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Diffstat (limited to 'libs/raylib/src/external/dr_mp3.h')
-rw-r--r-- | libs/raylib/src/external/dr_mp3.h | 3977 |
1 files changed, 0 insertions, 3977 deletions
diff --git a/libs/raylib/src/external/dr_mp3.h b/libs/raylib/src/external/dr_mp3.h deleted file mode 100644 index 0ecd0d3..0000000 --- a/libs/raylib/src/external/dr_mp3.h +++ /dev/null @@ -1,3977 +0,0 @@ -/* -MP3 audio decoder. Choice of public domain or MIT-0. See license statements at the end of this file. -dr_mp3 - v0.4.7 - 2019-07-28 - -David Reid - mackron@gmail.com - -Based off minimp3 (https://github.com/lieff/minimp3) which is where the real work was done. See the bottom of this file for -differences between minimp3 and dr_mp3. -*/ - -/* -USAGE -===== -dr_mp3 is a single-file library. To use it, do something like the following in one .c file. - #define DR_MP3_IMPLEMENTATION - #include "dr_mp3.h" - -You can then #include this file in other parts of the program as you would with any other header file. To decode audio data, -do something like the following: - - drmp3 mp3; - if (!drmp3_init_file(&mp3, "MySong.mp3", NULL)) { - // Failed to open file - } - - ... - - drmp3_uint64 framesRead = drmp3_read_pcm_frames_f32(pMP3, framesToRead, pFrames); - -The drmp3 object is transparent so you can get access to the channel count and sample rate like so: - - drmp3_uint32 channels = mp3.channels; - drmp3_uint32 sampleRate = mp3.sampleRate; - -The third parameter of drmp3_init_file() in the example above allows you to control the output channel count and sample rate. It -is a pointer to a drmp3_config object. Setting any of the variables of this object to 0 will cause dr_mp3 to use defaults. - -The example above initializes a decoder from a file, but you can also initialize it from a block of memory and read and seek -callbacks with drmp3_init_memory() and drmp3_init() respectively. - -You do not need to do any annoying memory management when reading PCM frames - this is all managed internally. You can request -any number of PCM frames in each call to drmp3_read_pcm_frames_f32() and it will return as many PCM frames as it can, up to the -requested amount. - -You can also decode an entire file in one go with drmp3_open_and_read_f32(), drmp3_open_memory_and_read_f32() and -drmp3_open_file_and_read_f32(). - - -OPTIONS -======= -#define these options before including this file. - -#define DR_MP3_NO_STDIO - Disable drmp3_init_file(), etc. - -#define DR_MP3_NO_SIMD - Disable SIMD optimizations. -*/ - -#ifndef dr_mp3_h -#define dr_mp3_h - -#ifdef __cplusplus -extern "C" { -#endif - -#include <stddef.h> - -#if defined(_MSC_VER) && _MSC_VER < 1600 -typedef signed char drmp3_int8; -typedef unsigned char drmp3_uint8; -typedef signed short drmp3_int16; -typedef unsigned short drmp3_uint16; -typedef signed int drmp3_int32; -typedef unsigned int drmp3_uint32; -typedef signed __int64 drmp3_int64; -typedef unsigned __int64 drmp3_uint64; -#else -#include <stdint.h> -typedef int8_t drmp3_int8; -typedef uint8_t drmp3_uint8; -typedef int16_t drmp3_int16; -typedef uint16_t drmp3_uint16; -typedef int32_t drmp3_int32; -typedef uint32_t drmp3_uint32; -typedef int64_t drmp3_int64; -typedef uint64_t drmp3_uint64; -#endif -typedef drmp3_uint8 drmp3_bool8; -typedef drmp3_uint32 drmp3_bool32; -#define DRMP3_TRUE 1 -#define DRMP3_FALSE 0 - -#define DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME 1152 -#define DRMP3_MAX_SAMPLES_PER_FRAME (DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME*2) - -#ifdef _MSC_VER -#define DRMP3_INLINE __forceinline -#else -#ifdef __GNUC__ -#define DRMP3_INLINE __inline__ __attribute__((always_inline)) -#else -#define DRMP3_INLINE -#endif -#endif - -/* -Low Level Push API -================== -*/ -typedef struct -{ - int frame_bytes, channels, hz, layer, bitrate_kbps; -} drmp3dec_frame_info; - -typedef struct -{ - float mdct_overlap[2][9*32], qmf_state[15*2*32]; - int