// Ogg Vorbis audio decoder - v1.05 - public domain // http://nothings.org/stb_vorbis/ // // Written by Sean Barrett in 2007, last updated in 2014 // Sponsored by RAD Game Tools. // // LICENSE // // This software is in the public domain. Where that dedication is not // recognized, you are granted a perpetual, irrevocable license to copy, // distribute, and modify this file as you see fit. // // No warranty for any purpose is expressed or implied by the author (nor // by RAD Game Tools). Report bugs and send enhancements to the author. // // Limitations: // // - seeking not supported except manually via PUSHDATA api // - floor 0 not supported (used in old ogg vorbis files pre-2004) // - lossless sample-truncation at beginning ignored // - cannot concatenate multiple vorbis streams // - sample positions are 32-bit, limiting seekable 192Khz // files to around 6 hours (Ogg supports 64-bit) // // Bugfix/warning contributors: // Terje Mathisen Niklas Frykholm Andy Hill // Casey Muratori John Bolton Gargaj // Laurent Gomila Marc LeBlanc Ronny Chevalier // Bernhard Wodo Evan Balster "alxprd"@github // Tom Beaumont Ingo Leitgeb Nicolas Guillemot // (If you reported a bug but do not appear in this list, it is because // someone else reported the bug before you. There were too many of you to // list them all because I was lax about updating for a long time, sorry.) // // Partial history: // 1.05 - 2015/04/19 - don't define __forceinline if it's redundant // 1.04 - 2014/08/27 - fix missing const-correct case in API // 1.03 - 2014/08/07 - warning fixes // 1.02 - 2014/07/09 - declare qsort comparison as explicitly _cdecl in Windows // 1.01 - 2014/06/18 - fix stb_vorbis_get_samples_float (interleaved was correct) // 1.0 - 2014/05/26 - fix memory leaks; fix warnings; fix bugs in >2-channel; // (API change) report sample rate for decode-full-file funcs // 0.99996 - - bracket #include for macintosh compilation // 0.99995 - - avoid alias-optimization issue in float-to-int conversion // // See end of file for full version history. ////////////////////////////////////////////////////////////////////////////// // // HEADER BEGINS HERE // #ifndef STB_VORBIS_INCLUDE_STB_VORBIS_H #define STB_VORBIS_INCLUDE_STB_VORBIS_H #if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO) #define STB_VORBIS_NO_STDIO 1 #endif #ifndef STB_VORBIS_NO_STDIO #include #endif #ifdef __cplusplus extern "C" { #endif /////////// THREAD SAFETY // Individual stb_vorbis* handles are not thread-safe; you cannot decode from // them from multiple threads at the same time. However, you can have multiple // stb_vorbis* handles and decode from them independently in multiple thrads. /////////// MEMORY ALLOCATION // normally stb_vorbis uses malloc() to allocate memory at startup, // and alloca() to allocate temporary memory during a frame on the // stack. (Memory consumption will depend on the amount of setup // data in the file and how you set the compile flags for speed // vs. size. In my test files the maximal-size usage is ~150KB.) // // You can modify the wrapper functions in the source (setup_malloc, // setup_temp_malloc, temp_malloc) to change this behavior, or you // can use a simpler allocation model: you pass in a buffer from // which stb_vorbis will allocate _all_ its memory (including the // temp memory). "open" may fail with a VORBIS_outofmem if you // do not pass in enough data; there is no way to determine how // much you do need except to succeed (at which point you can // query get_info to find the exact amount required. yes I know // this is lame). // // If you pass in a non-NULL buffer of the type below, allocation // will occur from it as described above. Otherwise just pass NULL // to use malloc()/alloca() typedef struct { char *alloc_buffer; int alloc_buffer_length_in_bytes; } stb_vorbis_alloc; /////////// FUNCTIONS USEABLE WITH ALL INPUT MODES typedef struct stb_vorbis stb_vorbis; typedef struct { unsigned int sample_rate; int channels; unsigned int setup_memory_required; unsigned int setup_temp_memory_required; unsigned int temp_memory_required; int max_frame_size; } stb_vorbis_info; // get general information about the file extern stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f); // get the last error detected (clears it, too) extern int stb_vorbis_get_error(stb_vorbis *f); // close an ogg vorbis file and free all memory in use extern void stb_vorbis_close(stb_vorbis *f); // this function returns the offset (in samples) from the beginning of the // file that will be returned by the next decode, if it is known, or -1 // otherwise. after a flush_pushdata() call, this may take a while before // it becomes valid again. // NOT WORKING YET after a seek with PULLDATA API extern int stb_vorbis_get_sample_offset(stb_vorbis *f); // returns the current seek point within the file, or offset from the beginning // of the memory buffer. In pushdata mode it returns 0. extern unsigned int stb_vorbis_get_file_offset(stb_vorbis *f); /////////// PUSHDATA API #ifndef STB_VORBIS_NO_PUSHDATA_API // this API allows you to get blocks of data from any source and hand // them to stb_vorbis. you have to buffer them; stb_vorbis will tell // you how much it used, and you have to give it the rest next time; // and stb_vorbis may not have enough data to work with and you will // need to give it the same data again PLUS more. Note that the Vorbis // specification does not bound the size of an individual frame. extern stb_vorbis *stb_vorbis_open_pushdata( unsigned char *datablock, int datablock_length_in_bytes, int *datablock_memory_consumed_in_bytes, int *error, stb_vorbis_alloc *alloc_buffer); // create a vorbis decoder by passing in the initial data block containing // the ogg&vorbis headers (you don't need to do parse them, just provide // the first N bytes of the file--you're told if it's not enough, see below) // on success, returns an stb_vorbis *, does not set error, returns the amount of // data parsed/consumed on this call in *datablock_memory_consumed_in_bytes; // on failure, returns NULL on error and sets *error, does not change *datablock_memory_consumed // if returns NULL and *error is VORBIS_need_more_data, then the input block was // incomplete and you need to pass in a larger block from the start of the file extern int stb_vorbis_decode_frame_pushdata( stb_vorbis *f, unsigned char *datablock, int datablock_length_in_bytes, int *channels, // place to write number of float * buffers float ***output, // place to write float ** array of float * buffers int *samples // place to write number of output samples ); // decode a frame of audio sample data if possible from the passed-in data block // // return value: number of bytes we used from datablock // // possible cases: // 0 bytes used, 0 samples output (need more data) // N bytes used, 0 samples output (resynching the stream, keep going) // N bytes used, M samples output (one frame of data) // note that after opening a file, you will ALWAYS get one N-bytes,0-sample // frame, because Vorbis always "discards" the first frame. // // Note that on resynch, stb_vorbis will rarely consume all of the buffer, // instead only datablock_length_in_bytes-3 or less. This is because it wants // to avoid missing parts of a page header if they cross a datablock boundary, // without writing state-machiney code to record a partial detection. // // The number of channels returned are stored in *channels (which can be // NULL--it is always the same as the number of channels reported by // get_info). *output will contain an array of float* buffers, one per // channel. In other words, (*output)[0][0] contains the first sample from // the first channel, and (*output)[1][0] contains the first sample from // the second channel. extern void stb_vorbis_flush_pushdata(stb_vorbis *f); // inform stb_vorbis that your next datablock will not be contiguous with // previous ones (e.g. you've seeked in the data); future attempts to decode // frames will cause stb_vorbis to resynchronize (as noted above), and // once it sees a valid Ogg page (typically 4-8KB, as large as 64KB), it // will begin decoding the _next_ frame. // // if you want to seek using pushdata, you need to seek in your file, then // call stb_vorbis_flush_pushdata(), then start calling decoding, then once // decoding is returning you data, call stb_vorbis_get_sample_offset, and // if you don't like the result, seek your file again and repeat. #endif ////////// PULLING INPUT API #ifndef STB_VORBIS_NO_PULLDATA_API // This API assumes stb_vorbis is allowed to pull data from a source-- // either a block of memory containing the _entire_ vorbis stream, or a // FILE * that you or it create, or possibly some other reading mechanism // if you go modify the source to replace the FILE * case with some kind // of callback to your code. (But if you don't support seeking, you may // just want to go ahead and use pushdata.) #if !defined(STB_VORBIS_NO_STDIO) && !defined(STB_VORBIS_NO_INTEGER_CONVERSION) extern int stb_vorbis_decode_filename(const char *filename, int *channels, int *sample_rate, short **output); #endif #if !defined(STB_VORBIS_NO_INTEGER_CONVERSION) extern int stb_vorbis_decode_memory(const unsigned char *mem, int len, int *channels, int *sample_rate, short **output); #endif // decode an entire file and output the data interleaved into a malloc()ed // buffer stored in *output. The return value is the number of samples // decoded, or -1 if the file could not be opened or was not an ogg vorbis file. // When you're done with it, just free() the pointer returned in *output. extern stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len, int *error, stb_vorbis_alloc *alloc_buffer); // create an ogg vorbis decoder from an ogg vorbis stream in memory (note // this must be the entire stream!). on failure, returns NULL and sets *error #ifndef STB_VORBIS_NO_STDIO extern stb_vorbis * stb_vorbis_open_filename(const char *filename, int *error, stb_vorbis_alloc *alloc_buffer); // create an ogg vorbis decoder from a filename via fopen(). on failure, // returns NULL and sets *error (possibly to VORBIS_file_open_failure). extern stb_vorbis * stb_vorbis_open_file(FILE *f, int close_handle_on_close, int *error, stb_vorbis_alloc *alloc_buffer); // create an ogg vorbis decoder from an open FILE *, looking for a stream at // the _current_ seek point (ftell). on failure, returns NULL and sets *error. // note that stb_vorbis must "own" this stream; if you seek it in between // calls to stb_vorbis, it will become confused. Morever, if you attempt to // perform stb_vorbis_seek_*() operations on this file, it will assume it // owns the _entire_ rest of the file after the start point. Use the next // function, stb_vorbis_open_file_section(), to limit it. extern stb_vorbis * stb_vorbis_open_file_section(FILE *f, int close_handle_on_close, int *error, stb_vorbis_alloc *alloc_buffer, unsigned int len); // create an ogg vorbis decoder from an open FILE *, looking for a stream at // the _current_ seek point (ftell); the stream will be of length 'len' bytes. // on failure, returns NULL and sets *error. note that stb_vorbis must "own" // this stream; if you seek it in between calls to stb_vorbis, it will become // confused. #endif extern int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number); extern int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number); // NOT WORKING YET // these functions seek in the Vorbis file to (approximately) 'sample_number'. // after calling seek_frame(), the next call to get_frame_*() will include // the specified sample. after calling stb_vorbis_seek(), the next call to // stb_vorbis_get_samples_* will start with the specified sample. If you // do not need to seek to EXACTLY the target sample when using get_samples_*, // you can also use seek_frame(). extern void stb_vorbis_seek_start(stb_vorbis *f); // this function is equivalent to stb_vorbis_seek(f,0), but it // actually works extern unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f); extern float stb_vorbis_stream_length_in_seconds(stb_vorbis *f); // these functions return the total length of the vorbis stream extern int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output); // decode the next frame and return the number of samples. the number of // channels returned are stored in *channels (which can be NULL--it is always // the same as the number of channels reported by get_info). *output will // contain an array of float* buffers, one per channel. These outputs will // be overwritten on the next call to stb_vorbis_get_frame_*. // // You generally should not intermix calls to stb_vorbis_get_frame_*() // and stb_vorbis_get_samples_*(), since the latter calls the former. #ifndef STB_VORBIS_NO_INTEGER_CONVERSION extern int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts); extern int stb_vorbis_get_frame_short (stb_vorbis *f, int num_c, short **buffer, int num_samples); #endif // decode the next frame and return the number of samples per channel. the // data is coerced to the number of channels you request according to the // channel coercion rules (see below). You must pass in the size of your // buffer(s) so that stb_vorbis will not overwrite the end of the buffer. // The maximum buffer size needed can be gotten from get_info(); however, // the Vorbis I specification implies an absolute maximum of 4096 samples // per channel. Note that for interleaved data, you pass in the number of // shorts (the size of your array), but the return value is the number of // samples per channel, not the total number of samples. // Channel coercion rules: // Let M be the number of channels requested, and N the number of channels present, // and Cn be the nth channel; let stereo L be the sum of all L and center channels, // and stereo R be the sum of all R and center channels (channel assignment from the // vorbis spec). // M N output // 1 k sum(Ck) for all k // 2 * stereo L, stereo R // k l k > l, the first l channels, then 0s // k l k <= l, the first k channels // Note that this is not _good_ surround etc. mixing at all! It's just so // you get something useful. extern int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats); extern int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples); // gets num_samples samples, not necessarily on a frame boundary--this requires // buffering so you have to supply the buffers. DOES NOT APPLY THE COERCION RULES. // Returns the number of samples stored per channel; it may be less than requested // at the end of the file. If there are no more samples in the file, returns 0. #ifndef STB_VORBIS_NO_INTEGER_CONVERSION extern int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts); extern int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int num_samples); #endif // gets num_samples samples, not necessarily on a frame boundary--this requires // buffering so you have to supply the buffers. Applies the coercion rules above // to produce 'channels' channels. Returns the number of samples stored per channel; // it may be less than requested at the end of the file. If there are no more // samples in the file, returns 0. #endif //////// ERROR CODES enum STBVorbisError { VORBIS__no_error, VORBIS_need_more_data=1, // not a real error VORBIS_invalid_api_mixing, // can't mix API modes VORBIS_outofmem, // not enough memory VORBIS_feature_not_supported, // uses floor 0 VORBIS_too_many_channels, // STB_VORBIS_MAX_CHANNELS is too small VORBIS_file_open_failure, // fopen() failed VORBIS_seek_without_length, // can't seek in unknown-length file VORBIS_unexpected_eof=10, // file is truncated? VORBIS_seek_invalid, // seek past EOF // decoding errors (corrupt/invalid stream) -- you probably // don't care about the exact details of these // vorbis errors: VORBIS_invalid_setup=20, VORBIS_invalid_stream, // ogg errors: VORBIS_missing_capture_pattern=30, VORBIS_invalid_stream_structure_version, VORBIS_continued_packet_flag_invalid, VORBIS_incorrect_stream_serial_number, VORBIS_invalid_first_page, VORBIS_bad_packet_type, VORBIS_cant_find_last_page, VORBIS_seek_failed, }; #ifdef __cplusplus } #endif #endif // STB_VORBIS_INCLUDE_STB_VORBIS_H // // HEADER ENDS HERE // ////////////////////////////////////////////////////////////////////////////// #ifndef STB_VORBIS_HEADER_ONLY // global configuration settings (e.g. set these in the project/makefile), // or just set them in this file at the top (although ideally the first few // should be visible when the header file is compiled too, although it's not // crucial) // STB_VORBIS_NO_PUSHDATA_API // does not compile the code for the various stb_vorbis_*_pushdata() // functions // #define STB_VORBIS_NO_PUSHDATA_API // STB_VORBIS_NO_PULLDATA_API // does not compile the code for the non-pushdata APIs // #define STB_VORBIS_NO_PULLDATA_API // STB_VORBIS_NO_STDIO // does not compile the code for the APIs that use FILE *s internally // or externally (implied by STB_VORBIS_NO_PULLDATA_API) // #define STB_VORBIS_NO_STDIO // STB_VORBIS_NO_INTEGER_CONVERSION // does not compile the code for converting audio sample data from // float to integer (implied by STB_VORBIS_NO_PULLDATA_API) // #define STB_VORBIS_NO_INTEGER_CONVERSION // STB_VORBIS_NO_FAST_SCALED_FLOAT // does not use a fast float-to-int trick to accelerate float-to-int on // most platforms which requires endianness be defined correctly. //#define STB_VORBIS_NO_FAST_SCALED_FLOAT // STB_VORBIS_MAX_CHANNELS [number] // globally define this to the maximum number of channels you need. // The spec does not put a restriction on channels except that // the count is stored in a byte, so 255 is the hard limit. // Reducing this saves about 16 bytes per value, so using 16 saves // (255-16)*16 or around 4KB. Plus anything other memory usage // I forgot to account for. Can probably go as low as 8 (7.1 audio), // 6 (5.1 audio), or 2 (stereo only). #ifndef STB_VORBIS_MAX_CHANNELS #define STB_VORBIS_MAX_CHANNELS 16 // enough for anyone? #endif // STB_VORBIS_PUSHDATA_CRC_COUNT [number] // after a flush_pushdata(), stb_vorbis begins scanning for the // next valid page, without backtracking. when it finds something // that looks like a page, it streams through it and verifies its // CRC32. Should that validation fail, it keeps scanning. But it's // possible that _while_ streaming through to check the CRC32 of // one candidate page, it sees another candidate page. This #define // determines how many "overlapping" candidate pages it can search // at once. Note that "real" pages are typically ~4KB to ~8KB, whereas // garbage pages could be as big as 64KB, but probably average ~16KB. // So don't hose ourselves by scanning an apparent 64KB page and // missing a ton of real ones in the interim; so minimum of 2 #ifndef STB_VORBIS_PUSHDATA_CRC_COUNT #define STB_VORBIS_PUSHDATA_CRC_COUNT 4 #endif // STB_VORBIS_FAST_HUFFMAN_LENGTH [number] // sets the log size of the huffman-acceleration table. Maximum // supported value is 24. with larger numbers, more decodings are O(1), // but the table size is larger so worse cache missing, so you'll have // to probe (and try multiple ogg vorbis files) to find the sweet spot. #ifndef STB_VORBIS_FAST_HUFFMAN_LENGTH #define STB_VORBIS_FAST_HUFFMAN_LENGTH 10 #endif // STB_VORBIS_FAST_BINARY_LENGTH [number] // sets the log size of the binary-search acceleration table. this // is used in similar fashion to the fast-huffman size to set initial // parameters for the binary search // STB_VORBIS_FAST_HUFFMAN_INT // The fast huffman tables are much more efficient if they can be // stored as 16-bit results instead of 32-bit results. This restricts // the codebooks to having only 65535 possible outcomes, though. // (At least, accelerated by the huffman table.) #ifndef STB_VORBIS_FAST_HUFFMAN_INT #define STB_VORBIS_FAST_HUFFMAN_SHORT #endif // STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH // If the 'fast huffman' search doesn't succeed, then stb_vorbis falls // back on binary searching for the correct one. This requires storing // extra tables with the huffman codes in sorted order. Defining this // symbol trades off space for speed by forcing a linear search in the // non-fast case, except for "sparse" codebooks. // #define STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH // STB_VORBIS_DIVIDES_IN_RESIDUE // stb_vorbis precomputes the result of the scalar residue decoding // that would otherwise require a divide per chunk. you can trade off // space for time by defining this symbol. // #define STB_VORBIS_DIVIDES_IN_RESIDUE // STB_VORBIS_DIVIDES_IN_CODEBOOK // vorbis VQ codebooks can be encoded two ways: with every case explicitly // stored, or with all elements being chosen from a small range of values, // and all values possible in all elements. By default, stb_vorbis expands // this latter kind out to look like the former kind for ease of decoding, // because otherwise an integer divide-per-vector-element is required to // unpack the index. If you define STB_VORBIS_DIVIDES_IN_CODEBOOK, you can // trade off storage for speed. //#define STB_VORBIS_DIVIDES_IN_CODEBOOK // STB_VORBIS_CODEBOOK_SHORTS // The vorbis file format encodes VQ codebook floats as ax+b where a and // b are floating point per-codebook constants, and x is a 16-bit int. // Normally, stb_vorbis decodes them to floats rather than leaving them // as 16-bit ints and computing ax+b while decoding. This is a speed/space // tradeoff; you can save space by defining this flag. #ifndef STB_VORBIS_CODEBOOK_SHORTS #define STB_VORBIS_CODEBOOK_FLOATS #endif // STB_VORBIS_DIVIDE_TABLE // this replaces small integer divides in the floor decode loop with // table lookups. made less than 1% difference, so disabled by default. // STB_VORBIS_NO_INLINE_DECODE // disables the inlining of the scalar codebook fast-huffman decode. // might save a little codespace; useful for debugging // #define STB_VORBIS_NO_INLINE_DECODE // STB_VORBIS_NO_DEFER_FLOOR // Normally we only decode the floor without synthesizing the actual // full curve. We can instead synthesize the curve immediately. This // requires more memory and is very likely slower, so I don't think // you'd ever want to do it except for debugging. // #define STB_VORBIS_NO_DEFER_FLOOR ////////////////////////////////////////////////////////////////////////////// #ifdef STB_VORBIS_NO_PULLDATA_API #define STB_VORBIS_NO_INTEGER_CONVERSION #define STB_VORBIS_NO_STDIO #endif #if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO) #define STB_VORBIS_NO_STDIO 1 #endif #ifndef STB_VORBIS_NO_INTEGER_CONVERSION #ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT // only need endianness for fast-float-to-int, which we don't // use for pushdata #ifndef STB_VORBIS_BIG_ENDIAN #define STB_VORBIS_ENDIAN 0 #else #define STB_VORBIS_ENDIAN 1 #endif #endif #endif #ifndef STB_VORBIS_NO_STDIO #include #endif #ifndef STB_VORBIS_NO_CRT #include #include #include #include #if !(defined(__APPLE__) || defined(MACOSX) || defined(macintosh) || defined(Macintosh)) #include #endif #else #define NULL 0 #endif #if !defined(_MSC_VER) && !(defined(__MINGW32__) && defined(__forceinline)) #if __GNUC__ #define __forceinline inline #else #define __forceinline #endif #endif #if STB_VORBIS_MAX_CHANNELS > 256 #error "Value of STB_VORBIS_MAX_CHANNELS outside of allowed range" #endif #if STB_VORBIS_FAST_HUFFMAN_LENGTH > 24 #error "Value of STB_VORBIS_FAST_HUFFMAN_LENGTH outside of allowed range" #endif #define MAX_BLOCKSIZE_LOG 13 // from specification #define MAX_BLOCKSIZE (1 << MAX_BLOCKSIZE_LOG) typedef unsigned char uint8; typedef signed char int8; typedef unsigned short uint16; typedef signed short int16; typedef unsigned int uint32; typedef signed int int32; #ifndef TRUE #define TRUE 1 #define FALSE 0 #endif #ifdef STB_VORBIS_CODEBOOK_FLOATS typedef float codetype; #else typedef uint16 codetype; #endif // @NOTE // // Some arrays below are tagged "//varies", which means it's actually // a variable-sized piece of data, but rather than malloc I assume it's // small enough it's better to just allocate it all together with the // main thing // // Most of the variables are specified with the smallest size I could pack // them into. It might give better performance to make them all full-sized // integers. It should be safe to freely rearrange the structures or change // the sizes larger--nothing relies on silently truncating etc., nor the // order of variables. #define FAST_HUFFMAN_TABLE_SIZE (1 << STB_VORBIS_FAST_HUFFMAN_LENGTH) #define FAST_HUFFMAN_TABLE_MASK (FAST_HUFFMAN_TABLE_SIZE - 1) typedef struct { int dimensions, entries; uint8 *codeword_lengths; float minimum_value; float delta_value; uint8 value_bits; uint8 lookup_type; uint8 sequence_p; uint8 sparse; uint32 lookup_values; codetype *multiplicands; uint32 *codewords; #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT int16 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; #else int32 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; #endif uint32 *sorted_codewords; int *sorted_values; int sorted_entries; } Codebook; typedef struct { uint8 order; uint16 rate; uint16 bark_map_size; uint8 amplitude_bits; uint8 amplitude_offset; uint8 number_of_books; uint8 book_list[16]; // varies } Floor0; typedef struct { uint8 partitions; uint8 partition_class_list[32]; // varies uint8 class_dimensions[16]; // varies uint8 class_subclasses[16]; // varies uint8 class_masterbooks[16]; // varies int16 subclass_books[16][8]; // varies uint16 Xlist[31*8+2]; // varies uint8 sorted_order[31*8+2]; uint8 neighbors[31*8+2][2]; uint8 floor1_multiplier; uint8 rangebits; int values; } Floor1; typedef union { Floor0 floor0; Floor1 floor1; } Floor; typedef struct { uint32 begin, end; uint32 part_size; uint8 classifications; uint8 classbook; uint8 **classdata; int16 (*residue_books)[8]; } Residue; typedef struct { uint8 magnitude; uint8 angle; uint8 mux; } MappingChannel; typedef struct { uint16 coupling_steps; MappingChannel *chan; uint8 submaps; uint8 submap_floor[15]; // varies uint8 submap_residue[15]; // varies } Mapping; typedef struct { uint8 blockflag; uint8 mapping; uint16 windowtype; uint16 transformtype; } Mode; typedef struct { uint32 goal_crc; // expected crc if match int bytes_left; // bytes left in packet uint32 crc_so_far; // running crc int bytes_done; // bytes processed in _current_ chunk uint32 sample_loc; // granule pos encoded in page } CRCscan; typedef struct { uint32 page_start, page_end; uint32 after_previous_page_start; uint32 first_decoded_sample; uint32 last_decoded_sample; } ProbedPage; struct stb_vorbis { // user-accessible info unsigned int sample_rate; int channels; unsigned int setup_memory_required; unsigned int temp_memory_required; unsigned int setup_temp_memory_required; // input config #ifndef STB_VORBIS_NO_STDIO FILE *f; uint32 f_start; int close_on_free; #endif uint8 *stream; uint8 *stream_start; uint8 *stream_end; uint32 stream_len; uint8 push_mode; uint32 first_audio_page_offset; ProbedPage p_first, p_last; // memory management stb_vorbis_alloc alloc; int setup_offset; int temp_offset; // run-time results int eof; enum STBVorbisError error; // user-useful data // header info int blocksize[2]; int blocksize_0, blocksize_1; int codebook_count; Codebook *codebooks; int floor_count; uint16 floor_types[64]; // varies Floor *floor_config; int residue_count; uint16 residue_types[64]; // varies Residue *residue_config; int mapping_count; Mapping *mapping; int mode_count; Mode mode_config[64]; // varies uint32 total_samples; // decode buffer float *channel_buffers[STB_VORBIS_MAX_CHANNELS]; float *outputs [STB_VORBIS_MAX_CHANNELS]; float *previous_window[STB_VORBIS_MAX_CHANNELS]; int previous_length; #ifndef STB_VORBIS_NO_DEFER_FLOOR int16 *finalY[STB_VORBIS_MAX_CHANNELS]; #else float *floor_buffers[STB_VORBIS_MAX_CHANNELS]; #endif uint32 current_loc; // sample location of next frame to decode int current_loc_valid; // per-blocksize precomputed data // twiddle factors float *A[2],*B[2],*C[2]; float *window[2]; uint16 *bit_reverse[2]; // current page/packet/segment streaming info uint32 serial; // stream serial number for verification int last_page; int segment_count; uint8 segments[255]; uint8 page_flag; uint8 bytes_in_seg; uint8 first_decode; int next_seg; int last_seg; // flag that we're on the last segment int last_seg_which; // what was the segment number of the last seg? uint32 acc; int valid_bits; int packet_bytes; int end_seg_with_known_loc; uint32 known_loc_for_packet; int discard_samples_deferred; uint32 samples_output; // push mode scanning int page_crc_tests; // only in push_mode: number of tests active; -1 if not searching #ifndef STB_VORBIS_NO_PUSHDATA_API CRCscan scan[STB_VORBIS_PUSHDATA_CRC_COUNT]; #endif // sample-access int channel_buffer_start; int channel_buffer_end; }; extern int my_prof(int slot); //#define stb_prof my_prof #ifndef stb_prof #define stb_prof(x) ((void) 0) #endif #if defined(STB_VORBIS_NO_PUSHDATA_API) #define IS_PUSH_MODE(f) FALSE #elif defined(STB_VORBIS_NO_PULLDATA_API) #define IS_PUSH_MODE(f) TRUE #else #define IS_PUSH_MODE(f) ((f)->push_mode) #endif typedef struct stb_vorbis vorb; static int error(vorb *f, enum STBVorbisError e) { f->error = e; if (!f->eof && e != VORBIS_need_more_data) { f->error=e; // breakpoint for debugging } return 0; } // these functions are used for allocating temporary memory // while decoding. if you can afford the stack space, use // alloca(); otherwise, provide a temp buffer and it will // allocate out of those. #define array_size_required(count,size) (count*(sizeof(void *)+(size))) #define temp_alloc(f,size) (f->alloc.alloc_buffer ? setup_temp_malloc(f,size) : alloca(size)) #ifdef dealloca #define temp_free(f,p) (f->alloc.alloc_buffer ? 0 : dealloca(size)) #else #define temp_free(f,p) 0 #endif #define temp_alloc_save(f) ((f)->temp_offset) #define temp_alloc_restore(f,p) ((f)->temp_offset = (p)) #define temp_block_array(f,count,size) make_block_array(temp_alloc(f,array_size_required(count,size)), count, size) // given a sufficiently large block of memory, make an array of pointers to subblocks of it static void *make_block_array(void *mem, int count, int size) { int i; void ** p = (void **) mem; char *q = (char *) (p + count); for (i=0; i < count; ++i) { p[i] = q; q += size; } return p; } static void *setup_malloc(vorb *f, int sz) { sz = (sz+3) & ~3; f->setup_memory_required += sz; if (f->alloc.alloc_buffer) { void *p = (char *) f->alloc.alloc_buffer + f->setup_offset; if (f->setup_offset + sz > f->temp_offset) return NULL; f->setup_offset += sz; return p; } return sz ? malloc(sz) : NULL; } static void setup_free(vorb *f, void *p) { if (f->alloc.alloc_buffer) return; // do nothing; setup mem is not a stack free(p); } static void *setup_temp_malloc(vorb *f, int sz) { sz = (sz+3) & ~3; if (f->alloc.alloc_buffer) { if (f->temp_offset - sz < f->setup_offset) return NULL; f->temp_offset -= sz; return (char *) f->alloc.alloc_buffer + f->temp_offset; } return malloc(sz); } static void setup_temp_free(vorb *f, void *p, int sz) { if (f->alloc.alloc_buffer) { f->temp_offset += (sz+3)&~3; return; } free(p); } #define CRC32_POLY 0x04c11db7 // from spec static uint32 crc_table[256]; static void crc32_init(void) { int i,j; uint32 s; for(i=0; i < 256; i++) { for (s=i<<24, j=0; j < 8; ++j) s = (s << 1) ^ (s >= (1U<<31) ? CRC32_POLY : 0); crc_table[i] = s; } } static __forceinline uint32 crc32_update(uint32 crc, uint8 byte) { return (crc << 8) ^ crc_table[byte ^ (crc >> 24)]; } // used in setup, and for huffman that doesn't go fast path static unsigned int bit_reverse(unsigned int n) { n = ((n & 0xAAAAAAAA) >> 1) | ((n & 0x55555555) << 1); n = ((n & 0xCCCCCCCC) >> 2) | ((n & 0x33333333) << 2); n = ((n & 0xF0F0F0F0) >> 4) | ((n & 0x0F0F0F0F) << 4); n = ((n & 0xFF00FF00) >> 8) | ((n & 0x00FF00FF) << 8); return (n >> 16) | (n << 16); } static float square(float x) { return x*x; } // this is a weird definition of log2() for which log2(1) = 1, log2(2) = 2, log2(4) = 3 // as required by the specification. fast(?) implementation from stb.h // @OPTIMIZE: called multiple times per-packet with "constants"; move to setup static int ilog(int32 n) { static signed char log2_4[16] = { 0,1,2,2,3,3,3,3,4,4,4,4,4,4,4,4 }; // 2 compares if n < 16, 3 compares otherwise (4 if signed or n > 1<<29) if (n < (1 << 14)) if (n < (1 << 4)) return 0 + log2_4[n ]; else if (n < (1 << 9)) return 5 + log2_4[n >> 5]; else return 10 + log2_4[n >> 10]; else if (n < (1 << 24)) if (n < (1 << 19)) return 15 + log2_4[n >> 15]; else return 20 + log2_4[n >> 20]; else if (n < (1 << 29)) return 25 + log2_4[n >> 25]; else if (n < (1 << 31)) return 30 + log2_4[n >> 30]; else return 0; // signed n returns 0 } #ifndef M_PI #define M_PI 3.14159265358979323846264f // from CRC #endif // code length assigned to a value with no huffman encoding #define NO_CODE 255 /////////////////////// LEAF SETUP FUNCTIONS ////////////////////////// // // these functions are only called at setup, and only a few times // per file static float float32_unpack(uint32 x) { // from the specification uint32 mantissa = x & 0x1fffff; uint32 sign = x & 0x80000000; uint32 exp = (x & 0x7fe00000) >> 21; double res = sign ? -(double)mantissa : (double)mantissa; return (float) ldexp((float)res, exp-788); } // zlib & jpeg huffman tables assume that the output symbols // can either be arbitrarily arranged, or have monotonically // increasing frequencies--they rely on the lengths being sorted; // this makes for a very simple generation algorithm. // vorbis allows a huffman table with non-sorted lengths. This // requires a more sophisticated construction, since symbols in // order do not map to huffman codes "in order". static void add_entry(Codebook *c, uint32 huff_code, int symbol, int count, int len, uint32 *values) { if (!c->sparse) { c->codewords [symbol] = huff_code; } else { c->codewords [count] = huff_code; c->codeword_lengths[count] = len; values [count] = symbol; } } static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values) { int i,k,m=0; uint32 available[32]; memset(available, 0, sizeof(available)); // find the first entry for (k=0; k < n; ++k) if (len[k] < NO_CODE) break; if (k == n) { assert(c->sorted_entries == 0); return TRUE; } // add to the list add_entry(c, 0, k, m++, len[k], values); // add all available leaves for (i=1; i <= len[k]; ++i) available[i] = 1 << (32-i); // note that the above code treats the first case specially, // but it's really the same as the following code, so they // could probably be combined (except the initial code is 0, // and I use 0 in available[] to mean 'empty') for (i=k+1; i < n; ++i) { uint32 res; int z = len[i], y; if (z == NO_CODE) continue; // find lowest available leaf (should always be earliest, // which is what the specification calls for) // note that this property, and the fact we can never have // more than one free leaf at a given level, isn't totally // trivial to prove, but it seems true and the assert never // fires, so! while (z > 0 && !available[z]) --z; if (z == 0) { assert(0); return FALSE; } res = available[z]; available[z] = 0; add_entry(c, bit_reverse(res), i, m++, len[i], values); // propogate availability up the tree if (z != len[i]) { for (y=len[i]; y > z; --y) { assert(available[y] == 0); available[y] = res + (1 << (32-y)); } } } return TRUE; } // accelerated huffman table allows fast O(1) match of all symbols // of length <= STB_VORBIS_FAST_HUFFMAN_LENGTH static void compute_accelerated_huffman(Codebook *c) { int i, len; for (i=0; i < FAST_HUFFMAN_TABLE_SIZE; ++i) c->fast_huffman[i] = -1; len = c->sparse ? c->sorted_entries : c->entries; #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT if (len > 32767) len = 32767; // largest possible value we can encode! #endif for (i=0; i < len; ++i) { if (c->codeword_lengths[i] <= STB_VORBIS_FAST_HUFFMAN_LENGTH) { uint32 z = c->sparse ? bit_reverse(c->sorted_codewords[i]) : c->codewords[i]; // set table entries for all bit combinations in the higher bits while (z < FAST_HUFFMAN_TABLE_SIZE) { c->fast_huffman[z] = i; z += 1 << c->codeword_lengths[i]; } } } } #ifdef _MSC_VER #define STBV_CDECL __cdecl #else #define STBV_CDECL #endif static int STBV_CDECL uint32_compare(const void *p, const void *q) { uint32 x = * (uint32 *) p; uint32 y = * (uint32 *) q; return x < y ? -1 : x > y; } static int include_in_sort(Codebook *c, uint8 len) { if (c->sparse) { assert(len != NO_CODE); return TRUE; } if (len == NO_CODE) return FALSE; if (len > STB_VORBIS_FAST_HUFFMAN_LENGTH) return TRUE; return FALSE; } // if the fast table above doesn't work, we want to binary // search them... need to reverse the bits static void compute_sorted_huffman(Codebook *c, uint8 *lengths, uint32 *values) { int i, len; // build a list of all the entries // OPTIMIZATION: don't include the short ones, since they'll be caught by FAST_HUFFMAN. // this is kind of a frivolous optimization--I don't see any performance improvement, // but it's like 4 extra lines of code, so. if (!c->sparse) { int k = 0; for (i=0; i < c->entries; ++i) if (include_in_sort(c, lengths[i])) c->sorted_codewords[k++] = bit_reverse(c->codewords[i]); assert(k == c->sorted_entries); } else { for (i=0; i < c->sorted_entries; ++i) c->sorted_codewords[i] = bit_reverse(c->codewords[i]); } qsort(c->sorted_codewords, c->sorted_entries, sizeof(c->sorted_codewords[0]), uint32_compare); c->sorted_codewords[c->sorted_entries] = 0xffffffff; len = c->sparse ? c->sorted_entries : c->entries; // now we need to indicate how they correspond; we could either // #1: sort a different data structure that says who they correspond to // #2: for each sorted entry, search the original list to find who corresponds // #3: for each original entry, find the sorted entry // #1 requires extra storage, #2 is slow, #3 can use binary search! for (i=0; i < len; ++i) { int huff_len = c->sparse ? lengths[values[i]] : lengths[i]; if (include_in_sort(c,huff_len)) { uint32 code = bit_reverse(c->codewords[i]); int x=0, n=c->sorted_entries; while (n > 1) { // invariant: sc[x] <= code < sc[x+n] int m = x + (n >> 1); if (c->sorted_codewords[m] <= code) { x = m; n -= (n>>1); } else { n >>= 1; } } assert(c->sorted_codewords[x] == code); if (c->sparse) { c->sorted_values[x] = values[i]; c->codeword_lengths[x] = huff_len; } else { c->sorted_values[x] = i; } } } } // only run while parsing the header (3 times) static int vorbis_validate(uint8 *data) { static uint8 vorbis[6] = { 'v', 'o', 'r', 'b', 'i', 's' }; return memcmp(data, vorbis, 6) == 0; } // called from setup only, once per code book // (formula implied by specification) static int lookup1_values(int entries, int dim) { int r = (int) floor(exp((float) log((float) entries) / dim)); if ((int) floor(pow((float) r+1, dim)) <= entries) // (int) cast for MinGW warning; ++r; // floor() to avoid _ftol() when non-CRT assert(pow((float) r+1, dim) > entries); assert((int) floor(pow((float) r, dim)) <= entries); // (int),floor() as above return r; } // called twice per file static void compute_twiddle_factors(int n, float *A, float *B, float *C) { int n4 = n >> 2, n8 = n >> 3; int k,k2; for (k=k2=0; k < n4; ++k,k2+=2) { A[k2 ] = (float) cos(4*k*M_PI/n); A[k2+1] = (float) -sin(4*k*M_PI/n); B[k2 ] = (float) cos((k2+1)*M_PI/n/2) * 