aboutsummaryrefslogtreecommitdiff
path: root/src
diff options
context:
space:
mode:
Diffstat (limited to 'src')
-rw-r--r--src/libs/stb/stb_vorbis.c8454
-rw-r--r--src/libs/tiny/tinysound.h2560
-rw-r--r--src/lua/audio/luaopen_audio.cpp12
-rw-r--r--src/lua/embed/debug.lua.h253
-rw-r--r--src/lua/embed/embed.h14
-rw-r--r--src/lua/embed/graphics.lua.h12
-rw-r--r--src/lua/embed/keyboard.lua.h31
-rw-r--r--src/lua/embed/mouse.lua.h29
-rw-r--r--src/lua/embed/path.lua.h36
-rw-r--r--src/lua/event/luaopen_event.cpp2
-rw-r--r--src/lua/filesystem/luaopen_filesystem.cpp4
-rw-r--r--src/lua/graphics/luaopen_graphics.cpp34
-rw-r--r--src/lua/luaopen_jin.cpp6
-rw-r--r--src/main.cpp7
-rw-r--r--src/render/font.cpp4
-rw-r--r--src/render/graphics.cpp2
-rw-r--r--src/render/jsl.cpp3
-rw-r--r--src/utils/matrix.cpp2
18 files changed, 4505 insertions, 6960 deletions
diff --git a/src/libs/stb/stb_vorbis.c b/src/libs/stb/stb_vorbis.c
index 1181e6d..3d338f0 100644
--- a/src/libs/stb/stb_vorbis.c
+++ b/src/libs/stb/stb_vorbis.c
@@ -1,11 +1,11 @@
-// Ogg Vorbis audio decoder - v1.10 - public domain
+// Ogg Vorbis audio decoder - v1.14 - public domain
// http://nothings.org/stb_vorbis/
//
// Original version written by Sean Barrett in 2007.
//
-// Originally sponsored by RAD Game Tools. Seeking sponsored
-// by Phillip Bennefall, Marc Andersen, Aaron Baker, Elias Software,
-// Aras Pranckevicius, and Sean Barrett.
+// Originally sponsored by RAD Game Tools. Seeking implementation
+// sponsored by Phillip Bennefall, Marc Andersen, Aaron Baker,
+// Elias Software, Aras Pranckevicius, and Sean Barrett.
//
// LICENSE
//
@@ -29,22 +29,27 @@
// Bernhard Wodo Evan Balster alxprd@github
// Tom Beaumont Ingo Leitgeb Nicolas Guillemot
// Phillip Bennefall Rohit Thiago Goulart
-// manxorist@github saga musix
+// manxorist@github saga musix github:infatum
+// Timur Gagiev
//
// Partial history:
-// 1.10 - 2017/03/03 - more robust seeking; fix negative ilog(); clear error in open_memory
-// 1.09 - 2016/04/04 - back out 'truncation of last frame' fix from previous version
-// 1.08 - 2016/04/02 - warnings; setup memory leaks; truncation of last frame
-// 1.07 - 2015/01/16 - fixes for crashes on invalid files; warning fixes; const
-// 1.06 - 2015/08/31 - full, correct support for seeking API (Dougall Johnson)
+// 1.14 - 2018-02-11 - delete bogus dealloca usage
+// 1.13 - 2018-01-29 - fix truncation of last frame (hopefully)
+// 1.12 - 2017-11-21 - limit residue begin/end to blocksize/2 to avoid large temp allocs in bad/corrupt files
+// 1.11 - 2017-07-23 - fix MinGW compilation
+// 1.10 - 2017-03-03 - more robust seeking; fix negative ilog(); clear error in open_memory
+// 1.09 - 2016-04-04 - back out 'truncation of last frame' fix from previous version
+// 1.08 - 2016-04-02 - warnings; setup memory leaks; truncation of last frame
+// 1.07 - 2015-01-16 - fixes for crashes on invalid files; warning fixes; const
+// 1.06 - 2015-08-31 - full, correct support for seeking API (Dougall Johnson)
// some crash fixes when out of memory or with corrupt files
// fix some inappropriately signed shifts
-// 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;
+// 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
//
// See end of file for full version history.
@@ -70,307 +75,307 @@
extern "C" {
#endif
-/////////// THREAD SAFETY
+ /////////// 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.
+ // 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
+ /////////// 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()
+ // 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;
+ typedef struct
+ {
+ char *alloc_buffer;
+ int alloc_buffer_length_in_bytes;
+ } stb_vorbis_alloc;
-/////////// FUNCTIONS USEABLE WITH ALL INPUT MODES
+ /////////// FUNCTIONS USEABLE WITH ALL INPUT MODES
-typedef struct stb_vorbis stb_vorbis;
+ typedef struct stb_vorbis stb_vorbis;
-typedef struct
-{
- unsigned int sample_rate;
- int channels;
+ typedef struct
+ {
+ unsigned int sample_rate;
+ int channels;
- unsigned int setup_memory_required;
- unsigned int setup_temp_memory_required;
- unsigned int temp_memory_required;
+ unsigned int setup_memory_required;
+ unsigned int setup_temp_memory_required;
+ unsigned int temp_memory_required;
- int max_frame_size;
-} stb_vorbis_info;
+ 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 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);
+ // 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);
+ // 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);
+ // 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);
+ // 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
+ /////////// 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(
- const unsigned char * datablock, int datablock_length_in_bytes,
- int *datablock_memory_consumed_in_bytes,
- int *error,
- const 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,
- const 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.
+ // 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(
+ const unsigned char * datablock, int datablock_length_in_bytes,
+ int *datablock_memory_consumed_in_bytes,
+ int *error,
+ const 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,
+ const 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
+ ////////// 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.)
+ // 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);
+ 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);
+ 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.
+ // 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, const 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
+ extern stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len,
+ int *error, const 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, const 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, const 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, const 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.
+ extern stb_vorbis * stb_vorbis_open_filename(const char *filename,
+ int *error, const 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, const 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, const 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);
-// 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 int stb_vorbis_seek_start(stb_vorbis *f);
-// this function is equivalent to stb_vorbis_seek(f,0)
-
-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.
+ 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);
+ // 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 int stb_vorbis_seek_start(stb_vorbis *f);
+ // this function is equivalent to stb_vorbis_seek(f,0)
+
+ 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);
+ 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.
-// 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.
-//
-// 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.
-
-// 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.
+ // decode the next frame and return the number of *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.
+ //
+ // 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.
+
+ // 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);
+ 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.
+ // 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
+ //////// ERROR CODES
-enum STBVorbisError
-{
- VORBIS__no_error,
+ enum STBVorbisError
+ {
+ VORBIS__no_error,
- VORBIS_need_more_data=1, // not a real 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_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
+ 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
+ // 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,
+ // 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
-};
+ // 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
@@ -516,25 +521,25 @@ enum STBVorbisError
//////////////////////////////////////////////////////////////////////////////
#ifdef STB_VORBIS_NO_PULLDATA_API
- #define STB_VORBIS_NO_INTEGER_CONVERSION
- #define STB_VORBIS_NO_STDIO
+#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
+#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
+// 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
+#ifndef STB_VORBIS_BIG_ENDIAN
+#define STB_VORBIS_ENDIAN 0
+#else
+#define STB_VORBIS_ENDIAN 1
+#endif
#endif
#endif
@@ -545,43 +550,44 @@ enum STBVorbisError
#endif
#ifndef STB_VORBIS_NO_CRT
- #include <stdlib.h>
- #include <string.h>
- #include <assert.h>
- #include <math.h>
-
- // find definition of alloca if it's not in stdlib.h:
- #ifdef _MSC_VER
- #include <malloc.h>
- #endif
- #if defined(__linux__) || defined(__linux) || defined(__EMSCRIPTEN__)
- #include <alloca.h>
- #endif
+#include <stdlib.h>
+#include <string.h>
+#include <assert.h>
+#include <math.h>
+
+// find definition of alloca if it's not in stdlib.h:
+#if defined(_MSC_VER) || defined(__MINGW32__)
+#include <malloc.h>
+#endif
+#if defined(__linux__) || defined(__linux) || defined(__EMSCRIPTEN__)
+#include <alloca.h>
+#endif
#else // STB_VORBIS_NO_CRT
- #define NULL 0
- #define malloc(s) 0
- #define free(s) ((void) 0)
- #define realloc(s) 0
+#define NULL 0
+#define malloc(s) 0
+#define free(s) ((void) 0)
+#define realloc(s) 0
#endif // STB_VORBIS_NO_CRT
#include <limits.h>
#ifdef __MINGW32__
- // eff you mingw:
- // "fixed":
- // http://sourceforge.net/p/mingw-w64/mailman/message/32882927/
- // "no that broke the build, reverted, who cares about C":
- // http://sourceforge.net/p/mingw-w64/mailman/message/32890381/
- #ifdef __forceinline
- #undef __forceinline
- #endif
- #define __forceinline
+// eff you mingw:
+// "fixed":
+// http://sourceforge.net/p/mingw-w64/mailman/message/32882927/
+// "no that broke the build, reverted, who cares about C":
+// http://sourceforge.net/p/mingw-w64/mailman/message/32890381/
+#ifdef __forceinline
+#undef __forceinline
+#endif
+#define __forceinline
+#define alloca __builtin_alloca
#elif !defined(_MSC_VER)
- #if __GNUC__
- #define __forceinline inline
- #else
- #define __forceinline
- #endif
+#if __GNUC__
+#define __forceinline inline
+#else
+#define __forceinline
+#endif
#endif
#if STB_VORBIS_MAX_CHANNELS > 256
@@ -636,237 +642,237 @@ typedef float codetype;
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
+ 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
+#else
int32 fast_huffman[FAST_HUFFMAN_TABLE_SIZE];
- #endif
- uint32 *sorted_codewords;
- int *sorted_values;
- int sorted_entries;
+#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
+ 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;
+ 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;
+ Floor0 floor0;
+ Floor1 floor1;
} Floor;
typedef struct
{
- uint32 begin, end;
- uint32 part_size;
- uint8 classifications;
- uint8 classbook;
- uint8 **classdata;
- int16 (*residue_books)[8];
+ 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;
+ 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
+ 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;
+ 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
+ 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 last_decoded_sample;
+ uint32 page_start, page_end;
+ uint32 last_decoded_sample;
} ProbedPage;
struct stb_vorbis
{
- // user-accessible info
- unsigned int sample_rate;
- int channels;
+ // 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;
+ unsigned int setup_memory_required;
+ unsigned int temp_memory_required;
+ unsigned int setup_temp_memory_required;
- // input config
+ // input config
#ifndef STB_VORBIS_NO_STDIO
- FILE *f;
- uint32 f_start;
- int close_on_free;
+ 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
- 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
+ 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];
+ CRCscan scan[STB_VORBIS_PUSHDATA_CRC_COUNT];
#endif
- // sample-access
- int channel_buffer_start;
- int channel_buffer_end;
+ // sample-access
+ int channel_buffer_start;
+ int channel_buffer_end;
};
#if defined(STB_VORBIS_NO_PUSHDATA_API)
- #define IS_PUSH_MODE(f) FALSE
+#define IS_PUSH_MODE(f) FALSE
#elif defined(STB_VORBIS_NO_PULLDATA_API)
- #define IS_PUSH_MODE(f) TRUE
+#define IS_PUSH_MODE(f) TRUE
#else
- #define IS_PUSH_MODE(f) ((f)->push_mode)
+#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;
+ f->error = e;
+ if (!f->eof && e != VORBIS_need_more_data) {
+ f->error = e; // breakpoint for debugging
+ }
+ return 0;
}
@@ -878,11 +884,7 @@ static int error(vorb *f, enum STBVorbisError e)
#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))
@@ -891,53 +893,53 @@ static int error(vorb *f, enum STBVorbisError e)
// 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;
+ 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;
+ 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 a stack
- free(p);
+ if (f->alloc.alloc_buffer) return; // do nothing; setup mem is 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);
+ 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);
+ if (f->alloc.alloc_buffer) {
+ f->temp_offset += (sz + 3)&~3;
+ return;
+ }
+ free(p);
}
#define CRC32_POLY 0x04c11db7 // from spec
@@ -945,34 +947,34 @@ static void setup_temp_free(vorb *f, void *p, int sz)
static uint32 crc_table[256];
static void crc32_init(void)
{
- int i,j;
- uint32 s;
- for(i=0; i < 256; i++) {
- for (s=(uint32) i << 24, j=0; j < 8; ++j)
- s = (s << 1) ^ (s >= (1U<<31) ? CRC32_POLY : 0);
- crc_table[i] = s;
- }
+ int i, j;
+ uint32 s;
+ for (i = 0; i < 256; i++) {
+ for (s = (uint32)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)];
+ 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);
+ 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;
+ return x*x;
}
// this is a weird definition of log2() for which log2(1) = 1, log2(2) = 2, log2(4) = 3
@@ -980,24 +982,24 @@ static float square(float x)
// @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 };
+ static signed char log2_4[16] = { 0,1,2,2,3,3,3,3,4,4,4,4,4,4,4,4 };
- if (n < 0) return 0; // signed n returns 0
+ if (n < 0) return 0; // signed n returns 0
- // 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 return 30 + log2_4[n >> 30];
+ // 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 return 30 + log2_4[n >> 30];
}
#ifndef M_PI
- #define M_PI 3.14159265358979323846264f // from CRC
+#define M_PI 3.14159265358979323846264f // from CRC
#endif
// code length assigned to a value with no huffman encoding
@@ -1010,12 +1012,12 @@ static int ilog(int32 n)
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);
+ // 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);
}
@@ -1028,83 +1030,84 @@ static float float32_unpack(uint32 x)
// 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;
- }
+ 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] = 1U << (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) { return FALSE; }
- res = available[z];
- assert(z >= 0 && z < 32);
- available[z] = 0;
- add_entry(c, bit_reverse(res), i, m++, len[i], values);
- // propogate availability up the tree
- if (z != len[i]) {
- assert(len[i] >= 0 && len[i] < 32);
- for (y=len[i]; y > z; --y) {
- assert(available[y] == 0);
- available[y] = res + (1 << (32-y));
- }
- }
- }
- return TRUE;
+ 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] = 1U << (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) { return FALSE; }
+ res = available[z];
+ assert(z >= 0 && z < 32);
+ available[z] = 0;
+ add_entry(c, bit_reverse(res), i, m++, len[i], values);
+ // propogate availability up the tree
+ if (z != len[i]) {
+ assert(len[i] >= 0 && len[i] < 32);
+ 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];
- }
- }
- }
+ 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
@@ -1115,165 +1118,168 @@ static void compute_accelerated_huffman(Codebook *c)
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;
+ 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 (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;
+ 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;
- }
- }
- }
+ 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;
+ 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;
+ 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);
- }
+ 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)));
+ 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;
+ 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;
+ 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]; }
- }
+ 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,id;
+ uint16 x, id;
} stbv__floor_ordering;
static int STBV_CDECL point_compare(const void *p, const void *q)
{
- stbv__floor_ordering *a = (stbv__floor_ordering *) p;
- stbv__floor_ordering *b = (stbv__floor_ordering *) q;
- return a->x < b->x ? -1 : a->x > b->x;
+ stbv__floor_ordering *a = (stbv__floor_ordering *)p;
+ stbv__floor_ordering *b = (stbv__floor_ordering *)q;
+ return a->x < b->x ? -1 : a->x > b->x;
}
//
@@ -1281,100 +1287,102 @@ static int STBV_CDECL point_compare(const void *p, const void *q)
#if defined(STB_VORBIS_NO_STDIO)
- #define USE_MEMORY(z) TRUE
+#define USE_MEMORY(z) TRUE
#else
- #define USE_MEMORY(z) ((z)->stream)
+#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++;
- }
+ 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
+#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 += (uint32) get8(f) << 24;
- return x;
+ uint32 x;
+ x = get8(f);
+ x += get8(f) << 8;
+ x += get8(f) << 16;
+ x += (uint32)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
+ 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
+ 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
+#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
}
@@ -1382,11 +1390,11 @@ 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;
+ 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
@@ -1395,118 +1403,118 @@ static int capture_pattern(vorb *f)
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.last_decoded_sample = loc0;
- f->p_first = p;
- }
- f->next_seg = 0;
- return TRUE;
+ 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.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);
+ 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;
+ 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);
+ 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;
+ 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)
@@ -1514,58 +1522,58 @@ static int next_segment(vorb *f)
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);
+ 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;
+ int x = get8_packet_raw(f);
+ f->valid_bits = 0;
+ return x;
}
static void flush_packet(vorb *f)
{
- while (get8_packet_raw(f) != EOP);
+ 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;
+ 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
@@ -1574,83 +1582,84 @@ static uint32 get_bits(vorb *f, int n)
// 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 += (unsigned) z << f->valid_bits;
- f->valid_bits += 8;
- } while (f->valid_bits <= 24);
- }
+ 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 += (unsigned)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
+ 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);
-
- if (c->codewords == NULL && c->sorted_codewords == NULL)
- return -1;
-
- // 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;
- }
+ int i;
+ prep_huffman(f);
- // 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;
- }
- }
+ if (c->codewords == NULL && c->sorted_codewords == NULL)
+ return -1;
- error(f, VORBIS_invalid_stream);
- f->valid_bits = 0;
- return -1;
+ // 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
@@ -1673,19 +1682,19 @@ static int codebook_decode_scalar_raw(vorb *f, Codebook *c)
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);
+ 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);
@@ -1697,9 +1706,9 @@ static int codebook_decode_scalar(vorb *f, Codebook *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)
+#define DECODE_VQ(var,f,c) DECODE_RAW(var,f,c)
#else
- #define DECODE_VQ(var,f,c) DECODE(var,f,c)
+#define DECODE_VQ(var,f,c) DECODE(var,f,c)
#endif
@@ -1715,241 +1724,244 @@ static int codebook_decode_scalar(vorb *f, Codebook *c)
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;
+ 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;
+ 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;
- }
+ 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;
+ 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;
+ 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;
- }
+ 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;
- }
+ 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;
+ 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;
+ 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; }
+ 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;
+ total_decode -= effective;
+ }
+ *c_inter_p = c_inter;
+ *p_inter_p = p_inter;
+ return TRUE;
}
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;
+ 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
+ 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
};
@@ -1974,289 +1986,303 @@ int8 integer_divide_table[DIVTAB_NUMER][DIVTAB_DENOM]; // 2KB
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;
+ 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;
- }
+ 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;
+ base = dy / adx;
+ if (dy < 0)
+ sy = base - 1;
+ else
+ sy = base + 1;
#endif
- ady -= abs(base) * adx;
- if (x1 > n) x1 = n;
- if (x < x1) {
- 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]);
- }
- }
+ ady -= abs(base) * adx;
+ if (x1 > n) x1 = n;
+ if (x < x1) {
+ 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;
+ 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;
}
+// n is 1/2 of the blocksize --
+// specification: "Correct per-vector decode length is [n]/2"
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
-
- CHECK(f);
-
- 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;
-
- for (pass=0; pass < 8; ++pass) {
- int pcount = 0, class_set = 0;
- if (ch == 2) {
- 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
- }
- 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;
- #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(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
- goto done;
- #endif
- } else {
- z += r->part_size;
- c_inter = z & 1;
- p_inter = z >> 1;
- }
- }
- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
- ++class_set;
- #endif
+ 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;
+ unsigned int actual_size = rtype == 2 ? n * 2 : n;
+ unsigned int limit_r_begin = (r->begin < actual_size ? r->begin : actual_size);
+ unsigned int limit_r_end = (r->end < actual_size ? r->end : actual_size);
+ int n_read = limit_r_end - limit_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
+
+ CHECK(f);
+
+ 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;
+
+ for (pass = 0; pass < 8; ++pass) {
+ int pcount = 0, class_set = 0;
+ if (ch == 2) {
+ 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
+ }
+ 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;
+#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(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+ goto done;
+#endif
+ }
+ else {
+ z += r->part_size;
+ c_inter = z & 1;
+ p_inter = z >> 1;
+ }
+ }
+#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;
- if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
- goto done;
- } else {
- z += r->part_size;
- c_inter = 0;
- p_inter = z;
- }
- }
- #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;
+ if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+ goto done;
+ }
+ 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;
- if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
- goto done;
- } else {
- z += r->part_size;
- c_inter = z % ch;
- p_inter = z / ch;
- }
- }
- #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;
+ if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+ goto done;
+ }
+ else {
+ z += r->part_size;
+ c_inter = z % ch;
+ p_inter = z / ch;
+ }
+ }
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+ ++class_set;
+#endif
+ }
}
- }
- }
- goto done;
- }
- CHECK(f);
-
- 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
- }
+ }
+ goto done;
+ }
+ CHECK(f);
+
+ 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;
- }
- }
+ 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:
- CHECK(f);
- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
- temp_free(f,part_classdata);
- #else
- temp_free(f,classifications);
- #endif
- temp_alloc_restore(f,temp_alloc_point);
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+ ++class_set;
+#endif
+ }
+ }
+done:
+ CHECK(f);
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+ temp_free(f, part_classdata);
+#else
+ temp_free(f, classifications);
+#endif
+ temp_alloc_restore(f, temp_alloc_point);
}
@@ -2264,76 +2290,76 @@ static void decode_residue(vorb *f, float *residue_buffers[], int ch, int n, int
// 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);
+ 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);
+ 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;
- }
+ 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];
+ 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'
+ 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
+ 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
@@ -2344,36 +2370,36 @@ void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype)
#if LIBVORBIS_MDCT
// directly call the vorbis MDCT using an interface documented
// by Jeff Roberts... useful for performance comparison
-typedef struct
+typedef struct
{
- int n;
- int log2n;
-
- float *trig;
- int *bitrev;
+ int n;
+ int log2n;
+
+ float *trig;
+ int *bitrev;
- float scale;
+ 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;
+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);
+ 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
@@ -2383,663 +2409,664 @@ void inverse_mdct(float *buffer, int n, vorb *f, int blocktype)
// 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;
- }
+ 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;
+ int i;
+ float k00_20, k01_21;
- float *e0 = e + d0;
- float *e2 = e0 + k_off;
+ 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];
+ 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;
+ 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];
+ 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;
+ 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];
+ 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;
+ 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];
+ 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;
+ e0 -= 8;
+ e2 -= 8;
- A += k1;
- }
+ 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;
- }
+ 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;
+ 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];
+ 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
+ z[-0] = y0 + y2; // z0 + z4 + z2 + z6
+ z[-2] = y0 - y2; // z0 + z4 - z2 - z6
- // done with y0,y2
+ // done with y0,y2
- k33 = z[-3] - z[-7];
+ k33 = z[-3] - z[-7];
- z[-4] = k00 + k33; // z0 - z4 + z3 - z7
- z[-6] = k00 - k33; // z0 - z4 - z3 + z7
+ z[-4] = k00 + k33; // z0 - z4 + z3 - z7
+ z[-6] = k00 - k33; // z0 - z4 - z3 + z7
- // done with k33
+ // done with k33
- k11 = z[-1] - z[-5];
- y1 = z[-1] + z[-5];
- y3 = z[-3] + z[-7];
+ 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
+ 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;
- }
+ 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
-
+ 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];
- // 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
+ d0[2] = p3;
+ d1[1] = -p3;
+ d2[2] = p2;
+ d3[1] = p2;
- // this cannot POSSIBLY be in place, so we refer to the buffers directly
+ p1 = e[0] * B[1] - e[1] * B[0];
+ p0 = -e[0] * B[0] - e[1] * B[1];
- {
- float *d0,*d1,*d2,*d3;
+ d0[3] = p1;
+ d1[0] = -p1;
+ d2[3] = p0;
+ d3[0] = p0;
- 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;
+ B -= 8;
+ e -= 8;
+ d0 += 4;
+ d2 += 4;
+ d1 -= 4;
+ d3 -= 4;
+ }
+ }
- 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_free(f,buf2);
- temp_alloc_restore(f,save_point);
+ temp_free(f, buf2);
+ 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];
+ 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;
+ 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
@@ -3049,39 +3076,40 @@ 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];
- if (lx != hx)
- draw_line(target, lx,ly, hx,hy, n2);
+ 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];
+ if (lx != hx)
+ draw_line(target, lx, ly, hx, hy, n2);
+ CHECK(f);
+ 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]);
CHECK(f);
- 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]);
- CHECK(f);
- }
- }
- return TRUE;
+ }
+ }
+ return TRUE;
}
// The meaning of "left" and "right"
@@ -3100,1340 +3128,1373 @@ static int do_floor(vorb *f, Mapping *map, int i, int n, float *target, YTYPE *f
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;
+ 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;
-
- CHECK(f);
-
- 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;
- }
- }
+ 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;
+
+ CHECK(f);
+
+ 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);
+ 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;
- }
+ // 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
- }
- }
- CHECK(f);
- // 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;
- }
-
- CHECK(f);
-// 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);
- CHECK(f);
-
-// INVERSE COUPLING
- 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 {
+ error:
+ zero_channel[i] = TRUE;
+ }
+ // So we just defer everything else to later
+
+ // at this point we've decoded the floor into buffer
+ }
+ }
+ CHECK(f);
+ // 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;
+ }
+
+ CHECK(f);
+ // 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);
+ CHECK(f);
+
+ // INVERSE COUPLING
+ 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
- a2 = m[j], m2 = m[j] - a[j];
- m[j] = m2;
- a[j] = a2;
- }
- }
- CHECK(f);
-
- // finish decoding the floors
+ 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;
+ }
+ }
+ CHECK(f);
+
+ // finish decoding the floors
#ifndef STB_VORBIS_NO_DEFER_FLOOR
- 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);
- }
- }
+ 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];
- }
- }
+ 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
- CHECK(f);
- for (i=0; i < f->channels; ++i)
- inverse_mdct(f->channel_buffers[i], n, f, m->blockflag);
- CHECK(f);
-
- // 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) {
- if (f->discard_samples_deferred >= right_start - left_start) {
- f->discard_samples_deferred -= (right_start - left_start);
- left_start = right_start;
- *p_left = left_start;
- } else {
- 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-left_start)) {
- if (current_end < f->current_loc) {
- // negative truncation, that's impossible!