reserv, free_format_bytes; - unsigned char header[4], reserv_buf[511]; -} drmp3dec; - -/* Initializes a low level decoder. */ -void drmp3dec_init(drmp3dec *dec); - -/* Reads a frame from a low level decoder. */ -int drmp3dec_decode_frame(drmp3dec *dec, const unsigned char *mp3, int mp3_bytes, void *pcm, drmp3dec_frame_info *info); - -/* Helper for converting between f32 and s16. */ -void drmp3dec_f32_to_s16(const float *in, drmp3_int16 *out, int num_samples); - - - -/* -Main API (Pull API) -=================== -*/ -#ifndef DR_MP3_DEFAULT_CHANNELS -#define DR_MP3_DEFAULT_CHANNELS 2 -#endif -#ifndef DR_MP3_DEFAULT_SAMPLE_RATE -#define DR_MP3_DEFAULT_SAMPLE_RATE 44100 -#endif - -typedef struct drmp3_src drmp3_src; -typedef drmp3_uint64 (* drmp3_src_read_proc)(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, void* pUserData); /* Returns the number of frames that were read. */ - -typedef enum -{ - drmp3_src_algorithm_none, - drmp3_src_algorithm_linear -} drmp3_src_algorithm; - -#define DRMP3_SRC_CACHE_SIZE_IN_FRAMES 512 -typedef struct -{ - drmp3_src* pSRC; - float pCachedFrames[2 * DRMP3_SRC_CACHE_SIZE_IN_FRAMES]; - drmp3_uint32 cachedFrameCount; - drmp3_uint32 iNextFrame; -} drmp3_src_cache; - -typedef struct -{ - drmp3_uint32 sampleRateIn; - drmp3_uint32 sampleRateOut; - drmp3_uint32 channels; - drmp3_src_algorithm algorithm; - drmp3_uint32 cacheSizeInFrames; /* The number of frames to read from the client at a time. */ -} drmp3_src_config; - -struct drmp3_src -{ - drmp3_src_config config; - drmp3_src_read_proc onRead; - void* pUserData; - float bin[256]; - drmp3_src_cache cache; /* <-- For simplifying and optimizing client -> memory reading. */ - union - { - struct - { - double alpha; - drmp3_bool32 isPrevFramesLoaded : 1; - drmp3_bool32 isNextFramesLoaded : 1; - } linear; - } algo; -}; - -typedef enum -{ - drmp3_seek_origin_start, - drmp3_seek_origin_current -} drmp3_seek_origin; - -typedef struct -{ - drmp3_uint64 seekPosInBytes; /* Points to the first byte of an MP3 frame. */ - drmp3_uint64 pcmFrameIndex; /* The index of the PCM frame this seek point targets. */ - drmp3_uint16 mp3FramesToDiscard; /* The number of whole MP3 frames to be discarded before pcmFramesToDiscard. */ - drmp3_uint16 pcmFramesToDiscard; /* The number of leading samples to read and discard. These are discarded after mp3FramesToDiscard. */ -} drmp3_seek_point; - -/* -Callback for when data is read. Return value is the number of bytes actually read. - -pUserData [in] The user data that was passed to drmp3_init(), drmp3_open() and family. -pBufferOut [out] The output buffer. -bytesToRead [in] The number of bytes to read. - -Returns the number of bytes actually read. - -A return value of less than bytesToRead indicates the end of the stream. Do _not_ return from this callback until -either the entire bytesToRead is filled or you have reached the end of the stream. -*/ -typedef size_t (* drmp3_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead); - -/* -Callback for when data needs to be seeked. - -pUserData [in] The user data that was passed to drmp3_init(), drmp3_open() and family. -offset [in] The number of bytes to move, relative to the origin. Will never be negative. -origin [in] The origin of the seek - the current position or the start of the stream. - -Returns whether or not the seek was successful. - -Whether or not it is relative to the beginning or current position is determined by the "origin" parameter which -will be either drmp3_seek_origin_start or drmp3_seek_origin_current. -*/ -typedef drmp3_bool32 (* drmp3_seek_proc)(void* pUserData, int offset, drmp3_seek_origin origin); - -typedef struct -{ - drmp3_uint32 outputChannels; - drmp3_uint32 outputSampleRate; -} drmp3_config; - -typedef struct -{ - drmp3dec decoder; - drmp3dec_frame_info frameInfo; - drmp3_uint32 channels; - drmp3_uint32 sampleRate; - drmp3_read_proc onRead; - drmp3_seek_proc