitContext {
+ const uint8 *buffer, *bufferEnd;
+ int index;
+ int sizeInBits;
+};
+
+struct QDM2SubPacket {
+ int type;
+ unsigned int size;
+ const uint8 *data; // pointer to subpacket data (points to input data buffer, it's not a private copy)
+};
+
+struct QDM2SubPNode {
+ QDM2SubPacket *packet;
+ struct QDM2SubPNode *next; // pointer to next packet in the list, NULL if leaf node
+};
+
+struct QDM2Complex {
+ float re;
+ float im;
+};
+
+struct FFTTone {
+ float level;
+ QDM2Complex *complex;
+ const float *table;
+ int phase;
+ int phase_shift;
+ int duration;
+ short time_index;
+ short cutoff;
+};
+
+struct FFTCoefficient {
+ int16 sub_packet;
+ uint8 channel;
+ int16 offset;
+ int16 exp;
+ uint8 phase;
+};
+
+struct VLC {
+ int32 bits;
+ int16 (*table)[2]; // code, bits
+ int32 table_size;
+ int32 table_allocated;
+};
+
+#include "common/pack-start.h"
+struct QDM2FFT {
+ QDM2Complex complex[MPA_MAX_CHANNELS][256];
+} PACKED_STRUCT;
+#include "common/pack-end.h"
+
+enum RDFTransformType {
+ RDFT,
+ IRDFT,
+ RIDFT,
+ IRIDFT
+};
+
+struct FFTComplex {
+ float re, im;
+};
+
+struct FFTContext {
+ int nbits;
+ int inverse;
+ uint16 *revtab;
+ FFTComplex *exptab;
+ FFTComplex *tmpBuf;
+ int mdctSize; // size of MDCT (i.e. number of input data * 2)
+ int mdctBits; // n = 2^nbits
+ // pre/post rotation tables
+ float *tcos;
+ float *tsin;
+ void (*fftPermute)(struct FFTContext *s, FFTComplex *z);
+ void (*fftCalc)(struct FFTContext *s, FFTComplex *z);
+ void (*imdctCalc)(struct FFTContext *s, float *output, const float *input);
+ void (*imdctHalf)(struct FFTContext *s, float *output, const float *input);
+ void (*mdctCalc)(struct FFTContext *s, float *output, const float *input);
+ int splitRadix;
+ int permutation;
+};
+
+enum {
+ FF_MDCT_PERM_NONE = 0,
+ FF_MDCT_PERM_INTERLEAVE = 1
+};
+
+struct RDFTContext {
+ int nbits;
+ int inverse;
+ int signConvention;
+
+ // pre/post rotation tables
+ float *tcos;
+ float *tsin;
+ FFTContext fft;
+};
+
+class QDM2Stream : public Audio::AudioStream {
+public:
+ QDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData);
+ ~QDM2Stream();
+
+ bool isStereo() const { return _channels == 2; }
+ bool endOfData() const { return ((_stream->pos() == _stream->size()) && (_outputSamples.size() == 0)); }
+ int getRate() const { return _sampleRate; }
+ int readBuffer(int16 *buffer, const int numSamples);
+
+private:
+ Common::SeekableReadStream *_stream;
+
+ // Parameters from codec header, do not change during playback
+ uint8 _channels;
+ uint16 _sampleRate;
+ uint16 _bitRate;
+ uint16 _blockSize; // Group
+ uint16 _frameSize; // FFT
+ uint16 _packetSize; // Checksum
+
+ // Parameters built from header parameters, do not change during playback
+ int _groupOrder; // order of frame group
+ int _fftOrder; // order of FFT (actually fft order+1)
+ int _fftFrameSize; // size of fft frame, in components (1 comples = re + im)
+ int _sFrameSize; // size of data frame
+ int _frequencyRange;
+ int _subSampling; // subsampling: 0=25%, 1=50%, 2=100% */
+ int _coeffPerSbSelect; // selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
+ int _cmTableSelect; // selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
+
+ // Packets and packet lists
+ QDM2SubPacket _subPackets[16]; // the packets themselves
+ QDM2SubPNode _subPacketListA[16]; // list of all packets
+ QDM2SubPNode _subPacketListB[16]; // FFT packets B are on list
+ int _subPacketsB; // number of packets on 'B' list
+ QDM2SubPNode _subPacketListC[16]; // packets with errors?