- *len = 0;
- } else {
- *len = current_end - f->current_loc;
+ // INVERSE MDCT
+ CHECK(f);
+ for (i = 0; i < f->channels; ++i)
+ inverse_mdct(f->channel_buffers[i], n, f, m->blockflag);
+ CHECK(f);
+
+ // 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) {
+ if (f->discard_samples_deferred >= right_start - left_start) {
+ f->discard_samples_deferred -= (right_start - left_start);
+ left_start = right_start;
+ *p_left = left_start;
+ }
+ else {
+ 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;
+ // then let's infer the size of the (probably) short final frame
+ if (current_end < f->current_loc + (right_end - left_start)) {
+ if (current_end < f->current_loc) {
+ // negative truncation, that's impossible!
+ *len = 0;
+ }
+ else {
+ *len = current_end - f->current_loc;
+ }
+ *len += left_start; // this doesn't seem right, but has no ill effect on my test files
+ if (*len > right_end) *len = right_end; // this should never happen
+ f->current_loc += *len;
+ return TRUE;
}
- *len += left_start;
- if (*len > right_end) *len = right_end; // this should never happen
- 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
- CHECK(f);
-
- 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
+ CHECK(f);
+
+ 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);
+ 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;
+ 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 int vorbis_pump_first_frame(stb_vorbis *f)
{
- int len, right, left, res;
- res = vorbis_decode_packet(f, &len, &left, &right);
- if (res)
- vorbis_finish_frame(f, len, left, right);
- return res;
+ int len, right, left, res;
+ res = vorbis_decode_packet(f, &len, &left, &right);
+ if (res)
+ vorbis_finish_frame(f, len, left, right);
+ return res;
}
#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;
+ // 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);
- }
+ 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;
- // 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;
- CHECK(f);
- 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->dimensions == 0 && c->entries != 0) return error(f, VORBIS_invalid_setup);
-
- 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;
- if (lengths[j] == 32)
- return error(f, VORBIS_invalid_setup);
- } else {
- lengths[j] = NO_CODE;
+#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;
+ CHECK(f);
+ 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->dimensions == 0 && c->entries != 0) return error(f, VORBIS_invalid_setup);
+
+ 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;
+ if (lengths[j] == 32)
+ return error(f, VORBIS_invalid_setup);
+ }
+ 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);
- if (c->codeword_lengths == NULL) return error(f, VORBIS_outofmem);
- 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;
-
- CHECK(f);
- 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->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);
+ if (c->codeword_lengths == NULL) return error(f, VORBIS_outofmem);
+ 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;
+
+ CHECK(f);
+ if (!c->sparse) {
+ c->codewords = (uint32 *)setup_malloc(f, sizeof(c->codewords[0]) * c->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));
- if (c->sorted_codewords == NULL) return error(f, VORBIS_outofmem);
- // 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 == NULL) return error(f, VORBIS_outofmem);
- ++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);
-
- CHECK(f);
- 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;
- }
- if (c->lookup_values == 0) return error(f, VORBIS_invalid_setup);
- 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;
- }
+ }
+ 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));
+ if (c->sorted_codewords == NULL) return error(f, VORBIS_outofmem);
+ // 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 == NULL) return error(f, VORBIS_outofmem);
+ ++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);
+
+ CHECK(f);
+ 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;
+ }
+ if (c->lookup_values == 0) return error(f, VORBIS_invalid_setup);
+ 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;
- float last=0;
- // 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) {
- unsigned int z = sparse ? c->sorted_values[j] : j;
- unsigned int div=1;
- for (k=0; k < c->dimensions; ++k) {
- int off = (z / div) % c->lookup_values;
- float val = mults[off];
- val = mults[off]*c->delta_value + c->minimum_value + last;
- c->multiplicands[j*c->dimensions + k] = val;
- if (c->sequence_p)
- last = val;
- if (k+1 < c->dimensions) {
- if (div > UINT_MAX / (unsigned int) c->lookup_values) {
- setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values);
- return error(f, VORBIS_invalid_setup);
- }
- div *= c->lookup_values;
- }
- }
+ if (c->lookup_type == 1) {
+ int len, sparse = c->sparse;
+ float last = 0;
+ // 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) {
+ unsigned int z = sparse ? c->sorted_values[j] : j;
+ unsigned int div = 1;
+ for (k = 0; k < c->dimensions; ++k) {
+ int off = (z / div) % c->lookup_values;
+ float val = mults[off];
+ val = mults[off] * c->delta_value + c->minimum_value + last;
+ c->multiplicands[j*c->dimensions + k] = val;
+ if (c->sequence_p)
+ last = val;
+ if (k + 1 < c->dimensions) {
+ if (div > UINT_MAX / (unsigned int)c->lookup_values) {
+ setup_temp_free(f, mults, sizeof(mults[0])*c->lookup_values);
+ return error(f, VORBIS_invalid_setup);
+ }
+ div *= c->lookup_values;
+ }
+ }
+ }
+ c->lookup_type = 2;
}
- c->lookup_type = 2;
- }
- else
+ else
#endif
- {
- float last=0;
- CHECK(f);
- c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values);
- if (c->multiplicands == NULL) { setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); }
- for (j=0; j < (int) c->lookup_values; ++j) {
- float val = mults[j] * c->delta_value + c->minimum_value + last;
- c->multiplicands[j] = val;
- if (c->sequence_p)
- last = val;
+ {
+ float last = 0;
+ CHECK(f);
+ c->multiplicands = (codetype *)setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values);
+ if (c->multiplicands == NULL) { setup_temp_free(f, mults, sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); }
+ for (j = 0; j < (int)c->lookup_values; ++j) {
+ float val = mults[j] * c->delta_value + c->minimum_value + last;
+ c->multiplicands[j] = val;
+ if (c->sequence_p)
+ last = val;
+ }
}
- }
#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
- skip:;
+ skip : ;
#endif
- setup_temp_free(f, mults, sizeof(mults[0])*c->lookup_values);
-
- CHECK(f);
- }
- CHECK(f);
- }
-
- // 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));
- if (f->floor_config == NULL) return error(f, VORBIS_outofmem);
- 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 {
- stbv__floor_ordering 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);
+ setup_temp_free(f, mults, sizeof(mults[0])*c->lookup_values);
+
+ CHECK(f);
+ }
+ CHECK(f);
+ }
+
+ // 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));
+ if (f->floor_config == NULL) return error(f, VORBIS_outofmem);
+ 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 {
+ stbv__floor_ordering 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 (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);
+ 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;
+ 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].id = j;
- }
- qsort(p, g->values, sizeof(p[0]), point_compare);
- for (j=0; j < g->values; ++j)
- g->sorted_order[j] = (uint8) p[j].id;
- // 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[0]));
- if (f->residue_config == NULL) return error(f, VORBIS_outofmem);
- memset(f->residue_config, 0, f->residue_count * sizeof(f->residue_config[0]));
- 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);
- if (r->end < r->begin) return error(f, VORBIS_invalid_setup);
- r->part_size = get_bits(f,24)+1;
- r->classifications = get_bits(f,6)+1;
- r->classbook = get_bits(f,8);
- if (r->classbook >= f->codebook_count) return error(f, VORBIS_invalid_setup);
- 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);
- if (r->residue_books == NULL) return error(f, VORBIS_outofmem);
- 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 sorting
+ for (j = 0; j < g->values; ++j) {
+ p[j].x = g->Xlist[j];
+ p[j].id = j;
+ }
+ qsort(p, g->values, sizeof(p[0]), point_compare);
+ for (j = 0; j < g->values; ++j)
+ g->sorted_order[j] = (uint8)p[j].id;
+ // 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;
}
- }
- }
- // 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);
- if (r->classdata[j] == NULL) return error(f, VORBIS_outofmem);
- 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));
- if (f->mapping == NULL) return error(f, VORBIS_outofmem);
- memset(f->mapping, 0, 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 (m->chan == NULL) return error(f, VORBIS_outofmem);
- 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);
- if (f->channel_buffers[i] == NULL || f->previous_window[i] == NULL || f->finalY[i] == NULL) return error(f, VORBIS_outofmem);
- #ifdef STB_VORBIS_NO_DEFER_FLOOR
- f->floor_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2);
- if (f->floor_buffers[i] == NULL) return error(f, VORBIS_outofmem);
- #endif
- }
+ 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[0]));
+ if (f->residue_config == NULL) return error(f, VORBIS_outofmem);
+ memset(f->residue_config, 0, f->residue_count * sizeof(f->residue_config[0]));
+ 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);
+ if (r->end < r->begin) return error(f, VORBIS_invalid_setup);
+ r->part_size = get_bits(f, 24) + 1;
+ r->classifications = get_bits(f, 6) + 1;
+ r->classbook = get_bits(f, 8);
+ if (r->classbook >= f->codebook_count) return error(f, VORBIS_invalid_setup);
+ 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);
+ if (r->residue_books == NULL) return error(f, VORBIS_outofmem);
+ 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);
+ if (r->classdata[j] == NULL) return error(f, VORBIS_outofmem);
+ 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));
+ if (f->mapping == NULL) return error(f, VORBIS_outofmem);
+ memset(f->mapping, 0, 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 (m->chan == NULL) return error(f, VORBIS_outofmem);
+ 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);
+ if (f->channel_buffers[i] == NULL || f->previous_window[i] == NULL || f->finalY[i] == NULL) return error(f, VORBIS_outofmem);
+ memset(f->channel_buffers[i], 0, sizeof(float) * f->blocksize_1);
+#ifdef STB_VORBIS_NO_DEFER_FLOOR
+ f->floor_buffers[i] = (float *)setup_malloc(f, sizeof(float) * f->blocksize_1 / 2);
+ if (f->floor_buffers[i] == NULL) return error(f, VORBIS_outofmem);
+#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;
+ 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;
+ 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;
- }
+ // 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;
+ unsigned int actual_size = f->blocksize_1 / 2;
+ unsigned int limit_r_begin = r->begin < actual_size ? r->begin : actual_size;
+ unsigned int limit_r_end = r->end < actual_size ? r->end : actual_size;
+ int n_read = limit_r_end - limit_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->first_decode = TRUE;
+ // maximum reasonable partition size is f->blocksize_1
- 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->temp_memory_required = classify_mem;
+ if (imdct_mem > f->temp_memory_required)
+ f->temp_memory_required = imdct_mem;
+ }
+
+ f->first_decode = TRUE;
- f->first_audio_page_offset = stb_vorbis_get_file_offset(f);
+ 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);
+ }
- return TRUE;
+ f->first_audio_page_offset = stb_vorbis_get_file_offset(f);
+
+ return TRUE;
}
static void vorbis_deinit(stb_vorbis *p)
{
- int i,j;
- if (p->residue_config) {
- 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) {
- CHECK(p);
- 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);
- if (p->mapping) {
- for (i=0; i < p->mapping_count; ++i)
- setup_free(p, p->mapping[i].chan);
- setup_free(p, p->mapping);
- }
- CHECK(p);
- for (i=0; i < p->channels && i < STB_VORBIS_MAX_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
+ int i, j;
+ if (p->residue_config) {
+ 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) {
+ CHECK(p);
+ 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);
+ if (p->mapping) {
+ for (i = 0; i < p->mapping_count; ++i)
+ setup_free(p, p->mapping[i].chan);
+ setup_free(p, p->mapping);
+ }
+ CHECK(p);
+ for (i = 0; i < p->channels && i < STB_VORBIS_MAX_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);
+ if (p == NULL) return;
+ vorbis_deinit(p);
+ setup_free(p, p);
}
static void vorbis_init(stb_vorbis *p, const 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
+ 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;
+ 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;
+ 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;
+ 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;
+ 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;
+ 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
+ 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;
+ }
+ }
+
+ 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
- const 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, (uint8 *) data, data_len);
- }
-
- f->stream = (uint8 *) data;
- f->stream_end = (uint8 *) 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 (int) (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
+ stb_vorbis *f, // the file we're decoding
+ const 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, (uint8 *)data, data_len);
+ }
+
+ f->stream = (uint8 *)data;
+ f->stream_end = (uint8 *)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;
+ if (f->eof) break;
*samples = 0;
- return (int) (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 (int) (f->stream - data);
+ return (int)(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 (int)(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 (int)(f->stream - data);
}
stb_vorbis *stb_vorbis_open_pushdata(
- const unsigned char *data, int data_len, // the memory available for decoding
- int *data_used, // only defined if result is not NULL
- int *error, const stb_vorbis_alloc *alloc)
-{
- stb_vorbis *f, p;
- vorbis_init(&p, alloc);
- p.stream = (uint8 *) data;
- p.stream_end = (uint8 *) 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 = (int) (f->stream - data);
- *error = 0;
- return f;
- } else {
- vorbis_deinit(&p);
- return NULL;
- }
+ const unsigned char *data, int data_len, // the memory available for decoding
+ int *data_used, // only defined if result is not NULL
+ int *error, const stb_vorbis_alloc *alloc)
+{
+ stb_vorbis *f, p;
+ vorbis_init(&p, alloc);
+ p.stream = (uint8 *)data;
+ p.stream_end = (uint8 *)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 = (int)(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 (unsigned int) (f->stream - f->stream_start);
- #ifndef STB_VORBIS_NO_STDIO
- return (unsigned int) (ftell(f->f) - f->f_start);
- #endif
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+ if (f->push_mode) return 0;
+#endif
+ if (USE_MEMORY(f)) return (unsigned int)(f->stream - f->stream_start);
+#ifndef STB_VORBIS_NO_STDIO
+ return (unsigned int)(ftell(f->f) - f->f_start);
+#endif
}
#ifndef STB_VORBIS_NO_PULLDATA_API
@@ -4443,72 +4504,72 @@ unsigned int stb_vorbis_get_file_offset(stb_vorbis *f)
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 candidate
- 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);
+ for (;;) {
+ int n;
+ if (f->eof) return 0;
+ n = get8(f);
+ if (n == 0x4f) { // page header candidate
+ 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 (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 (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;
+ }
}
- 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);
- }
- }
+ // not a valid page, so rewind and look for next one
+ set_file_offset(f, retry_loc);
+ }
+ }
}
@@ -4525,55 +4586,55 @@ static uint32 vorbis_find_page(stb_vorbis *f, uint32 *end, uint32 *last)
static int get_seek_page_info(stb_vorbis *f, ProbedPage *z)
{
- uint8 header[27], lacing[255];
- int i,len;
+ uint8 header[27], lacing[255];
+ int i, len;
- // record where the page starts
- z->page_start = stb_vorbis_get_file_offset(f);
+ // record where the page starts
+ z->page_start = stb_vorbis_get_file_offset(f);
- // parse the header
- getn(f, header, 27);
- if (header[0] != 'O' || header[1] != 'g' || header[2] != 'g' || header[3] != 'S')
- return 0;
- getn(f, lacing, header[26]);
+ // parse the header
+ getn(f, header, 27);
+ if (header[0] != 'O' || header[1] != 'g' || header[2] != 'g' || header[3] != 'S')
+ return 0;
+ getn(f, lacing, header[26]);
- // determine the length of the payload
- len = 0;
- for (i=0; i < header[26]; ++i)
- len += lacing[i];
+ // 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;
+ // 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] << 24);
+ // read the last-decoded sample out of the data
+ z->last_decoded_sample = header[6] + (header[7] << 8) + (header[8] << 16) + (header[9] << 24);
- // restore file state to where we were
- set_file_offset(f, z->page_start);
- return 1;
+ // restore file state to where we were
+ set_file_offset(f, z->page_start);
+ return 1;
}
// rarely used function to seek back to the preceeding page while finding the
// start of a packet
static int go_to_page_before(stb_vorbis *f, unsigned int limit_offset)
{
- unsigned int previous_safe, end;
+ unsigned int previous_safe, end;
- // now we want to seek back 64K from the limit
- if (limit_offset >= 65536 && limit_offset-65536 >= f->first_audio_page_offset)
- previous_safe = limit_offset - 65536;
- else
- previous_safe = f->first_audio_page_offset;
+ // now we want to seek back 64K from the limit
+ if (limit_offset >= 65536 && limit_offset - 65536 >= f->first_audio_page_offset)
+ previous_safe = limit_offset - 65536;
+ else
+ previous_safe = f->first_audio_page_offset;
- set_file_offset(f, previous_safe);
+ set_file_offset(f, previous_safe);
- while (vorbis_find_page(f, &end, NULL)) {
- if (end >= limit_offset && stb_vorbis_get_file_offset(f) < limit_offset)
- return 1;
- set_file_offset(f, end);
- }
+ while (vorbis_find_page(f, &end, NULL)) {
+ if (end >= limit_offset && stb_vorbis_get_file_offset(f) < limit_offset)
+ return 1;
+ set_file_offset(f, end);
+ }
- return 0;
+ return 0;
}
// implements the search logic for finding a page and starting decoding. if
@@ -4582,414 +4643,419 @@ static int go_to_page_before(stb_vorbis *f, unsigned int limit_offset)
// better).
static int seek_to_sample_coarse(stb_vorbis *f, uint32 sample_number)
{
- ProbedPage left, right, mid;
- int i, start_seg_with_known_loc, end_pos, page_start;
- uint32 delta, stream_length, padding;
- double offset, bytes_per_sample;
- int probe = 0;
-
- // find the last page and validate the target sample
- stream_length = stb_vorbis_stream_length_in_samples(f);
- if (stream_length == 0) return error(f, VORBIS_seek_without_length);
- if (sample_number > stream_length) return error(f, VORBIS_seek_invalid);
-
- // this is the maximum difference between the window-center (which is the
- // actual granule position value), and the right-start (which the spec
- // indicates should be the granule position (give or take one)).
- padding = ((f->blocksize_1 - f->blocksize_0) >> 2);
- if (sample_number < padding)
- sample_number = 0;
- else
- sample_number -= padding;
-
- left = f->p_first;
- while (left.last_decoded_sample == ~0U) {
- // (untested) the first page does not have a 'last_decoded_sample'
- set_file_offset(f, left.page_end);
- if (!get_seek_page_info(f, &left)) goto error;
- }
-
- right = f->p_last;
- assert(right.last_decoded_sample != ~0U);
-
- // starting from the start is handled differently
- if (sample_number <= left.last_decoded_sample) {
- if (stb_vorbis_seek_start(f))
- return 1;
- return 0;
- }
-
- while (left.page_end != right.page_start) {
- assert(left.page_end < right.page_start);
- // search range in bytes
- delta = right.page_start - left.page_end;
- if (delta <= 65536) {
- // there's only 64K left to search - handle it linearly
- set_file_offset(f, left.page_end);
- } else {
- if (probe < 2) {
- if (probe == 0) {
- // first probe (interpolate)
- double data_bytes = right.page_end - left.page_start;
- bytes_per_sample = data_bytes / right.last_decoded_sample;
- offset = left.page_start + bytes_per_sample * (sample_number - left.last_decoded_sample);
- } else {
- // second probe (try to bound the other side)
- double error = ((double) sample_number - mid.last_decoded_sample) * bytes_per_sample;
- if (error >= 0 && error < 8000) error = 8000;
- if (error < 0 && error > -8000) error = -8000;
- offset += error * 2;
+ ProbedPage left, right, mid;
+ int i, start_seg_with_known_loc, end_pos, page_start;
+ uint32 delta, stream_length, padding;
+ double offset, bytes_per_sample;
+ int probe = 0;
+
+ // find the last page and validate the target sample
+ stream_length = stb_vorbis_stream_length_in_samples(f);
+ if (stream_length == 0) return error(f, VORBIS_seek_without_length);
+ if (sample_number > stream_length) return error(f, VORBIS_seek_invalid);
+
+ // this is the maximum difference between the window-center (which is the
+ // actual granule position value), and the right-start (which the spec
+ // indicates should be the granule position (give or take one)).
+ padding = ((f->blocksize_1 - f->blocksize_0) >> 2);
+ if (sample_number < padding)
+ sample_number = 0;
+ else
+ sample_number -= padding;
+
+ left = f->p_first;
+ while (left.last_decoded_sample == ~0U) {
+ // (untested) the first page does not have a 'last_decoded_sample'
+ set_file_offset(f, left.page_end);
+ if (!get_seek_page_info(f, &left)) goto error;
+ }
+
+ right = f->p_last;
+ assert(right.last_decoded_sample != ~0U);
+
+ // starting from the start is handled differently
+ if (sample_number <= left.last_decoded_sample) {
+ if (stb_vorbis_seek_start(f))
+ return 1;
+ return 0;
+ }
+
+ while (left.page_end != right.page_start) {
+ assert(left.page_end < right.page_start);
+ // search range in bytes
+ delta = right.page_start - left.page_end;
+ if (delta <= 65536) {
+ // there's only 64K left to search - handle it linearly
+ set_file_offset(f, left.page_end);
+ }
+ else {
+ if (probe < 2) {
+ if (probe == 0) {
+ // first probe (interpolate)
+ double data_bytes = right.page_end - left.page_start;
+ bytes_per_sample = data_bytes / right.last_decoded_sample;
+ offset = left.page_start + bytes_per_sample * (sample_number - left.last_decoded_sample);
+ }
+ else {
+ // second probe (try to bound the other side)
+ double error = ((double)sample_number - mid.last_decoded_sample) * bytes_per_sample;
+ if (error >= 0 && error < 8000) error = 8000;
+ if (error < 0 && error > -8000) error = -8000;
+ offset += error * 2;
+ }
+
+ // ensure the offset is valid
+ if (offset < left.page_end)
+ offset = left.page_end;
+ if (offset > right.page_start - 65536)
+ offset = right.page_start - 65536;
+
+ set_file_offset(f, (unsigned int)offset);
+ }
+ else {
+ // binary search for large ranges (offset by 32K to ensure
+ // we don't hit the right page)
+ set_file_offset(f, left.page_end + (delta / 2) - 32768);
}
- // ensure the offset is valid
- if (offset < left.page_end)
- offset = left.page_end;
- if (offset > right.page_start - 65536)
- offset = right.page_start - 65536;
-
- set_file_offset(f, (unsigned int) offset);
- } else {
- // binary search for large ranges (offset by 32K to ensure
- // we don't hit the right page)
- set_file_offset(f, left.page_end + (delta / 2) - 32768);
- }
-
- if (!vorbis_find_page(f, NULL, NULL)) goto error;
- }
-
- for (;;) {
- if (!get_seek_page_info(f, &mid)) goto error;
- if (mid.last_decoded_sample != ~0U) break;
- // (untested) no frames end on this page
- set_file_offset(f, mid.page_end);
- assert(mid.page_start < right.page_start);
- }
-
- // if we've just found the last page again then we're in a tricky file,
- // and we're close enough.
- if (mid.page_start == right.page_start)
- break;
-
- if (sample_number < mid.last_decoded_sample)
- right = mid;
- else
- left = mid;
-
- ++probe;
- }
+ if (!vorbis_find_page(f, NULL, NULL)) goto error;
+ }
- // seek back to start of the last packet
- page_start = left.page_start;
- set_file_offset(f, page_start);
- if (!start_page(f)) return error(f, VORBIS_seek_failed);
- end_pos = f->end_seg_with_known_loc;
- assert(end_pos >= 0);
+ for (;;) {
+ if (!get_seek_page_info(f, &mid)) goto error;
+ if (mid.last_decoded_sample != ~0U) break;
+ // (untested) no frames end on this page
+ set_file_offset(f, mid.page_end);
+ assert(mid.page_start < right.page_start);
+ }
- for (;;) {
- for (i = end_pos; i > 0; --i)
- if (f->segments[i-1] != 255)
+ // if we've just found the last page again then we're in a tricky file,
+ // and we're close enough.