onSeek; - void* pUserData; - drmp3_uint32 mp3FrameChannels; /* The number of channels in the currently loaded MP3 frame. Internal use only. */ - drmp3_uint32 mp3FrameSampleRate; /* The sample rate of the currently loaded MP3 frame. Internal use only. */ - drmp3_uint32 pcmFramesConsumedInMP3Frame; - drmp3_uint32 pcmFramesRemainingInMP3Frame; - drmp3_uint8 pcmFrames[sizeof(float)*DRMP3_MAX_SAMPLES_PER_FRAME]; /* <-- Multipled by sizeof(float) to ensure there's enough room for DR_MP3_FLOAT_OUTPUT. */ - drmp3_uint64 currentPCMFrame; /* The current PCM frame, globally, based on the output sample rate. Mainly used for seeking. */ - drmp3_uint64 streamCursor; /* The current byte the decoder is sitting on in the raw stream. */ - drmp3_src src; - drmp3_seek_point* pSeekPoints; /* NULL by default. Set with drmp3_bind_seek_table(). Memory is owned by the client. dr_mp3 will never attempt to free this pointer. */ - drmp3_uint32 seekPointCount; /* The number of items in pSeekPoints. When set to 0 assumes to no seek table. Defaults to zero. */ - size_t dataSize; - size_t dataCapacity; - drmp3_uint8* pData; - drmp3_bool32 atEnd : 1; - struct - { - const drmp3_uint8* pData; - size_t dataSize; - size_t currentReadPos; - } memory; /* Only used for decoders that were opened against a block of memory. */ -} drmp3; - -/* -Initializes an MP3 decoder. - -onRead [in] The function to call when data needs to be read from the client. -onSeek [in] The function to call when the read position of the client data needs to move. -pUserData [in, optional] A pointer to application defined data that will be passed to onRead and onSeek. - -Returns true if successful; false otherwise. - -Close the loader with drmp3_uninit(). - -See also: drmp3_init_file(), drmp3_init_memory(), drmp3_uninit() -*/ -drmp3_bool32 drmp3_init(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, const drmp3_config* pConfig); - -/* -Initializes an MP3 decoder from a block of memory. - -This does not create a copy of the data. It is up to the application to ensure the buffer remains valid for -the lifetime of the drmp3 object. - -The buffer should contain the contents of the entire MP3 file. -*/ -drmp3_bool32 drmp3_init_memory(drmp3* pMP3, const void* pData, size_t dataSize, const drmp3_config* pConfig); - -#ifndef DR_MP3_NO_STDIO -/* -Initializes an MP3 decoder from a file. - -This holds the internal FILE object until drmp3_uninit() is called. Keep this in mind if you're caching drmp3 -objects because the operating system may restrict the number of file handles an application can have open at -any given time. -*/ -drmp3_bool32 drmp3_init_file(drmp3* pMP3, const char* filePath, const drmp3_config* pConfig); -#endif - -/* -Uninitializes an MP3 decoder. -*/ -void drmp3_uninit(drmp3* pMP3); - -/* -Reads PCM frames as interleaved 32-bit IEEE floating point PCM. - -Note that framesToRead specifies the number of PCM frames to read, _not_ the number of MP3 frames. -*/ -drmp3_uint64 drmp3_read_pcm_frames_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut); - -/* -Reads PCM frames as interleaved signed 16-bit integer PCM. - -Note that framesToRead specifies the number of PCM frames to read, _not_ the number of MP3 frames. -*/ -drmp3_uint64 drmp3_read_pcm_frames_s16(drmp3* pMP3, drmp3_uint64 framesToRead, drmp3_int16* pBufferOut); - -/* -Seeks to a specific frame. - -Note that this is _not_ an MP3 frame, but rather a PCM frame. -*/ -drmp3_bool32 drmp3_seek_to_pcm_frame(drmp3* pMP3, drmp3_uint64 frameIndex); - -/* -Calculates the total number of PCM frames in the MP3 stream. Cannot be used for infinite streams such as internet -radio. Runs in linear time. Returns 0 on error. -*/ -drmp3_uint64 drmp3_get_pcm_frame_count(drmp3* pMP3); - -/* -Calculates the total number of MP3 frames in the MP3 stream. Cannot be used for infinite streams such as internet -radio. Runs in linear time. Returns 0 on error. -*/ -drmp3_uint64 