+ QDM2SubPNode _subPacketListD[16]; // DCT packets
+
+ // FFT and tones
+ FFTTone _fftTones[1000];
+ int _fftToneStart;
+ int _fftToneEnd;
+ FFTCoefficient _fftCoefs[1000];
+ int _fftCoefsIndex;
+ int _fftCoefsMinIndex[5];
+ int _fftCoefsMaxIndex[5];
+ int _fftLevelExp[6];
+ //RDFTContext _rdftCtx;
+ QDM2FFT _fft;
+
+ // I/O data
+ uint8 *_compressedData;
+ float _outputBuffer[1024];
+ Common::Array<int16> _outputSamples;
+
+ // Synthesis filter
+ int16 ff_mpa_synth_window[512];
+ int16 _synthBuf[MPA_MAX_CHANNELS][512*2];
+ int _synthBufOffset[MPA_MAX_CHANNELS];
+ int32 _sbSamples[MPA_MAX_CHANNELS][128][32];
+
+ // Mixed temporary data used in decoding
+ float _toneLevel[MPA_MAX_CHANNELS][30][64];
+ int8 _codingMethod[MPA_MAX_CHANNELS][30][64];
+ int8 _quantizedCoeffs[MPA_MAX_CHANNELS][10][8];
+ int8 _toneLevelIdxBase[MPA_MAX_CHANNELS][30][8];
+ int8 _toneLevelIdxHi1[MPA_MAX_CHANNELS][3][8][8];
+ int8 _toneLevelIdxMid[MPA_MAX_CHANNELS][26][8];
+ int8 _toneLevelIdxHi2[MPA_MAX_CHANNELS][26];
+ int8 _toneLevelIdx[MPA_MAX_CHANNELS][30][64];
+ int8 _toneLevelIdxTemp[MPA_MAX_CHANNELS][30][64];
+
+ // Flags
+ bool _hasErrors; // packet has errors
+ int _superblocktype_2_3; // select fft tables and some algorithm based on superblock type
+ int _doSynthFilter; // used to perform or skip synthesis filter
+
+ uint8 _subPacket; // 0 to 15
+ int _noiseIdx; // index for dithering noise table
+
+ byte _emptyBuffer[FF_INPUT_BUFFER_PADDING_SIZE];
+
+ VLC _vlcTabLevel;
+ VLC _vlcTabDiff;
+ VLC _vlcTabRun;
+ VLC _fftLevelExpAltVlc;
+ VLC _fftLevelExpVlc;
+ VLC _fftStereoExpVlc;
+ VLC _fftStereoPhaseVlc;
+ VLC _vlcTabToneLevelIdxHi1;
+ VLC _vlcTabToneLevelIdxMid;
+ VLC _vlcTabToneLevelIdxHi2;
+ VLC _vlcTabType30;
+ VLC _vlcTabType34;
+ VLC _vlcTabFftToneOffset[5];
+ bool _vlcsInitialized;
+ void initVlc(void);
+
+ uint16 _softclipTable[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
+ void softclipTableInit(void);
+
+ float _noiseTable[4096];
+ byte _randomDequantIndex[256][5];
+ byte _randomDequantType24[128][3];
+ void rndTableInit(void);
+
+ float _noiseSamples[128];
+ void initNoiseSamples(void);
+
+ RDFTContext _rdftCtx;
+
+ void average_quantized_coeffs(void);
+ void build_sb_samples_from_noise(int sb);
+ void fix_coding_method_array(int sb, int channels, sb_int8_array coding_method);
+ void fill_tone_level_array(int flag);
+ void fill_coding_method_array(sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
+ sb_int8_array coding_method, int nb_channels,
+ int c, int superblocktype_2_3, int cm_table_select);
+ void synthfilt_build_sb_samples(GetBitContext *gb, int length, int sb_min, int sb_max);
+ void init_quantized_coeffs_elem0(int8 *quantized_coeffs, GetBitContext *gb, int length);