0.5f; B[k2+1] = (float) sin((k2+1)*M_PI/n/2) * 0.5f; } for (k=k2=0; k < n8; ++k,k2+=2) { C[k2 ] = (float) cos(2*(k2+1)*M_PI/n); C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n); } } static void compute_window(int n, float *window) { int n2 = n >> 1, i; for (i=0; i < n2; ++i) window[i] = (float) sin(0.5 * M_PI * square((float) sin((i - 0 + 0.5) / n2 * 0.5 * M_PI))); } static void compute_bitreverse(int n, uint16 *rev) { int ld = ilog(n) - 1; // ilog is off-by-one from normal definitions int i, n8 = n >> 3; for (i=0; i < n8; ++i) rev[i] = (bit_reverse(i) >> (32-ld+3)) << 2; } static int init_blocksize(vorb *f, int b, int n) { int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3; f->A[b] = (float *) setup_malloc(f, sizeof(float) * n2); f->B[b] = (float *) setup_malloc(f, sizeof(float) * n2); f->C[b] = (float *) setup_malloc(f, sizeof(float) * n4); if (!f->A[b] || !f->B[b] || !f->C[b]) return error(f, VORBIS_outofmem); compute_twiddle_factors(n, f->A[b], f->B[b], f->C[b]); f->window[b] = (float *) setup_malloc(f, sizeof(float) * n2); if (!f->window[b]) return error(f, VORBIS_outofmem); compute_window(n, f->window[b]); f->bit_reverse[b] = (uint16 *) setup_malloc(f, sizeof(uint16) * n8); if (!f->bit_reverse[b]) return error(f, VORBIS_outofmem); compute_bitreverse(n, f->bit_reverse[b]); return TRUE; } static void neighbors(uint16 *x, int n, int *plow, int *phigh) { int low = -1; int high = 65536; int i; for (i=0; i < n; ++i) { if (x[i] > low && x[i] < x[n]) { *plow = i; low = x[i]; } if (x[i] < high && x[i] > x[n]) { *phigh = i; high = x[i]; } } } // this has been repurposed so y is now the original index instead of y typedef struct { uint16 x,y; } Point; static int STBV_CDECL point_compare(const void *p, const void *q) { Point *a = (Point *) p; Point *b = (Point *) q; return a->x < b->x ? -1 : a->x > b->x; } // /////////////////////// END LEAF SETUP FUNCTIONS ////////////////////////// #if defined(STB_VORBIS_NO_STDIO) #define USE_MEMORY(z) TRUE #else #define USE_MEMORY(z) ((z)->stream) #endif static uint8 get8(vorb *z) { if (USE_MEMORY(z)) { if (z->stream >= z->stream_end) { z->eof = TRUE; return 0; } return *z->stream++; } #ifndef STB_VORBIS_NO_STDIO { int c = fgetc(z->f); if (c == EOF) { z->eof = TRUE; return 0; } return c; } #endif } static uint32 get32(vorb *f) { uint32 x; x = get8(f); x += get8(f) << 8; x += get8(f) << 16; x += get8(f) << 24; return x; } static int getn(vorb *z, uint8 *data, int n) { if (USE_MEMORY(z)) { if (z->stream+n > z->stream_end) { z->eof = 1; return 0; } memcpy(data, z->stream, n); z->stream += n; return 1; } #ifndef STB_VORBIS_NO_STDIO if (fread(data, n, 1, z->f) == 1) return 1; else { z->eof = 1; return 0; } #endif } static void skip(vorb *z, int n) { if (USE_MEMORY(z)) { z->stream += n; if (z->stream >= z->stream_end) z->eof = 1; return; } #ifndef STB_VORBIS_NO_STDIO { long x = ftell(z->f); fseek(z->f, x+n, SEEK_SET); } #endif } static int set_file_offset(stb_vorbis *f, unsigned int loc) { #ifndef STB_VORBIS_NO_PUSHDATA_API if (f->push_mode) return 0; #endif f->eof = 0; if (USE_MEMORY(f)) { if (f->stream_start + loc >= f->stream_end || f->stream_start + loc < f->stream_start) { f->stream = f->stream_end; f->eof = 1; return 0; } else { f->stream = f->stream_start + loc; return 1; } } #ifndef STB_VORBIS_NO_STDIO if (loc + f->f_start < loc || loc >= 0x80000000) { loc = 0x7fffffff; f->eof = 1; } else { loc += f->f_start; } if (!fseek(f->f, loc, SEEK_SET)) return 1; f->eof = 1; fseek(f->f, f->f_start, SEEK_END); return 0; #endif } static uint8 ogg_page_header[4] = { 0x4f, 0x67, 0x67, 0x53 }; static int capture_pattern(vorb *f) { if (0x4f != get8(f)) return FALSE; if (0x67 != get8(f)) return FALSE; if (0x67 != get8(f)) return FALSE; if (0x53 != get8(f)) return FALSE; return TRUE; } #define PAGEFLAG_continued_packet 1 #define PAGEFLAG_first_page 2 #define PAGEFLAG_last_page 4 static int start_page_no_capturepattern(vorb *f) { uint32 loc0,loc1,n; // stream structure version if (0 != get8(f)) return error(f, VORBIS_invalid_stream_structure_version); // header flag f->page_flag = get8(f); // absolute granule position loc0 = get32(f); loc1 = get32(f); // @TODO: validate loc0,loc1 as valid positions? // stream serial number -- vorbis doesn't interleave, so discard get32(f); //if (f->serial != get32(f)) return error(f, VORBIS_incorrect_stream_serial_number); // page sequence number n = get32(f); f->last_page = n; // CRC32 get32(f); // page_segments f->segment_count = get8(f); if (!getn(f, f->segments, f->segment_count)) return error(f, VORBIS_unexpected_eof); // assume we _don't_ know any the sample position of any segments f->end_seg_with_known_loc = -2; if (loc0 != ~0U || loc1 != ~0U) { int i; // determine which packet is the last one that will complete for (i=f->segment_count-1; i >= 0; --i) if (f->segments[i] < 255) break; // 'i' is now the index of the _last_ segment of a packet that ends if (i >= 0) { f->end_seg_with_known_loc = i; f->known_loc_for_packet = loc0; } } if (f->first_decode) { int i,len; ProbedPage p; len = 0; for (i=0; i < f->segment_count; ++i) len += f->segments[i]; len += 27 + f->segment_count; p.page_start = f->first_audio_page_offset; p.page_end = p.page_start + len; p.after_previous_page_start = p.page_start; p.first_decoded_sample = 0; p.last_decoded_sample = loc0; f->p_first = p; } f->next_seg = 0; return TRUE; } static int start_page(vorb *f) { if (!capture_pattern(f)) return error(f, VORBIS_missing_capture_pattern); return start_page_no_capturepattern(f); } static int start_packet(vorb *f) { while (f->next_seg == -1) { if (!start_page(f)) return FALSE; if (f->page_flag & PAGEFLAG_continued_packet) return error(f, VORBIS_continued_packet_flag_invalid); } f->last_seg = FALSE; f->valid_bits = 0; f->packet_bytes = 0; f->bytes_in_seg = 0; // f->next_seg is now valid return TRUE; } static int maybe_start_packet(vorb *f) { if (f->next_seg == -1) { int x = get8(f); if (f->eof) return FALSE; // EOF at page boundary is not an error! if (0x4f != x ) return error(f, VORBIS_missing_capture_pattern); if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern); if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern); if (0x53 != get8(f)) return error(f, VORBIS_missing_capture_pattern); if (!start_page_no_capturepattern(f)) return FALSE; if (f->page_flag & PAGEFLAG_continued_packet) { // set up enough state that we can read this packet if we want, // e.g. during recovery f->last_seg = FALSE; f->bytes_in_seg = 0; return error(f, VORBIS_continued_packet_flag_invalid); } } return start_packet(f); } static int next_segment(vorb *f) { int len; if (f->last_seg) return 0; if (f->next_seg == -1) { f->last_seg_which = f->segment_count-1; // in case start_page fails if (!start_page(f)) { f->last_seg = 1; return 0; } if (!(f->page_flag & PAGEFLAG_continued_packet)) return error(f, VORBIS_continued_packet_flag_invalid); } len = f->segments[f->next_seg++]; if (len < 255) { f->last_seg = TRUE; f->last_seg_which = f->next_seg-1; } if (f->next_seg >= f->segment_count) f->next_seg = -1; assert(f->bytes_in_seg == 0); f->bytes_in_seg = len; return len; } #define EOP (-1) #define INVALID_BITS (-1) static int get8_packet_raw(vorb *f) { if (!f->bytes_in_seg) { // CLANG! if (f->last_seg) return EOP; else if (!next_segment(f)) return EOP; } assert(f->bytes_in_seg > 0); --f->bytes_in_seg; ++f->packet_bytes; return get8(f); } static int get8_packet(vorb *f) { int x = get8_packet_raw(f); f->valid_bits = 0; return x; } static void flush_packet(vorb *f) { while (get8_packet_raw(f) != EOP); } // @OPTIMIZE: this is the secondary bit decoder, so it's probably not as important // as the huffman decoder? static uint32 get_bits(vorb *f, int n) { uint32 z; if (f->valid_bits < 0) return 0; if (f->valid_bits < n) { if (n > 24) { // the accumulator technique below would not work correctly in this case z = get_bits(f, 24); z += get_bits(f, n-24) << 24; return z; } if (f->valid_bits == 0) f->acc = 0; while (f->valid_bits < n) { int z = get8_packet_raw(f); if (z == EOP) { f->valid_bits = INVALID_BITS; return 0; } f->acc += z << f->valid_bits; f->valid_bits += 8; } } if (f->valid_bits < 0) return 0; z = f->acc & ((1 << n)-1); f->acc >>= n; f->valid_bits -= n; return z; } // @OPTIMIZE: primary accumulator for huffman // expand the buffer to as many bits as possible without reading off end of packet // it might be nice to allow f->valid_bits and f->acc to be stored in registers, // e.g. cache them locally and decode locally static __forceinline void prep_huffman(vorb *f) { if (f->valid_bits <= 24) { if (f->valid_bits == 0) f->acc = 0; do { int z; if (f->last_seg && !f->bytes_in_seg) return; z = get8_packet_raw(f); if (z == EOP) return; f->acc += z << f->valid_bits; f->valid_bits += 8; } while (f->valid_bits <= 24); } } enum { VORBIS_packet_id = 1, VORBIS_packet_comment = 3, VORBIS_packet_setup = 5, }; static int codebook_decode_scalar_raw(vorb *f, Codebook *c) { int i; prep_huffman(f); assert(c->sorted_codewords || c->codewords); // cases to use binary search: sorted_codewords && !c->codewords // sorted_codewords && c->entries > 8 if (c->entries > 8 ? c->sorted_codewords!=NULL : !c->codewords) { // binary search uint32 code = bit_reverse(f->acc); int x=0, n=c->sorted_entries, len; while (n > 1) { // invariant: sc[x] <= code < sc[x+n] int m = x + (n >> 1); if (c->sorted_codewords[m] <= code) { x = m; n -= (n>>1); } else { n >>= 1; } } // x is now the sorted index if (!c->sparse) x = c->sorted_values[x]; // x is now sorted index if sparse, or symbol otherwise len = c->codeword_lengths[x]; if (f->valid_bits >= len) { f->acc >>= len; f->valid_bits -= len; return x; } f->valid_bits = 0; return -1; } // if small, linear search assert(!c->sparse); for (i=0; i < c->entries; ++i) { if (c->codeword_lengths[i] == NO_CODE) continue; if (c->codewords[i] == (f->acc & ((1 << c->codeword_lengths[i])-1))) { if (f->valid_bits >= c->codeword_lengths[i]) { f->acc >>= c->codeword_lengths[i]; f->valid_bits -= c->codeword_lengths[i]; return i; } f->valid_bits = 0; return -1; } } error(f, VORBIS_invalid_stream); f->valid_bits = 0; return -1; } #ifndef STB_VORBIS_NO_INLINE_DECODE #define DECODE_RAW(var, f,c) \ if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH) \ prep_huffman(f); \ var = f->acc & FAST_HUFFMAN_TABLE_MASK; \ var = c->fast_huffman[var]; \ if (var >= 0) { \ int n = c->codeword_lengths[var]; \ f->acc >>= n; \ f->valid_bits -= n; \ if (f->valid_bits < 0) { f->valid_bits = 0; var = -1; } \ } else { \ var = codebook_decode_scalar_raw(f,c); \ } #else static int codebook_decode_scalar(vorb *f, Codebook *c) { int i; if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH) prep_huffman(f); // fast huffman table lookup i = f->acc & FAST_HUFFMAN_TABLE_MASK; i = c->fast_huffman[i]; if (i >= 0) { f->acc >>= c->codeword_lengths[i]; f->valid_bits -= c->codeword_lengths[i]; if (f->valid_bits < 0) { f->valid_bits = 0; return -1; } return i; } return codebook_decode_scalar_raw(f,c); } #define DECODE_RAW(var,f,c) var = codebook_decode_scalar(f,c); #endif #define DECODE(var,f,c) \ DECODE_RAW(var,f,c) \ if (c->sparse) var = c->sorted_values[var]; #ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK #define DECODE_VQ(var,f,c) DECODE_RAW(var,f,c) #else #define DECODE_VQ(var,f,c) DECODE(var,f,c) #endif // CODEBOOK_ELEMENT_FAST is an optimization for the CODEBOOK_FLOATS case // where we avoid one addition #ifndef STB_VORBIS_CODEBOOK_FLOATS #define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off] * c->delta_value + c->minimum_value) #define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off] * c->delta_value) #define CODEBOOK_ELEMENT_BASE(c) (c->minimum_value) #else #define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off]) #define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off]) #define CODEBOOK_ELEMENT_BASE(c) (0) #endif static int codebook_decode_start(vorb *f, Codebook *c) { int z = -1; // type 0 is only legal in a scalar context if (c->lookup_type == 0) error(f, VORBIS_invalid_stream); else { DECODE_VQ(z,f,c); if (c->sparse) assert(z < c->sorted_entries); if (z < 0) { // check for EOP if (!f->bytes_in_seg) if (f->last_seg) return z; error(f, VORBIS_invalid_stream); } } return z; } static int codebook_decode(vorb *f, Codebook *c, float *output, int len) { int i,z = codebook_decode_start(f,c); if (z < 0) return FALSE; if (len > c->dimensions) len = c->dimensions; #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK if (c->lookup_type == 1) { float last = CODEBOOK_ELEMENT_BASE(c); int div = 1; for (i=0; i < len; ++i) { int off = (z / div) % c->lookup_values; float val = CODEBOOK_ELEMENT_FAST(c,off) + last; output[i] += val; if (c->sequence_p) last = val + c->minimum_value; div *= c->lookup_values; } return TRUE; } #endif z *= c->dimensions; if (c->sequence_p) { float last = CODEBOOK_ELEMENT_BASE(c); for (i=0; i < len; ++i) { float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; output[i] += val; last = val + c->minimum_value; } } else { float last = CODEBOOK_ELEMENT_BASE(c); for (i=0; i < len; ++i) { output[i] += CODEBOOK_ELEMENT_FAST(c,z+i) + last; } } return TRUE; } static int codebook_decode_step(vorb *f, Codebook *c, float *output, int len, int step) { int i,z = codebook_decode_start(f,c); float last = CODEBOOK_ELEMENT_BASE(c); if (z < 0) return FALSE; if (len > c->dimensions) len = c->dimensions; #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK if (c->lookup_type == 1) { int div = 1; for (i=0; i < len; ++i) { int off = (z / div) % c->lookup_values; float val = CODEBOOK_ELEMENT_FAST(c,off) + last; output[i*step] += val; if (c->sequence_p) last = val; div *= c->lookup_values; } return TRUE; } #endif z *= c->dimensions; for (i=0; i < len; ++i) { float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; output[i*step] += val; if (c->sequence_p) last = val; } return TRUE; } static int codebook_decode_deinterleave_repeat(vorb *f, Codebook *c, float **outputs, int ch, int *c_inter_p, int *p_inter_p, int len, int total_decode) { int c_inter = *c_inter_p; int p_inter = *p_inter_p; int i,z, effective = c->dimensions; // type 0 is only legal in a scalar context if (c->lookup_type == 0) return error(f, VORBIS_invalid_stream); while (total_decode > 0) { float last = CODEBOOK_ELEMENT_BASE(c); DECODE_VQ(z,f,c); #ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK assert(!c->sparse || z < c->sorted_entries); #endif if (z < 0) { if (!f->bytes_in_seg) if (f->last_seg) return FALSE; return error(f, VORBIS_invalid_stream); } // if this will take us off the end of the buffers, stop short! // we check by computing the length of the virtual interleaved // buffer (len*ch), our current offset within it (p_inter*ch)+(c_inter), // and the length we'll be using (effective) if (c_inter + p_inter*ch + effective > len * ch) { effective = len*ch - (p_inter*ch - c_inter); } #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK if (c->lookup_type == 1) { int div = 1; for (i=0; i < effective; ++i) { int off = (z / div) % c->lookup_values; float val = CODEBOOK_ELEMENT_FAST(c,off) + last; if (outputs[c_inter]) outputs[c_inter][p_inter] += val; if (++c_inter == ch) { c_inter = 0; ++p_inter; } if (c->sequence_p) last = val; div *= c->lookup_values; } } else #endif { z *= c->dimensions; if (c->sequence_p) { for (i=0; i < effective; ++i) { float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; if (outputs[c_inter]) outputs[c_inter][p_inter] += val; if (++c_inter == ch) { c_inter = 0; ++p_inter; } last = val; } } else { for (i=0; i < effective; ++i) { float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; if (outputs[c_inter]) outputs[c_inter][p_inter] += val; if (++c_inter == ch) { c_inter = 0; ++p_inter; } } } } total_decode -= effective; } *c_inter_p = c_inter; *p_inter_p = p_inter; return TRUE; } #ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK static int codebook_decode_deinterleave_repeat_2(vorb *f, Codebook *c, float **outputs, int *c_inter_p, int *p_inter_p, int len, int total_decode) { int c_inter = *c_inter_p; int p_inter = *p_inter_p; int i,z, effective = c->dimensions; // type 0 is only legal in a scalar context if (c->lookup_type == 0) return error(f, VORBIS_invalid_stream); while (total_decode > 0) { float last = CODEBOOK_ELEMENT_BASE(c); DECODE_VQ(z,f,c); if (z < 0) { if (!f->bytes_in_seg) if (f->last_seg) return FALSE; return error(f, VORBIS_invalid_stream); } // if this will take us off the end of the buffers, stop short! // we check by computing the length of the virtual interleaved // buffer (len*ch), our current offset within it (p_inter*ch)+(c_inter), // and the length we'll be using (effective) if (c_inter + p_inter*2 + effective > len * 2) { effective = len*2 - (p_inter*2 - c_inter); } { z *= c->dimensions; stb_prof(11); if (c->sequence_p) { // haven't optimized this case because I don't have any examples for (i=0; i < effective; ++i) { float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; if (outputs[c_inter]) outputs[c_inter][p_inter] += val; if (++c_inter == 2) { c_inter = 0; ++p_inter; } last = val; } } else { i=0; if (c_inter == 1) { float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; if (outputs[c_inter]) outputs[c_inter][p_inter] += val; c_inter = 0; ++p_inter; ++i; } { float *z0 = outputs[0]; float *z1 = outputs[1]; for (; i+1 < effective;) { float v0 = CODEBOOK_ELEMENT_FAST(c,z+i) + last; float v1 = CODEBOOK_ELEMENT_FAST(c,z+i+1) + last; if (z0) z0[p_inter] += v0; if (z1) z1[p_inter] += v1; ++p_inter; i += 2; } } if (i < effective) { float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; if (outputs[c_inter]) outputs[c_inter][p_inter] += val; if (++c_inter == 2) { c_inter = 0; ++p_inter; } } } } total_decode -= effective; } *c_inter_p = c_inter; *p_inter_p = p_inter; return TRUE; } #endif static int predict_point(int x, int x0, int x1, int y0, int y1) { int dy = y1 - y0; int adx = x1 - x0; // @OPTIMIZE: force int division to round in the right direction... is this necessary on x86? int err = abs(dy) * (x - x0); int off = err / adx; return dy < 0 ? y0 - off : y0 + off; } // the following table is block-copied from the specification static float inverse_db_table[256] = { 1.0649863e-07f, 1.1341951e-07f, 1.2079015e-07f, 1.2863978e-07f, 1.3699951e-07f, 1.4590251e-07f, 1.5538408e-07f, 1.6548181e-07f, 1.7623575e-07f, 1.8768855e-07f, 1.9988561e-07f, 2.1287530e-07f, 2.2670913e-07f, 2.4144197e-07f, 2.5713223e-07f, 2.7384213e-07f, 2.9163793e-07f, 3.1059021e-07f, 3.3077411e-07f, 3.5226968e-07f, 3.7516214e-07f, 3.9954229e-07f, 4.2550680e-07f, 4.5315863e-07f, 4.8260743e-07f, 5.1396998e-07f, 5.4737065e-07f, 5.8294187e-07f, 6.2082472e-07f, 6.6116941e-07f, 7.0413592e-07f, 7.4989464e-07f, 7.9862701e-07f, 8.5052630e-07f, 9.0579828e-07f, 9.6466216e-07f, 1.0273513e-06f, 1.0941144e-06f, 1.1652161e-06f, 1.2409384e-06f, 1.3215816e-06f, 1.4074654e-06f, 1.4989305e-06f, 1.5963394e-06f, 1.7000785e-06f, 1.8105592e-06f, 1.9282195e-06f, 2.0535261e-06f, 2.1869758e-06f, 2.3290978e-06f, 2.4804557e-06f, 2.6416497e-06f, 2.8133190e-06f, 2.9961443e-06f, 3.1908506e-06f, 3.3982101e-06f, 3.6190449e-06f, 3.8542308e-06f, 4.1047004e-06f, 4.3714470e-06f, 4.6555282e-06f, 4.9580707e-06f, 5.2802740e-06f, 5.6234160e-06f, 5.9888572e-06f, 6.3780469e-06f, 6.7925283e-06f, 7.2339451e-06f, 7.7040476e-06f, 8.2047000e-06f, 8.7378876e-06f, 9.3057248e-06f, 9.9104632e-06f, 1.0554501e-05f, 1.1240392e-05f, 1.1970856e-05f, 1.2748789e-05f, 1.3577278e-05f, 1.4459606e-05f, 1.5399272e-05f, 1.6400004e-05f, 1.7465768e-05f, 1.8600792e-05f, 1.9809576e-05f, 2.1096914e-05f, 2.2467911e-05f, 2.3928002e-05f, 2.5482978e-05f, 2.7139006e-05f, 2.8902651e-05f, 3.0780908e-05f, 3.2781225e-05f, 