drmp3_get_mp3_frame_count(drmp3* pMP3); - -/* -Calculates the total number of MP3 and PCM frames in the MP3 stream. Cannot be used for infinite streams such as internet -radio. Runs in linear time. Returns 0 on error. - -This is equivalent to calling drmp3_get_mp3_frame_count() and drmp3_get_pcm_frame_count() except that it's more efficient. -*/ -drmp3_bool32 drmp3_get_mp3_and_pcm_frame_count(drmp3* pMP3, drmp3_uint64* pMP3FrameCount, drmp3_uint64* pPCMFrameCount); - -/* -Calculates the seekpoints based on PCM frames. This is slow. - -pSeekpoint count is a pointer to a uint32 containing the seekpoint count. On input it contains the desired count. -On output it contains the actual count. The reason for this design is that the client may request too many -seekpoints, in which case dr_mp3 will return a corrected count. - -Note that seektable seeking is not quite sample exact when the MP3 stream contains inconsistent sample rates. -*/ -drmp3_bool32 drmp3_calculate_seek_points(drmp3* pMP3, drmp3_uint32* pSeekPointCount, drmp3_seek_point* pSeekPoints); - -/* -Binds a seek table to the decoder. - -This does _not_ make a copy of pSeekPoints - it only references it. It is up to the application to ensure this -remains valid while it is bound to the decoder. - -Use drmp3_calculate_seek_points() to calculate the seek points. -*/ -drmp3_bool32 drmp3_bind_seek_table(drmp3* pMP3, drmp3_uint32 seekPointCount, drmp3_seek_point* pSeekPoints); - - -/* -Opens an decodes an entire MP3 stream as a single operation. - -pConfig is both an input and output. On input it contains what you want. On output it contains what you got. - -Free the returned pointer with drmp3_free(). -*/ -float* drmp3_open_and_read_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount); -drmp3_int16* drmp3_open_and_read_s16(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount); - -float* drmp3_open_memory_and_read_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount); -drmp3_int16* drmp3_open_memory_and_read_s16(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount); - -#ifndef DR_MP3_NO_STDIO -float* drmp3_open_file_and_read_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount); -drmp3_int16* drmp3_open_file_and_read_s16(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount); -#endif - -/* -Frees any memory that was allocated by a public drmp3 API. -*/ -void drmp3_free(void* p); - -#ifdef __cplusplus -} -#endif -#endif /* dr_mp3_h */ - - -/************************************************************************************************************************************************************ - ************************************************************************************************************************************************************ - - IMPLEMENTATION - - ************************************************************************************************************************************************************ - ************************************************************************************************************************************************************/ -#ifdef DR_MP3_IMPLEMENTATION -#include <stdlib.h> -#include <string.h> -#include <limits.h> /* For INT_MAX */ - -/* Disable SIMD when compiling with TCC for now. */ -#if defined(__TINYC__) -#define DR_MP3_NO_SIMD -#endif - -#define DRMP3_OFFSET_PTR(p, offset) ((void*)((drmp3_uint8*)(p) + (offset))) - -#define DRMP3_MAX_FREE_FORMAT_FRAME_SIZE 2304 /* more than ISO spec's */ -#ifndef DRMP3_MAX_FRAME_SYNC_MATCHES -#define DRMP3_MAX_FRAME_SYNC_MATCHES 10 -#endif - -#define DRMP3_MAX_L3_FRAME_PAYLOAD_BYTES DRMP3_MAX_FREE_FORMAT_FRAME_SIZE /* MUST be >= 320000/8/32000*1152 = 1440 */ - -#define DRMP3_MAX_BITRESERVOIR_BYTES 511 -#define DRMP3_SHORT_BLOCK_TYPE 2 -#define DRMP3_STOP_BLOCK_TYPE 3 -#define DRMP3_MODE_MONO 3 -#define DRMP3_MODE_JOINT_STEREO 1 -#define DRMP3_HDR_SIZE 4 -#define DRMP3_HDR_IS_MONO(h) (((h[3]) & 0xC0) == 0xC0) -#define DRMP3_HDR_IS_MS_STEREO(h) (((h[3]) & 0xE0) == 0x60) -#define DRMP3_HDR_IS_FREE_FORMAT(h) (((h[2]) & 0xF0) == 0) -#define DRMP3_HDR_IS_CRC(h) (!