+ void init_tone_level_dequantization(GetBitContext *gb, int length);
+ void process_subpacket_9(QDM2SubPNode *node);
+ void process_subpacket_10(QDM2SubPNode *node, int length);
+ void process_subpacket_11(QDM2SubPNode *node, int length);
+ void process_subpacket_12(QDM2SubPNode *node, int length);
+ void process_synthesis_subpackets(QDM2SubPNode *list);
+ void qdm2_decode_super_block(void);
+ void qdm2_fft_init_coefficient(int sub_packet, int offset, int duration,
+ int channel, int exp, int phase);
+ void qdm2_fft_decode_tones(int duration, GetBitContext *gb, int b);
+ void qdm2_decode_fft_packets(void);
+ void qdm2_fft_generate_tone(FFTTone *tone);
+ void qdm2_fft_tone_synthesizer(uint8 sub_packet);
+ void qdm2_calculate_fft(int channel);
+ void qdm2_synthesis_filter(uint8 index);
+ int qdm2_decodeFrame(Common::SeekableReadStream *in);
+};
+
// Fix compilation for non C99-compliant compilers, like MSVC
#ifndef int64_t
typedef signed long long int int64_t;
@@ -3062,6 +3318,10 @@ int QDM2Stream::readBuffer(int16 *buffer, const int numSamples) {
return decodedSamples;
}
+AudioStream *makeQDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData) {
+ return new QDM2Stream(stream, extraData);
+}
+
} // End of namespace Audio
#endif
diff --git a/sound/decoders/qdm2.h b/sound/decoders/qdm2.h
index 77208aec52..842ede3de0 100644
--- a/sound/decoders/qdm2.h
+++ b/sound/decoders/qdm2.h
@@ -29,266 +29,23 @@
#ifndef SOUND_QDM2_H
#define SOUND_QDM2_H
-#include "sound/audiostream.h"
-#include "common/array.h"
-#include "common/stream.h"
+namespace Common {
+ class SeekableReadStream;
+}
namespace Audio {
+ class AudioStream;
-enum {
- SOFTCLIP_THRESHOLD = 27600,
- HARDCLIP_THRESHOLD = 35716,
- MPA_MAX_CHANNELS = 2,
- MPA_FRAME_SIZE = 1152,
- FF_INPUT_BUFFER_PADDING_SIZE = 8
-};
-
-typedef int8 sb_int8_array[2][30][64];
-
-/* bit input */
-/* buffer, buffer_end and size_in_bits must be present and used by every reader */
-struct GetBitContext {
- const uint8 *buffer, *bufferEnd;
- int index;
- int sizeInBits;
-};
-
-struct QDM2SubPacket {
- int type;
- unsigned int size;
- const uint8 *data; // pointer to subpacket data (points to input data buffer, it's not a private copy)
-};
-
-struct QDM2SubPNode {
- QDM2SubPacket *packet;
- struct QDM2SubPNode *next; // pointer to next packet in the list, NULL if leaf node
-};
-
-struct QDM2Complex {
- float re;
- float im;
-};
-
-struct FFTTone {
- float level;
- QDM2Complex *complex;
- const float *table;
- int phase;
- int phase_shift;
- int duration;
- short time_index;
- short cutoff;
-};
-
-struct FFTCoefficient {
- int16 sub_packet;
- uint8 channel;
- int16 offset;
- int16 exp;
- uint8 phase;
-};
-
-struct VLC {
- int32 bits;
- int16 (*table)[2]; // code, bits