3.4911534e-05f, 3.7180282e-05f, 3.9596466e-05f, 4.2169667e-05f, 4.4910090e-05f, 4.7828601e-05f, 5.0936773e-05f, 5.4246931e-05f, 5.7772202e-05f, 6.1526565e-05f, 6.5524908e-05f, 6.9783085e-05f, 7.4317983e-05f, 7.9147585e-05f, 8.4291040e-05f, 8.9768747e-05f, 9.5602426e-05f, 0.00010181521f, 0.00010843174f, 0.00011547824f, 0.00012298267f, 0.00013097477f, 0.00013948625f, 0.00014855085f, 0.00015820453f, 0.00016848555f, 0.00017943469f, 0.00019109536f, 0.00020351382f, 0.00021673929f, 0.00023082423f, 0.00024582449f, 0.00026179955f, 0.00027881276f, 0.00029693158f, 0.00031622787f, 0.00033677814f, 0.00035866388f, 0.00038197188f, 0.00040679456f, 0.00043323036f, 0.00046138411f, 0.00049136745f, 0.00052329927f, 0.00055730621f, 0.00059352311f, 0.00063209358f, 0.00067317058f, 0.00071691700f, 0.00076350630f, 0.00081312324f, 0.00086596457f, 0.00092223983f, 0.00098217216f, 0.0010459992f, 0.0011139742f, 0.0011863665f, 0.0012634633f, 0.0013455702f, 0.0014330129f, 0.0015261382f, 0.0016253153f, 0.0017309374f, 0.0018434235f, 0.0019632195f, 0.0020908006f, 0.0022266726f, 0.0023713743f, 0.0025254795f, 0.0026895994f, 0.0028643847f, 0.0030505286f, 0.0032487691f, 0.0034598925f, 0.0036847358f, 0.0039241906f, 0.0041792066f, 0.0044507950f, 0.0047400328f, 0.0050480668f, 0.0053761186f, 0.0057254891f, 0.0060975636f, 0.0064938176f, 0.0069158225f, 0.0073652516f, 0.0078438871f, 0.0083536271f, 0.0088964928f, 0.009474637f, 0.010090352f, 0.010746080f, 0.011444421f, 0.012188144f, 0.012980198f, 0.013823725f, 0.014722068f, 0.015678791f, 0.016697687f, 0.017782797f, 0.018938423f, 0.020169149f, 0.021479854f, 0.022875735f, 0.024362330f, 0.025945531f, 0.027631618f, 0.029427276f, 0.031339626f, 0.033376252f, 0.035545228f, 0.037855157f, 0.040315199f, 0.042935108f, 0.045725273f, 0.048696758f, 0.051861348f, 0.055231591f, 0.058820850f, 0.062643361f, 0.066714279f, 0.071049749f, 0.075666962f, 0.080584227f, 0.085821044f, 0.091398179f, 0.097337747f, 0.10366330f, 0.11039993f, 0.11757434f, 0.12521498f, 0.13335215f, 0.14201813f, 0.15124727f, 0.16107617f, 0.17154380f, 0.18269168f, 0.19456402f, 0.20720788f, 0.22067342f, 0.23501402f, 0.25028656f, 0.26655159f, 0.28387361f, 0.30232132f, 0.32196786f, 0.34289114f, 0.36517414f, 0.38890521f, 0.41417847f, 0.44109412f, 0.46975890f, 0.50028648f, 0.53279791f, 0.56742212f, 0.60429640f, 0.64356699f, 0.68538959f, 0.72993007f, 0.77736504f, 0.82788260f, 0.88168307f, 0.9389798f, 1.0f }; // @OPTIMIZE: if you want to replace this bresenham line-drawing routine, // note that you must produce bit-identical output to decode correctly; // this specific sequence of operations is specified in the spec (it's // drawing integer-quantized frequency-space lines that the encoder // expects to be exactly the same) // ... also, isn't the whole point of Bresenham's algorithm to NOT // have to divide in the setup? sigh. #ifndef STB_VORBIS_NO_DEFER_FLOOR #define LINE_OP(a,b) a *= b #else #define LINE_OP(a,b) a = b #endif #ifdef STB_VORBIS_DIVIDE_TABLE #define DIVTAB_NUMER 32 #define DIVTAB_DENOM 64 int8 integer_divide_table[DIVTAB_NUMER][DIVTAB_DENOM]; // 2KB #endif static __forceinline void draw_line(float *output, int x0, int y0, int x1, int y1, int n) { int dy = y1 - y0; int adx = x1 - x0; int ady = abs(dy); int base; int x=x0,y=y0; int err = 0; int sy; #ifdef STB_VORBIS_DIVIDE_TABLE if (adx < DIVTAB_DENOM && ady < DIVTAB_NUMER) { if (dy < 0) { base = -integer_divide_table[ady][adx]; sy = base-1; } else { base = integer_divide_table[ady][adx]; sy = base+1; } } else { base = dy / adx; if (dy < 0) sy = base - 1; else sy = base+1; } #else base = dy / adx; if (dy < 0) sy = base - 1; else sy = base+1; #endif ady -= abs(base) * adx; if (x1 > n) x1 = n; LINE_OP(output[x], inverse_db_table[y]); for (++x; x < x1; ++x) { err += ady; if (err >= adx) { err -= adx; y += sy; } else y += base; LINE_OP(output[x], inverse_db_table[y]); } } static int residue_decode(vorb *f, Codebook *book, float *target, int offset, int n, int rtype) { int k; if (rtype == 0) { int step = n / book->dimensions; for (k=0; k < step; ++k) if (!codebook_decode_step(f, book, target+offset+k, n-offset-k, step)) return FALSE; } else { for (k=0; k < n; ) { if (!codebook_decode(f, book, target+offset, n-k)) return FALSE; k += book->dimensions; offset += book->dimensions; } } return TRUE; } static void decode_residue(vorb *f, float *residue_buffers[], int ch, int n, int rn, uint8 *do_not_decode) { int i,j,pass; Residue *r = f->residue_config + rn; int rtype = f->residue_types[rn]; int c = r->classbook; int classwords = f->codebooks[c].dimensions; int n_read = r->end - r->begin; int part_read = n_read / r->part_size; int temp_alloc_point = temp_alloc_save(f); #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE uint8 ***part_classdata = (uint8 ***) temp_block_array(f,f->channels, part_read * sizeof(**part_classdata)); #else int **classifications = (int **) temp_block_array(f,f->channels, part_read * sizeof(**classifications)); #endif stb_prof(2); for (i=0; i < ch; ++i) if (!do_not_decode[i]) memset(residue_buffers[i], 0, sizeof(float) * n); if (rtype == 2 && ch != 1) { for (j=0; j < ch; ++j) if (!do_not_decode[j]) break; if (j == ch) goto done; stb_prof(3); for (pass=0; pass < 8; ++pass) { int pcount = 0, class_set = 0; if (ch == 2) { stb_prof(13); while (pcount < part_read) { int z = r->begin + pcount*r->part_size; int c_inter = (z & 1), p_inter = z>>1; if (pass == 0) { Codebook *c = f->codebooks+r->classbook; int q; DECODE(q,f,c); if (q == EOP) goto done; #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE part_classdata[0][class_set] = r->classdata[q]; #else for (i=classwords-1; i >= 0; --i) { classifications[0][i+pcount] = q % r->classifications; q /= r->classifications; } #endif } stb_prof(5); for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { int z = r->begin + pcount*r->part_size; #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE int c = part_classdata[0][class_set][i]; #else int c = classifications[0][pcount]; #endif int b = r->residue_books[c][pass]; if (b >= 0) { Codebook *book = f->codebooks + b; stb_prof(20); // accounts for X time #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) goto done; #else // saves 1% if (!codebook_decode_deinterleave_repeat_2(f, book, residue_buffers, &c_inter, &p_inter, n, r->part_size)) goto done; #endif stb_prof(7); } else { z += r->part_size; c_inter = z & 1; p_inter = z >> 1; } } stb_prof(8); #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE ++class_set; #endif } } else if (ch == 1) { while (pcount < part_read) { int z = r->begin + pcount*r->part_size; int c_inter = 0, p_inter = z; if (pass == 0) { Codebook *c = f->codebooks+r->classbook; int q; DECODE(q,f,c); if (q == EOP) goto done; #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE part_classdata[0][class_set] = r->classdata[q]; #else for (i=classwords-1; i >= 0; --i) { classifications[0][i+pcount] = q % r->classifications; q /= r->classifications; } #endif } for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { int z = r->begin + pcount*r->part_size; #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE int c = part_classdata[0][class_set][i]; #else int c = classifications[0][pcount]; #endif int b = r->residue_books[c][pass]; if (b >= 0) { Codebook *book = f->codebooks + b; stb_prof(22); if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) goto done; stb_prof(3); } else { z += r->part_size; c_inter = 0; p_inter = z; } } #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE ++class_set; #endif } } else { while (pcount < part_read) { int z = r->begin + pcount*r->part_size; int c_inter = z % ch, p_inter = z/ch; if (pass == 0) { Codebook *c = f->codebooks+r->classbook; int q; DECODE(q,f,c); if (q == EOP) goto done; #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE part_classdata[0][class_set] = r->classdata[q]; #else for (i=classwords-1; i >= 0; --i) { classifications[0][i+pcount] = q % r->classifications; q /= r->classifications; } #endif } for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { int z = r->begin + pcount*r->part_size; #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE int c = part_classdata[0][class_set][i]; #else int c = classifications[0][pcount]; #endif int b = r->residue_books[c][pass]; if (b >= 0) { Codebook *book = f->codebooks + b; stb_prof(22); if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) goto done; stb_prof(3); } else { z += r->part_size; c_inter = z % ch; p_inter = z / ch; } } #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE ++class_set; #endif } } } goto done; } stb_prof(9); for (pass=0; pass < 8; ++pass) { int pcount = 0, class_set=0; while (pcount < part_read) { if (pass == 0) { for (j=0; j < ch; ++j) { if (!do_not_decode[j]) { Codebook *c = f->codebooks+r->classbook; int temp; DECODE(temp,f,c); if (temp == EOP) goto done; #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE part_classdata[j][class_set] = r->classdata[temp]; #else for (i=classwords-1; i >= 0; --i) { classifications[j][i+pcount] = temp % r->classifications; temp /= r->classifications; } #endif } } } for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { for (j=0; j < ch; ++j) { if (!do_not_decode[j]) { #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE int c = part_classdata[j][class_set][i]; #else int c = classifications[j][pcount]; #endif int b = r->residue_books[c][pass]; if (b >= 0) { float *target = residue_buffers[j]; int offset = r->begin + pcount * r->part_size; int n = r->part_size; Codebook *book = f->codebooks + b; if (!residue_decode(f, book, target, offset, n, rtype)) goto done; } } } } #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE ++class_set; #endif } } done: stb_prof(0); temp_alloc_restore(f,temp_alloc_point); } #if 0 // slow way for debugging void inverse_mdct_slow(float *buffer, int n) { int i,j; int n2 = n >> 1; float *x = (float *) malloc(sizeof(*x) * n2); memcpy(x, buffer, sizeof(*x) * n2); for (i=0; i < n; ++i) { float acc = 0; for (j=0; j < n2; ++j) // formula from paper: //acc += n/4.0f * x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1)); // formula from wikipedia //acc += 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5)); // these are equivalent, except the formula from the paper inverts the multiplier! // however, what actually works is NO MULTIPLIER!?! //acc += 64 * 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5)); acc += x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1)); buffer[i] = acc; } free(x); } #elif 0 // same as above, but just barely able to run in real time on modern machines void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype) { float mcos[16384]; int i,j; int n2 = n >> 1, nmask = (n << 2) -1; float *x = (float *) malloc(sizeof(*x) * n2); memcpy(x, buffer, sizeof(*x) * n2); for (i=0; i < 4*n; ++i) mcos[i] = (float) cos(M_PI / 2 * i / n); for (i=0; i < n; ++i) { float acc = 0; for (j=0; j < n2; ++j) acc += x[j] * mcos[(2 * i + 1 + n2)*(2*j+1) & nmask]; buffer[i] = acc; } free(x); } #elif 0 // transform to use a slow dct-iv; this is STILL basically trivial, // but only requires half as many ops void dct_iv_slow(float *buffer, int n) { float mcos[16384]; float x[2048]; int i,j; int n2 = n >> 1, nmask = (n << 3) - 1; memcpy(x, buffer, sizeof(*x) * n); for (i=0; i < 8*n; ++i) mcos[i] = (float) cos(M_PI / 4 * i / n); for (i=0; i < n; ++i) { float acc = 0; for (j=0; j < n; ++j) acc += x[j] * mcos[((2 * i + 1)*(2*j+1)) & nmask]; buffer[i] = acc; } } void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype) { int i, n4 = n >> 2, n2 = n >> 1, n3_4 = n - n4; float temp[4096]; memcpy(temp, buffer, n2 * sizeof(float)); dct_iv_slow(temp, n2); // returns -c'-d, a-b' for (i=0; i < n4 ; ++i) buffer[i] = temp[i+n4]; // a-b' for ( ; i < n3_4; ++i) buffer[i] = -temp[n3_4 - i - 1]; // b-a', c+d' for ( ; i < n ; ++i) buffer[i] = -temp[i - n3_4]; // c'+d } #endif #ifndef LIBVORBIS_MDCT #define LIBVORBIS_MDCT 0 #endif #if LIBVORBIS_MDCT // directly call the vorbis MDCT using an interface documented // by Jeff Roberts... useful for performance comparison typedef struct { int n; int log2n; float *trig; int *bitrev; float scale; } mdct_lookup; extern void mdct_init(mdct_lookup *lookup, int n); extern void mdct_clear(mdct_lookup *l); extern void mdct_backward(mdct_lookup *init, float *in, float *out); mdct_lookup M1,M2; void inverse_mdct(float *buffer, int n, vorb *f, int blocktype) { mdct_lookup *M; if (M1.n == n) M = &M1; else if (M2.n == n) M = &M2; else if (M1.n == 0) { mdct_init(&M1, n); M = &M1; } else { if (M2.n) __asm int 3; mdct_init(&M2, n); M = &M2; } mdct_backward(M, buffer, buffer); } #endif // the following were split out into separate functions while optimizing; // they could be pushed back up but eh. __forceinline showed no change; // they're probably already being inlined. static void imdct_step3_iter0_loop(int n, float *e, int i_off, int k_off, float *A) { float *ee0 = e + i_off; float *ee2 = ee0 + k_off; int i; assert((n & 3) == 0); for (i=(n>>2); i > 0; --i) { float k00_20, k01_21; k00_20 = ee0[ 0] - ee2[ 0]; k01_21 = ee0[-1] - ee2[-1]; ee0[ 0] += ee2[ 0];//ee0[ 0] = ee0[ 0] + ee2[ 0]; ee0[-1] += ee2[-1];//ee0[-1] = ee0[-1] + ee2[-1]; ee2[ 0] = k00_20 * A[0] - k01_21 * A[1]; ee2[-1] = k01_21 * A[0] + k00_20 * A[1]; A += 8; k00_20 = ee0[-2] - ee2[-2]; k01_21 = ee0[-3] - ee2[-3]; ee0[-2] += ee2[-2];//ee0[-2] = ee0[-2] + ee2[-2]; ee0[-3] += ee2[-3];//ee0[-3] = ee0[-3] + ee2[-3]; ee2[-2] = k00_20 * A[0] - k01_21 * A[1]; ee2[-3] = k01_21 * A[0] + k00_20 * A[1]; A += 8; k00_20 = ee0[-4] - ee2[-4]; k01_21 = ee0[-5] - ee2[-5]; ee0[-4] += ee2[-4];//ee0[-4] = ee0[-4] + ee2[-4]; ee0[-5] += ee2[-5];//ee0[-5] = ee0[-5] + ee2[-5]; ee2[-4] = k00_20 * A[0] - k01_21 * A[1]; ee2[-5] = k01_21 * A[0] + k00_20 * A[1]; A += 8; k00_20 = ee0[-6] - ee2[-6]; k01_21 = ee0[-7] - ee2[-7]; ee0[-6] += ee2[-6];//ee0[-6] = ee0[-6] + ee2[-6]; ee0[-7] += ee2[-7];//ee0[-7] = ee0[-7] + ee2[-7]; ee2[-6] = k00_20 * A[0] - k01_21 * A[1]; ee2[-7] = k01_21 * A[0] + k00_20 * A[1]; A += 8; ee0 -= 8; ee2 -= 8; } } static void imdct_step3_inner_r_loop(int lim, float *e, int d0, int k_off, float *A, int k1) { int i; float k00_20, k01_21; float *e0 = e + d0; float *e2 = e0 + k_off; for (i=lim >> 2; i > 0; --i) { k00_20 = e0[-0] - e2[-0]; k01_21 = e0[-1] - e2[-1]; e0[-0] += e2[-0];//e0[-0] = e0[-0] + e2[-0]; e0[-1] += e2[-1];//e0[-1] = e0[-1] + e2[-1]; e2[-0] = (k00_20)*A[0] - (k01_21) * A[1]; e2[-1] = (k01_21)*A[0] + (k00_20) * A[1]; A += k1; k00_20 = e0[-2] - e2[-2]; k01_21 = e0[-3] - e2[-3]; e0[-2] += e2[-2];//e0[-2] = e0[-2] + e2[-2]; e0[-3] += e2[-3];//e0[-3] = e0[-3] + e2[-3]; e2[-2] = (k00_20)*A[0] - (k01_21) * A[1]; e2[-3] = (k01_21)*A[0] + (k00_20) * A[1]; A += k1; k00_20 = e0[-4] - e2[-4]; k01_21 = e0[-5] - e2[-5]; e0[-4] += e2[-4];//e0[-4] = e0[-4] + e2[-4]; e0[-5] += e2[-5];//e0[-5] = e0[-5] + e2[-5]; e2[-4] = (k00_20)*A[0] - (k01_21) * A[1]; e2[-5] = (k01_21)*A[0] + (k00_20) * A[1]; A += k1; k00_20 = e0[-6] - e2[-6]; k01_21 = e0[-7] - e2[-7]; e0[-6] += e2[-6];//e0[-6] = e0[-6] + e2[-6]; e0[-7] += e2[-7];//e0[-7] = e0[-7] + e2[-7]; e2[-6] = (k00_20)*A[0] - (k01_21) * A[1]; e2[-7] = (k01_21)*A[0] + (k00_20) * A[1]; e0 -= 8; e2 -= 8; A += k1; } } static void imdct_step3_inner_s_loop(int n, float *e, int i_off, int k_off, float *A, int a_off, int k0) { int i; float A0 = A[0]; float A1 = A[0+1]; float A2 = A[0+a_off]; float A3 = A[0+a_off+1]; float A4 = A[0+a_off*2+0]; float A5 = A[0+a_off*2+1]; float A6 = A[0+a_off*3+0]; float A7 = A[0+a_off*3+1]; float k00,k11; float *ee0 = e +i_off; float *ee2 = ee0+k_off; for (i=n; i > 0; --i) { k00 = ee0[ 0] - ee2[ 0]; k11 = ee0[-1] - ee2[-1]; ee0[ 0] = ee0[ 0] + ee2[ 0]; ee0[-1] = ee0[-1] + ee2[-1]; ee2[ 0] = (k00) * A0 - (k11) * A1; ee2[-1] = (k11) * A0 + (k00) * A1; k00 = ee0[-2] - ee2[-2]; k11 = ee0[-3] - ee2[-3]; ee0[-2] = ee0[-2] + ee2[-2]; ee0[-3] = ee0[-3] + ee2[-3]; ee2[-2] = (k00) * A2 - (k11) * A3; ee2[-3] = (k11) * A2 + (k00) * A3; k00 = ee0[-4] - ee2[-4]; k11 = ee0[-5] - ee2[-5]; ee0[-4] = ee0[-4] + ee2[-4]; ee0[-5] = ee0[-5] + ee2[-5]; ee2[-4] = (k00) * A4 - (k11) * A5; ee2[-5] = (k11) * A4 + (k00) * A5; k00 = ee0[-6] - ee2[-6]; k11 = ee0[-7] - ee2[-7]; ee0[-6] = ee0[-6] + ee2[-6]; ee0[-7] = ee0[-7] + ee2[-7]; ee2[-6] = (k00) * A6 - (k11) * A7; ee2[-7] = (k11) * A6 + (k00) * A7; ee0 -= k0; ee2 -= k0; } } static __forceinline void iter_54(float *z) { float k00,k11,k22,k33; float y0,y1,y2,y3; k00 = z[ 0] - z[-4]; y0 = z[ 0] + z[-4]; y2 = z[-2] + z[-6]; k22 = z[-2] - z[-6]; z[-0] = y0 + y2; // z0 + z4 + z2 + z6 z[-2] = y0 - y2; // z0 + z4 - z2 - z6 // done with y0,y2 k33 = z[-3] - z[-7]; z[-4] = k00 + k33; // z0 - z4 + z3 - z7 z[-6] = k00 - k33; // z0 - z4 - z3 + z7 // done with k33 k11 = z[-1] - z[-5]; y1 = z[-1] + z[-5]; y3 = z[-3] + z[-7]; z[-1] = y1 + y3; // z1 + z5 + z3 + z7 z[-3] = y1 - y3; // z1 + z5 - z3 - z7 z[-5] = k11 - k22; // z1 - z5 + z2 - z6 z[-7] = k11 + k22; // z1 - z5 - z2 + z6 } static void imdct_step3_inner_s_loop_ld654(int n, float *e, int i_off, float *A, int base_n) { int a_off = base_n >> 3; float A2 = A[0+a_off]; float *z = e + i_off; float *base = z - 16 * n; while (z > base) { float k00,k11; k00 = z[-0] - z[-8]; k11 = z[-1] - z[-9]; z[-0] = z[-0] + z[-8]; z[-1] = z[-1] + z[-9]; z[-8] = k00; z[-9] = k11 ; k00 = z[ -2] - z[-10]; k11 = z[ -3] - z[-11]; z[ -2] = z[ -2] + z[-10]; z[ -3] = z[ -3] + z[-11]; z[-10] = (k00+k11) * A2; z[-11] = (k11-k00) * A2; k00 = z[-12] - z[ -4]; // reverse to avoid a unary negation k11 = z[ -5] - z[-13]; z[ -4] = z[ -4] + z[-12]; z[ -5] = z[ -5] + z[-13]; z[-12] = k11; z[-13] = k00; k00 = z[-14] - z[ -6]; // reverse to avoid a unary negation k11 = z[ -7] - z[-15]; z[ -6] = z[ -6] + z[-14]; z[ -7] = z[ -7] + z[-15]; z[-14] = (k00+k11) * A2; z[-15] = (k00-k11) * A2; iter_54(z); iter_54(z-8); z -= 16; } } static void inverse_mdct(float *buffer, int n, vorb *f, int blocktype) { int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l; int ld; // @OPTIMIZE: reduce register pressure by using fewer variables? int save_point = temp_alloc_save(f); float *buf2 = (float *) temp_alloc(f, n2 * sizeof(*buf2)); float *u=NULL,*v=NULL; // twiddle factors float *A = f->A[blocktype]; // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio" // See notes about bugs in that paper in less-optimal implementation 'inverse_mdct_old' after this function. // kernel from paper // merged: // copy and reflect spectral data // step 0 // note that it turns out that the items added together during // this step are, in fact, being added to themselves (as reflected // by step 0). inexplicable inefficiency! this became obvious // once I combined the passes. // so there's a missing 'times 2' here (for adding X to itself). // this propogates through linearly to the end, where the numbers // are 1/2 too small, and need to be compensated for. { float *d,*e, *AA, *e_stop; d = &buf2[n2-2]; AA = A; e = &buffer[0]; e_stop = &buffer[n2]; while (e != e_stop) { d[1] = (e[0] * AA[0] - e[2]*AA[1]); d[0] = (e[0] * AA[1] + e[2]*AA[0]); d -= 2; AA += 2; e += 4; } e = &buffer[n2-3]; while (d >= buf2) { d[1] = (-e[2] * AA[0] - -e[0]*AA[1]); d[0] = (-e[2] * AA[1] + -e[0]*AA[0]); d -= 2; AA += 2; e -= 4; } } // now we use symbolic names for these, so that we can // possibly swap their meaning as we change which operations // are in place u = buffer; v = buf2; // step 2 (paper output is w, now u) // this could be in place, but the data ends up in the wrong // place... _somebody_'s got to swap it, so this is nominated { float *AA = &A[n2-8]; float *d0,*d1, *e0, *e1; e0 = &v[n4]; e1 = &v[0]; d0 = &u[n4]; d1 = &u[0]; while (AA >= A) { float v40_20, v41_21; v41_21 = e0[1] - e1[1]; v40_20 = e0[0] - e1[0]; d0[1] = e0[1] + e1[1]; d0[0] = e0[0] + e1[0]; d1[1] = v41_21*AA[4] - v40_20*AA[5]; d1[0] = v40_20*AA[4] + v41_21*AA[5]; v41_21 = e0[3] - e1[3]; v40_20 = e0[2] - e1[2]; d0[3] = e0[3] + e1[3]; d0[2] = e0[2] + e1[2]; d1[3] = v41_21*AA[0] - v40_20*AA[1]; d1[2] = v40_20*AA[0] + v41_21*AA[1]; AA -= 8; d0 += 4; d1 += 4; e0 += 4; e1 += 4; } } // step 3 ld = ilog(n) - 1; // ilog is off-by-one from normal definitions // optimized step 3: // the original step3 loop can be nested r inside s or s inside r; // it's written originally as s inside r, but this is dumb when r // iterates many times, and s few. So I have two copies of it and // switch between them halfway. // this is iteration 0 of step 3 imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*0, -(n >> 3), A); imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*1, -(n >> 3), A); // this is iteration 1 of step 3 imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*0, -(n >> 4), A, 16); imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*1, -(n >> 4), A, 16); imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*2, -(n >> 4), A, 16); imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*3, -(n >> 4), A, 16); l=2; for (; l < (ld-3)>>1; ++l) { int k0 = n >> (l+2), k0_2 = k0>>1; int lim = 1 << (l+1); int i; for (i=0; i < lim; ++i) imdct_step3_inner_r_loop(n >> (l+4), u, n2-1 - k0*i, -k0_2, A, 1 << (l+3)); } for (; l < ld-6; ++l) { int k0 = n >> (l+2), k1 = 1 << (l+3), k0_2 = k0>>1; int rlim = n >> (l+6), r; int lim = 1 << (l+1); int i_off; float *A0 = A; i_off = n2-1; for (r=rlim; r > 0; --r) { imdct_step3_inner_s_loop(lim, u, i_off, -k0_2, A0, k1, k0); A0 += k1*4; i_off -= 8; } } // iterations with count: // ld-6,-5,-4 all interleaved together // the big win comes from getting rid of needless flops // due to the constants on pass 5 & 4 being all 1 and 0; // combining them to be simultaneous to improve cache made little difference imdct_step3_inner_s_loop_ld654(n >> 5, u, n2-1, A, n); // output is u // step 4, 5, and 6 // cannot be in-place because of step 5 { uint16 *bitrev = f->bit_reverse[blocktype]; // weirdly, I'd have thought reading sequentially and writing // erratically would have been better than vice-versa, but in // fact that's not what my testing showed. (That is, with // j = bitreverse(i), do you read i and write j, or read j and write i.) float *d0 = &v[n4-4]; float *d1 = &v[n2-4]; while (d0 >= v) { int k4; k4 = bitrev[0]; d1[3] = u[k4+0]; d1[2] = u[k4+1]; d0[3] = u[k4+2]; d0[2] = u[k4+3]; k4 = bitrev[1]; d1[1] = u[k4+0]; d1[0] = u[k4+1]; d0[1] = u[k4+2]; d0[0] = u[k4+3]; d0 -= 4; d1 -= 4; bitrev += 2; } } // (paper output is u, now v) // data must be in buf2 assert(v == buf2); // step 7 (paper output is v, now v) // this is now in place { float *C = f->C[blocktype]; float *d, *e; d = v; e = v + n2 - 4; while (d < e) { float a02,a11,b0,b1,b2,b3; a02 = d[0] - e[2]; a11 = d[1] + e[3]; b0 = C[1]*a02 + C[0]*a11; b1 = C[1]*a11 - C[0]*a02; b2 = d[0] + e[ 2]; b3 = d[1] - e[ 3]; d[0] = b2 + b0; d[1] = b3 + b1; e[2] = b2 - b0; e[3] = b1 - b3; a02 = d[2] - e[0]; a11 = d[3] + e[1]; b0 = C[3]*a02 + C[2]*a11; b1 = C[3]*a11 - C[2]*a02; b2 = d[2] + e[ 0]; b3 = d[3] - e[ 1]; d[2] = b2 + b0; d[3] = b3 + b1; e[0] = b2 - b0; e[1] = b1 - b3; C += 4; d += 4; e -= 4; } } // data must be in buf2 // step 8+decode (paper output is X, now buffer) // this generates pairs of data a la 8 and pushes them directly through // the decode kernel (pushing rather than pulling) to avoid having // to make another pass later // this cannot POSSIBLY be in place, so we refer to the buffers directly { float *d0,*d1,*d2,*d3; float *B = f->B[blocktype] + n2 - 8; float *e = buf2 + n2 - 8; d0 = &buffer[0]; d1 = &buffer[n2-4]; d2 = &buffer[n2]; d3 = &buffer[n-4]; while (e >= v) { float p0,p1,p2,p3; p3 = e[6]*B[7] - e[7]*B[6]; p2 = -e[6]*B[6] - e[7]*B[7]; d0[0] = p3; d1[3] = - p3; d2[0] = p2; d3[3] = p2; p1 = e[4]*B[5] - e[5]*B[4]; p0 = -e[4]*B[4] - e[5]*B[5]; d0[1] = p1; d1[2] = - p1; d2[1] = p0; d3[2] = p0; p3 = e[2]*B[3] - e[3]*B[2]; p2 = -e[2]*B[2] - e[3]*B[3]; d0[2] = p3; d1[1] = - p3; d2[2] = p2; d3[1] = p2; p1 = e[0]*B[1] - e[1]*B[0]; p0 = -e[0]*B[0] - e[1]*B[1]; d0[3] = p1; d1[0] = - p1; d2[3] = p0; d3[0] = p0; B -= 8; e -= 8; d0 += 4; d2 += 4; d1 -= 4; d3 -= 4; } } temp_alloc_restore(f,save_point); } #if 0 // this is the original version of the above code, if you want to optimize it from scratch void inverse_mdct_naive(float *buffer, int n) { float s; float A[1 << 12], B[1 << 12], C[1 << 11]; int i,k,k2,k4, n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l; int n3_4 = n - n4, ld; // how can they claim this only uses N words?! // oh, because they're only used sparsely, whoops float u[1 << 13], X[1 << 13], v[1 << 13], w[1 << 13]; // set up twiddle factors for (k=k2=0; k < n4; ++k,k2+=2) { A[k2 ] = (float) cos(4*k*M_PI/n); A[k2+1] = (float) -sin(4*k*M_PI/n); B[k2 ] = (float) cos((k2+1)*M_PI/n/2); B[k2+1] = (float) sin((k2+1)*M_PI/n/2); } for (k=k2=0; k < n8; ++k,k2+=2) { C[k2 ] = (float) cos(2*(k2+1)*M_PI/n); C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n); } // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio" // Note there are bugs in that pseudocode, presumably due to them attempting // to rename the arrays nicely rather than representing the way their actual // implementation bounces buffers back and forth. As a result, even in the // "some formulars corrected" version, a direct implementation fails. These // are noted below as "paper bug". // copy and reflect spectral data for (k=0; k < n2; ++k) u[k] = buffer[k]; for ( ; k < n ; ++k) u[k] = -buffer[n - k - 1]; // kernel from paper // step 1 for (k=k2=k4=0; k < n4; k+=1, k2+=2, k4+=4) { v[n-k4-1] = (u[k4] - u[n-k4-1]) * A[k2] - (u[k4+2] - u[n-k4-3])*A[k2+1]; v[n-k4-3] = (u[k4] - u[n-k4-1]) * A[k2+1] + (u[k4+2] - u[n-k4-3])*A[k2]; } // step 2 for (k=k4=0; k < n8; k+=1, k4+=4) { w[n2+3+k4] = v[n2+3+k4] + v[k4+3]; w[n2+1+k4] = v[n2+1+k4] + v[k4+1]; w[k4+3] = (v[n2+3+k4] - v[k4+3])*A[n2-4-k4] - (v[n2+1+k4]-v[k4+1])*A[n2-3-k4]; w[k4+1] = (v[n2+1+k4] - v[k4+1])*A[n2-4-k4] + (v[n2+3+k4]-v[k4+3])*A[n2-3-k4]; } // step 3 ld = ilog(n) - 1; // ilog is off-by-one from normal definitions for (l=0; l < ld-3; ++l) { int k0 = n >> (l+2), k1 = 1 << (l+3); int rlim = n >> (l+4), r4, r; int s2lim = 1 << (l+2), s2; for (r=r4=0; r < rlim; r4+=4,++r) { for (s2=0; s2 < s2lim; s2+=2) { u[n-1-k0*s2-r4] = w[n-1-k0*s2-r4] + w[n-1-k0*(s2+1)-r4]; u[n-3-k0*s2-r4] = w[n-3-k0*s2-r4] + w[n-3-k0*(s2+1)-r4]; u[n-1-k0*(s2+1)-r4] = (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1] - (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1+1]; u[n-3-k0*(s2+1)-r4] = (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1] + (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1+1]; } } if (l+1 < ld-3) { // paper bug: ping-ponging of u&w here is omitted memcpy(w, u, sizeof(u)); } } // step 4 for (i=0; i < n8; ++i) { int j = bit_reverse(i) >> (32-ld+3); assert(j < n8); if (i == j) { // paper bug: original code probably swapped in place; if copying, // need to directly copy in this case int i8 = i << 3; v[i8+1] = u[i8+1]; v[i8+3] = u[i8+3]; v[i8+5] = u[i8+5]; v[i8+7] = u[i8+7]; } else if (i < j) { int i8 = i << 3, j8 = j << 3; v[j8+1] = u[i8+1], v[i8+1] = u[j8 + 1]; v[j8+3] = u[i8+3], v[i8+3] = u[j8 + 3]; v[j8+5] = u[i8+5], v[i8+5] = u[j8 + 5]; v[j8+7] = u[i8+7], v[i8+7] = u[j8 + 7]; } } // step 5 for (k=0; k < n2; ++k) { w[k] = v[k*2+1]; } // step 6 for (k=k2=k4=0; k < n8; ++k, k2 += 2, k4 += 4) { u[n-1-k2] = w[k4]; u[n-2-k2] = w[k4+1]; u[n3_4 - 1 - k2] = w[k4+2]; u[n3_4 - 2 - k2] = w[k4+3]; } // step 7 for (k=k2=0; k < n8; ++k, k2 += 2) { v[n2 + k2 ] = ( u[n2 + k2] + u[n-2-k2] + C[k2+1]*(u[n2+k2]-u[n-2-k2]) + C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2; v[n-2 - k2] = ( u[n2 + k2] + u[n-2-k2] - C[k2+1]*(u[n2+k2]-u[n-2-k2]) - C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2; v[n2+1+ k2] = ( u[n2+1+k2] - u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2; v[n-1 - k2] = (-u[n2+1+k2] + u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2; } // step 8 for (k=k2=0; k < n4; ++k,k2 += 2) { X[k] = v[k2+n2]*B[k2 ] + v[k2+1+n2]*B[k2+1]; X[n2-1-k] = v[k2+n2]*B[k2+1] - v[k2+1+n2]*B[k2 ]; } // decode kernel to output // determined the following value experimentally // (by first figuring out what made inverse_mdct_slow work); then matching that here // (probably vorbis encoder premultiplies by n or n/2, to save it on the decoder?) s = 0.5; // theoretically would be n4 // [[[ note! the s value of 0.5 is compensated for by the B[] in the current code, // so it needs to use the "old" B values to behave correctly, or else // set s to 1.0 ]]] for (i=0; i < n4 ; ++i) buffer[i] = s * X[i+n4]; for ( ; i < n3_4; ++i) buffer[i] = -s * X[n3_4 - i - 1]; for ( ; i < n ; ++i) buffer[i] = -s * X[i - n3_4]; } #endif static float *get_window(vorb *f, int len) { len <<= 1; if (len == f->blocksize_0) return f->window[0]; if (len == f->blocksize_1) return f->window[1]; assert(0); return NULL; } #ifndef STB_VORBIS_NO_DEFER_FLOOR typedef int16 YTYPE; #else typedef int YTYPE; #endif static int do_floor(vorb *f, Mapping *map, int i, int n, float *target, YTYPE *finalY, uint8 *step2_flag) { int n2 = n >> 1; int s = map->chan[i].mux, floor; floor = map->submap_floor[s]; if (f->floor_types[floor] == 0) { return error(f, VORBIS_invalid_stream); } else { Floor1 *g = &f->floor_config[floor].floor1; int j,q; int lx = 0, ly = finalY[0] * g->floor1_multiplier; for (q=1; q < g->values; ++q) { j = g->sorted_order[q]; #ifndef STB_VORBIS_NO_DEFER_FLOOR if (finalY[j] >= 0) #else if (step2_flag[j]) #endif { int hy = finalY[j] * g->floor1_multiplier; int hx = g->Xlist[j]; draw_line(target, lx,ly, hx,hy, n2); lx = hx, ly = hy; } } if (lx < n2) // optimization of: draw_line(target, lx,ly, n,ly, n2); for (j=lx; j < n2; ++j) LINE_OP(target[j], inverse_db_table[ly]); } return TRUE; } // The meaning of "left" and "right" // // For a given frame: // we compute samples from 0..n // window_center is n/2 // we'll window and mix the samples from left_start to left_end with data from the previous frame // all of the samples from left_end to right_start can be output without mixing; however, // this interval is 0-length except when transitioning between short and long frames // all of the samples from right_start to right_end need to be mixed with the next frame, // which we don't have, so those get saved in a buffer // frame N's right_end-right_start, the number of samples to mix with the next frame, // has to be the same as frame N+1's left_end-left_start (which they are by // construction) static int vorbis_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode) { Mode *m; int i, n, prev, next, window_center; f->channel_buffer_start = f->channel_buffer_end = 0; retry: if (f->eof) return FALSE; if (!maybe_start_packet(f)) return FALSE; // check packet type if (get_bits(f,1) != 0) { if (IS_PUSH_MODE(f)) return error(f,VORBIS_bad_packet_type); while (EOP != get8_packet(f)); goto retry; } if (f->alloc.alloc_buffer) assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); i = get_bits(f, ilog(f->mode_count-1)); if (i == EOP) return FALSE; if (i >= f->mode_count) return FALSE; *mode = i; m = f->mode_config + i; if (m->blockflag) { n = f->blocksize_1; prev = get_bits(f,1); next = get_bits(f,1); } else { prev = next = 0; n = f->blocksize_0; } // WINDOWING window_center = n >> 1; if (m->blockflag && !prev) { *p_left_start = (n - f->blocksize_0) >> 2; *p_left_end = (n + f->blocksize_0) >> 2; } else { *p_left_start = 0; *p_left_end = window_center; } if (m->blockflag && !next) { *p_right_start = (n*3 - f->blocksize_0) >> 2; *p_right_end = (n*3 + f->blocksize_0) >> 2; } else { *p_right_start = window_center; *p_right_end = n; } return TRUE; } static int vorbis_decode_packet_rest(vorb *f, int *len, Mode *m, int left_start, int left_end, int right_start, int right_end, int *p_left) { Mapping *map; int i,j,k,n,n2; int zero_channel[256]; int really_zero_channel[256]; // WINDOWING n = f->blocksize[m->blockflag]; map = &f->mapping[m->mapping]; // FLOORS n2 = n >> 1; stb_prof(1); for (i=0; i < f->channels; ++i) { int s = map->chan[i].mux, floor; zero_channel[i] = FALSE; floor = map->submap_floor[s]; if (f->floor_types[floor] == 0) { return error(f, VORBIS_invalid_stream); } else { Floor1 *g = &f->floor_config[floor].floor1; if (get_bits(f, 1)) { short *finalY; uint8 step2_flag[256]; static int range_list[4] = { 256, 128, 86, 64 }; int range = range_list[g->floor1_multiplier-1]; int offset = 2; finalY = f->finalY[i]; finalY[0] = get_bits(f, ilog(range)-1); finalY[1] = get_bits(f, ilog(range)-1); for (j=0; j < g->partitions; ++j) { int pclass = g->partition_class_list[j]; int cdim = g->class_dimensions[pclass]; int cbits = g->class_subclasses[pclass]; int csub = (1 << cbits)-1; int cval = 0; if (cbits) { Codebook *c = f->codebooks + g->class_masterbooks[pclass]; DECODE(cval,f,c); } for (k=0; k < cdim; ++k) { int book = g->subclass_books[pclass][cval & csub]; cval = cval >> cbits; if (book >= 0) { int temp; Codebook *c = f->codebooks + book; DECODE(temp,f,c); finalY[offset++] = temp; } else finalY[offset++] = 0; } } if (f->valid_bits == INVALID_BITS) goto error; // behavior according to spec step2_flag[0] = step2_flag[1] = 1; for (j=2; j < g->values; ++j) { int low, high, pred, highroom, lowroom, room, val; low = g->neighbors[j][0]; high = g->neighbors[j][1]; //neighbors(g->Xlist, j, &low, &high); pred = predict_point(g->Xlist[j], g->Xlist[low], g->Xlist[high], finalY[low], finalY[high]); val = finalY[j]; highroom = range - pred; lowroom = pred; if (highroom < lowroom) room = highroom * 2; else room = lowroom * 2; if (val) { step2_flag[low] = step2_flag[high] = 1; step2_flag[j] = 1; if (val >= room) if (highroom > lowroom) finalY[j] = val - lowroom + pred; else finalY[j] = pred - val + highroom - 1; else if (val & 1) finalY[j] = pred - ((val+1)>>1); else finalY[j] = pred + (val>>1); } else { step2_flag[j] = 0; finalY[j] = pred; } } #ifdef STB_VORBIS_NO_DEFER_FLOOR do_floor(f, map, i, n, f->floor_buffers[i], finalY, step2_flag); #else // defer final floor computation until _after_ residue for (j=0; j < g->values; ++j) { if (!step2_flag[j]) finalY[j] = -1; } #endif } else { error: zero_channel[i] = TRUE; } // So we just defer everything else to later // at this point we've decoded the floor into buffer } } stb_prof(0); // at this point we've decoded all floors if (f->alloc.alloc_buffer) assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); // re-enable coupled channels if necessary memcpy(really_zero_channel, zero_channel, sizeof(really_zero_channel[0]) * f->channels); for (i=0; i < map->coupling_steps; ++i) if (!zero_channel[map->chan[i].magnitude] || !zero_channel[map->chan[i].angle]) { zero_channel[map->chan[i].magnitude] = zero_channel[map->chan[i].angle] = FALSE; } // RESIDUE DECODE for (i=0; i < map->submaps; ++i) { float *residue_buffers[STB_VORBIS_MAX_CHANNELS]; int r; uint8 do_not_decode[256]; int ch = 0; for (j=0; j < f->channels; ++j) { if (map->chan[j].mux == i) { if (zero_channel[j]) { do_not_decode[ch] = TRUE; residue_buffers[ch] = NULL; } else { do_not_decode[ch] = FALSE; residue_buffers[ch] = f->channel_buffers[j]; } ++ch; } } r = map->submap_residue[i]; decode_residue(f, residue_buffers, ch, n2, r, do_not_decode); } if (f->alloc.alloc_buffer) assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); // INVERSE COUPLING stb_prof(14); for (i = map->coupling_steps-1; i >= 0; --i) { int n2 = n >> 1; float *m = f->channel_buffers[map->chan[i].magnitude]; float *a = f->channel_buffers[map->chan[i].angle ]; for (j=0; j < n2; ++j) { float a2,m2; if (m[j] > 0) if (a[j] > 0) m2 = m[j], a2 = m[j] - a[j]; else a2 = m[j], m2 = m[j] + a[j]; else if (a[j] > 0) m2 = m[j], a2 = m[j] + a[j]; else a2 = m[j], m2 = m[j] - a[j]; m[j] = m2; a[j] = a2; } } // finish decoding the floors #ifndef STB_VORBIS_NO_DEFER_FLOOR stb_prof(15); for (i=0; i < f->channels; ++i) { if (really_zero_channel[i]) { memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2); } else { do_floor(f, map, i, n, f->channel_buffers[i], f->finalY[i], NULL); } } #else for (i=0; i < f->channels; ++i) { if (really_zero_channel[i]) { memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2); } else { for (j=0; j < n2; ++j) f->channel_buffers[i][j] *= f->floor_buffers[i][j]; } } #endif // INVERSE MDCT stb_prof(16); for (i=0; i < f->channels; ++i) inverse_mdct(f->channel_buffers[i], n, f, m->blockflag); stb_prof(0); // this shouldn't be necessary, unless we exited on an error // and want to flush to get to the next packet flush_packet(f); if (f->first_decode) { // assume we start so first non-discarded sample is sample 0 // this isn't to spec, but spec would require us to read ahead // and decode the size of all current frames--could be done, // but presumably it's not a commonly used feature f->current_loc = -n2; // start of first frame is positioned for discard // we might have to discard samples "from" the next frame too, // if we're lapping a large block then a small at the start? f->discard_samples_deferred = n - right_end; f->current_loc_valid = TRUE; f->first_decode = FALSE; } else if (f->discard_samples_deferred) { left_start += f->discard_samples_deferred; *p_left = left_start; f->discard_samples_deferred = 0; } else if (f->previous_length == 0 && f->current_loc_valid) { // we're recovering from a seek... that means we're going to discard // the samples from this packet even though we know our position from // the last page header, so we need to update the position based on // the discarded samples here // but wait, the code below is going to add this in itself even // on a discard, so we don't need to do it here... } // check if we have ogg information about the sample # for this packet if (f->last_seg_which == f->end_seg_with_known_loc) { // if we have a valid current loc, and this is final: if (f->current_loc_valid && (f->page_flag & PAGEFLAG_last_page)) { uint32 current_end = f->known_loc_for_packet - (n-right_end); // then let's infer the size of the (probably) short final frame if (current_end < f->current_loc + right_end) { if (current_end < f->current_loc) { // negative truncation, that's impossible! *len = 0; } else { *len = current_end - f->current_loc; } *len += left_start; f->current_loc += *len; return TRUE; } } // otherwise, just set our sample loc // guess that the ogg granule pos refers to the _middle_ of the // last frame? // set f->current_loc to the position of left_start f->current_loc = f->known_loc_for_packet - (n2-left_start); f->current_loc_valid = TRUE; } if (f->current_loc_valid) f->current_loc += (right_start - left_start); if (f->alloc.alloc_buffer) assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); *len = right_end; // ignore samples after the window goes to 0 return TRUE; } static int vorbis_decode_packet(vorb *f, int *len, int *p_left, int *p_right) { int mode, left_end, right_end; if (!vorbis_decode_initial(f, p_left, &left_end, p_right, &right_end, &mode)) return 0; return vorbis_decode_packet_rest(f, len, f->mode_config + mode, *p_left, left_end, *p_right, right_end, p_left); } static int vorbis_finish_frame(stb_vorbis *f, int len, int left, int right) { int prev,i,j; // we use right&left (the start of the right- and left-window sin()-regions) // to determine how much to return, rather than inferring from the rules // (same result, clearer code); 'left' indicates where our sin() window // starts, therefore where the previous window's right edge starts, and // therefore where to start mixing from the previous buffer. 