((h[1]) & 1)) -#define DRMP3_HDR_TEST_PADDING(h) ((h[2]) & 0x2) -#define DRMP3_HDR_TEST_MPEG1(h) ((h[1]) & 0x8) -#define DRMP3_HDR_TEST_NOT_MPEG25(h) ((h[1]) & 0x10) -#define DRMP3_HDR_TEST_I_STEREO(h) ((h[3]) & 0x10) -#define DRMP3_HDR_TEST_MS_STEREO(h) ((h[3]) & 0x20) -#define DRMP3_HDR_GET_STEREO_MODE(h) (((h[3]) >> 6) & 3) -#define DRMP3_HDR_GET_STEREO_MODE_EXT(h) (((h[3]) >> 4) & 3) -#define DRMP3_HDR_GET_LAYER(h) (((h[1]) >> 1) & 3) -#define DRMP3_HDR_GET_BITRATE(h) ((h[2]) >> 4) -#define DRMP3_HDR_GET_SAMPLE_RATE(h) (((h[2]) >> 2) & 3) -#define DRMP3_HDR_GET_MY_SAMPLE_RATE(h) (DRMP3_HDR_GET_SAMPLE_RATE(h) + (((h[1] >> 3) & 1) + ((h[1] >> 4) & 1))*3) -#define DRMP3_HDR_IS_FRAME_576(h) ((h[1] & 14) == 2) -#define DRMP3_HDR_IS_LAYER_1(h) ((h[1] & 6) == 6) - -#define DRMP3_BITS_DEQUANTIZER_OUT -1 -#define DRMP3_MAX_SCF (255 + DRMP3_BITS_DEQUANTIZER_OUT*4 - 210) -#define DRMP3_MAX_SCFI ((DRMP3_MAX_SCF + 3) & ~3) - -#define DRMP3_MIN(a, b) ((a) > (b) ? (b) : (a)) -#define DRMP3_MAX(a, b) ((a) < (b) ? (b) : (a)) - -#if !defined(DR_MP3_NO_SIMD) - -#if !defined(DR_MP3_ONLY_SIMD) && (defined(_M_X64) || defined(_M_ARM64) || defined(__x86_64__) || defined(__aarch64__)) -/* x64 always have SSE2, arm64 always have neon, no need for generic code */ -#define DR_MP3_ONLY_SIMD -#endif - -#if ((defined(_MSC_VER) && _MSC_VER >= 1400) && (defined(_M_IX86) || defined(_M_X64))) || ((defined(__i386__) || defined(__x86_64__)) && defined(__SSE2__)) -#if defined(_MSC_VER) -#include <intrin.h> -#endif -#include <emmintrin.h> -#define DRMP3_HAVE_SSE 1 -#define DRMP3_HAVE_SIMD 1 -#define DRMP3_VSTORE _mm_storeu_ps -#define DRMP3_VLD _mm_loadu_ps -#define DRMP3_VSET _mm_set1_ps -#define DRMP3_VADD _mm_add_ps -#define DRMP3_VSUB _mm_sub_ps -#define DRMP3_VMUL _mm_mul_ps -#define DRMP3_VMAC(a, x, y) _mm_add_ps(a, _mm_mul_ps(x, y)) -#define DRMP3_VMSB(a, x, y) _mm_sub_ps(a, _mm_mul_ps(x, y)) -#define DRMP3_VMUL_S(x, s) _mm_mul_ps(x, _mm_set1_ps(s)) -#define DRMP3_VREV(x) _mm_shuffle_ps(x, x, _MM_SHUFFLE(0, 1, 2, 3)) -typedef __m128 drmp3_f4; -#if defined(_MSC_VER) || defined(DR_MP3_ONLY_SIMD) -#define drmp3_cpuid __cpuid -#else -static __inline__ __attribute__((always_inline)) void drmp3_cpuid(int CPUInfo[], const int InfoType) -{ -#if defined(__PIC__) - __asm__ __volatile__( -#if defined(__x86_64__) - "push %%rbx\n" - "cpuid\n" - "xchgl %%ebx, %1\n" - "pop %%rbx\n" -#else - "xchgl %%ebx, %1\n" - "cpuid\n" - "xchgl %%ebx, %1\n" -#endif - : "=a" (CPUInfo[0]), "=r" (CPUInfo[1]), "=c" (CPUInfo[2]), "=d" (CPUInfo[3]) - : "a" (InfoType)); -#else - __asm__ __volatile__( - "cpuid" - : "=a" (CPUInfo[0]), "=b" (CPUInfo[1]), "=c" (CPUInfo[2]), "=d" (CPUInfo[3]) - : "a" (InfoType)); -#endif -} -#endif -static int drmp3_have_simd() -{ -#ifdef DR_MP3_ONLY_SIMD - return 1; -#else - static int g_have_simd; - int CPUInfo[4]; -#ifdef MINIMP3_TEST - static int g_counter; - if (g_counter++ > 100) - return 0; -#endif - if (g_have_simd) - goto end; - drmp3_cpuid(CPUInfo, 0); - if (CPUInfo[0] > 0) - { - drmp3_cpuid(CPUInfo, 1); - g_have_simd = (CPUInfo[3] & (1 << 26)) + 1; /* SSE2 */ - return g_have_simd - 1; - } - -end: - return g_have_simd - 1; -#endif -} -#elif defined(__ARM_NEON) || defined(__aarch64__) -#include <arm_neon.h> -#define DRMP3_HAVE_SIMD 1 -#define DRMP3_VSTORE vst1q_f32 -#define DRMP3_VLD vld1q_f32 -#define DRMP3_VSET vmovq_n_f32 -#define DRMP3_VADD vaddq_f32 -#define DRMP3_VSUB vsubq_f32 -#define DRMP3_VMUL vmulq_f32 -#define DRMP3_VMAC(a, x, y) vmlaq_f32(a, x, y) -#define DRMP3_VMSB(a, x, y) vmlsq_f32(a, x, y) -#define DRMP3_VMUL_S(x, s) vmulq_f32(x, vmovq_n_f32(s)) -#define DRMP3_VREV(x) vcombine_f32(vget_high_f32(vrev64q_f32(x)), vget_low_f32(vrev64q_f32(x))) -typedef float32x4_t drmp3_f4; -static int drmp3_have_simd() -{ /* TODO: detect neon