+ if (mid.page_start == right.page_start)
break;
- start_seg_with_known_loc = i;
+ if (sample_number < mid.last_decoded_sample)
+ right = mid;
+ else
+ left = mid;
- if (start_seg_with_known_loc > 0 || !(f->page_flag & PAGEFLAG_continued_packet))
- break;
+ ++probe;
+ }
- // (untested) the final packet begins on an earlier page
- if (!go_to_page_before(f, page_start))
- goto error;
+ // seek back to start of the last packet
+ page_start = left.page_start;
+ set_file_offset(f, page_start);
+ if (!start_page(f)) return error(f, VORBIS_seek_failed);
+ end_pos = f->end_seg_with_known_loc;
+ assert(end_pos >= 0);
- page_start = stb_vorbis_get_file_offset(f);
- if (!start_page(f)) goto error;
- end_pos = f->segment_count - 1;
- }
+ for (;;) {
+ for (i = end_pos; i > 0; --i)
+ if (f->segments[i - 1] != 255)
+ break;
+
+ start_seg_with_known_loc = i;
+
+ if (start_seg_with_known_loc > 0 || !(f->page_flag & PAGEFLAG_continued_packet))
+ break;
- // prepare to start decoding
- f->current_loc_valid = FALSE;
- f->last_seg = FALSE;
- f->valid_bits = 0;
- f->packet_bytes = 0;
- f->bytes_in_seg = 0;
- f->previous_length = 0;
- f->next_seg = start_seg_with_known_loc;
-
- for (i = 0; i < start_seg_with_known_loc; i++)
- skip(f, f->segments[i]);
-
- // start decoding (optimizable - this frame is generally discarded)
- if (!vorbis_pump_first_frame(f))
- return 0;
- if (f->current_loc > sample_number)
- return error(f, VORBIS_seek_failed);
- return 1;
+ // (untested) the final packet begins on an earlier page
+ if (!go_to_page_before(f, page_start))
+ goto error;
+
+ page_start = stb_vorbis_get_file_offset(f);
+ if (!start_page(f)) goto error;
+ end_pos = f->segment_count - 1;
+ }
+
+ // prepare to start decoding
+ f->current_loc_valid = FALSE;
+ f->last_seg = FALSE;
+ f->valid_bits = 0;
+ f->packet_bytes = 0;
+ f->bytes_in_seg = 0;
+ f->previous_length = 0;
+ f->next_seg = start_seg_with_known_loc;
+
+ for (i = 0; i < start_seg_with_known_loc; i++)
+ skip(f, f->segments[i]);
+
+ // start decoding (optimizable - this frame is generally discarded)
+ if (!vorbis_pump_first_frame(f))
+ return 0;
+ if (f->current_loc > sample_number)
+ return error(f, VORBIS_seek_failed);
+ return 1;
error:
- // try to restore the file to a valid state
- stb_vorbis_seek_start(f);
- return error(f, VORBIS_seek_failed);
+ // try to restore the file to a valid state
+ stb_vorbis_seek_start(f);
+ return error(f, VORBIS_seek_failed);
}
// the same as vorbis_decode_initial, but without advancing
static int peek_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode)
{
- int bits_read, bytes_read;
+ int bits_read, bytes_read;
- if (!vorbis_decode_initial(f, p_left_start, p_left_end, p_right_start, p_right_end, mode))
- return 0;
+ if (!vorbis_decode_initial(f, p_left_start, p_left_end, p_right_start, p_right_end, mode))
+ return 0;
- // either 1 or 2 bytes were read, figure out which so we can rewind
- bits_read = 1 + ilog(f->mode_count-1);
- if (f->mode_config[*mode].blockflag)
- bits_read += 2;
- bytes_read = (bits_read + 7) / 8;
+ // either 1 or 2 bytes were read, figure out which so we can rewind
+ bits_read = 1 + ilog(f->mode_count - 1);
+ if (f->mode_config[*mode].blockflag)
+ bits_read += 2;
+ bytes_read = (bits_read + 7) / 8;
- f->bytes_in_seg += bytes_read;
- f->packet_bytes -= bytes_read;
- skip(f, -bytes_read);
- if (f->next_seg == -1)
- f->next_seg = f->segment_count - 1;
- else
- f->next_seg--;
- f->valid_bits = 0;
+ f->bytes_in_seg += bytes_read;
+ f->packet_bytes -= bytes_read;
+ skip(f, -bytes_read);
+ if (f->next_seg == -1)
+ f->next_seg = f->segment_count - 1;
+ else
+ f->next_seg--;
+ f->valid_bits = 0;
- return 1;
+ return 1;
}
int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number)
{
- uint32 max_frame_samples;
-
- if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
-
- // fast page-level search
- if (!seek_to_sample_coarse(f, sample_number))
- return 0;
-
- assert(f->current_loc_valid);
- assert(f->current_loc <= sample_number);
-
- // linear search for the relevant packet
- max_frame_samples = (f->blocksize_1*3 - f->blocksize_0) >> 2;
- while (f->current_loc < sample_number) {
- int left_start, left_end, right_start, right_end, mode, frame_samples;
- if (!peek_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode))
- return error(f, VORBIS_seek_failed);
- // calculate the number of samples returned by the next frame
- frame_samples = right_start - left_start;
- if (f->current_loc + frame_samples > sample_number) {
- return 1; // the next frame will contain the sample
- } else if (f->current_loc + frame_samples + max_frame_samples > sample_number) {
- // there's a chance the frame after this could contain the sample
- vorbis_pump_first_frame(f);
- } else {
- // this frame is too early to be relevant
- f->current_loc += frame_samples;
- f->previous_length = 0;
- maybe_start_packet(f);
- flush_packet(f);
- }
- }
- // the next frame will start with the sample
- assert(f->current_loc == sample_number);
- return 1;
+ uint32 max_frame_samples;
+
+ if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+
+ // fast page-level search
+ if (!seek_to_sample_coarse(f, sample_number))
+ return 0;
+
+ assert(f->current_loc_valid);
+ assert(f->current_loc <= sample_number);
+
+ // linear search for the relevant packet
+ max_frame_samples = (f->blocksize_1 * 3 - f->blocksize_0) >> 2;
+ while (f->current_loc < sample_number) {
+ int left_start, left_end, right_start, right_end, mode, frame_samples;
+ if (!peek_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode))
+ return error(f, VORBIS_seek_failed);
+ // calculate the number of samples returned by the next frame
+ frame_samples = right_start - left_start;
+ if (f->current_loc + frame_samples > sample_number) {
+ return 1; // the next frame will contain the sample
+ }
+ else if (f->current_loc + frame_samples + max_frame_samples > sample_number) {
+ // there's a chance the frame after this could contain the sample
+ vorbis_pump_first_frame(f);
+ }
+ else {
+ // this frame is too early to be relevant
+ f->current_loc += frame_samples;
+ f->previous_length = 0;
+ maybe_start_packet(f);
+ flush_packet(f);
+ }
+ }
+ // the next frame will start with the sample
+ assert(f->current_loc == sample_number);
+ return 1;
}
int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number)
{
- if (!stb_vorbis_seek_frame(f, sample_number))
- return 0;
+ if (!stb_vorbis_seek_frame(f, sample_number))
+ return 0;
- if (sample_number != f->current_loc) {
- int n;
- uint32 frame_start = f->current_loc;
- stb_vorbis_get_frame_float(f, &n, NULL);
- assert(sample_number > frame_start);
- assert(f->channel_buffer_start + (int) (sample_number-frame_start) <= f->channel_buffer_end);
- f->channel_buffer_start += (sample_number - frame_start);
- }
+ if (sample_number != f->current_loc) {
+ int n;
+ uint32 frame_start = f->current_loc;
+ stb_vorbis_get_frame_float(f, &n, NULL);
+ assert(sample_number > frame_start);
+ assert(f->channel_buffer_start + (int)(sample_number - frame_start) <= f->channel_buffer_end);
+ f->channel_buffer_start += (sample_number - frame_start);
+ }
- return 1;
+ return 1;
}
int stb_vorbis_seek_start(stb_vorbis *f)
{
- if (IS_PUSH_MODE(f)) { return error(f, VORBIS_invalid_api_mixing); }
- set_file_offset(f, f->first_audio_page_offset);
- f->previous_length = 0;
- f->first_decode = TRUE;
- f->next_seg = -1;
- return vorbis_pump_first_frame(f);
+ if (IS_PUSH_MODE(f)) { return error(f, VORBIS_invalid_api_mixing); }
+ set_file_offset(f, f->first_audio_page_offset);
+ f->previous_length = 0;
+ f->first_decode = TRUE;
+ f->next_seg = -1;
+ return 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;
-
- done:
- set_file_offset(f, restore_offset);
- }
- return f->total_samples == SAMPLE_unknown ? 0 : f->total_samples;
+ 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;
+
+ 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;
+ 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);
+ 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;
- }
+ 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;
+ 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;
+ f->channel_buffer_start = left;
+ f->channel_buffer_end = left + len;
- if (channels) *channels = f->channels;
- if (output) *output = f->outputs;
- return 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, const stb_vorbis_alloc *alloc, unsigned int length)
{
- stb_vorbis *f, p;
- vorbis_init(&p, alloc);
- p.f = file;
- p.f_start = (uint32) 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 *f, p;
+ vorbis_init(&p, alloc);
+ p.f = file;
+ p.f_start = (uint32)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, const stb_vorbis_alloc *alloc)
{
- unsigned int len, start;
- start = (unsigned int) ftell(file);
- fseek(file, 0, SEEK_END);
- len = (unsigned int) (ftell(file) - start);
- fseek(file, start, SEEK_SET);
- return stb_vorbis_open_file_section(file, close_on_free, error, alloc, len);
+ unsigned int len, start;
+ start = (unsigned int)ftell(file);
+ fseek(file, 0, SEEK_END);
+ len = (unsigned int)(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, const 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;
+ 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, const 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);
- if (error) *error = VORBIS__no_error;
- return f;
- }
- }
- if (error) *error = p.error;
- vorbis_deinit(&p);
- return NULL;
+ 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);
+ if (error) *error = VORBIS__no_error;
+ return f;
+ }
+ }
+ if (error) *error = p.error;
+ vorbis_deinit(&p);
+ return NULL;
}
#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
@@ -5003,402 +5069,408 @@ stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len, int *err
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 },
+ { 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()
+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)
+#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;
- }
+ 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;
- }
- }
+#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];
+#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_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];
+ 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;
- }
- }
+ }
+ 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 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;
+ 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 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;
+ 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;
+ 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;
+ 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;
+ 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;
+ 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;
+ 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;
+ 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
- 1.10 - 2017/03/03 - more robust seeking; fix negative ilog(); clear error in open_memory
- 1.09 - 2016/04/04 - back out 'avoid discarding last frame' fix from previous version
- 1.08 - 2016/04/02 - fixed multiple warnings; fix setup memory leaks;
- avoid discarding last frame of audio data
- 1.07 - 2015/01/16 - fixed some warnings, fix mingw, const-correct API
- some more crash fixes when out of memory or with corrupt files
- 1.06 - 2015/08/31 - full, correct support for seeking API (Dougall Johnson)
- some crash fixes when out of memory or with corrupt files
- 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 <malloc.h> 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
+1.12 - 2017-11-21 - limit residue begin/end to blocksize/2 to avoid large temp allocs in bad/corrupt files
+1.11 - 2017-07-23 - fix MinGW compilation
+1.10 - 2017-03-03 - more robust seeking; fix negative ilog(); clear error in open_memory
+1.09 - 2016-04-04 - back out 'avoid discarding last frame' fix from previous version
+1.08 - 2016-04-02 - fixed multiple warnings; fix setup memory leaks;
+avoid discarding last frame of audio data
+1.07 - 2015-01-16 - fixed some warnings, fix mingw, const-correct API
+some more crash fixes when out of memory or with corrupt files
+1.06 - 2015-08-31 - full, correct support for seeking API (Dougall Johnson)
+some crash fixes when out of memory or with corrupt files
+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 <malloc.h> 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
@@ -5410,38 +5482,38 @@ This software is available under 2 licenses -- choose whichever you prefer.
------------------------------------------------------------------------------
ALTERNATIVE A - MIT License
Copyright (c) 2017 Sean Barrett
-Permission is hereby granted, free of charge, to any person obtaining a copy of
-this software and associated documentation files (the "Software"), to deal in
-the Software without restriction, including without limitation the rights to
-use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
-of the Software, and to permit persons to whom the Software is furnished to do
+Permission is hereby granted, free of charge, to any person obtaining a copy of
+this software and associated documentation files (the "Software"), to deal in
+the Software without restriction, including without limitation the rights to
+use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+of the Software, and to permit persons to whom the Software is furnished to do
so, subject to the following conditions:
-The above copyright notice and this permission notice shall be included in all
+The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software.
-THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
-IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
-FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
-AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
-LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
-OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
SOFTWARE.
------------------------------------------------------------------------------
ALTERNATIVE B - Public Domain (www.unlicense.org)
This is free and unencumbered software released into the public domain.
-Anyone is free to copy, modify, publish, use, compile, sell, or distribute this
-software, either in source code form or as a compiled binary, for any purpose,
+Anyone is free to copy, modify, publish, use, compile, sell, or distribute this
+software, either in source code form or as a compiled binary, for any purpose,
commercial or non-commercial, and by any means.
-In jurisdictions that recognize copyright laws, the author or authors of this
-software dedicate any and all copyright interest in the software to the public
-domain. We make this dedication for the benefit of the public at large and to
-the detriment of our heirs and successors. We intend this dedication to be an
-overt act of relinquishment in perpetuity of all present and future rights to
+In jurisdictions that recognize copyright laws, the author or authors of this
+software dedicate any and all copyright interest in the software to the public
+domain. We make this dedication for the benefit of the public at large and to
+the detriment of our heirs and successors. We intend this dedication to be an
+overt act of relinquishment in perpetuity of all present and future rights to
this software under copyright law.
-THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
-IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
-FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
-AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN
-ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN
+ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
------------------------------------------------------------------------------
-*/
+*/ \ No newline at end of file
diff --git a/src/libs/tiny/tinysound.h b/src/libs/tiny/tinysound.h
deleted file mode 100644
index 41d547d..0000000
--- a/src/libs/tiny/tinysound.h
+++ /dev/null
@@ -1,2560 +0,0 @@
-/*
-tinysound.h - v1.07
-
-Summary:
-tinysound is a C API for loading, playing, looping, panning and fading mono
-and stero sounds. This means tinysound imparts no external DLLs or large
-libraries that adversely effect shipping size. tinysound can also run on
-Windows XP since DirectSound ships with all recent versions of Windows.
-tinysound implements a custom SSE2 mixer by explicitly locking and unlocking
-portions of an internal. tinysound uses CoreAudio for Apple machines (like
-OSX and iOS). SDL is used for all other platforms. Define TS_FORCE_SDL
-before placaing the TS_IMPLEMENTATION in order to force the use of SDL.
-
-Revision history:
-1.0 (06/04/2016) initial release
-1.01 (06/06/2016) load WAV from memory
-separate portable and OS-specific code in tsMix
-fixed bug causing audio glitches when sounds ended
-added stb_vorbis loaders + demo example
-1.02 (06/08/2016) error checking + strings in vorbis loaders
-SSE2 implementation of mixer
-fix typos on docs/comments
-corrected volume bug introduced in 1.01
-1.03 (07/05/2016) size calculation helper (to know size of sound in
-bytes on the heap) tsSoundSize
-1.04 (12/06/2016) merged in Aaron Balint's contributions
-SFFT and pitch functions from Stephan M. Bernsee
-tsMix can run on its own thread with tsSpawnMixThread
-updated documentation, typo fixes
-fixed typo in malloc16 that caused heap corruption
-1.05 (12/08/2016) tsStopAllSounds, suggested by Aaron Balint
-1.06 (02/17/2017) port to CoreAudio for Apple machines
-1.07 (06/18/2017) SIMD the pitch shift code; swapped out old Bernsee
-code for a new re-write, updated docs as necessary,
-support for compiling as .c and .cpp on Windows,
-port for SDL (for Linux, or any other platform).
-Special thanks to DexP of github for 90% of the work
-on the SDL port!
-*/
-
-/*
-Contributors:
-Aaron Balint 1.04 - real time pitch
-1.04 - separate thread for tsMix
-1.04 - bugfix, removed extra free16 call for second channel
-DeXP 1.07 - initial work on SDL port
-*/
-
-/*
-To create implementation (the function definitions)
-#define TS_IMPLEMENTATION
-in *one* C/CPP file (translation unit) that includes this file
-
-DOCUMENTATION (very quick intro):
-1. create context
-2. load sounds from disk into memory
-3. play sounds
-4. free context
-
-1. tsContext* ctx = tsMakeContext( hwnd, frequency, latency, seconds, N );
-2. tsPlaySoundDef def = tsMakeDef( &tsLoadWAV( "path_to_file/filename.wav" ) );
-3. tsPlaySound( ctx, def );
-4. tsShutdownContext( ctx );
-
-DOCUMENTATION (longer introduction):
-tinysound consists of tsLoadedSounds, tsPlayingSounds and the tsContext.
-The tsContext encapsulates an OS sound API, as well as buffers + settings.
-tsLoadedSound holds raw samples of a sound. tsPlayingSound is an instance
-of a tsLoadedSound that represents a sound that can be played through the
-tsContext.
-
-There are two main versions of the API, the low-level and the high-level
-API. The low-level API does not manage any memory for tsPlayingSounds. The
-high level api holds a memory pool of playing sounds.
-
-High-level API:
-First create a context and pass in non-zero to the final parameter. This
-final parameter controls how large of a memory pool to use for tsPlayingSounds.
-Here's an example where N is the size of the internal pool:
-
-tsContext* ctx = tsMakeContext( hwnd, frequency, latency, seconds, N );
-
-We create tsPlayingSounds indirectly with tsPlayDef structs. tsPlayDef is a
-POD struct so feel free to make them straight on the stack. The tsPlayDef
-sets up initialization parameters. Here's an example to load a wav and
-play it:
-
-tsLoadedSound loaded = tsLoadWAV( "path_to_file/filename.wav" );
-tsPlaySoundDef def = tsMakeDef( &loaded );
-tsPlayingSound* sound = tsPlaySound( ctx, def );
-
-The same def can be used to play as many sounds as desired (even simultaneously)
-as long as the context playing sound pool is large enough.
-
-Low-level API:
-First create a context and pass 0 in the final parameter (0 here means
-the context will *not* allocate a tsPlayingSound memory pool):
-
-tsContext* ctx = tsMakeContext( hwnd, frequency, latency, seconds, 0 );
-
-parameters:
-hwnd -- HWND, handle to window (on OSX just pass in 0)
-frequency -- int, represents Hz frequency rate in which samples are played
-latency -- int, estimated latency in Hz from PlaySound call to speaker output
-seconds -- int, number of second of samples internal buffers can hold
-0 (last param) -- int, number of elements in tsPlayingSound pool
-
-We create a tsPlayingSound like so:
-tsLoadedSound loaded = tsLoadWAV( "path_to_file/filename.wav" );
-tsPlayingSound playing_sound = tsMakePlayingSound( &loaded );
-
-Then to play the sound we do:
-tsInsertSound( ctx, &playing_sound );
-
-The above tsInsertSound function call will place playing_sound into
-a singly-linked list inside the context. The context will remove
-the sound from its internal list when it finishes playing.
-
-WARNING: The high-level API cannot be mixed with the low-level API. If you
-try then the internal code will assert and crash. Pick one and stick with it.
-Usually he high-level API will be used, but if someone is *really* picky about
-their memory usage, or wants more control, the low-level API can be used.
-
-Here is the Low-Level API:
-tsPlayingSound tsMakePlayingSound( tsLoadedSound* loaded );
-void tsInsertSound( tsContext* ctx, tsPlayingSound* sound );
-
-Here is the High-Level API:
-tsPlayingSound* tsPlaySound( tsContext* ctx, tsPlaySoundDef def );
-tsPlaySoundDef tsMakeDef( tsLoadedSound* sound );
-void tsStopAllSounds( tsContext( ctx );
-
-Be sure to link against dsound.dll (or dsound.lib) on Windows.
-
-Read the rest of the header for specific details on all available functions
-and struct types.
-*/
-
-/*
-Known Limitations:
-
-* PCM mono/stereo format is the only formats the LoadWAV function supports. I don't
-guarantee it will work for all kinds of wav files, but it certainly does for the common
-kind (and can be changed fairly easily if someone wanted to extend it).
-* Only supports 16 bits per sample.
-* Mixer does not do any fancy clipping. The algorithm is to convert all 16 bit samples
-to float, mix all samples, and write back to audio API as 16 bit integers. In
-practice this works very well and clipping is not often a big problem.
-* I'm not super familiar with good ways to avoid the DirectSound play cursor from going
-past the write cursor. To mitigate this pass in a larger number to tsMakeContext's 4th
-parameter (buffer scale in seconds).
-* Pitch shifting code is pretty darn expensive. This is due to the use of a Fast Fourier Transform
-routine. The pitch shifting itself is written in rather efficient SIMD using SSE2 intrinsics,
-but the FFT routine is very basic. FFT is a big bottleneck for pitch shifting. There is a
-TODO optimization listed in this file for the FFT routine, but it's fairly low priority;
-optimizing FFT routines is difficult and requires a lot of specialized knowledge.
-*/
-
-/*
-FAQ
-Q : Why DirectSound instead of (insert API here) on Windows?
-A : Casey Muratori documented DS on Handmade Hero, other APIs do not have such good docs. DS has
-shipped on Windows XP all the way through Windows 10 -- using this header effectively intro-
-duces zero dependencies for the foreseeable future. The DS API itself is sane enough to quickly
-implement needed features, and users won't hear the difference between various APIs. Latency is
-not that great with DS but it is shippable. Additionally, many other APIs will in the end speak
-to Windows through the DS API.
-
-Q : Why not include Linux support?
-A : There have been a couple requests for ALSA support on Linux. For now the only option is to use
-SDL backend, which can indirectly support ALSA. SDL is used only in a very low-level manner;
-to get sound samples to the sound card via callback, so there shouldn't be much in the way of
-considering SDL a good option for "name your flavor" of Linux backend.
-
-Q : I would like to use my own memory management, how can I achieve this?
-A : This header makes a couple uses of malloc/free, and malloc16/free16. Simply find these bits
-and replace them with your own memory allocation routines. They can be wrapped up into a macro,
-or call your own functions directly -- it's up to you. Generally these functions allocate fairly
-large chunks of memory, and not very often (if at all), with one exception: tsSetPitch is a very
-expensive routine and requires frequent dynamic memory management.
-*/
-
-/*
-Some past discussion threads:
-https://www.reddit.com/r/gamedev/comments/6i39j2/tinysound_the_cutest_library_to_get_audio_into/
-https://www.reddit.com/r/gamedev/comments/4ml6l9/tinysound_singlefile_c_audio_library/
-https://forums.tigsource.com/index.php?topic=58706.0
-*/
-
-#if !defined( TINYSOUND_H )
-
-#define TS_WINDOWS 1
-#define TS_MAC 2
-#define TS_UNIX 3
-#define TS_SDL 4
-
-#if defined( _WIN32 )
-#define TS_PLATFORM TS_WINDOWS
-#elif defined( __APPLE__ )
-#define TS_PLATFORM TS_MAC
-#else
-#define TS_PLATFORM TS_SDL
-
-// please note TS_UNIX is not directly support
-// instead, unix-style OSes are encouraged to use SDL
-// see: https://www.libsdl.org/
-
-#endif
-
-// Use TS_FORCE_SDL to override the above macros and use
-// the SDL port.
-#ifdef TS_FORCE_SDL
-
-#undef TS_PLATFORM
-#define TS_PLATFORM TS_SDL
-
-#endif
-
-#include <stdint.h>
-
-// read this in the event of tsLoadWAV/tsLoadOGG errors
-// also read this in the event of certain errors from tsMakeContext
-extern const char* g_tsErrorReason;
-
-// stores a loaded sound in memory
-typedef struct
-{
- int sample_count;
- int channel_count;
- void* channels[2];
-} tsLoadedSound;
-
-struct tsPitchData;
-typedef struct tsPitchData tsPitchData;
-
-// represents an instance of a tsLoadedSound, can be played through the tsContext
-typedef struct tsPlayingSound
-{
- int active;
- int paused;
- int looped;
- float volume0;
- float volume1;
- float pan0;
- float pan1;
- float pitch;
- tsPitchData* pitch_filter[2];
- int sample_index;
- tsLoadedSound* loaded_sound;
- struct tsPlayingSound* next;
-} tsPlayingSound;
-
-// holds audio API info and other info
-struct tsContext;
-typedef struct tsContext tsContext;
-
-// The returned struct will contain a null pointer in tsLoadedSound::channel[ 0 ]
-// in the case of errors. Read g_tsErrorReason string for details on what happened.
-// Calls tsReadMemWAV internally.
-tsLoadedSound tsLoadWAV(const char* path);
-
-// Reads a WAV file from memory. Still allocates memory for the tsLoadedSound since
-// WAV format will interlace stereo, and we need separate data streams to do SIMD
-// properly.
-void tsReadMemWAV(const void* memory, tsLoadedSound* sound);
-
-// If stb_vorbis was included *before* tinysound go ahead and create
-// some functions for dealing with OGG files.