- int32 table_size;
- int32 table_allocated;
-};
-
-#include "common/pack-start.h"
-struct QDM2FFT {
- QDM2Complex complex[MPA_MAX_CHANNELS][256];
-} PACKED_STRUCT;
-#include "common/pack-end.h"
-
-enum RDFTransformType {
- RDFT,
- IRDFT,
- RIDFT,
- IRIDFT
-};
-
-struct FFTComplex {
- float re, im;
-};
-
-struct FFTContext {
- int nbits;
- int inverse;
- uint16 *revtab;
- FFTComplex *exptab;
- FFTComplex *tmpBuf;
- int mdctSize; // size of MDCT (i.e. number of input data * 2)
- int mdctBits; // n = 2^nbits
- // pre/post rotation tables
- float *tcos;
- float *tsin;
- void (*fftPermute)(struct FFTContext *s, FFTComplex *z);
- void (*fftCalc)(struct FFTContext *s, FFTComplex *z);
- void (*imdctCalc)(struct FFTContext *s, float *output, const float *input);
- void (*imdctHalf)(struct FFTContext *s, float *output, const float *input);
- void (*mdctCalc)(struct FFTContext *s, float *output, const float *input);
- int splitRadix;
- int permutation;
-};
-
-enum {
- FF_MDCT_PERM_NONE = 0,
- FF_MDCT_PERM_INTERLEAVE = 1
-};
-
-struct RDFTContext {
- int nbits;
- int inverse;
- int signConvention;
-
- // pre/post rotation tables
- float *tcos;
- float *tsin;
- FFTContext fft;
-};
-
-class QDM2Stream : public Audio::AudioStream {
-public:
- QDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData);
- ~QDM2Stream();
-
- bool isStereo() const { return _channels == 2; }
- bool endOfData() const { return ((_stream->pos() == _stream->size()) && (_outputSamples.size() == 0)); }
- int getRate() const { return _sampleRate; }
- int readBuffer(int16 *buffer, const int numSamples);
-
-private:
- Common::SeekableReadStream *_stream;
-
- // Parameters from codec header, do not change during playback
- uint8 _channels;
- uint16 _sampleRate;
- uint16 _bitRate;
- uint16 _blockSize; // Group
- uint16 _frameSize; // FFT
- uint16 _packetSize; // Checksum
-
- // Parameters built from header parameters, do not change during playback
- int _groupOrder; // order of frame group
- int _fftOrder; // order of FFT (actually fft order+1)
- int _fftFrameSize; // size of fft frame, in components (1 comples = re + im)
- int _sFrameSize; // size of data frame
- int _frequencyRange;
- int _subSampling; // subsampling: 0=25%, 1=50%, 2=100% */
- int _coeffPerSbSelect; // selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
- int _cmTableSelect; // selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
-
- // Packets and packet lists
- QDM2SubPacket _subPackets[16]; // the packets themselves
- QDM2SubPNode _subPacketListA[16]; // list of all packets
- QDM2SubPNode _subPacketListB[16]; // FFT packets B are on list
- int _subPacketsB; // number of packets on 'B' list
- QDM2SubPNode _subPacketListC[16]; // packets with errors?