for !DR_MP3_ONLY_SIMD */ - return 1; -} -#else -#define DRMP3_HAVE_SIMD 0 -#ifdef DR_MP3_ONLY_SIMD -#error DR_MP3_ONLY_SIMD used, but SSE/NEON not enabled -#endif -#endif - -#else - -#define DRMP3_HAVE_SIMD 0 - -#endif - -typedef struct -{ - const drmp3_uint8 *buf; - int pos, limit; -} drmp3_bs; - -typedef struct -{ - float scf[3*64]; - drmp3_uint8 total_bands, stereo_bands, bitalloc[64], scfcod[64]; -} drmp3_L12_scale_info; - -typedef struct -{ - drmp3_uint8 tab_offset, code_tab_width, band_count; -} drmp3_L12_subband_alloc; - -typedef struct -{ - const drmp3_uint8 *sfbtab; - drmp3_uint16 part_23_length, big_values, scalefac_compress; - drmp3_uint8 global_gain, block_type, mixed_block_flag, n_long_sfb, n_short_sfb; - drmp3_uint8 table_select[3], region_count[3], subblock_gain[3]; - drmp3_uint8 preflag, scalefac_scale, count1_table, scfsi; -} drmp3_L3_gr_info; - -typedef struct -{ - drmp3_bs bs; - drmp3_uint8 maindata[DRMP3_MAX_BITRESERVOIR_BYTES + DRMP3_MAX_L3_FRAME_PAYLOAD_BYTES]; - drmp3_L3_gr_info gr_info[4]; - float grbuf[2][576], scf[40], syn[18 + 15][2*32]; - drmp3_uint8 ist_pos[2][39]; -} drmp3dec_scratch; - -static void drmp3_bs_init(drmp3_bs *bs, const drmp3_uint8 *data, int bytes) -{ - bs->buf = data; - bs->pos = 0; - bs->limit = bytes*8; -} - -static drmp3_uint32 drmp3_bs_get_bits(drmp3_bs *bs, int n) -{ - drmp3_uint32 next, cache = 0, s = bs->pos & 7; - int shl = n + s; - const drmp3_uint8 *p = bs->buf + (bs->pos >> 3); - if ((bs->pos += n) > bs->limit) - return 0; - next = *p++ & (255 >> s); - while ((shl -= 8) > 0) - { - cache |= next << shl; - next = *p++; - } - return cache | (next >> -shl); -} - -static int drmp3_hdr_valid(const drmp3_uint8 *h) -{ - return h[0] == 0xff && - ((h[1] & 0xF0) == 0xf0 || (h[1] & 0xFE) == 0xe2) && - (DRMP3_HDR_GET_LAYER(h) != 0) && - (DRMP3_HDR_GET_BITRATE(h) != 15) && - (DRMP3_HDR_GET_SAMPLE_RATE(h) != 3); -} - -static int drmp3_hdr_compare(const drmp3_uint8 *h1, const drmp3_uint8 *h2) -{ - return drmp3_hdr_valid(h2) && - ((h1[1] ^ h2[1]) & 0xFE) == 0 && - ((h1[2] ^ h2[2]) & 0x0C) == 0 && - !(DRMP3_HDR_IS_FREE_FORMAT(h1) ^ DRMP3_HDR_IS_FREE_FORMAT(h2)); -} - -static unsigned drmp3_hdr_bitrate_kbps(const drmp3_uint8 *h) -{ - static const drmp3_uint8 halfrate[2][3][15] = { - { { 0,4,8,12,16,20,24,28,32,40,48,56,64,72,80 }, { 0,4,8,12,16,20,24,28,32,40,48,56,64,72,80 }, { 0,16,24,28,32,40,48,56,64,72,80,88,96,112,128 } }, - { { 0,16,20,24,28,32,40,48,56,64,80,96,112,128,160 }, { 0,16,24,28,32,40,48,56,64,80,96,112,128,160,192 }, { 0,16,32,48,64,80,96,112,128,144,160,176,192,208,224 } }, - }; - return 2*halfrate[!!DRMP3_HDR_TEST_MPEG1(h)][DRMP3_HDR_GET_LAYER(h) - 1][DRMP3_HDR_GET_BITRATE(h)]; -} - -static unsigned drmp3_hdr_sample_rate_hz(const drmp3_uint8 *h) -{ - static const unsigned g_hz[3] = { 44100, 48000, 32000 }; - return g_hz[DRMP3_HDR_GET_SAMPLE_RATE(h)] >> (int)!DRMP3_HDR_TEST_MPEG1(h) >> (int)!DRMP3_HDR_TEST_NOT_MPEG25(h); -} - -static unsigned drmp3_hdr_frame_samples(const drmp3_uint8 *h) -{ - return DRMP3_HDR_IS_LAYER_1(h) ? 384 : (1152 >> (int)DRMP3_HDR_IS_FRAME_576(h)); -} - -static int drmp3_hdr_frame_bytes(const drmp3_uint8 *h, int free_format_size) -{ - int frame_bytes = drmp3_hdr_frame_samples(h)*drmp3_hdr_bitrate_kbps(h)*125/drmp3_hdr_sample_rate_hz(h); - if (DRMP3_HDR_IS_LAYER_1(h)) - { - frame_bytes &= ~3; /* slot align */ - } - return frame_bytes ? frame_bytes : free_format_size; -} - -static int drmp3_hdr_padding(const drmp3_uint8 *h) -{ - return DRMP3_HDR_TEST_PADDING(h) ? (DRMP3_HDR_IS_LAYER_1(h) ? 4 : 1) : 0; -} - -#ifndef DR_MP3_ONLY_MP3 -static const drmp3_L12_subband_alloc *drmp3_L12_subband_alloc_table(const drmp3_uint8 *hdr, drmp3_L12_scale_info *sci) -{ - const drmp3_L12_subband_alloc *alloc; - int mode = DRMP3_HDR_GET_STEREO_MODE(hdr); - int nbands, stereo_bands = (mode == DRMP3_MODE_MONO) ? 