-#ifdef STB_VORBIS_INCLUDE_STB_VORBIS_H
-void tsReadMemOGG(const void* memory, int length, int* sample_rate, tsLoadedSound* sound);
-tsLoadedSound tsLoadOGG(const char* path, int* sample_rate);
-#endif
-
-// Uses free16 (aligned free, implemented later in this file) to free up both of
-// the channels stored within sound
-void tsFreeSound(tsLoadedSound* sound);
-
-// Returns the size, in bytes, of all heap-allocated memory for this particular
-// loaded sound
-int tsSoundSize(tsLoadedSound* sound);
-
-// playing_pool_count -- 0 to setup low-level API, non-zero to size the internal
-// memory pool for tsPlayingSound instances
-tsContext* tsMakeContext(void* hwnd, unsigned play_frequency_in_Hz, int latency_factor_in_Hz, int num_buffered_seconds, int playing_pool_count);
-void tsShutdownContext(tsContext* ctx);
-
-// Call tsSpawnMixThread once to setup a separate thread for the context to run
-// upon. The separate thread will continually call tsMix and perform mixing
-// operations.
-void tsSpawnMixThread(tsContext* ctx);
-
-// Use tsThreadSleepDelay to specify a custom sleep delay time.
-// A sleep will occur after each call to tsMix. By default YieldProcessor
-// is used, and no sleep occurs. Use a sleep delay to conserve CPU bandwidth.
-// A recommended sleep time is a little less than 1/2 your predicted 1/FPS.
-// 60 fps is 16 ms, so about 1-5 should work well in most cases.
-void tsThreadSleepDelay(tsContext* ctx, int milliseconds);
-
-// Call this manually, once per game tick recommended, if you haven't ever
-// called tsSpawnMixThread. Otherwise the thread will call tsMix itself.
-// num_samples_to_write is not used on Windows. On Mac it is used to push
-// samples into a circular buffer while CoreAudio simultaneously pulls samples
-// off of the buffer. num_samples_to_write should be computed each update tick
-// as delta_time * play_frequency_in_Hz + 1.
-void tsMix(tsContext* ctx);
-
-// All of the functions in this next section should only be called if tsIsActive
-// returns true. Calling them otherwise probably won't do anything bad, but it
-// won't do anything at all. If a sound is active it resides in the context's
-// internal list of playing sounds.
-int tsIsActive(tsPlayingSound* sound);
-
-// Flags sound for removal. Upon next tsMix call will remove sound from playing
-// list. If high-level API used sound is placed onto the internal free list.
-void tsStopSound(tsPlayingSound* sound);
-
-void tsLoopSound(tsPlayingSound* sound, int zero_for_no_loop);
-void tsPauseSound(tsPlayingSound* sound, int one_for_paused);
-
-// lerp from 0 to 1, 0 full left, 1 full right
-void tsSetPan(tsPlayingSound* sound, float pan);
-
-// explicitly set volume of each channel. Can be used as panning (but it's
-// recommended to use the tsSetPan function for panning).
-void tsSetVolume(tsPlayingSound* sound, float volume_left, float volume_right);
-
-// Change pitch (not duration) of sound. pitch = 0.5f for one octave lower, pitch = 2.0f for one octave higher.
-// pitch at 1.0f applies no change. pitch settings farther away from 1.0f create more distortion and lower
-// the output sample quality. pitch can be adjusted in real-time for doppler effects and the like. Going beyond
-// 0.5f and 2.0f may require some tweaking the pitch shifting parameters, and is not recommended.
-
-// Additional important information about performance: This function
-// is quite expensive -- you have been warned! Try it out and be aware of how much CPU consumption it uses.
-// To avoid destroying the originally loaded sound samples, tsSetPitch will do a one-time allocation to copy
-// sound samples into a new buffer. The new buffer contains the pitch adjusted samples, and these will be played
-// through tsMix. This lets the pitch be modulated at run-time, but requires dynamically allocated memory. The
-// memory is freed once the sound finishes playing. If a one-time pitch adjustment is desired, for performance
-// reasons please consider doing an off-line pitch adjustment manually as a pre-processing step for your sounds.
-// Also, consider changing malloc16 and free16 to match your custom memory allocation needs. Try adjusting
-// TS_PITCH_QUALITY (must be a power of two) and see how this affects your performance.
-void tsSetPitch(tsPlayingSound* sound, float pitch);
-
-// Delays sound before actually playing it. Requires context to be passed in
-// since there's a conversion from seconds to samples per second.
-// If one were so inclined another version could be implemented like:
-// void tsSetDelay( tsPlayingSound* sound, float delay, int samples_per_second )
-void tsSetDelay(tsContext* ctx, tsPlayingSound* sound, float delay_in_seconds);
-
-// Portable sleep function
-void tsSleep(int milliseconds);
-
-// LOW-LEVEL API
-tsPlayingSound tsMakePlayingSound(tsLoadedSound* loaded);
-void tsInsertSound(tsContext* ctx, tsPlayingSound* sound);
-
-// HIGH-LEVEL API
-typedef struct
-{
- int paused;
- int looped;
- float volume_left;
- float volume_right;
- float pan;
- float pitch;
- float delay;
- tsLoadedSound* loaded;
-} tsPlaySoundDef;
-
-tsPlayingSound* tsPlaySound(tsContext* ctx, tsPlaySoundDef def);
-tsPlaySoundDef tsMakeDef(tsLoadedSound* sound);
-void tsStopAllSounds(tsContext* ctx);
-
-#define TINYSOUND_H
-#endif
-
-#ifdef TS_IMPLEMENTATION
-
-#define _CRT_SECURE_NO_WARNINGS FUCK_YOU
-#include <stdlib.h> // malloc, free
-#include <stdio.h> // fopen, fclose
-#include <string.h> // memcmp, memset, memcpy
-#include <xmmintrin.h>
-#include <emmintrin.h>
-
-#if TS_PLATFORM == TS_WINDOWS
-
-#include <dsound.h>
-#undef PlaySound
-
-#if defined( _MSC_VER )
-#pragma comment( lib, "dsound.lib" )
-#endif
-
-#elif TS_PLATFORM == TS_MAC
-
-#include <CoreAudio/CoreAudio.h>
-#include <AudioUnit/AudioUnit.h>
-#include <pthread.h>
-#include <mach/mach_time.h>
-
-#else
-
-#include "SDL2/SDL.h"
-
-#endif
-
-#define TS_CHECK( X, Y ) do { if ( !(X) ) { g_tsErrorReason = Y; goto ts_err; } } while ( 0 )
-#if TS_PLATFORM == TS_MAC && defined( __clang__ )
-#define TS_ASSERT_INTERNAL __builtin_trap( )
-#else
-#define TS_ASSERT_INTERNAL *(int*)0 = 0
-#endif
-#define TS_ASSERT( X ) do { if ( !(X) ) TS_ASSERT_INTERNAL; } while ( 0 )
-#define TS_ALIGN( X, Y ) ((((size_t)X) + ((Y) - 1)) & ~((Y) - 1))
-#define TS_TRUNC( X, Y ) ((size_t)(X) & ~((Y) - 1))
-
-const char* g_tsErrorReason;
-
-static void* tsReadFileToMemory(const char* path, int* size)
-{
- void* data = 0;
- FILE* fp = fopen(path, "rb");
- int sizeNum = 0;
-
- if (fp)
- {
- fseek(fp, 0, SEEK_END);
- sizeNum = (int)ftell(fp);
- fseek(fp, 0, SEEK_SET);
- data = malloc(sizeNum);
- fread(data, sizeNum, 1, fp);
- fclose(fp);
- }
-
- if (size) *size = sizeNum;
- return data;
-}
-
-static int tsFourCC(const char* CC, void* memory)
-{
- if (!memcmp(CC, memory, 4)) return 1;
- return 0;
-}
-
-static char* tsNext(char* data)
-{
- uint32_t size = *(uint32_t*)(data + 4);
- size = (size + 1) & ~1;
- return data + 8 + size;
-}
-
-static void* malloc16(size_t size)
-{
- void* p = malloc(size + 16);
- if (!p) return 0;
- unsigned char offset = (size_t)p & 15;
- p = (void*)TS_ALIGN(p + 1, 16);
- *((char*)p - 1) = 16 - offset;
- TS_ASSERT(!((size_t)p & 15));
- return p;
-}
-
-static void free16(void* p)
-{
- if (!p) return;
- free((char*)p - (size_t)*((char*)p - 1));
-}
-
-static void tsLastElement(__m128* a, int i, int j, int16_t* samples, int offset)
-{
- switch (offset)
- {
- case 1:
- a[i] = _mm_set_ps(samples[j], 0.0f, 0.0f, 0.0f);
- break;
-
- case 2:
- a[i] = _mm_set_ps(samples[j], samples[j + 1], 0.0f, 0.0f);
- break;
-
- case 3:
- a[i] = _mm_set_ps(samples[j], samples[j + 1], samples[j + 2], 0.0f);
- break;
-
- case 0:
- a[i] = _mm_set_ps(samples[j], samples[j + 1], samples[j + 2], samples[j + 3]);
- break;
- }
-}
-
-void tsReadMemWAV(const void* memory, tsLoadedSound* sound)
-{
-#pragma pack( push, 1 )
- typedef struct
- {
- uint16_t wFormatTag;
- uint16_t nChannels;
- uint32_t nSamplesPerSec;
- uint32_t nAvgBytesPerSec;
- uint16_t nBlockAlign;
- uint16_t wBitsPerSample;
- uint16_t cbSize;
- uint16_t wValidBitsPerSample;
- uint32_t dwChannelMask;
- uint8_t SubFormat[18];
- } Fmt;
-#pragma pack( pop )
-
- char* data = (char*)memory;
- TS_CHECK(data, "Unable to read input file (file doesn't exist, or could not allocate heap memory.");
- TS_CHECK(tsFourCC("RIFF", data), "Incorrect file header; is this a WAV file?");
- TS_CHECK(tsFourCC("WAVE", data + 8), "Incorrect file header; is this a WAV file?");
-
- data += 12;
-
- TS_CHECK(tsFourCC("fmt ", data), "fmt chunk not found.");
- Fmt fmt;
- fmt = *(Fmt*)(data + 8);
- TS_CHECK(fmt.wFormatTag == 1, "Only PCM WAV files are supported.");
- TS_CHECK(fmt.nChannels == 1 || fmt.nChannels == 2, "Only mono or stereo supported (too many channels detected).");
- TS_CHECK(fmt.wBitsPerSample == 16, "Only 16 bits per sample supported.");
- TS_CHECK(fmt.nBlockAlign == fmt.nChannels * 2, "implementation error");
-
- data = tsNext(data);
- TS_CHECK(tsFourCC("data", data), "data chunk not found.");
- int sample_size = *((uint32_t*)(data + 4));
- int sample_count = sample_size / (fmt.nChannels * sizeof(uint16_t));
- sound->sample_count = sample_count;
- sound->channel_count = fmt.nChannels;
-
- int wide_count = (int)TS_ALIGN(sample_count, 4);
- wide_count /= 4;
- int wide_offset = sample_count & 3;
- int16_t* samples = (int16_t*)(data + 8);
- float* sample = (float*)alloca(sizeof(float) * 4 + 16);
- sample = (float*)TS_ALIGN(sample, 16);
-
- switch (sound->channel_count)
- {
- case 1:
- {
- sound->channels[0] = malloc16(wide_count * sizeof(__m128));
- sound->channels[1] = 0;
- __m128* a = (__m128*)sound->channels[0];
-
- for (int i = 0, j = 0; i < wide_count - 1; ++i, j += 4)
- {
- sample[0] = (float)samples[j];
- sample[1] = (float)samples[j + 1];
- sample[2] = (float)samples[j + 2];
- sample[3] = (float)samples[j + 3];
- a[i] = _mm_load_ps(sample);
- }
-
- tsLastElement(a, wide_count - 1, (wide_count - 1) * 4, samples, wide_offset);
- } break;
-
- case 2:
- {
- __m128* a = (__m128*)malloc16(wide_count * sizeof(__m128) * 2);
- __m128* b = a + wide_count;
-
- for (int i = 0, j = 0; i < wide_count - 1; ++i, j += 8)
- {
- sample[0] = (float)samples[j];
- sample[1] = (float)samples[j + 2];
- sample[2] = (float)samples[j + 4];
- sample[3] = (float)samples[j + 6];
- a[i] = _mm_load_ps(sample);
-
- sample[0] = (float)samples[j + 1];
- sample[1] = (float)samples[j + 3];
- sample[2] = (float)samples[j + 5];
- sample[3] = (float)samples[j + 7];
- b[i] = _mm_load_ps(sample);
- }
-
- tsLastElement(a, wide_count - 1, (wide_count - 1) * 4, samples, wide_offset);
- tsLastElement(b, wide_count - 1, (wide_count - 1) * 4 + 4, samples, wide_offset);
- sound->channels[0] = a;
- sound->channels[1] = b;
- } break;
-
- default:
- TS_CHECK(0, "unsupported channel count (only support mono and stereo).");
- }
-
- return;
-
-ts_err:
- memset(&sound, 0, sizeof(sound));
-}
-
-tsLoadedSound tsLoadWAV(const char* path)
-{
- tsLoadedSound sound = { 0 };
- char* wav = (char*)tsReadFileToMemory(path, 0);
- tsReadMemWAV(wav, &sound);
- free(wav);
- return sound;
-}
-
-// If stb_vorbis was included *before* tinysound go ahead and create
-// some functions for dealing with OGG files.
-#ifdef STB_VORBIS_INCLUDE_STB_VORBIS_H
-void tsReadMemOGG(const void* memory, int length, int* sample_rate, tsLoadedSound* sound)
-{
- int16_t* samples = 0;
- int channel_count;
- int sample_count = stb_vorbis_decode_memory((const unsigned char*)memory, length, &channel_count, sample_rate, &samples);
-
- TS_CHECK(sample_count > 0, "stb_vorbis_decode_memory failed. Make sure your file exists and is a valid OGG file.");
-
- int wide_count = (int)TS_ALIGN(sample_count, 4) / 4;
- int wide_offset = sample_count & 3;
- float* sample = (float*)alloca(sizeof(float) * 4 + 16);
- sample = (float*)TS_ALIGN(sample, 16);
- __m128* a;
- __m128* b;
-
- switch (channel_count)
- {
- case 1:
- {
- a = (__m128*)malloc16(wide_count * sizeof(__m128));
- b = 0;
-
- for (int i = 0, j = 0; i < wide_count - 1; ++i, j += 4)
- {
- sample[0] = (float)samples[j];
- sample[1] = (float)samples[j + 1];
- sample[2] = (float)samples[j + 2];
- sample[3] = (float)samples[j + 3];
- a[i] = _mm_load_ps(sample);
- }
-
- tsLastElement(a, wide_count - 1, (wide_count - 1) * 4, samples, wide_offset);
- } break;
-
- case 2:
- a = (__m128*)malloc16(wide_count * sizeof(__m128) * 2);
- b = a + wide_count;
-
- for (int i = 0, j = 0; i < wide_count - 1; ++i, j += 8)
- {
- sample[0] = (float)samples[j];
- sample[1] = (float)samples[j + 2];
- sample[2] = (float)samples[j + 4];
- sample[3] = (float)samples[j + 6];
- a[i] = _mm_load_ps(sample);
-
- sample[0] = (float)samples[j + 1];
- sample[1] = (float)samples[j + 3];
- sample[2] = (float)samples[j + 5];
- sample[3] = (float)samples[j + 7];
- b[i] = _mm_load_ps(sample);
- }
-
- tsLastElement(a, wide_count - 1, (wide_count - 1) * 4, samples, wide_offset);
- tsLastElement(b, wide_count - 1, (wide_count - 1) * 4 + 4, samples, wide_offset);
- break;
-
- default:
- TS_CHECK(0, "Unsupported channel count.");
- }
-
- sound->sample_count = sample_count;
- sound->channel_count = channel_count;
- sound->channels[0] = a;
- sound->channels[1] = b;
- free(samples);
- return;
-
-ts_err:
- free(samples);
- memset(sound, 0, sizeof(tsLoadedSound));
-}
-
-tsLoadedSound tsLoadOGG(const char* path, int* sample_rate)
-{
- int length;
- void* memory = tsReadFileToMemory(path, &length);
- tsLoadedSound sound;
- tsReadMemOGG(memory, length, sample_rate, &sound);
- free(memory);
-
- return sound;
-}
-#endif
-
-void tsFreeSound(tsLoadedSound* sound)
-{
- free16(sound->channels[0]);
- memset(sound, 0, sizeof(tsLoadedSound));
-}
-
-int tsSoundSize(tsLoadedSound* sound)
-{
- return sound->sample_count * sound->channel_count * sizeof(uint16_t);
-}
-
-tsPlayingSound tsMakePlayingSound(tsLoadedSound* loaded)
-{
- tsPlayingSound playing;
- playing.active = 0;
- playing.paused = 0;
- playing.looped = 0;
- playing.volume0 = 1.0f;
- playing.volume1 = 1.0f;
- playing.pan0 = 0.5f;
- playing.pan1 = 0.5f;
- playing.pitch = 1.0f;
- playing.pitch_filter[0] = 0;
- playing.pitch_filter[1] = 0;
- playing.sample_index = 0;
- playing.loaded_sound = loaded;
- playing.next = 0;
- return playing;
-}
-
-int tsIsActive(tsPlayingSound* sound)
-{
- return sound->active;
-}
-
-void tsStopSound(tsPlayingSound* sound)
-{
- sound->active = 0;
-}
-
-void tsLoopSound(tsPlayingSound* sound, int zero_for_no_loop)
-{
- sound->looped = zero_for_no_loop;
-}
-
-void tsPauseSound(tsPlayingSound* sound, int one_for_paused)
-{
- sound->paused = one_for_paused;
-}
-
-void tsSetPan(tsPlayingSound* sound, float pan)
-{
- if (pan > 1.0f) pan = 1.0f;
- else if (pan < 0.0f) pan = 0.0f;
- float left = 1.0f - pan;
- float right = pan;
- sound->pan0 = left;
- sound->pan1 = right;
-}
-
-void tsSetPitch(tsPlayingSound* sound, float pitch)
-{
- sound->pitch = pitch;
-}
-
-void tsSetVolume(tsPlayingSound* sound, float volume_left, float volume_right)
-{
- if (volume_left < 0.0f) volume_left = 0.0f;
- if (volume_right < 0.0f) volume_right = 0.0f;
- sound->volume0 = volume_left;
- sound->volume1 = volume_right;
-}
-
-static void tsRemoveFilter(tsPlayingSound* playing);
-
-#if TS_PLATFORM == TS_WINDOWS
-
-void tsSleep(int milliseconds)
-{
- Sleep(milliseconds);
-}
-
-struct tsContext
-{
- unsigned latency_samples;
- unsigned running_index;
- int Hz;
- int bps;
- int buffer_size;
- int wide_count;
- tsPlayingSound* playing;
- __m128* floatA;
- __m128* floatB;
- __m128i* samples;
- tsPlayingSound* playing_pool;
- tsPlayingSound* playing_free;
-
- // platform specific stuff
- LPDIRECTSOUND dsound;
- LPDIRECTSOUNDBUFFER buffer;
- LPDIRECTSOUNDBUFFER primary;
-
- // data for tsMix thread, enable these with tsSpawnMixThread
- CRITICAL_SECTION critical_section;
- int separate_thread;
- int running;
- int sleep_milliseconds;
-};
-
-static void tsReleaseContext(tsContext* ctx)
-{
- if (ctx->separate_thread) DeleteCriticalSection(&ctx->critical_section);
-#ifdef __cplusplus
- ctx->buffer->Release();
- ctx->primary->Release();
- ctx->dsound->Release();
-#else
- ctx->buffer->lpVtbl->Release(ctx->buffer);
- ctx->primary->lpVtbl->Release(ctx->primary);
- ctx->dsound->lpVtbl->Release(ctx->dsound);
-#endif
- tsPlayingSound* playing = ctx->playing;
- while (playing)
- {
- tsRemoveFilter(playing);
- playing = playing->next;
- }
- free(ctx);
-}
-
-static DWORD WINAPI tsCtxThread(LPVOID lpParameter)
-{
- tsContext* ctx = (tsContext*)lpParameter;
-
- while (ctx->running)
- {
- tsMix(ctx);
- if (ctx->sleep_milliseconds) tsSleep(ctx->sleep_milliseconds);
- else YieldProcessor();
- }
-
- ctx->separate_thread = 0;
- return 0;
-}
-
-static void tsLock(tsContext* ctx)
-{
- if (ctx->separate_thread) EnterCriticalSection(&ctx->critical_section);
-}
-
-static void tsUnlock(tsContext* ctx)
-{
- if (ctx->separate_thread) LeaveCriticalSection(&ctx->critical_section);
-}
-
-tsContext* tsMakeContext(void* hwnd, unsigned play_frequency_in_Hz, int latency_factor_in_Hz, int num_buffered_seconds, int playing_pool_count)
-{
- int bps = sizeof(INT16) * 2;
- int buffer_size = play_frequency_in_Hz * bps * num_buffered_seconds;
- tsContext* ctx = 0;
- WAVEFORMATEX format = { 0 };
- DSBUFFERDESC bufdesc = { 0 };
- LPDIRECTSOUND dsound;
-
- TS_CHECK(hwnd, "Invalid hwnd passed to tsMakeContext.");
-
- HRESULT res = DirectSoundCreate(0, &dsound, 0);
- TS_CHECK(res == DS_OK, "DirectSoundCreate failed");
-#ifdef __cplusplus
- dsound->SetCooperativeLevel((HWND)hwnd, DSSCL_PRIORITY);
-#else
- dsound->lpVtbl->SetCooperativeLevel(dsound, (HWND)hwnd, DSSCL_PRIORITY);
-#endif
- bufdesc.dwSize = sizeof(bufdesc);
- bufdesc.dwFlags = DSBCAPS_PRIMARYBUFFER;
-
- LPDIRECTSOUNDBUFFER primary_buffer;
-#ifdef __cplusplus
- res = dsound->CreateSoundBuffer(&bufdesc, &primary_buffer, 0);
-#else
- res = dsound->lpVtbl->CreateSoundBuffer(dsound, &bufdesc, &primary_buffer, 0);
-#endif
- TS_CHECK(res == DS_OK, "Failed to create primary sound buffer");
-
- format.wFormatTag = WAVE_FORMAT_PCM;
- format.nChannels = 2;
- format.nSamplesPerSec = play_frequency_in_Hz;
- format.wBitsPerSample = 16;
- format.nBlockAlign = (format.nChannels * format.wBitsPerSample) / 8;
- format.nAvgBytesPerSec = format.nSamplesPerSec * format.nBlockAlign;
- format.cbSize = 0;
-#ifdef __cplusplus
- res = primary_buffer->SetFormat(&format);
-#else
- res = primary_buffer->lpVtbl->SetFormat(primary_buffer, &format);
-#endif
- TS_CHECK(res == DS_OK, "Failed to set format on primary buffer");
-
- LPDIRECTSOUNDBUFFER secondary_buffer;
- bufdesc.dwSize = sizeof(bufdesc);
- bufdesc.dwFlags = 0;
- bufdesc.dwBufferBytes = buffer_size;
- bufdesc.lpwfxFormat = &format;
-#ifdef __cplusplus
- res = dsound->CreateSoundBuffer(&bufdesc, &secondary_buffer, 0);
-#else
- res = dsound->lpVtbl->CreateSoundBuffer(dsound, &bufdesc, &secondary_buffer, 0);
-#endif
- TS_CHECK(res == DS_OK, "Failed to set format on secondary buffer");
-
- int sample_count = play_frequency_in_Hz * num_buffered_seconds;
- int wide_count = (int)TS_ALIGN(sample_count, 4);
- int pool_size = playing_pool_count * sizeof(tsPlayingSound);
- int mix_buffers_size = sizeof(__m128) * wide_count * 2;
- int sample_buffer_size = sizeof(__m128i) * wide_count;
- ctx = (tsContext*)malloc(sizeof(tsContext) + mix_buffers_size + sample_buffer_size + 16 + pool_size);
- ctx->latency_samples = (unsigned)TS_ALIGN(play_frequency_in_Hz / latency_factor_in_Hz, 4);
- ctx->running_index = 0;
- ctx->Hz = play_frequency_in_Hz;
- ctx->bps = bps;
- ctx->buffer_size = buffer_size;
- ctx->wide_count = wide_count;
- ctx->dsound = dsound;
- ctx->buffer = secondary_buffer;
- ctx->primary = primary_buffer;
- ctx->playing = 0;
- ctx->floatA = (__m128*)(ctx + 1);
- ctx->floatA = (__m128*)TS_ALIGN(ctx->floatA, 16);
- TS_ASSERT(!((size_t)ctx->floatA & 15));