'right' // indicates where our sin() ending-window starts, therefore that's where // we start saving, and where our returned-data ends. // mixin from previous window if (f->previous_length) { int i,j, n = f->previous_length; float *w = get_window(f, n); for (i=0; i < f->channels; ++i) { for (j=0; j < n; ++j) f->channel_buffers[i][left+j] = f->channel_buffers[i][left+j]*w[ j] + f->previous_window[i][ j]*w[n-1-j]; } } prev = f->previous_length; // last half of this data becomes previous window f->previous_length = len - right; // @OPTIMIZE: could avoid this copy by double-buffering the // output (flipping previous_window with channel_buffers), but // then previous_window would have to be 2x as large, and // channel_buffers couldn't be temp mem (although they're NOT // currently temp mem, they could be (unless we want to level // performance by spreading out the computation)) for (i=0; i < f->channels; ++i) for (j=0; right+j < len; ++j) f->previous_window[i][j] = f->channel_buffers[i][right+j]; if (!prev) // there was no previous packet, so this data isn't valid... // this isn't entirely true, only the would-have-overlapped data // isn't valid, but this seems to be what the spec requires return 0; // truncate a short frame if (len < right) right = len; f->samples_output += right-left; return right - left; } static void vorbis_pump_first_frame(stb_vorbis *f) { int len, right, left; if (vorbis_decode_packet(f, &len, &left, &right)) vorbis_finish_frame(f, len, left, right); } #ifndef STB_VORBIS_NO_PUSHDATA_API static int is_whole_packet_present(stb_vorbis *f, int end_page) { // make sure that we have the packet available before continuing... // this requires a full ogg parse, but we know we can fetch from f->stream // instead of coding this out explicitly, we could save the current read state, // read the next packet with get8() until end-of-packet, check f->eof, then // reset the state? but that would be slower, esp. since we'd have over 256 bytes // of state to restore (primarily the page segment table) int s = f->next_seg, first = TRUE; uint8 *p = f->stream; if (s != -1) { // if we're not starting the packet with a 'continue on next page' flag for (; s < f->segment_count; ++s) { p += f->segments[s]; if (f->segments[s] < 255) // stop at first short segment break; } // either this continues, or it ends it... if (end_page) if (s < f->segment_count-1) return error(f, VORBIS_invalid_stream); if (s == f->segment_count) s = -1; // set 'crosses page' flag if (p > f->stream_end) return error(f, VORBIS_need_more_data); first = FALSE; } for (; s == -1;) { uint8 *q; int n; // check that we have the page header ready if (p + 26 >= f->stream_end) return error(f, VORBIS_need_more_data); // validate the page if (memcmp(p, ogg_page_header, 4)) return error(f, VORBIS_invalid_stream); if (p[4] != 0) return error(f, VORBIS_invalid_stream); if (first) { // the first segment must NOT have 'continued_packet', later ones MUST if (f->previous_length) if ((p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream); // if no previous length, we're resynching, so we can come in on a continued-packet, // which we'll just drop } else { if (!(p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream); } n = p[26]; // segment counts q = p+27; // q points to segment table p = q + n; // advance past header // make sure we've read the segment table if (p > f->stream_end) return error(f, VORBIS_need_more_data); for (s=0; s < n; ++s) { p += q[s]; if (q[s] < 255) break; } if (end_page) if (s < n-1) return error(f, VORBIS_invalid_stream); if (s == n) s = -1; // set 'crosses page' flag if (p > f->stream_end) return error(f, VORBIS_need_more_data); first = FALSE; } return TRUE; } #endif // !STB_VORBIS_NO_PUSHDATA_API static int start_decoder(vorb *f) { uint8 header[6], x,y; int len,i,j,k, max_submaps = 0; int longest_floorlist=0; // first page, first packet if (!start_page(f)) return FALSE; // validate page flag if (!(f->page_flag & PAGEFLAG_first_page)) return error(f, VORBIS_invalid_first_page); if (f->page_flag & PAGEFLAG_last_page) return error(f, VORBIS_invalid_first_page); if (f->page_flag & PAGEFLAG_continued_packet) return error(f, VORBIS_invalid_first_page); // check for expected packet length if (f->segment_count != 1) return error(f, VORBIS_invalid_first_page); if (f->segments[0] != 30) return error(f, VORBIS_invalid_first_page); // read packet // check packet header if (get8(f) != VORBIS_packet_id) return error(f, VORBIS_invalid_first_page); if (!getn(f, header, 6)) return error(f, VORBIS_unexpected_eof); if (!vorbis_validate(header)) return error(f, VORBIS_invalid_first_page); // vorbis_version if (get32(f) != 0) return error(f, VORBIS_invalid_first_page); f->channels = get8(f); if (!f->channels) return error(f, VORBIS_invalid_first_page); if (f->channels > STB_VORBIS_MAX_CHANNELS) return error(f, VORBIS_too_many_channels); f->sample_rate = get32(f); if (!f->sample_rate) return error(f, VORBIS_invalid_first_page); get32(f); // bitrate_maximum get32(f); // bitrate_nominal get32(f); // bitrate_minimum x = get8(f); { int log0,log1; log0 = x & 15; log1 = x >> 4; f->blocksize_0 = 1 << log0; f->blocksize_1 = 1 << log1; if (log0 < 6 || log0 > 13) return error(f, VORBIS_invalid_setup); if (log1 < 6 || log1 > 13) return error(f, VORBIS_invalid_setup); if (log0 > log1) return error(f, VORBIS_invalid_setup); } // framing_flag x = get8(f); if (!(x & 1)) return error(f, VORBIS_invalid_first_page); // second packet! if (!start_page(f)) return FALSE; if (!start_packet(f)) return FALSE; do { len = next_segment(f); skip(f, len); f->bytes_in_seg = 0; } while (len); // third packet! if (!start_packet(f)) return FALSE; #ifndef STB_VORBIS_NO_PUSHDATA_API if (IS_PUSH_MODE(f)) { if (!is_whole_packet_present(f, TRUE)) { // convert error in ogg header to write type if (f->error == VORBIS_invalid_stream) f->error = VORBIS_invalid_setup; return FALSE; } } #endif crc32_init(); // always init it, to avoid multithread race conditions if (get8_packet(f) != VORBIS_packet_setup) return error(f, VORBIS_invalid_setup); for (i=0; i < 6; ++i) header[i] = get8_packet(f); if (!vorbis_validate(header)) return error(f, VORBIS_invalid_setup); // codebooks f->codebook_count = get_bits(f,8) + 1; f->codebooks = (Codebook *) setup_malloc(f, sizeof(*f->codebooks) * f->codebook_count); if (f->codebooks == NULL) return error(f, VORBIS_outofmem); memset(f->codebooks, 0, sizeof(*f->codebooks) * f->codebook_count); for (i=0; i < f->codebook_count; ++i) { uint32 *values; int ordered, sorted_count; int total=0; uint8 *lengths; Codebook *c = f->codebooks+i; x = get_bits(f, 8); if (x != 0x42) return error(f, VORBIS_invalid_setup); x = get_bits(f, 8); if (x != 0x43) return error(f, VORBIS_invalid_setup); x = get_bits(f, 8); if (x != 0x56) return error(f, VORBIS_invalid_setup); x = get_bits(f, 8); c->dimensions = (get_bits(f, 8)<<8) + x; x = get_bits(f, 8); y = get_bits(f, 8); c->entries = (get_bits(f, 8)<<16) + (y<<8) + x; ordered = get_bits(f,1); c->sparse = ordered ? 0 : get_bits(f,1); if (c->sparse) lengths = (uint8 *) setup_temp_malloc(f, c->entries); else lengths = c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries); if (!lengths) return error(f, VORBIS_outofmem); if (ordered) { int current_entry = 0; int current_length = get_bits(f,5) + 1; while (current_entry < c->entries) { int limit = c->entries - current_entry; int n = get_bits(f, ilog(limit)); if (current_entry + n > (int) c->entries) { return error(f, VORBIS_invalid_setup); } memset(lengths + current_entry, current_length, n); current_entry += n; ++current_length; } } else { for (j=0; j < c->entries; ++j) { int present = c->sparse ? get_bits(f,1) : 1; if (present) { lengths[j] = get_bits(f, 5) + 1; ++total; } else { lengths[j] = NO_CODE; } } } if (c->sparse && total >= c->entries >> 2) { // convert sparse items to non-sparse! if (c->entries > (int) f->setup_temp_memory_required) f->setup_temp_memory_required = c->entries; c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries); memcpy(c->codeword_lengths, lengths, c->entries); setup_temp_free(f, lengths, c->entries); // note this is only safe if there have been no intervening temp mallocs! lengths = c->codeword_lengths; c->sparse = 0; } // compute the size of the sorted tables if (c->sparse) { sorted_count = total; } else { sorted_count = 0; #ifndef STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH for (j=0; j < c->entries; ++j) if (lengths[j] > STB_VORBIS_FAST_HUFFMAN_LENGTH && lengths[j] != NO_CODE) ++sorted_count; #endif } c->sorted_entries = sorted_count; values = NULL; if (!c->sparse) { c->codewords = (uint32 *) setup_malloc(f, sizeof(c->codewords[0]) * c->entries); if (!c->codewords) return error(f, VORBIS_outofmem); } else { unsigned int size; if (c->sorted_entries) { c->codeword_lengths = (uint8 *) setup_malloc(f, c->sorted_entries); if (!c->codeword_lengths) return error(f, VORBIS_outofmem); c->codewords = (uint32 *) setup_temp_malloc(f, sizeof(*c->codewords) * c->sorted_entries); if (!c->codewords) return error(f, VORBIS_outofmem); values = (uint32 *) setup_temp_malloc(f, sizeof(*values) * c->sorted_entries); if (!values) return error(f, VORBIS_outofmem); } size = c->entries + (sizeof(*c->codewords) + sizeof(*values)) * c->sorted_entries; if (size > f->setup_temp_memory_required) f->setup_temp_memory_required = size; } if (!compute_codewords(c, lengths, c->entries, values)) { if (c->sparse) setup_temp_free(f, values, 0); return error(f, VORBIS_invalid_setup); } if (c->sorted_entries) { // allocate an extra slot for sentinels c->sorted_codewords = (uint32 *) setup_malloc(f, sizeof(*c->sorted_codewords) * (c->sorted_entries+1)); // allocate an extra slot at the front so that c->sorted_values[-1] is defined // so that we can catch that case without an extra if c->sorted_values = ( int *) setup_malloc(f, sizeof(*c->sorted_values ) * (c->sorted_entries+1)); if (c->sorted_values) { ++c->sorted_values; c->sorted_values[-1] = -1; } compute_sorted_huffman(c, lengths, values); } if (c->sparse) { setup_temp_free(f, values, sizeof(*values)*c->sorted_entries); setup_temp_free(f, c->codewords, sizeof(*c->codewords)*c->sorted_entries); setup_temp_free(f, lengths, c->entries); c->codewords = NULL; } compute_accelerated_huffman(c); c->lookup_type = get_bits(f, 4); if (c->lookup_type > 2) return error(f, VORBIS_invalid_setup); if (c->lookup_type > 0) { uint16 *mults; c->minimum_value = float32_unpack(get_bits(f, 32)); c->delta_value = float32_unpack(get_bits(f, 32)); c->value_bits = get_bits(f, 4)+1; c->sequence_p = get_bits(f,1); if (c->lookup_type == 1) { c->lookup_values = lookup1_values(c->entries, c->dimensions); } else { c->lookup_values = c->entries * c->dimensions; } mults = (uint16 *) setup_temp_malloc(f, sizeof(mults[0]) * c->lookup_values); if (mults == NULL) return error(f, VORBIS_outofmem); for (j=0; j < (int) c->lookup_values; ++j) { int q = get_bits(f, c->value_bits); if (q == EOP) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_invalid_setup); } mults[j] = q; } #ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK if (c->lookup_type == 1) { int len, sparse = c->sparse; // pre-expand the lookup1-style multiplicands, to avoid a divide in the inner loop if (sparse) { if (c->sorted_entries == 0) goto skip; c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->sorted_entries * c->dimensions); } else c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->entries * c->dimensions); if (c->multiplicands == NULL) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); } len = sparse ? c->sorted_entries : c->entries; for (j=0; j < len; ++j) { int z = sparse ? c->sorted_values[j] : j, div=1; for (k=0; k < c->dimensions; ++k) { int off = (z / div) % c->lookup_values; c->multiplicands[j*c->dimensions + k] = #ifndef STB_VORBIS_CODEBOOK_FLOATS mults[off]; #else mults[off]*c->delta_value + c->minimum_value; // in this case (and this case only) we could pre-expand c->sequence_p, // and throw away the decode logic for it; have to ALSO do // it in the case below, but it can only be done if // STB_VORBIS_CODEBOOK_FLOATS // !STB_VORBIS_DIVIDES_IN_CODEBOOK #endif div *= c->lookup_values; } } setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); c->lookup_type = 2; } else #endif { c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values); #ifndef STB_VORBIS_CODEBOOK_FLOATS memcpy(c->multiplicands, mults, sizeof(c->multiplicands[0]) * c->lookup_values); #else for (j=0; j < (int) c->lookup_values; ++j) c->multiplicands[j] = mults[j] * c->delta_value + c->minimum_value; #endif setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); } #ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK skip:; #endif #ifdef STB_VORBIS_CODEBOOK_FLOATS if (c->lookup_type == 2 && c->sequence_p) { for (j=1; j < (int) c->lookup_values; ++j) c->multiplicands[j] = c->multiplicands[j-1]; c->sequence_p = 0; } #endif } } // time domain transfers (notused) x = get_bits(f, 6) + 1; for (i=0; i < x; ++i) { uint32 z = get_bits(f, 16); if (z != 0) return error(f, VORBIS_invalid_setup); } // Floors f->floor_count = get_bits(f, 6)+1; f->floor_config = (Floor *) setup_malloc(f, f->floor_count * sizeof(*f->floor_config)); for (i=0; i < f->floor_count; ++i) { f->floor_types[i] = get_bits(f, 16); if (f->floor_types[i] > 1) return error(f, VORBIS_invalid_setup); if (f->floor_types[i] == 0) { Floor0 *g = &f->floor_config[i].floor0; g->order = get_bits(f,8); g->rate = get_bits(f,16); g->bark_map_size = get_bits(f,16); g->amplitude_bits = get_bits(f,6); g->amplitude_offset = get_bits(f,8); g->number_of_books = get_bits(f,4) + 1; for (j=0; j < g->number_of_books; ++j) g->book_list[j] = get_bits(f,8); return error(f, VORBIS_feature_not_supported); } else { Point p[31*8+2]; Floor1 *g = &f->floor_config[i].floor1; int max_class = -1; g->partitions = get_bits(f, 5); for (j=0; j < g->partitions; ++j) { g->partition_class_list[j] = get_bits(f, 4); if (g->partition_class_list[j] > max_class) max_class = g->partition_class_list[j]; } for (j=0; j <= max_class; ++j) { g->class_dimensions[j] = get_bits(f, 3)+1; g->class_subclasses[j] = get_bits(f, 2); if (g->class_subclasses[j]) { g->class_masterbooks[j] = get_bits(f, 8); if (g->class_masterbooks[j] >= f->codebook_count) return error(f, VORBIS_invalid_setup); } for (k=0; k < 1 << g->class_subclasses[j]; ++k) { g->subclass_books[j][k] = get_bits(f,8)-1; if (g->subclass_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup); } } g->floor1_multiplier = get_bits(f,2)+1; g->rangebits = get_bits(f,4); g->Xlist[0] = 0; g->Xlist[1] = 1 << g->rangebits; g->values = 2; for (j=0; j < g->partitions; ++j) { int c = g->partition_class_list[j]; for (k=0; k < g->class_dimensions[c]; ++k) { g->Xlist[g->values] = get_bits(f, g->rangebits); ++g->values; } } // precompute the sorting for (j=0; j < g->values; ++j) { p[j].x = g->Xlist[j]; p[j].y = j; } qsort(p, g->values, sizeof(p[0]), point_compare); for (j=0; j < g->values; ++j) g->sorted_order[j] = (uint8) p[j].y; // precompute the neighbors for (j=2; j < g->values; ++j) { int low,hi; neighbors(g->Xlist, j, &low,&hi); g->neighbors[j][0] = low; g->neighbors[j][1] = hi; } if (g->values > longest_floorlist) longest_floorlist = g->values; } } // Residue f->residue_count = get_bits(f, 6)+1; f->residue_config = (Residue *) setup_malloc(f, f->residue_count * sizeof(*f->residue_config)); for (i=0; i < f->residue_count; ++i) { uint8 residue_cascade[64]; Residue *r = f->residue_config+i; f->residue_types[i] = get_bits(f, 16); if (f->residue_types[i] > 2) return error(f, VORBIS_invalid_setup); r->begin = get_bits(f, 24); r->end = get_bits(f, 24); r->part_size = get_bits(f,24)+1; r->classifications = get_bits(f,6)+1; r->classbook = get_bits(f,8); for (j=0; j < r->classifications; ++j) { uint8 high_bits=0; uint8 low_bits=get_bits(f,3); if (get_bits(f,1)) high_bits = get_bits(f,5); residue_cascade[j] = high_bits*8 + low_bits; } r->residue_books = (short (*)[8]) setup_malloc(f, sizeof(r->residue_books[0]) * r->classifications); for (j=0; j < r->classifications; ++j) { for (k=0; k < 8; ++k) { if (residue_cascade[j] & (1 << k)) { r->residue_books[j][k] = get_bits(f, 8); if (r->residue_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup); } else { r->residue_books[j][k] = -1; } } } // precompute the classifications[] array to avoid inner-loop mod/divide // call it 'classdata' since we already have r->classifications r->classdata = (uint8 **) setup_malloc(f, sizeof(*r->classdata) * f->codebooks[r->classbook].entries); if (!r->classdata) return error(f, VORBIS_outofmem); memset(r->classdata, 0, sizeof(*r->classdata) * f->codebooks[r->classbook].entries); for (j=0; j < f->codebooks[r->classbook].entries; ++j) { int classwords = f->codebooks[r->classbook].dimensions; int temp = j; r->classdata[j] = (uint8 *) setup_malloc(f, sizeof(r->classdata[j][0]) * classwords); for (k=classwords-1; k >= 0; --k) { r->classdata[j][k] = temp % r->classifications; temp /= r->classifications; } } } f->mapping_count = get_bits(f,6)+1; f->mapping = (Mapping *) setup_malloc(f, f->mapping_count * sizeof(*f->mapping)); for (i=0; i < f->mapping_count; ++i) { Mapping *m = f->mapping + i; int mapping_type = get_bits(f,16); if (mapping_type != 0) return error(f, VORBIS_invalid_setup); m->chan = (MappingChannel *) setup_malloc(f, f->channels * sizeof(*m->chan)); if (get_bits(f,1)) m->submaps = get_bits(f,4)+1; else m->submaps = 1; if (m->submaps > max_submaps) max_submaps = m->submaps; if (get_bits(f,1)) { m->coupling_steps = get_bits(f,8)+1; for (k=0; k < m->coupling_steps; ++k) { m->chan[k].magnitude = get_bits(f, ilog(f->channels-1)); m->chan[k].angle = get_bits(f, ilog(f->channels-1)); if (m->chan[k].magnitude >= f->channels) return error(f, VORBIS_invalid_setup); if (m->chan[k].angle >= f->channels) return error(f, VORBIS_invalid_setup); if (m->chan[k].magnitude == m->chan[k].angle) return error(f, VORBIS_invalid_setup); } } else m->coupling_steps = 0; // reserved field if (get_bits(f,2)) return error(f, VORBIS_invalid_setup); if (m->submaps > 1) { for (j=0; j < f->channels; ++j) { m->chan[j].mux = get_bits(f, 4); if (m->chan[j].mux >= m->submaps) return error(f, VORBIS_invalid_setup); } } else // @SPECIFICATION: this case is missing from the spec for (j=0; j < f->channels; ++j) m->chan[j].mux = 0; for (j=0; j < m->submaps; ++j) { get_bits(f,8); // discard m->submap_floor[j] = get_bits(f,8); m->submap_residue[j] = get_bits(f,8); if (m->submap_floor[j] >= f->floor_count) return error(f, VORBIS_invalid_setup); if (m->submap_residue[j] >= f->residue_count) return error(f, VORBIS_invalid_setup); } } // Modes f->mode_count = get_bits(f, 6)+1; for (i=0; i < f->mode_count; ++i) { Mode *m = f->mode_config+i; m->blockflag = get_bits(f,1); m->windowtype = get_bits(f,16); m->transformtype = get_bits(f,16); m->mapping = get_bits(f,8); if (m->windowtype != 0) return error(f, VORBIS_invalid_setup); if (m->transformtype != 0) return error(f, VORBIS_invalid_setup); if (m->mapping >= f->mapping_count) return error(f, VORBIS_invalid_setup); } flush_packet(f); f->previous_length = 0; for (i=0; i < f->channels; ++i) { f->channel_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1); f->previous_window[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2); f->finalY[i] = (int16 *) setup_malloc(f, sizeof(int16) * longest_floorlist); #ifdef STB_VORBIS_NO_DEFER_FLOOR f->floor_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2); #endif } if (!init_blocksize(f, 0, f->blocksize_0)) return FALSE; if (!init_blocksize(f, 1, f->blocksize_1)) return FALSE; f->blocksize[0] = f->blocksize_0; f->blocksize[1] = f->blocksize_1; #ifdef