0 : (mode == DRMP3_MODE_JOINT_STEREO) ? (DRMP3_HDR_GET_STEREO_MODE_EXT(hdr) << 2) + 4 : 32; - - if (DRMP3_HDR_IS_LAYER_1(hdr)) - { - static const drmp3_L12_subband_alloc g_alloc_L1[] = { { 76, 4, 32 } }; - alloc = g_alloc_L1; - nbands = 32; - } else if (!DRMP3_HDR_TEST_MPEG1(hdr)) - { - static const drmp3_L12_subband_alloc g_alloc_L2M2[] = { { 60, 4, 4 }, { 44, 3, 7 }, { 44, 2, 19 } }; - alloc = g_alloc_L2M2; - nbands = 30; - } else - { - static const drmp3_L12_subband_alloc g_alloc_L2M1[] = { { 0, 4, 3 }, { 16, 4, 8 }, { 32, 3, 12 }, { 40, 2, 7 } }; - int sample_rate_idx = DRMP3_HDR_GET_SAMPLE_RATE(hdr); - unsigned kbps = drmp3_hdr_bitrate_kbps(hdr) >> (int)(mode != DRMP3_MODE_MONO); - if (!kbps) /* free-format */ - { - kbps = 192; - } - - alloc = g_alloc_L2M1; - nbands = 27; - if (kbps < 56) - { - static const drmp3_L12_subband_alloc g_alloc_L2M1_lowrate[] = { { 44, 4, 2 }, { 44, 3, 10 } }; - alloc = g_alloc_L2M1_lowrate; - nbands = sample_rate_idx == 2 ? 12 : 8; - } else if (kbps >= 96 && sample_rate_idx != 1) - { - nbands = 30; - } - } - - sci->total_bands = (drmp3_uint8)nbands; - sci->stereo_bands = (drmp3_uint8)DRMP3_MIN(stereo_bands, nbands); - - return alloc; -} - -static void drmp3_L12_read_scalefactors(drmp3_bs *bs, drmp3_uint8 *pba, drmp3_uint8 *scfcod, int bands, float *scf) -{ - static const float g_deq_L12[18*3] = { -#define DRMP3_DQ(x) 9.53674316e-07f/x, 7.56931807e-07f/x, 6.00777173e-07f/x - DRMP3_DQ(3),DRMP3_DQ(7),DRMP3_DQ(15),DRMP3_DQ(31),DRMP3_DQ(63),DRMP3_DQ(127),DRMP3_DQ(255),DRMP3_DQ(511),DRMP3_DQ(1023),DRMP3_DQ(2047),DRMP3_DQ(4095),DRMP3_DQ(8191),DRMP3_DQ(16383),DRMP3_DQ(32767),DRMP3_DQ(65535),DRMP3_DQ(3),DRMP3_DQ(5),DRMP3_DQ(9) - }; - int i, m; - for (i = 0; i < bands; i++) - { - float s = 0; - int ba = *pba++; - int mask = ba ? 4 + ((19 >> scfcod[i]) & 3) : 0; - for (m = 4; m; m >>= 1) - { - if (mask & m) - { - int b = drmp3_bs_get_bits(bs, 6); - s = g_deq_L12[ba*3 - 6 + b % 3]*(1 << 21 >> b/3); - } - *scf++ = s; - } - } -} - -static void drmp3_L12_read_scale_info(const drmp3_uint8 *hdr, drmp3_bs *bs, drmp3_L12_scale_info *sci) -{ - static const drmp3_uint8 g_bitalloc_code_tab[] = { - 0,17, 3, 4, 5,6,7, 8,9,10,11,12,13,14,15,16, - 0,17,18, 3,19,4,5, 6,7, 8, 9,10,11,12,13,16, - 0,17,18, 3,19,4,5,16, - 0,17,18,16, - 0,17,18,19, 4,5,6, 7,8, 9,10,11,12,13,14,15, - 0,17,18, 3,19,4,5, 6,7, 8, 9,10,11,12,13,14, - 0, 2, 3, 4, 5,6,7, 8,9,10,11,12,13,14,15,16 - }; - const drmp3_L12_subband_alloc *subband_alloc = drmp3_L12_subband_alloc_table(hdr, sci); - - int i, k = 0, ba_bits = 0; - const drmp3_uint8 *ba_code_tab = g_bitalloc_code_tab; - - for (i = 0; i < sci->total_bands; i++) - { - drmp3_uint8 ba; - if (i == k) - { - k += subband_alloc->band_count; - ba_bits = subband_alloc->code_tab_width; - ba_code_tab = g_bitalloc_code_tab + subband_alloc->tab_offset; - subband_alloc++; - } - ba = ba_code_tab[drmp3_bs_get_bits(bs, ba_bits)]; - sci->bitalloc[2*i] = ba; - if (i < sci->stereo_bands) - { - ba = ba_code_tab[drmp3_bs_get_bits(bs, ba_bits)]; - } - sci->bitalloc[2*i + 1] = sci->stereo_bands ? ba : 0; - } - - for (i = 0; i < 2*sci->total_bands; i++) - { - sci->scfcod[i] = (drmp3_uint8)(sci->bitalloc[i] ? DRMP3_HDR_IS_LAYER_1(hdr) ? 2 : drmp3_bs_get_bits(bs, 2) : 6); - } - - drmp3_L12_read_scalefactors(bs, sci->bitalloc, sci->scfcod, sci->total_bands*2, sci->scf); - - for (i = sci->stereo_bands; i < sci->total_bands; i++) - { - sci->bitalloc[2*i + 1] = 0; - } -} - -static int drmp3_L12_dequantize_granule(float *grbuf, drmp3_bs *bs, drmp3_L12_scale_info *sci, int group_size) -{ - int i, j, k, choff = 576; - for (j = 0; j < 4; j++) - { - float *dst = grbuf + group_size*j; - for (i = 0; i < 2*sci->total_bands; i++) - { - int ba = sci->bitalloc[i]; - if (ba != 0) - { - if (ba < 17) - { - int half = (1 << (ba - 1)) - 1; - for (k = 0; k < group_size; k++) - { - dst[k] = (float)((int)drmp3_bs_get_bits(bs, ba) - half); - } - } else - { - unsigned mod = (2 << (ba - 17)) + 1; /* 3, 5, 9 */ - unsigned code = drmp3_bs_get_bits(bs, mod + 2 - (mod >> 3)); /* 