- ctx->floatB = ctx->floatA + wide_count;
- ctx->samples = (__m128i*)ctx->floatB + wide_count;
- ctx->running = 1;
- ctx->separate_thread = 0;
- ctx->sleep_milliseconds = 0;
-
- if (playing_pool_count)
- {
- ctx->playing_pool = (tsPlayingSound*)(ctx->samples + wide_count);
- for (int i = 0; i < playing_pool_count - 1; ++i)
- ctx->playing_pool[i].next = ctx->playing_pool + i + 1;
- ctx->playing_pool[playing_pool_count - 1].next = 0;
- ctx->playing_free = ctx->playing_pool;
- }
-
- else
- {
- ctx->playing_pool = 0;
- ctx->playing_free = 0;
- }
-
- return ctx;
-
-ts_err:
- free(ctx);
- return 0;
-}
-
-void tsSpawnMixThread(tsContext* ctx)
-{
- if (ctx->separate_thread) return;
- InitializeCriticalSectionAndSpinCount(&ctx->critical_section, 0x00000400);
- ctx->separate_thread = 1;
- CreateThread(0, 0, tsCtxThread, ctx, 0, 0);
-}
-
-#elif TS_PLATFORM == TS_MAC
-
-void tsSleep(int milliseconds)
-{
- usleep(milliseconds * 1000);
-}
-
-struct tsContext
-{
- unsigned latency_samples;
- unsigned index0; // read
- unsigned index1; // write
- int Hz;
- int bps;
- int wide_count;
- int sample_count;
- tsPlayingSound* playing;
- __m128* floatA;
- __m128* floatB;
- __m128i* samples;
- tsPlayingSound* playing_pool;
- tsPlayingSound* playing_free;
-
- // platform specific stuff
- AudioComponentInstance inst;
-
- // data for tsMix thread, enable these with tsSpawnMixThread
- pthread_t thread;
- pthread_mutex_t mutex;
- int separate_thread;
- int running;
- int sleep_milliseconds;
-};
-
-static void tsReleaseContext(tsContext* ctx)
-{
- if (ctx->separate_thread) pthread_mutex_destroy(&ctx->mutex);
- AudioOutputUnitStop(ctx->inst);
- AudioUnitUninitialize(ctx->inst);
- AudioComponentInstanceDispose(ctx->inst);
- tsPlayingSound* playing = ctx->playing;
- while (playing)
- {
- tsRemoveFilter(playing);
- playing = playing->next;
- }
- free(ctx);
-}
-
-static void* tsCtxThread(void* udata)
-{
- tsContext* ctx = (tsContext*)udata;
-
- while (ctx->running)
- {
- tsMix(ctx);
- if (ctx->sleep_milliseconds) tsSleep(ctx->sleep_milliseconds);
- else pthread_yield_np();
- }
-
- ctx->separate_thread = 0;
- pthread_exit(0);
- return 0;
-}
-
-static void tsLock(tsContext* ctx)
-{
- if (ctx->separate_thread) pthread_mutex_lock(&ctx->mutex);
-}
-
-static void tsUnlock(tsContext* ctx)
-{
- if (ctx->separate_thread) pthread_mutex_unlock(&ctx->mutex);
-}
-
-static OSStatus tsMemcpyToCA(void* udata, AudioUnitRenderActionFlags* ioActionFlags, const AudioTimeStamp* inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList* ioData);
-
-tsContext* tsMakeContext(void* unused, unsigned play_frequency_in_Hz, int latency_factor_in_Hz, int num_buffered_seconds, int playing_pool_count)
-{
- int bps = sizeof(uint16_t) * 2;
-
- AudioComponentDescription comp_desc = { 0 };
- comp_desc.componentType = kAudioUnitType_Output;
- comp_desc.componentSubType = kAudioUnitSubType_DefaultOutput;
- comp_desc.componentFlags = 0;
- comp_desc.componentFlagsMask = 0;
- comp_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
-
- AudioComponent comp = AudioComponentFindNext(NULL, &comp_desc);
- if (!comp)
- {
- g_tsErrorReason = "Failed to create output unit from AudioComponentFindNext.";
- return 0;
- }
-
- AudioStreamBasicDescription stream_desc = { 0 };
- stream_desc.mSampleRate = (double)play_frequency_in_Hz;
- stream_desc.mFormatID = kAudioFormatLinearPCM;
- stream_desc.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
- stream_desc.mFramesPerPacket = 1;
- stream_desc.mChannelsPerFrame = 2;
- stream_desc.mBitsPerChannel = sizeof(uint16_t) * 8;
- stream_desc.mBytesPerPacket = bps;
- stream_desc.mBytesPerFrame = bps;
- stream_desc.mReserved = 0;
-
- AudioComponentInstance inst;
- OSStatus ret;
- AURenderCallbackStruct input;
-
- ret = AudioComponentInstanceNew(comp, &inst);
-
- int sample_count = play_frequency_in_Hz * num_buffered_seconds;
- int latency_count = (unsigned)TS_ALIGN(play_frequency_in_Hz / latency_factor_in_Hz, 4);
- TS_ASSERT(sample_count > latency_count);
- int wide_count = (int)TS_ALIGN(sample_count, 4) / 4;
- int pool_size = playing_pool_count * sizeof(tsPlayingSound);
- int mix_buffers_size = sizeof(__m128) * wide_count * 2;
- int sample_buffer_size = sizeof(__m128i) * wide_count;
- tsContext* ctx = (tsContext*)malloc(sizeof(tsContext) + mix_buffers_size + sample_buffer_size + 16 + pool_size);
- TS_CHECK(ret == noErr, "AudioComponentInstanceNew failed");
- ctx->latency_samples = latency_count;
- ctx->index0 = 0;
- ctx->index1 = 0;
- ctx->Hz = play_frequency_in_Hz;
- ctx->bps = bps;
- ctx->wide_count = wide_count;
- ctx->sample_count = wide_count * 4;
- ctx->inst = inst;
- ctx->playing = 0;
- ctx->floatA = (__m128*)(ctx + 1);
- ctx->floatA = (__m128*)TS_ALIGN(ctx->floatA, 16);
- TS_ASSERT(!((size_t)ctx->floatA & 15));
- ctx->floatB = ctx->floatA + wide_count;
- ctx->samples = (__m128i*)ctx->floatB + wide_count;
- ctx->running = 1;
- ctx->separate_thread = 0;
- ctx->sleep_milliseconds = 0;
-
- ret = AudioUnitSetProperty(inst, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &stream_desc, sizeof(stream_desc));
- TS_CHECK(ret == noErr, "Failed to set stream forat");
-
- input.inputProc = tsMemcpyToCA;
- input.inputProcRefCon = ctx;
- ret = AudioUnitSetProperty(inst, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(input));
- TS_CHECK(ret == noErr, "AudioUnitSetProperty failed");
-
- ret = AudioUnitInitialize(inst);
- TS_CHECK(ret == noErr, "Couldn't initialize output unit");
-
- ret = AudioOutputUnitStart(inst);
- TS_CHECK(ret == noErr, "Couldn't start output unit");
-
- if (playing_pool_count)
- {
- ctx->playing_pool = (tsPlayingSound*)(ctx->samples + wide_count);
- for (int i = 0; i < playing_pool_count - 1; ++i)
- ctx->playing_pool[i].next = ctx->playing_pool + i + 1;
- ctx->playing_pool[playing_pool_count - 1].next = 0;
- ctx->playing_free = ctx->playing_pool;
- }
-
- else
- {
- ctx->playing_pool = 0;
- ctx->playing_free = 0;
- }
-
- return ctx;
-
-ts_err:
- free(ctx);
- return 0;
-}
-
-void tsSpawnMixThread(tsContext* ctx)
-{
- if (ctx->separate_thread) return;
- pthread_mutex_init(&ctx->mutex, 0);
- ctx->separate_thread = 1;
- pthread_create(&ctx->thread, 0, tsCtxThread, ctx);
-}
-
-#else
-
-void tsSleep(int milliseconds)
-{
- SDL_Delay(milliseconds);
-}
-
-struct tsContext
-{
- unsigned latency_samples;
- unsigned index0; // read
- unsigned index1; // write
- unsigned running_index;
- int Hz;
- int bps;
- int buffer_size;
- int wide_count;
- int sample_count;
- tsPlayingSound* playing;
- __m128* floatA;
- __m128* floatB;
- __m128i* samples;
- tsPlayingSound* playing_pool;
- tsPlayingSound* playing_free;
-
- // data for tsMix thread, enable these with tsSpawnMixThread
- SDL_Thread* thread;
- SDL_mutex* mutex;
- int separate_thread;
- int running;
- int sleep_milliseconds;
-};
-
-static void tsReleaseContext(tsContext* ctx)
-{
- if (ctx->separate_thread) SDL_DestroyMutex(ctx->mutex);
- tsPlayingSound* playing = ctx->playing;
- while (playing)
- {
- tsRemoveFilter(playing);
- playing = playing->next;
- }
- SDL_CloseAudio();
- free(ctx);
-}
-
-int tsCtxThread(void* udata)
-{
- tsContext* ctx = (tsContext*)udata;
-
- while (ctx->running)
- {
- tsMix(ctx);
- if (ctx->sleep_milliseconds) tsSleep(ctx->sleep_milliseconds);
- else tsSleep(1);
- }
-
- ctx->separate_thread = 0;
- return 0;
-}
-
-static void tsLock(tsContext* ctx)
-{
- if (ctx->separate_thread) SDL_LockMutex(ctx->mutex);
-}
-
-static void tsUnlock(tsContext* ctx)
-{
- if (ctx->separate_thread) SDL_UnlockMutex(ctx->mutex);
-}
-
-void tsSDL_AudioCallback(void* udata, Uint8* stream, int len);
-
-tsContext* tsMakeContext(void* unused, unsigned play_frequency_in_Hz, int latency_factor_in_Hz, int num_buffered_seconds, int playing_pool_count)
-{
- (void)unused;
- int bps = sizeof(uint16_t) * 2;
- int sample_count = play_frequency_in_Hz * num_buffered_seconds;
- int latency_count = (unsigned)TS_ALIGN(play_frequency_in_Hz / latency_factor_in_Hz, 4);
- TS_ASSERT(sample_count > latency_count);
- int wide_count = (int)TS_ALIGN(sample_count, 4) / 4;
- int pool_size = playing_pool_count * sizeof(tsPlayingSound);
- int mix_buffers_size = sizeof(__m128) * wide_count * 2;
- int sample_buffer_size = sizeof(__m128i) * wide_count;
- tsContext* ctx = 0;
- SDL_AudioSpec wanted;
- int ret = SDL_Init(SDL_INIT_AUDIO);
- TS_CHECK(ret >= 0, "Can't init SDL audio");
-
- ctx = (tsContext*)malloc(sizeof(tsContext) + mix_buffers_size + sample_buffer_size + 16 + pool_size);
- TS_CHECK(ctx != NULL, "Can't create audio context");
- ctx->latency_samples = latency_count;
- ctx->index0 = 0;
- ctx->index1 = 0;
- ctx->Hz = play_frequency_in_Hz;
- ctx->bps = bps;
- ctx->wide_count = wide_count;
- ctx->sample_count = wide_count * 4;
- ctx->playing = 0;
- ctx->floatA = (__m128*)(ctx + 1);
- ctx->floatA = (__m128*)TS_ALIGN(ctx->floatA, 16);
- TS_ASSERT(!((size_t)ctx->floatA & 15));
- ctx->floatB = ctx->floatA + wide_count;
- ctx->samples = (__m128i*)ctx->floatB + wide_count;
- ctx->running = 1;
- ctx->separate_thread = 0;
- ctx->sleep_milliseconds = 0;
-
- SDL_memset(&wanted, 0, sizeof(wanted));
- wanted.freq = play_frequency_in_Hz;
- wanted.format = AUDIO_S16SYS;
- wanted.channels = 2; /* 1 = mono, 2 = stereo */
- wanted.samples = 1024;
- wanted.callback = tsSDL_AudioCallback;
- wanted.userdata = ctx;
- ret = SDL_OpenAudio(&wanted, NULL);
- TS_CHECK(ret >= 0, "Can't open SDL audio");
- SDL_PauseAudio(0);
-
- if (playing_pool_count)
- {
- ctx->playing_pool = (tsPlayingSound*)(ctx->samples + wide_count);
- for (int i = 0; i < playing_pool_count - 1; ++i)
- ctx->playing_pool[i].next = ctx->playing_pool + i + 1;
- ctx->playing_pool[playing_pool_count - 1].next = 0;
- ctx->playing_free = ctx->playing_pool;
- }
-
- else
- {
- ctx->playing_pool = 0;
- ctx->playing_free = 0;
- }
-
- return ctx;
-
-ts_err:
- if (ctx) free(ctx);
- return 0;
-}
-
-void tsSpawnMixThread(tsContext* ctx)
-{
- if (ctx->separate_thread) return;
- ctx->mutex = SDL_CreateMutex();
- ctx->separate_thread = 1;
- ctx->thread = SDL_CreateThread(&tsCtxThread, "TinySoundThread", ctx);
-}
-
-#endif
-
-#if TS_PLATFORM == TS_SDL || TS_PLATFORM == TS_MAC
-
-static int tsSamplesWritten(tsContext* ctx)
-{
- int index0 = ctx->index0;
- int index1 = ctx->index1;
- if (index0 <= index1) return index1 - index0;
- else return ctx->sample_count - index0 + index1;
-}
-
-static int tsSamplesUnwritten(tsContext* ctx)
-{
- int index0 = ctx->index0;
- int index1 = ctx->index1;
- if (index0 <= index1) return ctx->sample_count - index1 + index0;
- else return index0 - index1;
-}
-
-static int tsSamplesToMix(tsContext* ctx)
-{
- int lat = ctx->latency_samples;
- int written = tsSamplesWritten(ctx);
- int dif = lat - written;
- TS_ASSERT(dif >= 0);
- if (dif)
- {
- int unwritten = tsSamplesUnwritten(ctx);
- return dif < unwritten ? dif : unwritten;
- }
- return 0;
-}
-
-#define TS_SAMPLES_TO_BYTES( interleaved_sample_count ) ((interleaved_sample_count) * ctx->bps)
-#define TS_BYTES_TO_SAMPLES( byte_count ) ((byte_count) / ctx->bps)
-
-static void tsPushBytes(tsContext* ctx, void* data, int size)
-{
- int index0 = ctx->index0;
- int index1 = ctx->index1;
- int samples = TS_BYTES_TO_SAMPLES(size);
- int sample_count = ctx->sample_count;
-
- int unwritten = tsSamplesUnwritten(ctx);
- if (unwritten < samples) samples = unwritten;
- int can_overflow = index0 <= index1;
- int would_overflow = index1 + samples > sample_count;
-
- if (can_overflow && would_overflow)
- {
- int first_size = TS_SAMPLES_TO_BYTES(sample_count - index1);
- int second_size = size - first_size;
- memcpy((char*)ctx->samples + TS_SAMPLES_TO_BYTES(index1), data, first_size);
- memcpy(ctx->samples, (char*)data + first_size, second_size);
- ctx->index1 = TS_BYTES_TO_SAMPLES(second_size);
- }
-
- else
- {
- memcpy((char*)ctx->samples + TS_SAMPLES_TO_BYTES(index1), data, size);
- ctx->index1 += TS_BYTES_TO_SAMPLES(size);
- }
-}
-
-static int tsPullBytes(tsContext* ctx, void* dst, int size)
-{
- int index0 = ctx->index0;
- int index1 = ctx->index1;
- int allowed_size = TS_SAMPLES_TO_BYTES(tsSamplesWritten(ctx));
- int zeros = 0;
-
- if (allowed_size < size)
- {
- zeros = size - allowed_size;
- size = allowed_size;
- }
-
- if (index1 >= index0)
- {
- memcpy(dst, ((char*)ctx->samples) + TS_SAMPLES_TO_BYTES(index0), size);
- ctx->index0 += TS_BYTES_TO_SAMPLES(size);
- }
-
- else
- {
- int first_size = TS_SAMPLES_TO_BYTES(ctx->sample_count) - TS_SAMPLES_TO_BYTES(index0);
- if (first_size > size) first_size = size;
- int second_size = size - first_size;
- memcpy(dst, ((char*)ctx->samples) + TS_SAMPLES_TO_BYTES(index0), first_size);
- memcpy(((char*)dst) + first_size, ctx->samples, second_size);
- if (second_size) ctx->index0 = TS_BYTES_TO_SAMPLES(second_size);
- else ctx->index0 += TS_BYTES_TO_SAMPLES(first_size);
- }
-
- return zeros;
-}
-
-#endif
-
-void tsShutdownContext(tsContext* ctx)
-{
- if (ctx->separate_thread)
- {
- tsLock(ctx);
- ctx->running = 0;
- tsUnlock(ctx);
- }
-
- while (ctx->separate_thread) tsSleep(1);
- tsReleaseContext(ctx);
-}
-
-void tsThreadSleepDelay(tsContext* ctx, int milliseconds)
-{
- ctx->sleep_milliseconds = milliseconds;
-}
-
-void tsInsertSound(tsContext* ctx, tsPlayingSound* sound)
-{
- // Cannot use tsPlayingSound if tsMakeContext was passed non-zero for playing_pool_count
- // since non-zero playing_pool_count means the context is doing some memory-management
- // for a playing sound pool. InsertSound assumes the pool does not exist, and is apart
- // of the lower-level API (see top of this header for documentation details).
- TS_ASSERT(ctx->playing_pool == 0);
-
- if (sound->active) return;
- tsLock(ctx);
- sound->next = ctx->playing;
- ctx->playing = sound;
- sound->active = 1;
- tsUnlock(ctx);
-}
-
-// NOTE: does not allow delay_in_seconds to be negative (clamps at 0)
-void tsSetDelay(tsContext* ctx, tsPlayingSound* sound, float delay_in_seconds)
-{
- if (delay_in_seconds < 0.0f) delay_in_seconds = 0.0f;
- sound->sample_index = (int)(delay_in_seconds * (float)ctx->Hz);
- sound->sample_index = -(int)TS_ALIGN(sound->sample_index, 4);
-}
-
-tsPlaySoundDef tsMakeDef(tsLoadedSound* sound)
-{
- tsPlaySoundDef def;
- def.paused = 0;
- def.looped = 0;
- def.volume_left = 1.0f;
- def.volume_right = 1.0f;
- def.pan = 0.5f;
- def.pitch = 1.0f;
- def.delay = 0.0f;
- def.loaded = sound;
- return def;
-}
-
-tsPlayingSound* tsPlaySound(tsContext* ctx, tsPlaySoundDef def)
-{
- tsLock(ctx);
-
- tsPlayingSound* playing = ctx->playing_free;
- if (!playing) return 0;
- ctx->playing_free = playing->next;
- *playing = tsMakePlayingSound(def.loaded);
- playing->active = 1;
- playing->paused = def.paused;
- playing->looped = def.looped;
- tsSetVolume(playing, def.volume_left, def.volume_right);
- tsSetPan(playing, def.pan);
- tsSetPitch(playing, def.pitch);
- tsSetDelay(ctx, playing, def.delay);
- playing->next = ctx->playing;
- ctx->playing = playing;
-
- tsUnlock(ctx);
-
- return playing;
-}
-
-void tsStopAllSounds(tsContext* ctx)
-{
- // This is apart of the high level API, not the low level API.
- // If using the low level API you must write your own function to
- // stop playing all sounds.
- TS_ASSERT(ctx->playing_pool == 0);
-
- tsPlayingSound* sound = ctx->playing;
- ctx->playing = 0;
-
- while (sound)
- {
- tsPlayingSound* next = sound->next;
- sound->next = ctx->playing_free;
- ctx->playing_free = sound;
- sound = next;
- }
-}
-
-#if TS_PLATFORM == TS_WINDOWS
-
-static void tsPosition(tsContext* ctx, int* byte_to_lock, int* bytes_to_write)
-{
- // compute bytes to be written to direct sound
- DWORD play_cursor;
- DWORD write_cursor;
-#ifdef __cplusplus
- HRESULT hr = ctx->buffer->GetCurrentPosition(&play_cursor, &write_cursor);
-#else
- HRESULT hr = ctx->buffer->lpVtbl->GetCurrentPosition(ctx->buffer, &play_cursor, &write_cursor);
-#endif
- TS_ASSERT(hr == DS_OK);
-
- DWORD lock = (ctx->running_index * ctx->bps) % ctx->buffer_size;
- DWORD target_cursor = (write_cursor + ctx->latency_samples * ctx->bps) % ctx->buffer_size;
- target_cursor = (DWORD)TS_ALIGN(target_cursor, 16);
- DWORD write;
-
- if (lock > target_cursor)
- {
- write = (ctx->buffer_size - lock) + target_cursor;
- }
-
- else
- {
- write = target_cursor - lock;
- }
-
- *byte_to_lock = lock;
- *bytes_to_write = write;
-}
-
-static void tsMemcpyToDS(tsContext* ctx, int16_t* samples, int byte_to_lock, int bytes_to_write)
-{
- // copy mixer buffers to direct sound
- void* region1;
- DWORD size1;
- void* region2;
- DWORD size2;
-#ifdef __cplusplus
- HRESULT hr = ctx->buffer->Lock(byte_to_lock, bytes_to_write, &region1, &size1, &region2, &size2, 0);
-
- if (hr == DSERR_BUFFERLOST)
- {
- ctx->buffer->Restore();
- hr = ctx->buffer->Lock(byte_to_lock, bytes_to_write, &region1, &size1, &region2, &size2, 0);
- }
-#else
- HRESULT hr = ctx->buffer->lpVtbl->Lock(ctx->buffer, byte_to_lock, bytes_to_write, &region1, &size1, &region2, &size2, 0);
-
- if (hr == DSERR_BUFFERLOST)
- {
- ctx->buffer->lpVtbl->Restore(ctx->buffer);
- hr = ctx->buffer->lpVtbl->Lock(ctx->buffer, byte_to_lock, bytes_to_write, &region1, &size1, &region2, &size2, 0);
- }
-#endif
-
- if (!SUCCEEDED(hr))
- return;
-
- unsigned running_index = ctx->running_index;
- INT16* sample1 = (INT16*)region1;
- DWORD sample1_count = size1 / ctx->bps;
- memcpy(sample1, samples, sample1_count * sizeof(INT16) * 2);
- samples += sample1_count * 2;
- running_index += sample1_count;
-
- INT16* sample2 = (INT16*)region2;
- DWORD sample2_count = size2 / ctx->bps;
- memcpy(sample2, samples, sample2_count * sizeof(INT16) * 2);
- samples += sample2_count * 2;
- running_index += sample2_count;
-
-#ifdef __cplusplus
- ctx->buffer->Unlock(region1, size1, region2, size2);
-#else
- ctx->buffer->lpVtbl->Unlock(ctx->buffer, region1, size1, region2, size2);
-#endif
- ctx->running_index = running_index;
-
- // meager hack to fill out sound buffer before playing
- static int first;
- if (!first)
- {
-#ifdef __cplusplus
- ctx->buffer->Play(0, 0, DSBPLAY_LOOPING);
-#else
- ctx->buffer->lpVtbl->Play(ctx->buffer, 0, 0, DSBPLAY_LOOPING);
-#endif
- first = 1;
- }
-}
-
-#elif TS_PLATFORM == TS_MAC
-
-static OSStatus tsMemcpyToCA(void* udata, AudioUnitRenderActionFlags* ioActionFlags, const AudioTimeStamp* inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList* ioData)
-{
- tsContext* ctx = (tsContext*)udata;
- int bps = ctx->bps;
- int samples_requested_to_consume = inNumberFrames;
- AudioBuffer* buffer = ioData->mBuffers;
-
- TS_ASSERT(ioData->mNumberBuffers == 1);
- TS_ASSERT(buffer->mNumberChannels == 2);
- int byte_size = buffer->mDataByteSize;
- TS_ASSERT(byte_size == samples_requested_to_consume * bps);
-
- int zero_bytes = tsPullBytes(ctx, buffer->mData, byte_size);
- memset(((char*)buffer->mData) + (byte_size - zero_bytes), 0, zero_bytes);
-
- return noErr;
-}
-
-#elif TS_PLATFORM == TS_SDL
-
-static void tsSDL_AudioCallback(void* udata, Uint8* stream, int len)
-{
- tsContext* ctx = (tsContext*)udata;
- int zero_bytes = tsPullBytes(ctx, stream, len);
- memset(stream + (len - zero_bytes), 0, zero_bytes);
-}
-
-#endif
-
-static void tsPitchShift(float pitchShift, int num_samples_to_process, float sampleRate, float* indata, tsPitchData** pitch_filter);
-
-// Pitch processing tunables
-#define TS_MAX_FRAME_LENGTH 4096
-#define TS_PITCH_FRAME_SIZE 512
-#define TS_PITCH_QUALITY 8
-
-// interals
-#define TS_STEPSIZE (TS_PITCH_FRAME_SIZE / TS_PITCH_QUALITY)
-#define TS_OVERLAP (TS_PITCH_FRAME_SIZE - TS_STEPSIZE)
-#define TS_EXPECTED_FREQUENCY (2.0f * 3.14159265359f * (float)TS_STEPSIZE / (float)TS_PITCH_FRAME_SIZE)
-
-// TODO:
-// Use a memory pool for these things. For now they are just malloc16'd/free16'd
-// Not high priority to use a pool, since pitch shifting is already really expensive,
-// and cost of malloc is dwarfed. But would be a nice-to-have for potential memory
-// fragmentation issues.