- QDM2SubPNode _subPacketListD[16]; // DCT packets
-
- // FFT and tones
- FFTTone _fftTones[1000];
- int _fftToneStart;
- int _fftToneEnd;
- FFTCoefficient _fftCoefs[1000];
- int _fftCoefsIndex;
- int _fftCoefsMinIndex[5];
- int _fftCoefsMaxIndex[5];
- int _fftLevelExp[6];
- //RDFTContext _rdftCtx;
- QDM2FFT _fft;
-
- // I/O data
- uint8 *_compressedData;
- float _outputBuffer[1024];
- Common::Array<int16> _outputSamples;
-
- // Synthesis filter
- int16 ff_mpa_synth_window[512];
- int16 _synthBuf[MPA_MAX_CHANNELS][512*2];
- int _synthBufOffset[MPA_MAX_CHANNELS];
- int32 _sbSamples[MPA_MAX_CHANNELS][128][32];
-
- // Mixed temporary data used in decoding
- float _toneLevel[MPA_MAX_CHANNELS][30][64];
- int8 _codingMethod[MPA_MAX_CHANNELS][30][64];
- int8 _quantizedCoeffs[MPA_MAX_CHANNELS][10][8];
- int8 _toneLevelIdxBase[MPA_MAX_CHANNELS][30][8];
- int8 _toneLevelIdxHi1[MPA_MAX_CHANNELS][3][8][8];
- int8 _toneLevelIdxMid[MPA_MAX_CHANNELS][26][8];
- int8 _toneLevelIdxHi2[MPA_MAX_CHANNELS][26];
- int8 _toneLevelIdx[MPA_MAX_CHANNELS][30][64];
- int8 _toneLevelIdxTemp[MPA_MAX_CHANNELS][30][64];
-
- // Flags
- bool _hasErrors; // packet has errors
- int _superblocktype_2_3; // select fft tables and some algorithm based on superblock type
- int _doSynthFilter; // used to perform or skip synthesis filter
-
- uint8 _subPacket; // 0 to 15
- int _noiseIdx; // index for dithering noise table
-
- byte _emptyBuffer[FF_INPUT_BUFFER_PADDING_SIZE];
-
- VLC _vlcTabLevel;
- VLC _vlcTabDiff;
- VLC _vlcTabRun;
- VLC _fftLevelExpAltVlc;
- VLC _fftLevelExpVlc;
- VLC _fftStereoExpVlc;
- VLC _fftStereoPhaseVlc;
- VLC _vlcTabToneLevelIdxHi1;
- VLC _vlcTabToneLevelIdxMid;
- VLC _vlcTabToneLevelIdxHi2;
- VLC _vlcTabType30;
- VLC _vlcTabType34;
- VLC _vlcTabFftToneOffset[5];
- bool _vlcsInitialized;
- void initVlc(void);
-
- uint16 _softclipTable[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
- void softclipTableInit(void);
-
- float _noiseTable[4096];
- byte _randomDequantIndex[256][5];
- byte _randomDequantType24[128][3];
- void rndTableInit(void);
-
- float _noiseSamples[128];
- void initNoiseSamples(void);
-
- RDFTContext _rdftCtx;
-
- void average_quantized_coeffs(void);
- void build_sb_samples_from_noise(int sb);
- void fix_coding_method_array(int sb, int channels, sb_int8_array coding_method);
- void fill_tone_level_array(int flag);
- void fill_coding_method_array(sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
- sb_int8_array coding_method, int nb_channels,
- int c, int superblocktype_2_3, int cm_table_select);
- void synthfilt_build_sb_samples(GetBitContext *gb, int length, int sb_min, int sb_max);
- void init_quantized_coeffs_elem0(int8 *quantized_coeffs, GetBitContext *gb, int length);
- void init_tone_level_dequantization(GetBitContext *gb, int length);
- void process_subpacket_9(QDM2SubPNode *node);
- void process_subpacket_10(QDM2SubPNode *node, int length);
- void process_subpacket_11(QDM2SubPNode *node, int length);
- void process_subpacket_12(QDM2SubPNode *node, int length);
- void process_synthesis_subpackets(QDM2SubPNode *list);
- void qdm2_decode_super_block(void);
- void qdm2_fft_init_coefficient(int sub_packet, int offset, int duration,
- int channel, int exp, int phase);
- void qdm2_fft_decode_tones(int duration, GetBitContext *gb, int b);
- void qdm2_decode_fft_packets(void);
- void qdm2_fft_generate_tone(FFTTone *tone);
- void qdm2_fft_tone_synthesizer(uint8 sub_packet);
- void qdm2_calculate_fft(int channel);
- void qdm2_synthesis_filter(uint8 index);
- int qdm2_decodeFrame(Common::SeekableReadStream *in);
-};
+/**
+ * Create a new AudioStream from the QDM2 data in the given stream.
+ *
+ * @param stream the SeekableReadStream from which to read the FLAC data
+ * @param extraData the QuickTime extra data stream
+ * @return a new AudioStream, or NULL, if an error occured
+ */
+AudioStream *makeQDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData);
} // End of namespace Audio
#endif // SOUND_QDM2_H
-
#endif // Mohawk/Plugins guard
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-30 21:38:03 +0000