STB_VORBIS_DIVIDE_TABLE if (integer_divide_table[1][1]==0) for (i=0; i < DIVTAB_NUMER; ++i) for (j=1; j < DIVTAB_DENOM; ++j) integer_divide_table[i][j] = i / j; #endif // compute how much temporary memory is needed // 1. { uint32 imdct_mem = (f->blocksize_1 * sizeof(float) >> 1); uint32 classify_mem; int i,max_part_read=0; for (i=0; i < f->residue_count; ++i) { Residue *r = f->residue_config + i; int n_read = r->end - r->begin; int part_read = n_read / r->part_size; if (part_read > max_part_read) max_part_read = part_read; } #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(uint8 *)); #else classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(int *)); #endif f->temp_memory_required = classify_mem; if (imdct_mem > f->temp_memory_required) f->temp_memory_required = imdct_mem; } f->first_decode = TRUE; if (f->alloc.alloc_buffer) { assert(f->temp_offset == f->alloc.alloc_buffer_length_in_bytes); // check if there's enough temp memory so we don't error later if (f->setup_offset + sizeof(*f) + f->temp_memory_required > (unsigned) f->temp_offset) return error(f, VORBIS_outofmem); } f->first_audio_page_offset = stb_vorbis_get_file_offset(f); return TRUE; } static void vorbis_deinit(stb_vorbis *p) { int i,j; for (i=0; i < p->residue_count; ++i) { Residue *r = p->residue_config+i; if (r->classdata) { for (j=0; j < p->codebooks[r->classbook].entries; ++j) setup_free(p, r->classdata[j]); setup_free(p, r->classdata); } setup_free(p, r->residue_books); } if (p->codebooks) { for (i=0; i < p->codebook_count; ++i) { Codebook *c = p->codebooks + i; setup_free(p, c->codeword_lengths); setup_free(p, c->multiplicands); setup_free(p, c->codewords); setup_free(p, c->sorted_codewords); // c->sorted_values[-1] is the first entry in the array setup_free(p, c->sorted_values ? c->sorted_values-1 : NULL); } setup_free(p, p->codebooks); } setup_free(p, p->floor_config); setup_free(p, p->residue_config); for (i=0; i < p->mapping_count; ++i) setup_free(p, p->mapping[i].chan); setup_free(p, p->mapping); for (i=0; i < p->channels; ++i) { setup_free(p, p->channel_buffers[i]); setup_free(p, p->previous_window[i]); #ifdef STB_VORBIS_NO_DEFER_FLOOR setup_free(p, p->floor_buffers[i]); #endif setup_free(p, p->finalY[i]); } for (i=0; i < 2; ++i) { setup_free(p, p->A[i]); setup_free(p, p->B[i]); setup_free(p, p->C[i]); setup_free(p, p->window[i]); setup_free(p, p->bit_reverse[i]); } #ifndef STB_VORBIS_NO_STDIO if (p->close_on_free) fclose(p->f); #endif } void stb_vorbis_close(stb_vorbis *p) { if (p == NULL) return; vorbis_deinit(p); setup_free(p,p); } static void vorbis_init(stb_vorbis *p, stb_vorbis_alloc *z) { memset(p, 0, sizeof(*p)); // NULL out all malloc'd pointers to start if (z) { p->alloc = *z; p->alloc.alloc_buffer_length_in_bytes = (p->alloc.alloc_buffer_length_in_bytes+3) & ~3; p->temp_offset = p->alloc.alloc_buffer_length_in_bytes; } p->eof = 0; p->error = VORBIS__no_error; p->stream = NULL; p->codebooks = NULL; p->page_crc_tests = -1; #ifndef STB_VORBIS_NO_STDIO p->close_on_free = FALSE; p->f = NULL; #endif } int stb_vorbis_get_sample_offset(stb_vorbis *f) { if (f->current_loc_valid) return f->current_loc; else return -1; } stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f) { stb_vorbis_info d; d.channels = f->channels; d.sample_rate = f->sample_rate; d.setup_memory_required = f->setup_memory_required; d.setup_temp_memory_required = f->setup_temp_memory_required; d.temp_memory_required = f->temp_memory_required; d.max_frame_size = f->blocksize_1 >> 1; return d; } int stb_vorbis_get_error(stb_vorbis *f) { int e = f->error; f->error = VORBIS__no_error; return e; } static stb_vorbis * vorbis_alloc(stb_vorbis *f) { stb_vorbis *p = (stb_vorbis *) setup_malloc(f, sizeof(*p)); return p; } #ifndef STB_VORBIS_NO_PUSHDATA_API void stb_vorbis_flush_pushdata(stb_vorbis *f) { f->previous_length = 0; f->page_crc_tests = 0; f->discard_samples_deferred = 0; f->current_loc_valid = FALSE; f->first_decode = FALSE; f->samples_output = 0; f->channel_buffer_start = 0; f->channel_buffer_end = 0; } static int vorbis_search_for_page_pushdata(vorb *f, uint8 *data, int data_len) { int i,n; for (i=0; i < f->page_crc_tests; ++i) f->scan[i].bytes_done = 0; // if we have room for more scans, search for them first, because // they may cause us to stop early if their header is incomplete if (f->page_crc_tests < STB_VORBIS_PUSHDATA_CRC_COUNT) { if (data_len < 4) return 0; data_len -= 3; // need to look for 4-byte sequence, so don't miss // one that straddles a boundary for (i=0; i < data_len; ++i) { if (data[i] == 0x4f) { if (0==memcmp(data+i, ogg_page_header, 4)) { int j,len; uint32 crc; // make sure we have the whole page header if (i+26 >= data_len || i+27+data[i+26] >= data_len) { // only read up to this page start, so hopefully we'll // have the whole page header start next time data_len = i; break; } // ok, we have it all; compute the length of the page len = 27 + data[i+26]; for (j=0; j < data[i+26]; ++j) len += data[i+27+j]; // scan everything up to the embedded crc (which we must 0) crc = 0; for (j=0; j < 22; ++j) crc = crc32_update(crc, data[i+j]); // now process 4 0-bytes for ( ; j < 26; ++j) crc = crc32_update(crc, 0); // len is the total number of bytes we need to scan n = f->page_crc_tests++; f->scan[n].bytes_left = len-j; f->scan[n].crc_so_far = crc; f->scan[n].goal_crc = data[i+22] + (data[i+23] << 8) + (data[i+24]<<16) + (data[i+25]<<24); // if the last frame on a page is continued to the next, then // we can't recover the sample_loc immediately if (data[i+27+data[i+26]-1] == 255) f->scan[n].sample_loc = ~0; else f->scan[n].sample_loc = data[i+6] + (data[i+7] << 8) + (data[i+ 8]<<16) + (data[i+ 9]<<24); f->scan[n].bytes_done = i+j; if (f->page_crc_tests == STB_VORBIS_PUSHDATA_CRC_COUNT) break; // keep going if we still have room for more } } } } for (i=0; i < f->page_crc_tests;) { uint32 crc; int j; int n = f->scan[i].bytes_done; int m = f->scan[i].bytes_left; if (m > data_len - n) m = data_len - n; // m is the bytes to scan in the current chunk crc = f->scan[i].crc_so_far; for (j=0; j < m; ++j) crc = crc32_update(crc, data[n+j]); f->scan[i].bytes_left -= m; f->scan[i].crc_so_far = crc; if (f->scan[i].bytes_left == 0) { // does it match? if (f->scan[i].crc_so_far == f->scan[i].goal_crc) { // Houston, we have page data_len = n+m; // consumption amount is wherever that scan ended f->page_crc_tests = -1; // drop out of page scan mode f->previous_length = 0; // decode-but-don't-output one frame f->next_seg = -1; // start a new page f->current_loc = f->scan[i].sample_loc; // set the current sample location // to the amount we'd have decoded had we decoded this page f->current_loc_valid = f->current_loc != ~0U; return data_len; } // delete entry f->scan[i] = f->scan[--f->page_crc_tests]; } else { ++i; } } return data_len; } // return value: number of bytes we used int stb_vorbis_decode_frame_pushdata( stb_vorbis *f, // the file we're decoding uint8 *data, int data_len, // the memory available for decoding int *channels, // place to write number of float * buffers float ***output, // place to write float ** array of float * buffers int *samples // place to write number of output samples ) { int i; int len,right,left; if (!IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); if (f->page_crc_tests >= 0) { *samples = 0; return vorbis_search_for_page_pushdata(f, data, data_len); } f->stream = data; f->stream_end = data + data_len; f->error = VORBIS__no_error; // check that we have the entire packet in memory if (!is_whole_packet_present(f, FALSE)) { *samples = 0; return 0; } if (!vorbis_decode_packet(f, &len, &left, &right)) { // save the actual error we encountered enum STBVorbisError error = f->error; if (error == VORBIS_bad_packet_type) { // flush and resynch f->error = VORBIS__no_error; while (get8_packet(f) != EOP) if (f->eof) break; *samples = 0; return f->stream - data; } if (error == VORBIS_continued_packet_flag_invalid) { if (f->previous_length == 0) { // we may be resynching, in which case it's ok to hit one // of these; just discard the packet f->error = VORBIS__no_error; while (get8_packet(f) != EOP) if (f->eof) break; *samples = 0; return f->stream - data; } } // if we get an error while parsing, what to do? // well, it DEFINITELY won't work to continue from where we are! stb_vorbis_flush_pushdata(f); // restore the error that actually made us bail f->error = error; *samples = 0; return 1; } // success! len = vorbis_finish_frame(f, len, left, right); for (i=0; i < f->channels; ++i) f->outputs[i] = f->channel_buffers[i] + left; if (channels) *channels = f->channels; *samples = len; *output = f->outputs; return f->stream - data; } stb_vorbis *stb_vorbis_open_pushdata( unsigned char *data, int data_len, // the memory available for decoding int *data_used, // only defined if result is not NULL int *error, stb_vorbis_alloc *alloc) { stb_vorbis *f, p; vorbis_init(&p, alloc); p.stream = data; p.stream_end = data + data_len; p.push_mode = TRUE; if (!start_decoder(&p)) { if (p.eof) *error = VORBIS_need_more_data; else *error = p.error; return NULL; } f = vorbis_alloc(&p); if (f) { *f = p; *data_used = f->stream - data; *error = 0; return f; } else { vorbis_deinit(&p); return NULL; } } #endif // STB_VORBIS_NO_PUSHDATA_API unsigned int stb_vorbis_get_file_offset(stb_vorbis *f) { #ifndef STB_VORBIS_NO_PUSHDATA_API if (f->push_mode) return 0; #endif if (USE_MEMORY(f)) return f->stream - f->stream_start; #ifndef STB_VORBIS_NO_STDIO return ftell(f->f) - f->f_start; #endif } #ifndef STB_VORBIS_NO_PULLDATA_API // // DATA-PULLING API // static uint32 vorbis_find_page(stb_vorbis *f, uint32 *end, uint32 *last) { for(;;) { int n; if (f->eof) return 0; n = get8(f); if (n == 0x4f) { // page header unsigned int retry_loc = stb_vorbis_get_file_offset(f); int i; // check if we're off the end of a file_section stream if (retry_loc - 25 > f->stream_len) return 0; // check the rest of the header for (i=1; i < 4; ++i) if (get8(f) != ogg_page_header[i]) break; if (f->eof) return 0; if (i == 4) { uint8 header[27]; uint32 i, crc, goal, len; for (i=0; i < 4; ++i) header[i] = ogg_page_header[i]; for (; i < 27; ++i) header[i] = get8(f); if (f->eof) return 0; if (header[4] != 0) goto invalid; goal = header[22] + (header[23] << 8) + (header[24]<<16) + (header[25]<<24); for (i=22; i < 26; ++i) header[i] = 0; crc = 0; for (i=0; i < 27; ++i) crc = crc32_update(crc, header[i]); len = 0; for (i=0; i < header[26]; ++i) { int s = get8(f); crc = crc32_update(crc, s); len += s; } if (len && f->eof) return 0; for (i=0; i < len; ++i) crc = crc32_update(crc, get8(f)); // finished parsing probable page if (crc == goal) { // we could now check that it's either got the last // page flag set, OR it's followed by the capture // pattern, but I guess TECHNICALLY you could have // a file with garbage between each ogg page and recover // from it automatically? So even though that paranoia // might decrease the chance of an invalid decode by // another 2^32, not worth it since it would hose those // invalid-but-useful files? if (end) *end = stb_vorbis_get_file_offset(f); if (last) { if (header[5] & 0x04) *last = 1; else *last = 0; } set_file_offset(f, retry_loc-1); return 1; } } invalid: // not a valid page, so rewind and look for next one set_file_offset(f, retry_loc); } } } // seek is implemented with 'interpolation search'--this is like // binary search, but we use the data values to estimate the likely // location of the data item (plus a bit of a bias so when the // estimation is wrong we don't waste overly much time) #define SAMPLE_unknown 0xffffffff // ogg vorbis, in its insane infinite wisdom, only provides // information about the sample at the END of the page. // therefore we COULD have the data we need in the current // page, and not know it. we could just use the end location // as our only knowledge for bounds, seek back, and eventually // the binary search finds it. or we can try to be smart and // not waste time trying to locate more pages. we try to be // smart, since this data is already in memory anyway, so // doing needless I/O would be crazy! static int vorbis_analyze_page(stb_vorbis *f, ProbedPage *z) { uint8 header[27], lacing[255]; uint8 packet_type[255]; int num_packet, packet_start; int i,len; uint32 samples; // record where the page starts z->page_start = stb_vorbis_get_file_offset(f); // parse the header getn(f, header, 27); assert(header[0] == 'O' && header[1] == 'g' && header[2] == 'g' && header[3] == 'S'); getn(f, lacing, header[26]); // determine the length of the payload len = 0; for (i=0; i < header[26]; ++i) len += lacing[i]; // this implies where the page ends z->page_end = z->page_start + 27 + header[26] + len; // read the last-decoded sample out of the data z->last_decoded_sample = header[6] + (header[7] << 8) + (header[8] << 16) + (header[9] << 16); if (header[5] & 4) { // if this is the last page, it's not possible to work // backwards to figure out the first sample! whoops! fuck. z->first_decoded_sample = SAMPLE_unknown; set_file_offset(f, z->page_start); return 1; } // scan through the frames to determine the sample-count of each one... // our goal is the sample # of the first fully-decoded sample on the // page, which is the first decoded sample of the 2nd packet num_packet=0; packet_start = ((header[5] & 1) == 0); for (i=0; i < header[26]; ++i) { if (packet_start) { uint8 n,b; if (lacing[i] == 0) goto bail; // trying to read from zero-length packet n = get8(f); // if bottom bit is non-zero, we've got corruption if (n & 1) goto bail; n >>= 1; b = ilog(f->mode_count-1); n &= (1 << b)-1; if (n >= f->mode_count) goto bail; packet_type[num_packet++] = f->mode_config[n].blockflag; skip(f, lacing[i]-1); } else skip(f, lacing[i]); packet_start = (lacing[i] < 255); } // now that we know the sizes of all the pages, we can start determining // how much sample data there is. samples = 0; // for the last packet, we step by its whole length, because the definition // is that we encoded the end sample loc of the 'last packet completed', // where 'completed' refers to packets being split, and we are left to guess // what 'end sample loc' means. we assume it means ignoring the fact that // the last half of the data is useless without windowing against the next // packet... (so it's not REALLY complete in that sense) if (num_packet > 1) samples += f->blocksize[packet_type[num_packet-1]]; for (i=num_packet-2; i >= 1; --i) { // now, for this packet, how many samples do we have that // do not overlap the following packet? if (packet_type[i] == 1) if (packet_type[i+1] == 1) samples += f->blocksize_1 >> 1; else samples += ((f->blocksize_1 - f->blocksize_0) >> 2) + (f->blocksize_0 >> 1); else samples += f->blocksize_0 >> 1; } // now, at this point, we've rewound to the very beginning of the // _second_ packet. if we entirely discard the first packet after // a seek, this will be exactly the right sample number. HOWEVER! // we can't as easily compute this number for the LAST page. The // only way to get the sample offset of the LAST page is to use // the end loc from the previous page. But what that returns us // is _exactly_ the place where we get our first non-overlapped // sample. (I think. Stupid spec for being ambiguous.) So for // consistency it's better to do that here, too. However, that // will then require us to NOT discard all of the first frame we // decode, in some cases, which means an even weirder frame size // and extra code. what a fucking pain. // we're going to discard the first packet if we // start the seek here, so we don't care about it. (we could actually // do better; if the first packet is long, and the previous packet // is short, there's actually data in the first half of the first // packet that doesn't need discarding... but not worth paying the // effort of tracking that of that here and in the seeking logic) // except crap, if we infer it from the _previous_ packet's end // location, we DO need to use that definition... and we HAVE to // infer the start loc of the LAST packet from the previous packet's // end location. fuck you, ogg vorbis. z->first_decoded_sample = z->last_decoded_sample - samples; // restore file state to where we were set_file_offset(f, z->page_start); return 1; // restore file state to where we were bail: set_file_offset(f, z->page_start); return 0; } static int vorbis_seek_frame_from_page(stb_vorbis *f, uint32 page_start, uint32 first_sample, uint32 target_sample, int fine) { int left_start, left_end, right_start, right_end, mode,i; int frame=0; uint32 frame_start; int frames_to_skip, data_to_skip; // first_sample is the sample # of the first sample that doesn't // overlap the previous page... note that this requires us to // _partially_ discard the first packet! bleh. set_file_offset(f, page_start); f->next_seg = -1; // force page resync frame_start = first_sample; // frame start is where the previous packet's last decoded sample // was, which corresponds to left_end... EXCEPT if the previous // packet was long and this packet is short? Probably a bug here. // now, we can start decoding frames... we'll only FAKE decode them, // until we find the frame that contains our sample; then we'll rewind, // and try again for (;;) { int start; if (!vorbis_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode)) return error(f, VORBIS_seek_failed); if (frame == 0) start = left_end; else start = left_start; // the window starts at left_start; the last valid sample we generate // before the next frame's window start is right_start-1 if (target_sample < frame_start + right_start-start) break; flush_packet(f); if (f->eof) return error(f, VORBIS_seek_failed); frame_start += right_start - start; ++frame; } // ok, at this point, the sample we want is contained in frame #'frame' // to decode frame #'frame' normally, we have to decode the // previous frame first... but if it's the FIRST frame of the page // we can't. if it's the first frame, it means it falls in the part // of the first frame that doesn't overlap either of the other frames. // so, if we have to handle that case for the first frame, we might // as well handle it for all of them, so: if (target_sample > frame_start + (left_end - left_start)) { // so what we want to do is go ahead and just immediately decode // this frame, but then make it so the next get_frame_float() uses // this already-decoded data? or do we want to go ahead and rewind, // and leave a flag saying to skip the first N data? let's do that frames_to_skip = frame; // if this is frame #1, skip 1 frame (#0) data_to_skip = left_end - left_start; } else { // otherwise, we want to skip frames 0, 1, 2, ... frame-2 // (which means frame-2+1 total frames) then decode frame-1, // then leave frame pending frames_to_skip = frame - 1; assert(frames_to_skip >= 0); data_to_skip = -1; } set_file_offset(f, page_start); f->next_seg = - 1; // force page resync for (i=0; i < frames_to_skip; ++i) { maybe_start_packet(f); flush_packet(f); } if (data_to_skip >= 0) { int i,j,n = f->blocksize_0 >> 1; f->discard_samples_deferred = data_to_skip; for (i=0; i < f->channels; ++i) for (j=0; j < n; ++j) f->previous_window[i][j] = 0; f->previous_length = n; frame_start += data_to_skip; } else { f->previous_length = 0; vorbis_pump_first_frame(f); } // at this point, the NEXT decoded frame will generate the desired sample if (fine) { // so if we're doing sample accurate streaming, we want to go ahead and decode it! if (target_sample != frame_start) { int n; stb_vorbis_get_frame_float(f, &n, NULL); assert(target_sample > frame_start); assert(f->channel_buffer_start + (int) (target_sample-frame_start) < f->channel_buffer_end); f->channel_buffer_start += (target_sample - frame_start); } } return 0; } static int vorbis_seek_base(stb_vorbis *f, unsigned int sample_number, int fine) { ProbedPage p[2],q; if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); // do we know the location of the last page? if (f->p_last.page_start == 0) { uint32 z = stb_vorbis_stream_length_in_samples(f); if (z == 0) return error(f, VORBIS_cant_find_last_page); } p[0] = f->p_first; p[1] = f->p_last; if (sample_number >= f->p_last.last_decoded_sample) sample_number = f->p_last.last_decoded_sample-1; if (sample_number < f->p_first.last_decoded_sample) { vorbis_seek_frame_from_page(f, p[0].page_start, 0, sample_number, fine); return 0; } else { int attempts=0; while (p[0].page_end < p[1].page_start) { uint32 probe; uint32 start_offset, end_offset; uint32 start_sample, end_sample; // copy these into local variables so we can tweak them // if any are unknown start_offset = p[0].page_end; end_offset = p[1].after_previous_page_start; // an address known to seek to page p[1] start_sample = p[0].last_decoded_sample; end_sample = p[1].last_decoded_sample; // currently there is no such tweaking logic needed/possible? if (start_sample == SAMPLE_unknown || end_sample == SAMPLE_unknown) return error(f, VORBIS_seek_failed); // now we want to lerp between these for the target samples... // step 1: we need to bias towards the page start... if (start_offset + 4000 < end_offset) end_offset -= 4000; // now compute an interpolated search loc probe = start_offset + (int) floor((float) (end_offset - start_offset) / (end_sample - start_sample) * (sample_number - start_sample)); // next we need to bias towards binary search... // code is a little wonky to allow for full 32-bit unsigned values if (attempts >= 4) { uint32 probe2 = start_offset + ((end_offset - start_offset) >> 1); if (attempts >= 8) probe = probe2; else if (probe < probe2) probe = probe + ((probe2 - probe) >> 1); else probe = probe2 + ((probe - probe2) >> 1); } ++attempts; set_file_offset(f, probe); if (!vorbis_find_page(f, NULL, NULL)) return error(f, VORBIS_seek_failed); if (!vorbis_analyze_page(f, &q)) return error(f, VORBIS_seek_failed); q.after_previous_page_start = probe; // it's possible we've just found the last page again if (q.page_start == p[1].page_start) { p[1] = q; continue; } if (sample_number < q.last_decoded_sample) p[1] = q; else p[0] = q; } if (p[0].last_decoded_sample <= sample_number && sample_number < p[1].last_decoded_sample) { vorbis_seek_frame_from_page(f, p[1].page_start, p[0].last_decoded_sample, sample_number, fine); return 0; } return error(f, VORBIS_seek_failed); } } int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number) { return vorbis_seek_base(f, sample_number, FALSE); } int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number) { return vorbis_seek_base(f, sample_number, TRUE); } void stb_vorbis_seek_start(stb_vorbis *f) { if (IS_PUSH_MODE(f)) { error(f, VORBIS_invalid_api_mixing); return; } set_file_offset(f, f->first_audio_page_offset); f->previous_length = 0; f->first_decode = TRUE; f->next_seg = -1; vorbis_pump_first_frame(f); } unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f) { unsigned int restore_offset, previous_safe; unsigned int end, last_page_loc; if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); if (!f->total_samples) { unsigned int last; uint32 lo,hi; char header[6]; // first, store the current decode position so we can restore it restore_offset = stb_vorbis_get_file_offset(f); // now we want to seek back 64K from the end (the last page must // be at most a little less than 64K, but let's allow a little slop) if (f->stream_len >= 65536 && f->stream_len-65536 >= f->first_audio_page_offset) previous_safe = f->stream_len - 65536; else previous_safe = f->first_audio_page_offset; set_file_offset(f, previous_safe); // previous_safe is now our candidate 'earliest known place that seeking // to will lead to the final page' if (!vorbis_find_page(f, &end, &last)) { // if we can't find a page, we're hosed! f->error = VORBIS_cant_find_last_page; f->total_samples = 0xffffffff; goto done; } // check if there are more pages last_page_loc = stb_vorbis_get_file_offset(f); // stop when the last_page flag is set, not when we reach eof; // this allows us to stop short of a 'file_section' end without // explicitly checking the length of the section while (!last) { set_file_offset(f, end); if (!vorbis_find_page(f, &end, &last)) { // the last page we found didn't have the 'last page' flag // set. whoops! break; } previous_safe = last_page_loc+1; last_page_loc = stb_vorbis_get_file_offset(f); } set_file_offset(f, last_page_loc); // parse the header getn(f, (unsigned char *)header, 6); // extract the absolute granule position lo = get32(f); hi = get32(f); if (lo == 0xffffffff && hi == 0xffffffff) { f->error = VORBIS_cant_find_last_page; f->total_samples = SAMPLE_unknown; goto done; } if (hi) lo = 0xfffffffe; // saturate f->total_samples = lo; f->p_last.page_start = last_page_loc; f->p_last.page_end = end; f->p_last.last_decoded_sample = lo; f->p_last.first_decoded_sample = SAMPLE_unknown; f->p_last.after_previous_page_start = previous_safe; done: set_file_offset(f, restore_offset); } return f->total_samples == SAMPLE_unknown ? 0 : f->total_samples; } float stb_vorbis_stream_length_in_seconds(stb_vorbis *f) { return stb_vorbis_stream_length_in_samples(f) / (float) f->sample_rate; } int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output) { int len, right,left,i; if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); if (!vorbis_decode_packet(f, &len, &left, &right)) { f->channel_buffer_start = f->channel_buffer_end = 0; return 0; } len = vorbis_finish_frame(f, len, left, right); for (i=0; i < f->channels; ++i) f->outputs[i] = f->channel_buffers[i] + left; f->channel_buffer_start = left; f->channel_buffer_end = left+len; if (channels) *channels = f->channels; if (output) *output = f->outputs; return len; } #ifndef STB_VORBIS_NO_STDIO stb_vorbis * stb_vorbis_open_file_section(FILE *file, int close_on_free, int *error, stb_vorbis_alloc *alloc, unsigned int length) { stb_vorbis *f, p; vorbis_init(&p, alloc); p.f = file; p.f_start = ftell(file); p.stream_len = length; p.close_on_free = close_on_free; if (start_decoder(&p)) { f = vorbis_alloc(&p); if (f) { *f = p; vorbis_pump_first_frame(f); return f; } } if (error) *error = p.error; vorbis_deinit(&p); return NULL; } stb_vorbis * stb_vorbis_open_file(FILE *file, int close_on_free, int *error, stb_vorbis_alloc *alloc) { unsigned int len, start; start = ftell(file); fseek(file, 0, SEEK_END); len = ftell(file) - start; fseek(file, start, SEEK_SET); return stb_vorbis_open_file_section(file, close_on_free, error, alloc, len); } stb_vorbis * stb_vorbis_open_filename(const char *filename, int *error, stb_vorbis_alloc *alloc) { FILE *f = fopen(filename, "rb"); if (f) return stb_vorbis_open_file(f, TRUE, error, alloc); if (error) *error = VORBIS_file_open_failure; return NULL; } #endif // STB_VORBIS_NO_STDIO stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len, int *error, stb_vorbis_alloc *alloc) { stb_vorbis *f, p; if (data == NULL) return NULL; vorbis_init(&p, alloc); p.stream = (uint8 *) data; p.stream_end = (uint8 *) data + len; p.stream_start = (uint8 *) p.stream; p.stream_len = len; p.push_mode = FALSE; if (start_decoder(&p)) { f = vorbis_alloc(&p); if (f) { *f = p; vorbis_pump_first_frame(f); return f; } } if (error) *error = p.error; vorbis_deinit(&p); return NULL; } #ifndef STB_VORBIS_NO_INTEGER_CONVERSION #define PLAYBACK_MONO 1 #define PLAYBACK_LEFT 2 #define PLAYBACK_RIGHT 4 #define L (PLAYBACK_LEFT | PLAYBACK_MONO) #define C (PLAYBACK_LEFT | PLAYBACK_RIGHT | PLAYBACK_MONO) #define R (PLAYBACK_RIGHT | PLAYBACK_MONO) static int8 channel_position[7][6] = { { 0 }, { C }, { L, R }, { L, C, R }, { L, R, L, R }, { L, C, R, L, R }, { L, C, R, L, R, C }, }; #ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT typedef union { float f; int i; } float_conv; typedef char stb_vorbis_float_size_test[sizeof(float)==4 && sizeof(int) == 4]; #define FASTDEF(x) float_conv x // add (1<<23) to convert to int, then divide by 2^SHIFT, then add 0.5/2^SHIFT to round #define MAGIC(SHIFT) (1.5f * (1 << (23-SHIFT)) + 0.5f/(1 << SHIFT)) #define ADDEND(SHIFT) (((150-SHIFT) << 23) + (1 << 22)) #define FAST_SCALED_FLOAT_TO_INT(temp,x,s) (temp.f = (x) + MAGIC(s), temp.i - ADDEND(s)) #define check_endianness() #else #define FAST_SCALED_FLOAT_TO_INT(temp,x,s) ((int) ((x) * (1 << (s)))) #define check_endianness() #define FASTDEF(x) #endif static void copy_samples(short *dest, float *src, int len) { int i; check_endianness(); for (i=0; i < len; ++i) { FASTDEF(temp); int v = FAST_SCALED_FLOAT_TO_INT(temp, src[i],15); if ((unsigned int) (v + 32768) > 65535) v = v < 0 ? -32768 : 32767; dest[i] = v; } } static void compute_samples(int mask, short *output, int num_c, float **data, int d_offset, int len) { #define BUFFER_SIZE 32 float buffer[BUFFER_SIZE]; int i,j,o,n = BUFFER_SIZE; check_endianness(); for (o = 0; o < len; o += BUFFER_SIZE) { memset(buffer, 0, sizeof(buffer)); if (o + n > len) n = len - o; for (j=0; j < num_c; ++j) { if (channel_position[num_c][j] & mask) { for (i=0; i < n; ++i) buffer[i] += data[j][d_offset+o+i]; } } for (i=0; i < n; ++i) { FASTDEF(temp); int v = FAST_SCALED_FLOAT_TO_INT(temp,buffer[i],15); if ((unsigned int) (v + 32768) > 65535) v = v < 0 ? -32768 : 32767; output[o+i] = v; } } } static void compute_stereo_samples(short *output, int num_c, float **data, int d_offset, int len) { #define BUFFER_SIZE 32 float buffer[BUFFER_SIZE]; int i,j,o,n = BUFFER_SIZE >> 1; // o is the offset in the source data check_endianness(); for (o = 0; o < len; o += BUFFER_SIZE >> 1) { // o2 is the offset in the output data int o2 = o << 1; memset(buffer, 0, sizeof(buffer)); if (o + n > len) n = len - o; for (j=0; j < num_c; ++j) { int m = channel_position[num_c][j] & (PLAYBACK_LEFT | PLAYBACK_RIGHT); if (m == (PLAYBACK_LEFT | PLAYBACK_RIGHT)) { for (i=0; i < n; ++i) { buffer[i*2+0] += data[j][d_offset+o+i]; buffer[i*2+1] += data[j][d_offset+o+i]; } } else if (m == PLAYBACK_LEFT) { for (i=0; i < n; ++i) { buffer[i*2+0] += data[j][d_offset+o+i]; } } else if (m == PLAYBACK_RIGHT) { for (i=0; i < n; ++i) { buffer[i*2+1] += data[j][d_offset+o+i]; } } } for (i=0; i < (n<<1); ++i) { FASTDEF(temp); int v = FAST_SCALED_FLOAT_TO_INT(temp,buffer[i],15); if ((unsigned int) (v + 32768) > 65535) v = v < 0 ? -32768 : 32767; output[o2+i] = v; } } } static void convert_samples_short(int buf_c, short **buffer, int b_offset, int data_c, float **data, int d_offset, int samples) { int i; if (buf_c != data_c && buf_c <= 2 && data_c <= 6) { static int channel_selector[3][2] = { {0}, {PLAYBACK_MONO}, {PLAYBACK_LEFT, PLAYBACK_RIGHT} }; for (i=0; i < buf_c; ++i) compute_samples(channel_selector[buf_c][i], buffer[i]+b_offset, data_c, data, d_offset, samples); } else { int limit = buf_c < data_c ? buf_c : data_c; for (i=0; i < limit; ++i) copy_samples(buffer[i]+b_offset, data[i]+d_offset, samples); for ( ; i < buf_c; ++i) memset(buffer[i]+b_offset, 0, sizeof(short) * samples); } } int stb_vorbis_get_frame_short(stb_vorbis *f, int num_c, short **buffer, int num_samples) { float **output; int len = stb_vorbis_get_frame_float(f, NULL, &output); if (len > num_samples) len = num_samples; if (len) convert_samples_short(num_c, buffer, 0, f->channels, output, 0, len); return len; } static void convert_channels_short_interleaved(int buf_c, short *buffer, int data_c, float **data, int d_offset, int len) { int i; check_endianness(); if (buf_c != data_c && buf_c <= 2 && data_c <= 6) { assert(buf_c == 2); for (i=0; i < buf_c; ++i) compute_stereo_samples(buffer, data_c, data, d_offset, len); } else { int limit = buf_c < data_c ? buf_c : data_c; int j; for (j=0; j < len; ++j) { for (i=0; i < limit; ++i) { FASTDEF(temp); float f = data[i][d_offset+j]; int v = FAST_SCALED_FLOAT_TO_INT(temp, f,15);//data[i][d_offset+j],15); if ((unsigned int) (v + 32768) > 65535) v = v < 0 ? -32768 : 32767; *buffer++ = v; } for ( ; i < buf_c; ++i) *buffer++ = 0; } } } int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts) { float **output; int len; if (num_c == 1) return stb_vorbis_get_frame_short(f,num_c,&buffer, num_shorts); len = stb_vorbis_get_frame_float(f, NULL, &output); if (len) { if (len*num_c > num_shorts) len = num_shorts / num_c; convert_channels_short_interleaved(num_c, buffer, f->channels, output, 0, len); } return len; } int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts) { float **outputs; int len = num_shorts / channels; int n=0; int z = f->channels; if (z > channels) z = channels; while (n < len) { int k = f->channel_buffer_end - f->channel_buffer_start; if (n+k >= len) k = len - n; if (k) convert_channels_short_interleaved(channels, buffer, f->channels, f->channel_buffers, f->channel_buffer_start, k); buffer += k*channels; n += k; f->channel_buffer_start += k; if (n == len) break; if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break; } return n; } int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int len) { float **outputs; int n=0; int z = f->channels; if (z > channels) z = channels; while (n < len) { int k = f->channel_buffer_end - f->channel_buffer_start; if (n+k >= len) k = len - n; if (k) convert_samples_short(channels, buffer, n, f->channels, f->channel_buffers, f->channel_buffer_start, k); n += k; f->channel_buffer_start += k; if (n == len) break; if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break; } return n; } #ifndef STB_VORBIS_NO_STDIO int stb_vorbis_decode_filename(const char *filename, int *channels, int *sample_rate, short **output) { int data_len, offset, total, limit, error; short *data; stb_vorbis *v = stb_vorbis_open_filename(filename, &error, NULL); if (v == NULL) return -1; limit = v->channels * 4096; *channels = v->channels; if (sample_rate) *sample_rate = v->sample_rate; offset = data_len = 0; total = limit; data = (short *) malloc(total * sizeof(*data)); if (data == NULL) { stb_vorbis_close(v); return -2; } for (;;) { int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset); if (n == 0) break; data_len += n; offset += n * v->channels; if (offset + limit > total) { short *data2; total *= 2; data2 = (short *) realloc(data, total * sizeof(*data)); if (data2 == NULL) { free(data); stb_vorbis_close(v); return -2; } data = data2; } } *output = data; stb_vorbis_close(v); return data_len; } #endif // NO_STDIO int stb_vorbis_decode_memory(const uint8 *mem, int len, int *channels, int *sample_rate, short **output) { int data_len, offset, total, limit, error; short *data; stb_vorbis *v = stb_vorbis_open_memory(mem, len, &error, NULL); if (v == NULL) return -1; limit = v->channels * 4096; *channels = v->channels; if (sample_rate) *sample_rate = v->sample_rate; offset = data_len = 0; total = limit; data = (short *) malloc(total * sizeof(*data)); if (data == NULL) { stb_vorbis_close(v); return -2; } for (;;) { int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset); if (n == 0) break; data_len += n; offset += n * v->channels; if (offset + limit > total) { short *data2; total *= 2; data2 = (short *) realloc(data, total * sizeof(*data)); if (data2 == NULL) { free(data); stb_vorbis_close(v); return -2; } data = data2; } } *output = data; stb_vorbis_close(v); return data_len; } #endif // STB_VORBIS_NO_INTEGER_CONVERSION int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats) { float **outputs; int len = num_floats / channels; int n=0; int z = f->channels; if (z > channels) z = channels; while (n < len) { int i,j; int k = f->channel_buffer_end - f->channel_buffer_start; if (n+k >= len) k = len - n; for (j=0; j < k; ++j) { for (i=0; i < z; ++i) *buffer++ = f->channel_buffers[i][f->channel_buffer_start+j]; for ( ; i < channels; ++i) *buffer++ = 0; } n += k; f->channel_buffer_start += k; if (n == len) break; if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break; } return n; } int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples) { float **outputs; int n=0; int z = f->channels; if (z > channels) z = channels; while (n < num_samples) { int i; int k = f->channel_buffer_end - f->channel_buffer_start; if (n+k >= num_samples) k = num_samples - n; if (k) { for (i=0; i < z; ++i) memcpy(buffer[i]+n, f->channel_buffers[i]+f->channel_buffer_start, sizeof(float)*k); for ( ; i < channels; ++i) memset(buffer[i]+n, 0, sizeof(float) * k); } n += k; f->channel_buffer_start += k; if (n == num_samples) break; if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break; } return n; } #endif // STB_VORBIS_NO_PULLDATA_API /* Version history 1.05 - 2015/04/19 - don't define __forceinline if it's redundant 1.04 - 2014/08/27 - fix missing const-correct case in API 1.03 - 2014/08/07 - Warning fixes 1.02 - 2014/07/09 - Declare qsort compare function _cdecl on windows 1.01 - 2014/06/18 - fix stb_vorbis_get_samples_float 1.0 - 2014/05/26 - fix memory leaks; fix warnings; fix bugs in multichannel (API change) report sample rate for decode-full-file funcs 0.99996 - bracket #include for macintosh compilation by Laurent Gomila 0.99995 - use union instead of pointer-cast for fast-float-to-int to avoid alias-optimization problem 0.99994 - change fast-float-to-int to work in single-precision FPU mode, remove endian-dependence 0.99993 - remove assert that fired on legal files with empty tables 0.99992 - rewind-to-start 0.99991 - bugfix to stb_vorbis_get_samples_short by Bernhard Wodo 0.9999 - (should have been 0.99990) fix no-CRT support, compiling as C++ 0.9998 - add a full-decode function with a memory source 0.9997 - fix a bug in the read-from-FILE case in 0.9996 addition 0.9996 - query length of vorbis stream in samples/seconds 0.9995 - bugfix to another optimization that only happened in certain files 0.9994 - bugfix to one of the optimizations that caused significant (but inaudible?) errors 0.9993 - performance improvements; runs in 99% to 104% of time of reference implementation 0.9992 - performance improvement of IMDCT; now performs close to reference implementation 0.9991 - performance improvement of IMDCT 0.999 - (should have been 0.9990) performance improvement of IMDCT 0.998 - no-CRT support from Casey Muratori 0.997 - bugfixes for bugs found by Terje Mathisen 0.996 - bugfix: fast-huffman decode initialized incorrectly for sparse codebooks; fixing gives 10% speedup - found by Terje Mathisen 0.995 - bugfix: fix to 'effective' overrun detection - found by Terje Mathisen 0.994 - bugfix: garbage decode on final VQ symbol of a non-multiple - found by Terje Mathisen 0.993 - bugfix: pushdata API required 1 extra byte for empty page (failed to consume final page if empty) - found by Terje Mathisen 0.992 - fixes for MinGW warning 0.991 - turn fast-float-conversion on by default 0.990 - fix push-mode seek recovery if you seek into the headers 0.98b - fix to bad release of 0.98 0.98 - fix push-mode seek recovery; robustify float-to-int and support non-fast mode 0.97 - builds under c++ (typecasting, don't use 'class' keyword) 0.96 - somehow MY 0.95 was right, but the web one was wrong, so here's my 0.95 rereleased as 0.96, fixes a typo in the clamping code 0.95 - clamping code for 16-bit functions 0.94 - not publically released 0.93 - fixed all-zero-floor case (was decoding garbage) 0.92 - fixed a memory leak 0.91 - conditional compiles to omit parts of the API and the infrastructure to support them: STB_VORBIS_NO_PULLDATA_API, STB_VORBIS_NO_PUSHDATA_API, STB_VORBIS_NO_STDIO, STB_VORBIS_NO_INTEGER_CONVERSION 0.90 - first public release */ #endif // STB_VORBIS_HEADER_ONLY