-typedef struct tsPitchData
-{
- float pitch_shifted_output_samples[TS_MAX_FRAME_LENGTH];
- float in_FIFO[TS_STEPSIZE + TS_PITCH_FRAME_SIZE];
- float out_FIFO[TS_STEPSIZE + TS_PITCH_FRAME_SIZE];
- float fft_data[2 * TS_PITCH_FRAME_SIZE];
- float previous_phase[TS_PITCH_FRAME_SIZE / 2 + 4];
- float sum_phase[TS_PITCH_FRAME_SIZE / 2 + 4];
- float window_accumulator[TS_STEPSIZE + TS_PITCH_FRAME_SIZE];
- float freq[TS_PITCH_FRAME_SIZE];
- float mag[TS_PITCH_FRAME_SIZE];
- float pitch_shift_workspace[TS_PITCH_FRAME_SIZE];
- int index;
-} tsPitchData;
-
-static void tsRemoveFilter(tsPlayingSound* playing)
-{
- for (int i = 0; i < 2; i++)
- {
- if (playing->pitch_filter[i])
- {
- free16(playing->pitch_filter[i]);
- playing->pitch_filter[i] = 0;
- }
- }
-}
-
-void tsMix(tsContext* ctx)
-{
- tsLock(ctx);
-
-#if TS_PLATFORM == TS_WINDOWS
-
- int byte_to_lock;
- int bytes_to_write;
- tsPosition(ctx, &byte_to_lock, &bytes_to_write);
-
- if (!bytes_to_write) goto unlock;
- int samples_to_write = bytes_to_write / ctx->bps;
-
-#elif TS_PLATFORM == TS_MAC || TS_PLATFORM == TS_SDL
-
- int samples_to_write = tsSamplesToMix(ctx);
- if (!samples_to_write) goto unlock;
- int bytes_to_write = samples_to_write * ctx->bps;
-
-#else
-#endif
-
- // clear mixer buffers
- int wide_count = samples_to_write / 4;
- TS_ASSERT(!(samples_to_write & 3));
-
- __m128* floatA = ctx->floatA;
- __m128* floatB = ctx->floatB;
- __m128 zero = _mm_set1_ps(0.0f);
-
- for (int i = 0; i < wide_count; ++i)
- {
- floatA[i] = zero;
- floatB[i] = zero;
- }
-
- // mix all playing sounds into the mixer buffers
- tsPlayingSound** ptr = &ctx->playing;
- while (*ptr)
- {
- tsPlayingSound* playing = *ptr;
- tsLoadedSound* loaded = playing->loaded_sound;
- __m128* cA = (__m128*)loaded->channels[0];
- __m128* cB = (__m128*)loaded->channels[1];
-
- // Attempted to play a sound with no audio.
- // Make sure the audio file was loaded properly. Check for
- // error messages in g_tsErrorReason.
- TS_ASSERT(cA);
-
- int mix_count = samples_to_write;
- int offset = playing->sample_index;
- int remaining = loaded->sample_count - offset;
- if (remaining < mix_count) mix_count = remaining;
- TS_ASSERT(remaining > 0);
-
- float vA0 = playing->volume0 * playing->pan0;
- float vB0 = playing->volume1 * playing->pan1;
- __m128 vA = _mm_set1_ps(vA0);
- __m128 vB = _mm_set1_ps(vB0);
-
- // skip sound if it's delay is longer than mix_count and
- // handle various delay cases
- int delay_offset = 0;
- if (offset < 0)
- {
- int samples_till_positive = -offset;
- int mix_leftover = mix_count - samples_till_positive;
-
- if (mix_leftover <= 0)
- {
- playing->sample_index += mix_count;
- goto get_next_playing_sound;
- }
-
- else
- {
- offset = 0;
- delay_offset = samples_till_positive;
- mix_count = mix_leftover;
- }
- }
- TS_ASSERT(!(delay_offset & 3));
-
- // immediately remove any inactive elements
- if (!playing->active || !ctx->running)
- goto remove;
-
- // skip all paused sounds
- if (playing->paused)
- goto get_next_playing_sound;
-
- // SIMD offets
- int mix_wide = (int)TS_ALIGN(mix_count, 4) / 4;
- int offset_wide = (int)TS_TRUNC(offset, 4) / 4;
- int delay_wide = (int)TS_ALIGN(delay_offset, 4) / 4;
-
- // use tsPitchShift to on-the-fly pitch shift some samples
- // only call this function if the user set a custom pitch value
- if (playing->pitch != 1.0f)
- {
- int sample_count = (mix_wide - 2 * delay_wide) * 4;
- int falling_behind = sample_count > TS_MAX_FRAME_LENGTH;
-
- // TS_MAX_FRAME_LENGTH represents max samples we can pitch shift in one go. In the event
- // that this process takes longer than the time required to play the actual sound, just
- // fall back to the original sound (non-pitch shifted). This will sound very ugly. To
- // prevent falling behind, make sure not to pitch shift too many sounds at once. Try tweaking
- // TS_PITCH_QUALITY to make it lower (must be a power of 2).
- if (!falling_behind)
- {
- tsPitchShift(playing->pitch, sample_count, (float)ctx->Hz, (float*)(cA + delay_wide + offset_wide), playing->pitch_filter);
- cA = (__m128 *)playing->pitch_filter[0]->pitch_shifted_output_samples;
-
- if (loaded->channel_count == 2)
- {
- tsPitchShift(playing->pitch, sample_count, (float)ctx->Hz, (float*)(cB + delay_wide + offset_wide), playing->pitch_filter + 1);
- cB = (__m128 *)playing->pitch_filter[1]->pitch_shifted_output_samples;
- }
-
- offset_wide = -delay_wide;
- }
- }
-
- // apply volume, load samples into float buffers
- switch (loaded->channel_count)
- {
- case 1:
- for (int i = delay_wide; i < mix_wide - delay_wide; ++i)
- {
- __m128 A = cA[i + offset_wide];
- __m128 B = _mm_mul_ps(A, vB);
- A = _mm_mul_ps(A, vA);
- floatA[i] = _mm_add_ps(floatA[i], A);
- floatB[i] = _mm_add_ps(floatB[i], B);
- }
- break;
-
- case 2:
- {
- for (int i = delay_wide; i < mix_wide - delay_wide; ++i)
- {
- __m128 A = cA[i + offset_wide];
- __m128 B = cB[i + offset_wide];
-
- A = _mm_mul_ps(A, vA);
- B = _mm_mul_ps(B, vB);
- floatA[i] = _mm_add_ps(floatA[i], A);
- floatB[i] = _mm_add_ps(floatB[i], B);
- }
- } break;
- }
-
- // playing list logic
- playing->sample_index += mix_count;
- if (playing->sample_index == loaded->sample_count)
- {
- if (playing->looped)
- {
- playing->sample_index = 0;
- goto get_next_playing_sound;
- }
-
- remove:
- playing->sample_index = 0;
- *ptr = (*ptr)->next;
- playing->next = 0;
- playing->active = 0;
-
- tsRemoveFilter(playing);
-
- // if using high-level API manage the tsPlayingSound memory ourselves
- if (ctx->playing_pool)
- {
- playing->next = ctx->playing_free;
- ctx->playing_free = playing;
- }
-
- // we already incremented next pointer, so don't do it again
- continue;
- }
-
- get_next_playing_sound:
- if (*ptr) ptr = &(*ptr)->next;
- else break;
- }
-
- // load all floats into 16 bit packed interleaved samples
-#if TS_PLATFORM == TS_WINDOWS
-
- __m128i* samples = ctx->samples;
- for (int i = 0; i < wide_count; ++i)
- {
- __m128i a = _mm_cvtps_epi32(floatA[i]);
- __m128i b = _mm_cvtps_epi32(floatB[i]);
- __m128i a0b0a1b1 = _mm_unpacklo_epi32(a, b);
- __m128i a2b2a3b3 = _mm_unpackhi_epi32(a, b);
- samples[i] = _mm_packs_epi32(a0b0a1b1, a2b2a3b3);
- }
- tsMemcpyToDS(ctx, (int16_t*)samples, byte_to_lock, bytes_to_write);
-
-#elif TS_PLATFORM == TS_MAC || TS_PLATFORM == TS_SDL
-
- // Since the ctx->samples array is already in use as a ring buffer
- // reusing floatA to store output is a good way to temporarly store
- // the final samples. Then a single ring buffer push can be used
- // afterwards. Pretty hacky, but whatever :)
- __m128i* samples = (__m128i*)floatA;
- memset(samples, 0, sizeof(__m128i) * wide_count);
- for (int i = 0; i < wide_count; ++i)
- {
- __m128i a = _mm_cvtps_epi32(floatA[i]);
- __m128i b = _mm_cvtps_epi32(floatB[i]);
- __m128i a0b0a1b1 = _mm_unpacklo_epi32(a, b);
- __m128i a2b2a3b3 = _mm_unpackhi_epi32(a, b);
- samples[i] = _mm_packs_epi32(a0b0a1b1, a2b2a3b3);
- }
- tsPushBytes(ctx, samples, bytes_to_write);
-
-#else
-#endif
-
-unlock:
- tsUnlock(ctx);
-}
-
-// TODO:
-// Try this optimization out (2N POINT REAL FFT USING AN N POINT COMPLEX FFT)
-// http://www.fftguru.com/fftguru.com.tutorial2.pdf
-
-#include <math.h>
-
-static uint32_t tsRev32(uint32_t x)
-{
- uint32_t a = ((x & 0xAAAAAAAA) >> 1) | ((x & 0x55555555) << 1);
- uint32_t b = ((a & 0xCCCCCCCC) >> 2) | ((a & 0x33333333) << 2);
- uint32_t c = ((b & 0xF0F0F0F0) >> 4) | ((b & 0x0F0F0F0F) << 4);
- uint32_t d = ((c & 0xFF00FF00) >> 8) | ((c & 0x00FF00FF) << 8);
- return (d >> 16) | (d << 16);
-}
-
-static uint32_t tsPopCount(uint32_t x)
-{
- uint32_t a = x - ((x >> 1) & 0x55555555);
- uint32_t b = (((a >> 2) & 0x33333333) + (a & 0x33333333));
- uint32_t c = (((b >> 4) + b) & 0x0F0F0F0F);
- uint32_t d = c + (c >> 8);
- uint32_t e = d + (d >> 16);
- uint32_t f = e & 0x0000003F;
- return f;
-}
-
-static uint32_t tsLog2(uint32_t x)
-{
- uint32_t a = x | (x >> 1);
- uint32_t b = a | (a >> 2);
- uint32_t c = b | (b >> 4);
- uint32_t d = c | (c >> 8);
- uint32_t e = d | (d >> 16);
- uint32_t f = e >> 1;
- return tsPopCount(f);
-}
-
-// x contains real inputs
-// y contains imaginary inputs
-// count must be a power of 2
-// sign must be 1.0 (forward transform) or -1.0f (inverse transform)
-static void tsFFT(float* x, float* y, int count, float sign)
-{
- int exponent = (int)tsLog2((uint32_t)count);
-
- // bit reversal stage
- // swap all elements with their bit reversed index within the
- // lowest level of the Cooley-Tukey recursion tree
- for (int i = 1; i < count - 1; i++)
- {
- uint32_t j = tsRev32((uint32_t)i);
- j >>= (32 - exponent);
- if (i < (int)j)
- {
- float tx = x[i];
- float ty = y[i];
- x[i] = x[j];
- y[i] = y[j];
- x[j] = tx;
- y[j] = ty;
- }
- }
-
- // for each recursive iteration
- for (int iter = 0, L = 1; iter < exponent; ++iter)
- {
- int Ls = L;
- L <<= 1;
- float ur = 1.0f; // cos( pi / 2 )
- float ui = 0; // sin( pi / 2 )
- float arg = 3.14159265359f / (float)Ls;
- float wr = cosf(arg);
- float wi = -sign * sinf(arg);
-
- // rows in DFT submatrix
- for (int j = 0; j < Ls; ++j)
- {
- // do butterflies upon DFT row elements
- for (int i = j; i < count; i += L)
- {
- int index = i + Ls;
- float x_index = x[index];
- float y_index = y[index];
- float x_i = x[i];
- float y_i = y[i];
-
- float tr = ur * x_index - ui * y_index;
- float ti = ur * y_index + ui * x_index;
- float x_low = x_i - tr;
- float x_high = x_i + tr;
- float y_low = y_i - ti;
- float y_high = y_i + ti;
-
- x[index] = x_low;
- y[index] = y_low;
- x[i] = x_high;
- y[i] = y_high;
- }
-
- // Rotate u1 and u2 via Givens rotations (2d planar rotation).
- // This keeps cos/sin calls in the outermost loop.
- // Floating point error is scaled proportionally to Ls.
- float t = ur * wr - ui * wi;
- ui = ur * wi + ui * wr;
- ur = t;
- }
- }
-
- // scale factor for forward transform
- if (sign > 0)
- {
- float inv_count = 1.0f / (float)count;
- for (int i = 0; i < count; i++)
- {
- x[i] *= inv_count;
- y[i] *= inv_count;
- }
- }
-}
-
-#ifdef _MSC_VER
-
-#define TS_ALIGN16_0 __declspec( align( 16 ) )
-#define TS_ALIGN16_1
-#define TS_SELECTANY extern const __declspec( selectany )
-
-#else
-
-#define TS_ALIGN16_0
-#define TS_ALIGN16_1 __attribute__( (aligned( 16 )) )
-#define TS_SELECTANY const __attribute__( (selectany) )
-
-#endif
-
-// SSE2 trig funcs from https://github.com/to-miz/sse_mathfun_extension/
-#define _PS_CONST( Name, Val ) \
- TS_SELECTANY TS_ALIGN16_0 float _ps_##Name[ 4 ] TS_ALIGN16_1 = { Val, Val, Val, Val }
-
-#define _PS_CONST_TYPE( Name, Type, Val ) \
- TS_SELECTANY TS_ALIGN16_0 Type _ps_##Name[ 4 ] TS_ALIGN16_1 = { Val, Val, Val, Val }
-
-#define _PI32_CONST( Name, Val ) \
- TS_SELECTANY TS_ALIGN16_0 int _pi32_##Name[ 4 ] TS_ALIGN16_1 = { Val, Val, Val, Val }
-
-_PS_CONST_TYPE(sign_mask, int, (int)0x80000000);
-_PS_CONST_TYPE(inv_sign_mask, int, (int)~0x80000000);
-
-_PS_CONST(atanrange_hi, 2.414213562373095f);
-_PS_CONST(atanrange_lo, 0.4142135623730950f);
-_PS_CONST(cephes_PIO2F, 1.5707963267948966192f);
-_PS_CONST(cephes_PIO4F, 0.7853981633974483096f);
-_PS_CONST(1, 1.0f);
-_PS_CONST(0p5, 0.5f);
-_PS_CONST(0, 0);
-_PS_CONST(sincof_p0, -1.9515295891E-4f);
-_PS_CONST(sincof_p1, 8.3321608736E-3f);
-_PS_CONST(sincof_p2, -1.6666654611E-1f);
-_PS_CONST(atancof_p0, 8.05374449538e-2f);
-_PS_CONST(atancof_p1, 1.38776856032E-1f);
-_PS_CONST(atancof_p2, 1.99777106478E-1f);
-_PS_CONST(atancof_p3, 3.33329491539E-1f);
-_PS_CONST(cephes_PIF, 3.141592653589793238f);
-_PS_CONST(cephes_2PIF, 2.0f * 3.141592653589793238f);
-_PS_CONST(cephes_FOPI, 1.27323954473516f); // 4 / M_PI
-_PS_CONST(minus_cephes_DP1, -0.78515625f);
-_PS_CONST(minus_cephes_DP2, -2.4187564849853515625e-4f);
-_PS_CONST(minus_cephes_DP3, -3.77489497744594108e-8f);
-_PS_CONST(coscof_p0, 2.443315711809948E-005f);
-_PS_CONST(coscof_p1, -1.388731625493765E-003f);
-_PS_CONST(coscof_p2, 4.166664568298827E-002f);
-_PS_CONST(frame_size, (float)TS_PITCH_FRAME_SIZE);
-
-_PI32_CONST(1, 1);
-_PI32_CONST(inv1, ~1);
-_PI32_CONST(2, 2);
-_PI32_CONST(4, 4);
-
-static __m128 _mm_atan_ps(__m128 x)
-{
- __m128 sign_bit, y;
-
- sign_bit = x;
- /* take the absolute value */
- x = _mm_and_ps(x, *(__m128*)_ps_inv_sign_mask);
- /* extract the sign bit (upper one) */
- sign_bit = _mm_and_ps(sign_bit, *(__m128*)_ps_sign_mask);
-
- /* range reduction, init x and y depending on range */
- /* x > 2.414213562373095 */
- __m128 cmp0 = _mm_cmpgt_ps(x, *(__m128*)_ps_atanrange_hi);
- /* x > 0.4142135623730950 */
- __m128 cmp1 = _mm_cmpgt_ps(x, *(__m128*)_ps_atanrange_lo);
-
- /* x > 0.4142135623730950 && !( x > 2.414213562373095 ) */
- __m128 cmp2 = _mm_andnot_ps(cmp0, cmp1);
-
- /* -( 1.0/x ) */
- __m128 y0 = _mm_and_ps(cmp0, *(__m128*)_ps_cephes_PIO2F);
- __m128 x0 = _mm_div_ps(*(__m128*)_ps_1, x);
- x0 = _mm_xor_ps(x0, *(__m128*)_ps_sign_mask);
-
- __m128 y1 = _mm_and_ps(cmp2, *(__m128*)_ps_cephes_PIO4F);
- /* (x-1.0)/(x+1.0) */
- __m128 x1_o = _mm_sub_ps(x, *(__m128*)_ps_1);
- __m128 x1_u = _mm_add_ps(x, *(__m128*)_ps_1);
- __m128 x1 = _mm_div_ps(x1_o, x1_u);
-
- __m128 x2 = _mm_and_ps(cmp2, x1);
- x0 = _mm_and_ps(cmp0, x0);
- x2 = _mm_or_ps(x2, x0);
- cmp1 = _mm_or_ps(cmp0, cmp2);
- x2 = _mm_and_ps(cmp1, x2);
- x = _mm_andnot_ps(cmp1, x);
- x = _mm_or_ps(x2, x);
-
- y = _mm_or_ps(y0, y1);
-
- __m128 zz = _mm_mul_ps(x, x);
- __m128 acc = *(__m128*)_ps_atancof_p0;
- acc = _mm_mul_ps(acc, zz);
- acc = _mm_sub_ps(acc, *(__m128*)_ps_atancof_p1);
- acc = _mm_mul_ps(acc, zz);
- acc = _mm_add_ps(acc, *(__m128*)_ps_atancof_p2);
- acc = _mm_mul_ps(acc, zz);
- acc = _mm_sub_ps(acc, *(__m128*)_ps_atancof_p3);
- acc = _mm_mul_ps(acc, zz);
- acc = _mm_mul_ps(acc, x);
- acc = _mm_add_ps(acc, x);
- y = _mm_add_ps(y, acc);
-
- /* update the sign */
- y = _mm_xor_ps(y, sign_bit);
-
- return y;
-}
-
-static __m128 _mm_atan2_ps(__m128 y, __m128 x)
-{
- __m128 x_eq_0 = _mm_cmpeq_ps(x, *(__m128*)_ps_0);
- __m128 x_gt_0 = _mm_cmpgt_ps(x, *(__m128*)_ps_0);
- __m128 x_le_0 = _mm_cmple_ps(x, *(__m128*)_ps_0);
- __m128 y_eq_0 = _mm_cmpeq_ps(y, *(__m128*)_ps_0);
- __m128 x_lt_0 = _mm_cmplt_ps(x, *(__m128*)_ps_0);
- __m128 y_lt_0 = _mm_cmplt_ps(y, *(__m128*)_ps_0);
-
- __m128 zero_mask = _mm_and_ps(x_eq_0, y_eq_0);
- __m128 zero_mask_other_case = _mm_and_ps(y_eq_0, x_gt_0);
- zero_mask = _mm_or_ps(zero_mask, zero_mask_other_case);
-
- __m128 pio2_mask = _mm_andnot_ps(y_eq_0, x_eq_0);
- __m128 pio2_mask_sign = _mm_and_ps(y_lt_0, *(__m128*)_ps_sign_mask);
- __m128 pio2_result = *(__m128*)_ps_cephes_PIO2F;
- pio2_result = _mm_xor_ps(pio2_result, pio2_mask_sign);
- pio2_result = _mm_and_ps(pio2_mask, pio2_result);
-
- __m128 pi_mask = _mm_and_ps(y_eq_0, x_le_0);
- __m128 pi = *(__m128*)_ps_cephes_PIF;
- __m128 pi_result = _mm_and_ps(pi_mask, pi);
-
- __m128 swap_sign_mask_offset = _mm_and_ps(x_lt_0, y_lt_0);
- swap_sign_mask_offset = _mm_and_ps(swap_sign_mask_offset, *(__m128*)_ps_sign_mask);
-
- __m128 offset0 = _mm_setzero_ps();
- __m128 offset1 = *(__m128*)_ps_cephes_PIF;
- offset1 = _mm_xor_ps(offset1, swap_sign_mask_offset);
-
- __m128 offset = _mm_andnot_ps(x_lt_0, offset0);
- offset = _mm_and_ps(x_lt_0, offset1);
-
- __m128 arg = _mm_div_ps(y, x);
- __m128 atan_result = _mm_atan_ps(arg);
- atan_result = _mm_add_ps(atan_result, offset);
-
- /* select between zero_result, pio2_result and atan_result */
-
- __m128 result = _mm_andnot_ps(zero_mask, pio2_result);
- atan_result = _mm_andnot_ps(pio2_mask, atan_result);
- atan_result = _mm_andnot_ps(pio2_mask, atan_result);
- result = _mm_or_ps(result, atan_result);
- result = _mm_or_ps(result, pi_result);
-
- return result;
-}
-
-static void _mm_sincos_ps(__m128 x, __m128 *s, __m128 *c)
-{
- __m128 xmm1, xmm2, xmm3 = _mm_setzero_ps(), sign_bit_sin, y;
- __m128i emm0, emm2, emm4;
- sign_bit_sin = x;
- /* take the absolute value */
- x = _mm_and_ps(x, *(__m128*)_ps_inv_sign_mask);
- /* extract the sign bit (upper one) */
- sign_bit_sin = _mm_and_ps(sign_bit_sin, *(__m128*)_ps_sign_mask);
-
- /* scale by 4/Pi */
- y = _mm_mul_ps(x, *(__m128*)_ps_cephes_FOPI);
-
- /* store the integer part of y in emm2 */
- emm2 = _mm_cvttps_epi32(y);
-
- /* j=(j+1) & (~1) (see the cephes sources) */
- emm2 = _mm_add_epi32(emm2, *(__m128i*)_pi32_1);
- emm2 = _mm_and_si128(emm2, *(__m128i*)_pi32_inv1);
- y = _mm_cvtepi32_ps(emm2);
-
- emm4 = emm2;
-
- /* get the swap sign flag for the sine */
- emm0 = _mm_and_si128(emm2, *(__m128i*)_pi32_4);
- emm0 = _mm_slli_epi32(emm0, 29);
- __m128 swap_sign_bit_sin = _mm_castsi128_ps(emm0);
-
- /* get the polynom selection mask for the sine*/
- emm2 = _mm_and_si128(emm2, *(__m128i*)_pi32_2);
- emm2 = _mm_cmpeq_epi32(emm2, _mm_setzero_si128());
- __m128 poly_mask = _mm_castsi128_ps(emm2);
-
- /* The magic pass: "Extended precision modular arithmetic"
- x = ((x - y * DP1) - y * DP2) - y * DP3; */
- xmm1 = *(__m128*)_ps_minus_cephes_DP1;
- xmm2 = *(__m128*)_ps_minus_cephes_DP2;
- xmm3 = *(__m128*)_ps_minus_cephes_DP3;
- xmm1 = _mm_mul_ps(y, xmm1);
- xmm2 = _mm_mul_ps(y, xmm2);
- xmm3 = _mm_mul_ps(y, xmm3);
- x = _mm_add_ps(x, xmm1);
- x = _mm_add_ps(x, xmm2);
- x = _mm_add_ps(x, xmm3);
-
- emm4 = _mm_sub_epi32(emm4, *(__m128i*)_pi32_2);
- emm4 = _mm_andnot_si128(emm4, *(__m128i*)_pi32_4);
- emm4 = _mm_slli_epi32(emm4, 29);
- __m128 sign_bit_cos = _mm_castsi128_ps(emm4);
-
- sign_bit_sin = _mm_xor_ps(sign_bit_sin, swap_sign_bit_sin);
-
-
- /* Evaluate the first polynom (0 <= x <= Pi/4) */
- __m128 z = _mm_mul_ps(x, x);
- y = *(__m128*)_ps_coscof_p0;
-
- y = _mm_mul_ps(y, z);
- y = _mm_add_ps(y, *(__m128*)_ps_coscof_p1);
- y = _mm_mul_ps(y, z);
- y = _mm_add_ps(y, *(__m128*)_ps_coscof_p2);
- y = _mm_mul_ps(y, z);
- y = _mm_mul_ps(y, z);
- __m128 tmp = _mm_mul_ps(z, *(__m128*)_ps_0p5);
- y = _mm_sub_ps(y, tmp);
- y = _mm_add_ps(y, *(__m128*)_ps_1);
-
- /* Evaluate the second polynom (Pi/4 <= x <= 0) */
-
- __m128 y2 = *(__m128*)_ps_sincof_p0;
- y2 = _mm_mul_ps(y2, z);
- y2 = _mm_add_ps(y2, *(__m128*)_ps_sincof_p1);
- y2 = _mm_mul_ps(y2, z);
- y2 = _mm_add_ps(y2, *(__m128*)_ps_sincof_p2);
- y2 = _mm_mul_ps(y2, z);
- y2 = _mm_mul_ps(y2, x);
- y2 = _mm_add_ps(y2, x);
-
- /* select the correct result from the two polynoms */
- xmm3 = poly_mask;
- __m128 ysin2 = _mm_and_ps(xmm3, y2);
- __m128 ysin1 = _mm_andnot_ps(xmm3, y);
- y2 = _mm_sub_ps(y2, ysin2);
- y = _mm_sub_ps(y, ysin1);
-
- xmm1 = _mm_add_ps(ysin1, ysin2);
- xmm2 = _mm_add_ps(y, y2);
-
- /* update the sign */
- *s = _mm_xor_ps(xmm1, sign_bit_sin);
- *c = _mm_xor_ps(xmm2, sign_bit_cos);
-}
-
-static __m128i select_si(__m128i a, __m128i b, __m128i mask)
-{
- return _mm_xor_si128(a, _mm_and_si128(mask, _mm_xor_si128(b, a)));
-}
-
-#define tsVonHann( i ) (-0.5f * cosf( 2.0f * 3.14159265359f * (float)(i) / (float)TS_PITCH_FRAME_SIZE ) + 0.5f)
-
-static __m128 tsVonHann4(int i)
-{
- __m128 k4 = _mm_set_ps((float)(i * 4 + 3), (float)(i * 4 + 2), (float)(i * 4 + 1), (float)(i * 4));
- k4 = _mm_mul_ps(*(__m128*)_ps_cephes_2PIF, k4);
- k4 = _mm_div_ps(k4, *(__m128*)_ps_frame_size);
-
- // Seems like _mm_cos_ps and _mm_sincos_ps was causing some audio popping...
- // I'm not really skilled enough to fix it, but feel free to try: http://gruntthepeon.free.fr/ssemath/sse_mathfun.h
- // My guess is some large negative or positive values were causing some
- // precision trouble. In this case manually calling 4 cosines is not
- // really a big deal, since this function is not a bottleneck.
-
-#if 0
- __m128 c = _mm_cos_ps(k4);
-#elif 0
- __m128 s, c;
- _mm_sincos_ps(k4, &s, &c);
-#else
- __m128 c = k4;
- float* cf = (float*)&c;
- cf[0] = cosf(cf[0]);
- cf[1] = cosf(cf[1]);
- cf[2] = cosf(cf[2]);
- cf[3] = cosf(cf[3]);
-#endif
-
- __m128 von_hann = _mm_add_ps(_mm_mul_ps(_mm_set_ps1(-0.5f), c), _mm_set_ps1(0.5f));
- return von_hann;
-}
-
-// Analysis and synthesis steps learned from Bernsee's wonderful blog post:
-// http://blogs.zynaptiq.com/bernsee/pitch-shifting-using-the-ft/
-static void tsPitchShift(float pitchShift, int num_samples_to_process, float sampleRate, float* indata, tsPitchData** pitch_filter)
-{
- TS_ASSERT(num_samples_to_process <= TS_MAX_FRAME_LENGTH);
-
- // make sure compiler didn't do anything weird with the member
- // offsets of tsPitchData. All arrays must be 16 byte aligned
- TS_ASSERT(!((size_t)&(((tsPitchData*)0)->pitch_shifted_output_samples) & 15));
- TS_ASSERT(!((size_t)&(((tsPitchData*)0)->fft_data) & 15));
- TS_ASSERT(!((size_t)&(((tsPitchData*)0)->previous_phase) & 15));
- TS_ASSERT(!((size_t)&(((tsPitchData*)0)->sum_phase) & 15));
- TS_ASSERT(!((size_t)&(((tsPitchData*)0)->window_accumulator) & 15));
- TS_ASSERT(!((size_t)&(((tsPitchData*)0)->freq) & 15));
- TS_ASSERT(!((size_t)&(((tsPitchData*)0)->mag) & 15));
- TS_ASSERT(!((size_t)&(((tsPitchData*)0)->pitch_shift_workspace) & 15));
-
- tsPitchData* pf;
-
- if (*pitch_filter == NULL)
- {
- pf = (tsPitchData*)malloc16(sizeof(tsPitchData));
- memset(pf, 0, sizeof(tsPitchData));
- *pitch_filter = pf;
- }
- else
- {
- pf = *pitch_filter;
- }
-
- float freqPerBin = sampleRate / (float)TS_PITCH_FRAME_SIZE;
- __m128 freq_per_bin = _mm_set_ps1(sampleRate / (float)TS_PITCH_FRAME_SIZE);
- __m128 pi = *(__m128*)_ps_cephes_PIF;
- __m128 two_pi = *(__m128*)_ps_cephes_2PIF;
- __m128 pitch_quality = _mm_set_ps1((float)TS_PITCH_QUALITY);
- float* out_samples = pf->pitch_shifted_output_samples;
- if (pf->index == 0) pf->index = TS_OVERLAP;
-
- while (num_samples_to_process)
- {
- int copy_count = TS_PITCH_FRAME_SIZE - pf->index;
- if (num_samples_to_process < copy_count) copy_count = num_samples_to_process;
-
- memcpy(pf->in_FIFO + pf->index, indata, sizeof(float) * copy_count);
- memcpy(out_samples, pf->out_FIFO + pf->index - TS_OVERLAP, sizeof(float) * copy_count);
-
- int start_index = pf->index;
- int offset = start_index & 3;
- start_index += 4 - offset;
-
- for (int i = 0; i < offset; ++i)
- pf->in_FIFO[pf->index + i] /= 32768.0f;
-
- int extra = copy_count & 3;
- copy_count = copy_count / 4 - extra;
- __m128* in_FIFO = (__m128*)(pf->in_FIFO + pf->index + offset);
- TS_ASSERT(!((size_t)in_FIFO & 15));
- __m128 int16_max = _mm_set_ps1(32768.0f);
-
- for (int i = 0; i < copy_count; ++i)
- {
- __m128 val = in_FIFO[i];
- __m128 div = _mm_div_ps(val, int16_max);
- in_FIFO[i] = div;
- }
-
- for (int i = 0, copy_count4 = copy_count * 4; i < extra; ++i)
- {
- int index = copy_count4 + i;
- pf->in_FIFO[pf->index + index] /= 32768.0f;
- }
-
- TS_ASSERT(!((size_t)out_samples & 15));
- __m128* out_samples4 = (__m128*)out_samples;
- for (int i = 0; i < copy_count; ++i)
- {
- __m128 val = out_samples4[i];
- __m128 mul = _mm_mul_ps(val, int16_max);
- out_samples4[i] = mul;
- }
-
- for (int i = 0, copy_count4 = copy_count * 4; i < extra; ++i)
- {
- int index = copy_count4 + i;
- out_samples[index] *= 32768.0f;
- }
-
- copy_count = copy_count * 4 + extra;
- num_samples_to_process -= copy_count;
- pf->index += copy_count;
- indata += copy_count;
- out_samples += copy_count;
-
- if (pf->index >= TS_PITCH_FRAME_SIZE)
- {
- pf->index = TS_OVERLAP;
- {
- __m128* fft_data = (__m128*)pf->fft_data;
- __m128* in_FIFO = (__m128*)pf->in_FIFO;
-
- for (int k = 0; k < TS_PITCH_FRAME_SIZE / 4; k++)
- {
- __m128 von_hann = tsVonHann4(k);
- __m128 sample = in_FIFO[k];
- __m128 windowed_sample = _mm_mul_ps(sample, von_hann);
- fft_data[k] = windowed_sample;
- }
- }
-
- memset(pf->fft_data + TS_PITCH_FRAME_SIZE, 0, TS_PITCH_FRAME_SIZE * sizeof(float));
- tsFFT(pf->fft_data, pf->fft_data + TS_PITCH_FRAME_SIZE, TS_PITCH_FRAME_SIZE, 1.0f);
-
- {
- __m128* fft_data = (__m128*)pf->fft_data;
- __m128* previous_phase = (__m128*)pf->previous_phase;
- __m128* magnitudes = (__m128*)pf->mag;
- __m128* frequencies = (__m128*)pf->freq;
- int simd_count = (TS_PITCH_FRAME_SIZE / 2) / 4;
-
- for (int k = 0; k <= simd_count; k++)
- {
- __m128 real = fft_data[k];
- __m128 imag = fft_data[(TS_PITCH_FRAME_SIZE / 4) + k];
- __m128 overlap_phase = _mm_set_ps((float)(k * 4 + 3) * TS_EXPECTED_FREQUENCY, (float)(k * 4 + 2) * TS_EXPECTED_FREQUENCY, (float)(k * 4 + 1) * TS_EXPECTED_FREQUENCY, (float)(k * 4) * TS_EXPECTED_FREQUENCY);
- __m128 k4 = _mm_set_ps((float)(k * 4 + 3), (float)(k * 4 + 2), (float)(k * 4 + 1), (float)(k * 4));
-
- __m128 mag = _mm_mul_ps(_mm_set_ps1(2.0f), _mm_sqrt_ps(_mm_add_ps(_mm_mul_ps(real, real), _mm_mul_ps(imag, imag))));
- __m128 phase = _mm_atan2_ps(imag, real);
- __m128 phase_dif = _mm_sub_ps(phase, previous_phase[k]);
-
- previous_phase[k] = phase;
- phase_dif = _mm_sub_ps(phase_dif, overlap_phase);
-
- // map delta phase into +/- pi interval
- __m128i qpd = _mm_cvttps_epi32(_mm_div_ps(phase_dif, pi));
- __m128i zero = _mm_setzero_si128();
- __m128i ltzero_mask = _mm_cmplt_epi32(qpd, zero);
- __m128i ones_bit = _mm_and_si128(qpd, _mm_set1_epi32(1));
- __m128i neg_qpd = _mm_sub_epi32(qpd, ones_bit);
- __m128i pos_qpd = _mm_add_epi32(qpd, ones_bit);
- qpd = select_si(pos_qpd, neg_qpd, ltzero_mask);
- __m128 pi_range_offset = _mm_mul_ps(pi, _mm_cvtepi32_ps(qpd));
- phase_dif = _mm_sub_ps(phase_dif, pi_range_offset);
-
- __m128 deviation = _mm_div_ps(_mm_mul_ps(_mm_set_ps1((float)TS_PITCH_QUALITY), phase_dif), two_pi);
- __m128 true_freq_estimated = _mm_add_ps(_mm_mul_ps(k4, freq_per_bin), _mm_mul_ps(deviation, freq_per_bin));
-
- magnitudes[k] = mag;
- frequencies[k] = true_freq_estimated;
- }
- }
-
- // actual pitch shifting work
- // shift frequencies into workspace
- memset(pf->pitch_shift_workspace, 0, (TS_PITCH_FRAME_SIZE / 2) * sizeof(float));
- for (int k = 0; k <= TS_PITCH_FRAME_SIZE / 2; k++)
- {
- int index = (int)(k * pitchShift);
- if (index <= TS_PITCH_FRAME_SIZE / 2)
- pf->pitch_shift_workspace[index] = pf->freq[k] * pitchShift;
- }
-
- // swap buffers around to reuse old pf->preq buffer as the new workspace
- float* frequencies = pf->pitch_shift_workspace;
- float* pitch_shift_workspace = pf->freq;
- float* magnitudes = pf->mag;
-
- // shift magnitudes into workspace
- memset(pitch_shift_workspace, 0, TS_PITCH_FRAME_SIZE * sizeof(float));
- for (int k = 0; k <= TS_PITCH_FRAME_SIZE / 2; k++)
- {
- int index = (int)(k * pitchShift);
- if (index <= TS_PITCH_FRAME_SIZE / 2)
- pitch_shift_workspace[index] += magnitudes[k];
- }
-
- // track where the shifted magnitudes are
- magnitudes = pitch_shift_workspace;
-
- {
- __m128* magnitudes4 = (__m128*)magnitudes;
- __m128* frequencies4 = (__m128*)frequencies;
- __m128* fft_data = (__m128*)pf->fft_data;
- __m128* sum_phase = (__m128*)pf->sum_phase;
- int simd_count = (TS_PITCH_FRAME_SIZE / 2) / 4;
-
- for (int k = 0; k <= simd_count; k++)
- {
- __m128 mag = magnitudes4[k];
- __m128 freq = frequencies4[k];
- __m128 freq_per_bin_k = _mm_set_ps((float)(k * 4 + 3) * freqPerBin, (float)(k * 4 + 2) * freqPerBin, (float)(k * 4 + 1) * freqPerBin, (float)(k * 4) * freqPerBin);
-
- freq = _mm_sub_ps(freq, freq_per_bin_k);
- freq = _mm_div_ps(freq, freq_per_bin);
-
- freq = _mm_mul_ps(two_pi, freq);
- freq = _mm_div_ps(freq, pitch_quality);
-
- __m128 overlap_phase = _mm_set_ps((float)(k * 4 + 3) * TS_EXPECTED_FREQUENCY, (float)(k * 4 + 2) * TS_EXPECTED_FREQUENCY, (float)(k * 4 + 1) * TS_EXPECTED_FREQUENCY, (float)(k * 4) * TS_EXPECTED_FREQUENCY);
- freq = _mm_add_ps(freq, overlap_phase);
-
- __m128 phase = sum_phase[k];
- phase = _mm_add_ps(phase, freq);
- sum_phase[k] = phase;
-
- __m128 c, s;
- _mm_sincos_ps(phase, &s, &c);
- __m128 real = _mm_mul_ps(mag, c);
- __m128 imag = _mm_mul_ps(mag, s);
-
- fft_data[k] = real;
- fft_data[(TS_PITCH_FRAME_SIZE / 4) + k] = imag;
- }
- }
-
- for (int k = TS_PITCH_FRAME_SIZE + 2; k < 2 * TS_PITCH_FRAME_SIZE - 2; ++k)
- pf->fft_data[k] = 0;
-
- tsFFT(pf->fft_data, pf->fft_data + TS_PITCH_FRAME_SIZE, TS_PITCH_FRAME_SIZE, -1);
-
- {
- __m128* fft_data = (__m128*)pf->fft_data;
- __m128* window_accumulator = (__m128*)pf->window_accumulator;
-
- for (int k = 0; k < TS_PITCH_FRAME_SIZE / 4; ++k)
- {
- __m128 von_hann = tsVonHann4(k);
- __m128 fft_data_segment = fft_data[k];
- __m128 accumulator_segment = window_accumulator[k];
- __m128 divisor = _mm_div_ps(pitch_quality, _mm_set_ps1(8.0f));
- fft_data_segment = _mm_mul_ps(von_hann, fft_data_segment);
- fft_data_segment = _mm_div_ps(fft_data_segment, divisor);
- accumulator_segment = _mm_add_ps(accumulator_segment, fft_data_segment);
- window_accumulator[k] = accumulator_segment;
- }
- }
-
- memcpy(pf->out_FIFO, pf->window_accumulator, TS_STEPSIZE * sizeof(float));
- memmove(pf->window_accumulator, pf->window_accumulator + TS_STEPSIZE, TS_PITCH_FRAME_SIZE * sizeof(float));
- memmove(pf->in_FIFO, pf->in_FIFO + TS_STEPSIZE, TS_OVERLAP * sizeof(float));
- }
- }
-}
-
-/*
-zlib license:
-
-Copyright (c) 2017 Randy Gaul http://www.randygaul.net
-
-This software is provided 'as-is', without any express or implied warranty.
-In no event will the authors be held liable for any damages arising from
-the use of this software.
-
-Permission is granted to anyone to use this software for any purpose,
-including commercial applications, and to alter it and redistribute it
-freely, subject to the following restrictions:
-1. The origin of this software must not be misrepresented; you must not
-claim that you wrote the original software. If you use this software
-in a product, an acknowledgment in the product documentation would be
-appreciated but is not required.
-2. Altered source versions must be plainly marked as such, and must not
-be misrepresented as being the original software.
-3. This notice may not be removed or altered from any source distribution.
-*/
-
-#endif
diff --git a/src/lua/audio/luaopen_audio.cpp b/src/lua/audio/luaopen_audio.cpp
index 1378f89..4c9b5a7 100644
--- a/src/lua/audio/luaopen_audio.cpp
+++ b/src/lua/audio/luaopen_audio.cpp
@@ -1,3 +1,5 @@
+#include <SDL2/SDL.h>
+
#include "libs/luax/luax.h"
#include "audio/audio.h"
@@ -7,8 +9,12 @@ namespace lua
{
static int l_init(lua_State* L)
{
-
- return 0;
+ if (SDL_Init(SDL_INIT_AUDIO) < 0)
+ {
+ luax_error(L, "could not init audio");
+ luax_pushboolean(L, false);
+ return 1;
+ }
}
static int l_newSound(lua_State* L)
@@ -25,7 +31,7 @@ namespace lua
int luaopen_audio(lua_State* L)
{
-
+
return 1;
}
}
diff --git a/src/lua/embed/debug.lua.h b/src/lua/embed/debug.lua.h
index f3838a0..7ccc99d 100644
--- a/src/lua/embed/debug.lua.h
+++ b/src/lua/embed/debug.lua.h
@@ -1,125 +1,132 @@
/* debug.lua */
-static const char debug_lua[] =
-{45,45,91,91,32,13,10,32,32,32,32,102,111,114,32,100,101,98,117,103,32,112,117,
-114,112,111,115,101,32,13,10,32,32,32,32,43,45,45,45,45,45,45,45,45,45,45,45,
-45,45,45,45,45,45,45,45,43,13,10,32,32,32,32,124,100,101,98,117,103,32,109,
-115,103,32,111,108,100,32,32,32,32,32,32,124,13,10,32,32,32,32,124,46,46,46,
-32,32,32,32,32,32,32,32,32,32,32,32,32,32,32,32,124,13,10,32,32,32,32,124,46,
-46,46,32,32,32,32,32,32,32,32,32,32,32,32,32,32,32,32,124,13,10,32,32,32,32,
-124,46,46,46,32,32,32,32,32,32,32,32,32,32,32,32,32,32,32,32,124,13,10,32,32,
-32,32,124,100,101,98,117,103,32,109,115,103,32,110,101,119,32,32,32,32,32,32,
-124,13,10,32,32,32,32,43,45,45,45,45,45,45,45,45,45,45,45,45,45,45,45,45,45,
-45,45,43,13,10,93,93,32,13,10,13,10,106,105,110,46,100,101,98,117,103,32,61,
-32,106,105,110,46,100,101,98,117,103,32,111,114,32,123,125,32,13,10,13,10,45,
-45,32,114,101,110,100,101,114,32,112,97,110,101,108,32,13,10,108,111,99,97,
-108,32,112,97,110,101,108,32,61,32,110,105,108,32,13,10,13,10,108,111,99,97,
-108,32,100,101,98,117,103,32,61,32,102,97,108,115,101,13,10,13,10,45,45,32,
-100,101,98,117,103,32,109,115,103,32,98,117,102,102,101,114,32,13,10,108,111,
-99,97,108,32,98,117,102,102,101,114,32,61,32,123,125,32,13,10,13,10,45,45,32,
-99,111,110,102,105,103,117,114,101,32,13,10,108,111,99,97,108,32,98,115,105,
-122,101,32,32,32,61,32,49,48,13,10,108,111,99,97,108,32,102,115,105,122,101,
-32,32,32,61,32,49,53,13,10,108,111,99,97,108,32,108,104,101,105,103,104,116,
-32,61,32,49,56,13,10,108,111,99,97,108,32,97,108,112,104,97,32,32,32,61,32,50,
-50,48,13,10,108,111,99,97,108,32,109,97,114,103,105,110,32,32,61,32,49,48,13,
-10,13,10,45,45,32,114,101,102,114,101,115,104,32,98,117,102,102,101,114,32,
-111,114,32,110,111,116,32,13,10,108,111,99,97,108,32,114,101,102,114,101,115,
-104,32,61,32,116,114,117,101,32,13,10,13,10,102,117,110,99,116,105,111,110,32,
-106,105,110,46,100,101,98,117,103,46,105,110,105,116,40,41,13,10,32,32,32,32,
-100,101,98,117,103,32,61,32,116,114,117,101,13,10,9,112,97,110,101,108,32,61,
-32,106,105,110,46,103,114,97,112,104,105,99,115,46,67,97,110,118,97,115,40,
-106,105,110,46,103,114,97,112,104,105,99,115,46,115,105,122,101,40,41,41,32,
-13,10,101,110,100,13,10,13,10,45,45,32,115,101,116,32,98,117,102,102,101,114,
-32,115,105,122,101,32,13,10,102,117,110,99,116,105,111,110,32,106,105,110,46,
-100,101,98,117,103,46,115,105,122,101,40,99,41,13,10,9,98,115,105,122,101,32,
-61,32,99,13,10,101,110,100,32,13,10,13,10,102,117,110,99,116,105,111,110,32,
-106,105,110,46,100,101,98,117,103,46,112,114,105,110,116,40,109,115,103,41,13,
-10,9,105,102,32,110,111,116,32,100,101,98,117,103,32,116,104,101,110,32,114,
-101,116,117,114,110,32,101,110,100,32,13,10,13,10,9,109,115,103,32,61,32,116,
-111,115,116,114,105,110,103,40,109,115,103,41,13,10,9,108,111,99,97,108,32,
-116,112,32,61,32,116,121,112,101,40,109,115,103,41,13,10,9,105,102,32,116,112,
-32,126,61,32,34,115,116,114,105,110,103,34,32,97,110,100,32,116,112,32,126,61,
-32,34,110,117,109,98,101,114,34,32,116,104,101,110,32,13,10,9,9,109,115,103,
-32,61,32,115,116,114,105,110,103,46,102,111,114,109,97,116,40,34,112,114,105,
-110,116,32,102,97,105,108,101,100,44,32,101,120,112,101,99,116,32,115,116,114,
-105,110,103,32,111,114,32,110,117,109,98,101,114,32,98,117,116,32,103,101,116,
-32,97,32,37,115,34,44,32,116,112,41,13,10,9,101,110,100,13,10,9,13,10,32,32,
-32,32,45,45,32,114,101,109,111,118,101,32,116,104,101,32,102,105,114,115,116,
-32,111,110,101,32,40,111,108,100,32,109,115,103,41,13,10,9,105,102,32,35,98,
-117,102,102,101,114,32,62,61,32,98,115,105,122,101,32,116,104,101,110,32,13,
-10,9,9,116,97,98,108,101,46,114,101,109,111,118,101,40,98,117,102,102,101,114,
-44,32,49,41,13,10,9,101,110,100,32,13,10,32,32,32,32,13,10,32,32,32,32,98,117,
-102,102,101,114,91,35,98,117,102,102,101,114,32,43,32,49,93,32,61,32,109,115,
-103,13,10,32,32,32,32,114,101,102,114,101,115,104,32,61,32,116,114,117,101,13,
-10,101,110,100,13,10,13,10,45,45,32,99,108,101,97,114,32,100,101,98,117,103,
-32,98,117,102,102,101,114,32,13,10,102,117,110,99,116,105,111,110,32,106,105,
-110,46,100,101,98,117,103,46,99,108,101,97,114,40,41,13,10,9,98,117,102,102,
-101,114,32,61,32,123,125,32,13,10,101,110,100,13,10,13,10,108,111,99,97,108,
-32,102,117,110,99,116,105,111,110,32,103,101,116,83,116,114,72,101,105,103,
-104,116,40,115,116,114,44,32,108,104,101,105,103,104,116,41,32,13,10,9,108,
-111,99,97,108,32,104,32,61,32,108,104,101,105,103,104,116,13,10,9,105,102,32,
-35,115,116,114,32,61,61,32,48,32,116,104,101,110,32,13,10,9,9,104,32,61,32,48,
-13,10,9,101,110,100,32,13,10,9,102,111,114,32,105,32,61,32,49,44,32,35,115,
-116,114,32,100,111,32,13,10,9,9,108,111,99,97,108,32,99,32,61,32,115,116,114,
-105,110,103,46,115,117,98,40,115,116,114,44,32,105,44,32,105,41,13,10,9,9,105,
-102,32,99,32,61,61,32,39,92,110,39,32,116,104,101,110,32,13,10,9,9,9,104,32,
-61,32,104,32,43,32,108,104,101,105,103,104,116,13,10,9,9,101,110,100,32,13,10,
-9,101,110,100,32,13,10,9,114,101,116,117,114,110,32,104,32,13,10,101,110,100,
-13,10,13,10,108,111,99,97,108,32,32,102,117,110,99,116,105,111,110,32,103,101,
-116,66,103,81,117,97,100,40,41,32,13,10,9,108,111,99,97,108,32,119,105,100,
-116,104,44,32,104,101,105,103,104,116,32,61,32,48,44,32,48,32,9,13,10,9,102,
-111,114,32,105,32,61,32,49,44,32,35,98,117,102,102,101,114,32,100,111,13,10,9,
-9,108,111,99,97,108,32,119,44,32,104,32,61,32,106,105,110,46,103,114,97,112,
-104,105,99,115,46,98,111,120,40,32,98,117,102,102,101,114,91,105,93,44,32,102,
-115,105,122,101,44,32,49,44,32,108,104,101,105,103,104,116,41,13,10,9,9,104,
-101,105,103,104,116,32,61,32,104,101,105,103,104,116,32,43,32,104,32,13,10,9,
-9,105,102,32,119,105,100,116,104,32,60,32,119,32,116,104,101,110,13,10,9,9,9,
-119,105,100,116,104,32,61,32,119,32,13,10,9,9,101,110,100,32,13,10,9,101,110,
-100,9,13,10,9,114,101,116,117,114,110,32,119,105,100,116,104,44,32,104,101,
-105,103,104,116,13,10,101,110,100,32,13,10,13,10,45,45,32,114,101,110,100,101,
-114,32,116,111,32,115,99,114,101,101,110,13,10,102,117,110,99,116,105,111,110,
-32,106,105,110,46,100,101,98,117,103,46,114,101,110,100,101,114,40,41,32,13,
-10,32,32,32,32,105,102,32,110,111,116,32,100,101,98,117,103,32,116,104,101,
-110,32,114,101,116,117,114,110,32,101,110,100,13,10,32,32,32,32,13,10,32,32,
-32,32,105,102,32,114,101,102,114,101,115,104,32,116,104,101,110,32,13,10,32,
-32,32,32,32,32,32,13,10,32,32,32,32,32,32,32,32,106,105,110,46,103,114,97,112,
-104,105,99,115,46,98,105,110,100,40,112,97,110,101,108,41,13,10,13,10,32,32,
-32,32,32,32,32,32,32,32,32,32,106,105,110,46,103,114,97,112,104,105,99,115,46,
-99,108,101,97,114,40,48,44,32,48,44,32,48,44,32,48,41,13,10,32,32,32,32,32,32,
-32,32,32,32,32,32,13,10,32,32,32,32,32,32,32,32,32,32,32,32,106,105,110,46,
-103,114,97,112,104,105,99,115,46,115,116,117,100,121,40,41,13,10,32,32,32,32,
-32,32,32,32,32,32,32,32,13,10,32,32,32,32,32,32,32,32,32,32,32,32,108,111,99,
-97,108,32,119,119,44,32,119,104,32,61,32,106,105,110,46,103,114,97,112,104,
-105,99,115,46,115,105,122,101,40,41,13,10,32,32,32,32,32,32,32,32,32,32,32,32,
-108,111,99,97,108,32,98,103,119,44,32,98,103,104,32,61,32,103,101,116,66,103,
-81,117,97,100,40,41,13,10,32,32,32,32,32,32,32,32,32,32,32,32,106,105,110,46,
-103,114,97,112,104,105,99,115,46,99,111,108,111,114,40,48,44,32,48,44,32,48,
-44,32,97,108,112,104,97,41,13,10,32,32,32,32,32,32,32,32,32,32,32,32,106,105,
-110,46,103,114,97,112,104,105,99,115,46,114,101,99,116,40,34,102,105,108,108,
-34,44,32,48,44,32,119,104,32,45,32,98,103,104,32,45,32,109,97,114,103,105,110,
-44,32,98,103,119,32,43,32,109,97,114,103,105,110,44,32,98,103,104,32,43,32,
-109,97,114,103,105,110,41,13,10,32,32,32,32,32,32,32,32,32,32,32,32,13,10,32,
-32,32,32,32,32,32,32,32,32,32,32,106,105,110,46,103,114,97,112,104,105,99,115,
-46,99,111,108,111,114,40,41,13,10,32,32,32,32,32,32,32,32,32,32,32,32,108,111,
-99,97,108,32,121,32,61,32,119,104,32,13,10,32,32,32,32,32,32,32,32,32,32,32,
-32,102,111,114,32,105,32,61,32,35,98,117,102,102,101,114,44,32,49,44,32,45,49,
-32,100,111,13,10,32,32,32,32,32,32,32,32,32,32,32,32,32,32,32,32,108,111,99,
-97,108,32,109,115,103,32,61,32,98,117,102,102,101,114,91,105,93,32,13,10,32,
-32,32,32,32,32,32,32,32,32,32,32,32,32,32,32,108,111,99,97,108,32,104,32,61,
-32,103,101,116,83,116,114,72,101,105,103,104,116,40,109,115,103,44,32,108,104,
-101,105,103,104,116,41,13,10,32,32,32,32,32,32,32,32,32,32,32,32,32,32,32,32,
-121,32,61,32,121,32,45,32,104,32,13,10,32,32,32,32,32,32,32,32,32,32,32,32,32,
-32,32,32,106,105,110,46,103,114,97,112,104,105,99,115,46,119,114,105,116,101,
-40,109,115,103,44,32,109,97,114,103,105,110,32,47,32,50,44,32,121,32,45,32,
-109,97,114,103,105,110,47,32,50,44,32,102,115,105,122,101,44,32,49,44,32,108,
-104,101,105,103,104,116,41,32,13,10,32,32,32,32,32,32,32,32,32,32,32,32,101,
-110,100,13,10,9,13,10,32,32,32,32,32,32,32,32,106,105,110,46,103,114,97,112,
-104,105,99,115,46,98,105,110,100,40,41,13,10,32,32,32,32,13,10,32,32,32,32,32,
-32,32,32,114,101,102,114,101,115,104,32,61,32,102,97,108,115,101,13,10,32,32,
-32,32,101,110,100,32,13,10,32,32,32,32,13,10,32,32,32,32,106,105,110,46,103,
-114,97,112,104,105,99,115,46,99,111,108,111,114,40,41,13,10,32,32,32,32,106,
-105,110,46,103,114,97,112,104,105,99,115,46,100,114,97,119,40,112,97,110,101,
-108,44,32,48,44,32,48,41,13,10,101,110,100,13,10,13,10,102,117,110,99,116,105,
-111,110,32,106,105,110,46,100,101,98,117,103,46,115,116,97,116,117,115,40,41,
-32,13,10,9,114,101,116,117,114,110,32,100,101,98,117,103,32,13,10,101,110,100,
-13,10};
+static const char* debug_lua = R"(
+--[[
+ for debug purpose
+ +-------------------+
+ |debug msg old |
+ |... |
+ |... |
+ |... |
+ |debug msg new |
+ +-------------------+
+]]
+jin.debug = jin.debug or {}
+
+-- render panel
+local panel = nil
+
+local debug = false
+
+-- debug msg buffer
+local buffer = {}
+
+-- configure
+local bsize = 10
+local fsize = 15
+local lheight = 18
+local alpha = 220
+local margin = 10
+
+-- refresh buffer or not
+local refresh = true
+
+function jin.debug.init()
+ debug = true
+ panel = jin.graphics.Canvas(jin.graphics.size())
+end
+
+-- set buffer size
+function jin.debug.size(c)
+ bsize = c
+end
+
+function jin.debug.print(msg)
+ if not debug then return end
+
+ msg = tostring(msg)
+ local tp = type(msg)
+ if tp ~= "string" and tp ~= "number" then
+ msg = string.format("print failed, expect string or number but get a %s", tp)
+ end
+
+ -- remove the first one (old msg)
+ if #buffer >= bsize then
+ table.remove(buffer, 1)
+ end
+
+ buffer[#buffer + 1] = msg
+ refresh = true
+end
+
+-- clear debug buffer
+function jin.debug.clear()
+ buffer = {}
+end
+
+local function getStrHeight(str, lheight)
+ local h = lheight
+ if #str == 0 then
+ h = 0
+ end
+ for i = 1, #str do
+ local c = string.sub(str, i, i)
+ if c == '\n' then
+ h = h + lheight
+ end
+ end
+ return h
+end
+
+local function getBgQuad()
+ local width, height = 0, 0
+ for i = 1, #buffer do
+ local w, h = jin.graphics.box( buffer[i], fsize, 1, lheight)
+ height = height + h
+ if width < w then
+ width = w
+ end
+ end
+ return width, height
+end
+
+-- render to screen
+function jin.debug.render()
+ if not debug then return end
+
+ if refresh then
+
+ jin.graphics.bind(panel)
+
+ jin.graphics.clear(0, 0, 0, 0)
+
+ jin.graphics.study()
+
+ local ww, wh = jin.graphics.size()
+ local bgw, bgh = getBgQuad()
+ jin.graphics.color(0, 0, 0, alpha)
+ jin.graphics.rect("fill", 0, wh - bgh - margin, bgw + margin, bgh + margin)
+
+ jin.graphics.color()
+ local y = wh
+ for i = #buffer, 1, -1 do
+ local msg = buffer[i]
+ local h = getStrHeight(msg, lheight)
+ y = y - h
+ jin.graphics.write(msg, margin / 2, y - margin/ 2, fsize, 1, lheight)
+ end
+
+ jin.graphics.bind()
+
+ refresh = false
+ end
+
+ jin.graphics.color()
+ jin.graphics.draw(panel, 0, 0)
+end
+
+function jin.debug.status()
+ return debug
+end
+
+)"; \ No newline at end of file
diff --git a/src/lua/embed/embed.h b/src/lua/embed/embed.h
index 685355f..2ef8b75 100644
--- a/src/lua/embed/embed.h
+++ b/src/lua/embed/embed.h
@@ -33,16 +33,16 @@ namespace embed
// embed scripts
const jin_Embed scripts[] = {
- { "graphics.lua", graphics_lua },
- { "keyboard.lua", keyboard_lua },
- { "mouse.lua", mouse_lua },
- { "debug.lua", debug_lua},
- { "boot.lua", boot_lua },
- { 0, 0 }
+ {"graphics.lua", graphics_lua},
+ {"keyboard.lua", keyboard_lua},
+ {"mouse.lua", mouse_lua},
+ {"debug.lua", debug_lua},
+ {"boot.lua", boot_lua},
+ {0, 0}
};
// load all emebd lua scripts
- for (int i = 0; scripts[i].fname; i++)
+ for (int i = 0; scripts[i].fname; ++i)
embed(L, scripts[i].source, scripts[i].fname);
}
}
diff --git a/src/lua/embed/graphics.lua.h b/src/lua/embed/graphics.lua.h
index 0e10e97..85cf979 100644
--- a/src/lua/embed/graphics.lua.h
+++ b/src/lua/embed/graphics.lua.h
@@ -1,8 +1,8 @@
/* graphics.lua */
-static const char graphics_lua[] =
-{45,45,45,45,45,45,45,45,45,45,45,45,45,45,45,45,45,13,10,45,45,32,106,105,110,
-46,103,114,97,112,104,105,99,115,32,13,10,45,45,45,45,45,45,45,45,45,45,45,45,
-45,45,45,45,45,13,10,13,10,106,105,110,46,103,114,97,112,104,105,99,115,32,61,
-32,106,105,110,46,103,114,97,112,104,105,99,115,32,111,114,32,123,125,32,13,
-10,13,10};
+static const char* graphics_lua = R"(
+-----------------
+-- jin.graphics
+-----------------
+jin.graphics = jin.graphics or {}
+)";
diff --git a/src/lua/embed/keyboard.lua.h b/src/lua/embed/keyboard.lua.h
index 037c255..66e3c2a 100644
--- a/src/lua/embed/keyboard.lua.h
+++ b/src/lua/embed/keyboard.lua.h
@@ -1,13 +1,20 @@
-static const char keyboard_lua[] =
-{ 45,45,91,91,32,13,10,9,107,101,121,98,111,97,114,100,32,101,120,116,101,110,
- 115,105,111,110,32,13,10,93,93,32,13,10,13,10,106,105,110,46,107,101,121,98,
- 111,97,114,100,32,61,32,106,105,110,46,107,101,121,98,111,97,114,100,32,111,
- 114,32,123,125,32,13,10,13,10,108,111,99,97,108,32,107,101,121,115,32,61,32,
- 123,125,32,13,10,13,10,102,117,110,99,116,105,111,110,32,106,105,110,46,107,
- 101,121,98,111,97,114,100,46,105,115,68,111,119,110,40,107,41,32,13,10,32,32,
- 32,32,114,101,116,117,114,110,32,107,101,121,115,91,107,93,13,10,101,110,100,
- 32,32,13,10,13,10,102,117,110,99,116,105,111,110,32,106,105,110,46,107,101,
- 121,98,111,97,114,100,46,115,101,116,40,107,44,32,115,116,97,116,117,115,41,
- 32,13,10,9,107,101,121,115,91,107,93,32,61,32,115,116,97,116,117,115,32,13,10,
- 101,110,100,32,13,10 };
+static const char* keyboard_lua = R"(
+--[[
+ jin.keyboard extension
+]]
+
+jin.keyboard = jin.keyboard or {}
+
+local keys = {}
+
+function jin.keyboard.isDown(k)
+ return keys[k]
+end
+
+function jin.keyboard.set(k, status)
+ keys[k] = status
+end
+
+
+)";
diff --git a/src/lua/embed/mouse.lua.h b/src/lua/embed/mouse.lua.h
index eb5a3ad..f57d08c 100644
--- a/src/lua/embed/mouse.lua.h
+++ b/src/lua/embed/mouse.lua.h
@@ -1,11 +1,18 @@
-static const char mouse_lua[] =
-{45,45,91,91,32,13,10,9,109,111,117,115,101,32,101,120,116,101,110,115,105,111,
-110,13,10,93,93,32,13,10,13,10,106,105,110,46,109,111,117,115,101,32,61,32,
-106,105,110,46,109,111,117,115,101,32,111,114,32,123,125,32,13,10,13,10,108,
-111,99,97,108,32,98,117,116,116,111,110,32,61,32,123,125,32,13,10,13,10,102,
-117,110,99,116,105,111,110,32,106,105,110,46,109,111,117,115,101,46,105,115,
-68,111,119,110,40,98,116,110,41,32,13,10,9,114,101,116,117,114,110,32,98,117,
-116,116,111,110,91,98,116,110,93,13,10,101,110,100,32,13,10,13,10,102,117,110,
-99,116,105,111,110,32,106,105,110,46,109,111,117,115,101,46,115,101,116,40,98,
-116,110,44,32,115,116,97,116,117,115,41,32,13,10,9,98,117,116,116,111,110,91,
-98,116,110,93,32,61,32,115,116,97,116,117,115,13,10,101,110,100,32,13,10};
+static const char* mouse_lua = R"(
+--[[
+ jin.mouse extension
+]]
+
+jin.mouse = jin.mouse or {}
+
+local button = {}
+
+function jin.mouse.isDown(btn)
+ return button[btn]
+end
+
+function jin.mouse.set(btn, status)
+ button[btn] = status
+end
+
+)"; \ No newline at end of file
diff --git a/src/lua/embed/path.lua.h b/src/lua/embed/path.lua.h
index 8a2968a..3ebeab1 100644
--- a/src/lua/embed/path.lua.h
+++ b/src/lua/embed/path.lua.h
@@ -1,21 +1,19 @@
/* path.lua */
-static const char path_lua[] =
-{45,45,91,91,32,13,10,32,32,32,32,106,105,110,46,112,97,116,104,32,101,120,116,
-101,110,115,105,111,110,13,10,93,93,32,13,10,13,10,106,105,110,46,112,97,116,
-104,32,61,32,106,105,110,46,112,97,116,104,32,111,114,32,123,125,32,13,10,13,
-10,45,45,32,103,97,109,101,32,114,111,111,116,32,100,105,114,101,99,116,111,
-114,121,32,13,10,106,105,110,46,95,114,111,111,116,32,61,32,110,105,108,32,13,
-10,13,10,108,111,99,97,108,32,102,117,110,99,116,105,111,110,32,105,115,102,
-117,108,108,40,112,97,116,104,41,32,13,10,32,32,32,32,13,10,101,110,100,32,13,
-10,13,10,45,45,32,109,101,114,103,101,32,115,117,98,32,112,97,116,104,32,105,
-110,116,111,32,111,110,101,32,13,10,108,111,99,97,108,32,102,117,110,99,116,
-105,111,110,32,109,101,114,103,101,40,46,46,46,41,32,13,10,32,32,32,32,13,10,
-101,110,100,32,13,10,13,10,45,45,32,114,101,116,117,114,110,32,102,117,108,
-108,32,112,97,116,104,32,111,102,32,97,32,103,105,118,101,110,32,112,97,116,
-104,32,13,10,102,117,110,99,116,105,111,110,32,106,105,110,46,112,97,116,104,
-46,102,117,108,108,40,112,97,116,104,41,13,10,32,32,32,32,108,111,99,97,108,
-32,114,111,111,116,32,61,32,106,105,110,46,95,100,105,114,32,46,46,32,39,47,
-39,32,46,46,32,106,105,110,46,95,97,114,103,118,91,50,93,13,10,32,32,32,32,
-114,101,116,117,114,110,32,114,111,111,116,32,46,46,32,39,47,39,32,46,46,32,
-112,97,116,104,32,13,10,101,110,100,13,10,13,10};
+static const char* path_lua = R"(
+--[[
+ jin.path extension
+]]
+jin.path = jin.path or {}
+
+-- game root directory
+jin._root = nil
+
+-- return full path of a given path
+function jin.path.full(path)
+ local root = jin._dir .. '/' .. jin._argv[2]
+ return root .. '/' .. path
+end
+
+
+)";
diff --git a/src/lua/event/luaopen_event.cpp b/src/lua/event/luaopen_event.cpp
index c12a969..a417b60 100644
--- a/src/lua/event/luaopen_event.cpp
+++ b/src/lua/event/luaopen_event.cpp
@@ -33,7 +33,7 @@ namespace lua
case SDL_QUIT:
luax_setfield_string(L, "type", "quit");
break;
-
+
case SDL_KEYDOWN:
luax_setfield_string(L, "type", "keydown");
luax_setfield_string(L, "key", SDL_GetKeyName(e.key.keysym.sym));
diff --git a/src/lua/filesystem/luaopen_filesystem.cpp b/src/lua/filesystem/luaopen_filesystem.cpp
index bad293e..daf858c 100644
--- a/src/lua/filesystem/luaopen_filesystem.cpp
+++ b/src/lua/filesystem/luaopen_filesystem.cpp
@@ -80,7 +80,7 @@ namespace lua
int size = tmp.size();
- for (int i = 0; i<size - 4; i++)
+ for (int i = 0; i<size - 4; ++i)
{
if (tmp[i] == '.')
{
@@ -99,7 +99,7 @@ namespace lua
tmp = filename;
size = tmp.size();
- for (int i = 0; i<size; i++)
+ for (int i = 0; i<size; ++i)
{
if (tmp[i] == '.')
{
diff --git a/src/lua/graphics/luaopen_graphics.cpp b/src/lua/graphics/luaopen_graphics.cpp
index 490f560..8ddd455 100644
--- a/src/lua/graphics/luaopen_graphics.cpp
+++ b/src/lua/graphics/luaopen_graphics.cpp
@@ -1,24 +1,25 @@
+#include <SDL2/SDL.h>
+
#include "libs/luax/luax.h"
+
#include "render/image.h"
#include "render/canvas.h"
#include "render/jsl.h"
#include "render/graphics.h"
#include "render/window.h"
#include "render/font.h"
-#include "../luaopen_types.h"
-#include "lua/embed/graphics.lua.h"
-#include "libs/GLee/GLee.h"
#include "fs/filesystem.h"
-#include <SDL2/SDL.h>
-using namespace jin::render;
-using namespace jin::fs;
+#include "../luaopen_types.h"
+#include "../embed/graphics.lua.h"
namespace jin
{
namespace lua
{
-
+ using namespace render;
+ using namespace fs;
+
/**
* jin.graphics context, storge some module
* shared variables.
@@ -224,14 +225,24 @@ namespace lua
return 0;
}
+ static int l_unbindCanvas(lua_State* L)
+ {
+ Canvas::unbind();
+ return 0;
+ }
+
static int l_useShader(lua_State* L)
{
+ if (luax_gettop(L) == 0)
+ {
+ JSLProgram::unuse();
+ return 0;
+ }
if (luax_istype(L, 1, TYPE_JSL))
{
/* is image */
JSLProgram* jsl = (JSLProgram*)luax_toudata(L, 1);
jsl->use();
-
}
else
{
@@ -378,7 +389,7 @@ namespace lua
return 1;
}
float* p = new float[2 * n];
- for (int i = 1; i <= 2 * n; i++)
+ for (int i = 1; i <= 2 * n; ++i)
p[i - 1] = luax_rawgetnumber(L, 3, i);
render::polygon(mode, p, n);
delete[] p;
@@ -494,6 +505,7 @@ namespace lua
{"study", l_study},
// bind canvas
{"bind", l_bindCanvas},
+ {"unbind", l_unbindCanvas},
// use shader
{"use", l_useShader},
{"unuse", l_unuseShader},
@@ -529,5 +541,5 @@ namespace lua
return 1;
}
-}
-}
+}// lua
+}// jin
diff --git a/src/lua/luaopen_jin.cpp b/src/lua/luaopen_jin.cpp
index 5c4edc8..e6d98a4 100644
--- a/src/lua/luaopen_jin.cpp
+++ b/src/lua/luaopen_jin.cpp
@@ -62,9 +62,7 @@ namespace lua
{"keyboard", luaopen_keyboard},
{"filesystem", luaopen_filesystem},
{"net", luaopen_net},
- /*
- {"audio", luaopen_audio}
- */
+ //{"audio", luaopen_audio},
{0, 0}
};
@@ -77,7 +75,7 @@ namespace lua
luax_justglobal(L, -1, MODULE_NAME);
// register submodules
- for (int i = 0; mods[i].name; i++)
+ for (int i = 0; mods[i].name; ++i)
{
// open submodules
mods[i].func(L);
diff --git a/src/main.cpp b/src/main.cpp
index fb77f6d..7417591 100644
--- a/src/main.cpp
+++ b/src/main.cpp
@@ -7,7 +7,6 @@
#include "libs/luax/luax.h"
#include "lua/luaopen_jin.h"
-using namespace jin::lua;
#include "fs/filesystem.h"
@@ -23,11 +22,11 @@ int main(int argc, char* argv[])
* open jin module, jin module is on the top
* of stack
*/
- luaopen_jin(L);
+ jin::lua::luaopen_jin(L);
// add args to global field
luax_newtable(L);
- for (int i = 0; i < argc; i++)
+ for (int i = 0; i < argc; ++i)
luax_setraw_string(L, -2, i + 1, argv[i]);
luax_setfield(L, -2, "_argv");
@@ -45,7 +44,7 @@ int main(int argc, char* argv[])
luax_setfield_string(L, "_dir", buffer);
// boot
- boot(L);
+ jin::lua::boot(L);
return 0;
}
diff --git a/src/render/font.cpp b/src/render/font.cpp
index 1c13a34..39d1e4e 100644
--- a/src/render/font.cpp
+++ b/src/render/font.cpp
@@ -97,7 +97,7 @@ namespace render
float factor = fheight / (float)PIXEL_HEIGHT;
- for (int i = 0; i < len; i++)
+ for (int i = 0; i < len; ++i)
{
char c = text[i];
if (c == '\n')
@@ -166,7 +166,7 @@ namespace render
float factor = fheight / (float)PIXEL_HEIGHT;
- for (int i = 0; i < len; i++)
+ for (int i = 0; i < len; ++i)
{
char c = str[i];
if (c == '\n')
diff --git a/src/render/graphics.cpp b/src/render/graphics.cpp
index 21bb0ec..15d8a9c 100644
--- a/src/render/graphics.cpp
+++ b/src/render/graphics.cpp
@@ -79,7 +79,7 @@ namespace render
void polygon_line(float* p, int count)
{
float* verts = new float[count * 4];
- for (int i = 0; i < count; i++)
+ for (int i = 0; i < count; ++i)
{
// each line has two point n,n+1
verts[i * 4] = p[i * 2];
diff --git a/src/render/jsl.cpp b/src/render/jsl.cpp
index 5bb9347..6fcee53 100644
--- a/src/render/jsl.cpp
+++ b/src/render/jsl.cpp
@@ -55,10 +55,9 @@ namespace render
glUseProgram(pid);
}
- void JSLProgram::unuse()
+ shared void JSLProgram::unuse()
{
glUseProgram(0);
-
}
void JSLProgram::sendFloat(const char* variable, float number)
diff --git a/src/utils/matrix.cpp b/src/utils/matrix.cpp
index 6a9c69e..b970ec0 100644
--- a/src/utils/matrix.cpp
+++ b/src/utils/matrix.cpp
@@ -162,7 +162,7 @@ namespace util
void Matrix::transform(vertex * dst, const vertex * src, int size) const
{
- for (int i = 0; i<size; i++)
+ for (int i = 0; i<size; ++i)
{
// Store in temp variables in case src = dst
float x = (e[0] * src[i].x) + (e[4] * src[i].y) + (0) + (e[12]);
generated by cgit v1.1-26-g67d0 at 2025-05-21 12:13:31 +0000