5, 7, 10 */ - for (k = 0; k < group_size; k++, code /= mod) - { - dst[k] = (float)((int)(code % mod - mod/2)); - } - } - } - dst += choff; - choff = 18 - choff; - } - } - return group_size*4; -} - -static void drmp3_L12_apply_scf_384(drmp3_L12_scale_info *sci, const float *scf, float *dst) -{ - int i, k; - memcpy(dst + 576 + sci->stereo_bands*18, dst + sci->stereo_bands*18, (sci->total_bands - sci->stereo_bands)*18*sizeof(float)); - for (i = 0; i < sci->total_bands; i++, dst += 18, scf += 6) - { - for (k = 0; k < 12; k++) - { - dst[k + 0] *= scf[0]; - dst[k + 576] *= scf[3]; - } - } -} -#endif - -static int drmp3_L3_read_side_info(drmp3_bs *bs, drmp3_L3_gr_info *gr, const drmp3_uint8 *hdr) -{ - static const drmp3_uint8 g_scf_long[8][23] = { - { 6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54,0 }, - { 12,12,12,12,12,12,16,20,24,28,32,40,48,56,64,76,90,2,2,2,2,2,0 }, - { 6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54,0 }, - { 6,6,6,6,6,6,8,10,12,14,16,18,22,26,32,38,46,54,62,70,76,36,0 }, - { 6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54,0 }, - { 4,4,4,4,4,4,6,6,8,8,10,12,16,20,24,28,34,42,50,54,76,158,0 }, - { 4,4,4,4,4,4,6,6,6,8,10,12,16,18,22,28,34,40,46,54,54,192,0 }, - { 4,4,4,4,4,4,6,6,8,10,12,16,20,24,30,38,46,56,68,84,102,26,0 } - }; - static const drmp3_uint8 g_scf_short[8][40] = { - { 4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 }, - { 8,8,8,8,8,8,8,8,8,12,12,12,16,16,16,20,20,20,24,24,24,28,28,28,36,36,36,2,2,2,2,2,2,2,2,2,26,26,26,0 }, - { 4,4,4,4,4,4,4,4,4,6,6,6,6,6,6,8,8,8,10,10,10,14,14,14,18,18,18,26,26,26,32,32,32,42,42,42,18,18,18,0 }, - { 4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,32,32,32,44,44,44,12,12,12,0 }, - { 4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 }, - { 4,4,4,4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,22,22,22,30,30,30,56,56,56,0 }, - { 4,4,4,4,4,4,4,4,4,4,4,4,6,6,6,6,6,6,10,10,10,12,12,12,14,14,14,16,16,16,20,20,20,26,26,26,66,66,66,0 }, - { 4,4,4,4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,12,12,12,16,16,16,20,20,20,26,26,26,34,34,34,42,42,42,12,12,12,0 } - }; - static const drmp3_uint8 g_scf_mixed[8][40] = { - { 6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 }, - { 12,12,12,4,4,4,8,8,8,12,12,12,16,16,16,20,20,20,24,24,24,28,28,28,36,36,36,2,2,2,2,2,2,2,2,2,26,26,26,0 }, - { 6,6,6,6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,14,14,14,18,18,18,26,26,26,32,32,32,42,42,42,18,18,18,0 }, - { 6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,32,32,32,44,44,44,12,12,12,0 }, - { 6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 }, - { 4,4,4,4,4,4,6,6,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,22,22,22,30,30,30,56,56,56,0 }, - { 4,4,4,4,4,4,6,6,4,4,4,6,6,6,6,6,6,10,10,10,12,12,12,14,14,14,16,16,16,20,20,20,26,26,26,66,66,66,0 }, - { 4,4,4,4,4,4,6,6,4,4,4,6,6,6,8,8,8,12,12,12,16,16,16,20,20,20,26,26,26,34,34,34,42,42,42,12,12,12,0 } - }; - - unsigned tables, scfsi = 0; - int main_data_begin, part_23_sum = 0; - int gr_count = DRMP3_HDR_IS_MONO(hdr) ? 1 : 2; - int sr_idx = DRMP3_HDR_GET_MY_SAMPLE_RATE(hdr); sr_idx -= (sr_idx != 0); - - if (DRMP3_HDR_TEST_MPEG1(hdr)) - { - gr_count *= 2; - main_data_begin = drmp3_bs_get_bits(bs, 9); - scfsi = drmp3_bs_get_bits(bs, 7 + gr_count); - } else - { - main_data_begin = drmp3_bs_get_bits(bs, 8 + gr_count) >> gr_count; - } - - do - { - if (DRMP3_HDR_IS_MONO(hdr)) - { - scfsi <<= 4; - } - gr->part_23_length = (drmp3_uint16)drmp3_bs_get_bits(bs, 12); - part_23_sum += gr->part_23_length; - gr->big_values = (drmp3_uint16)drmp3_bs_get_bits(bs, 9); - if (gr->big_values > 288) - { - return -1; - } - gr->global_gain = (drmp3_uint8)drmp3_bs_get_bits(bs, 8); - gr->scalefac_compress = (drmp3_uint16)drmp3_bs_get_bits(bs, DRMP3_HDR_TEST_MPEG1(hdr) ? 4 : 9); - gr->sfbtab = g_scf_long[sr_idx]; - gr->n_long_sfb = 22; - gr->n_short_sfb = 0; - if (drmp3_bs_get_bits(bs, 1)) - { - gr->block_type = (drmp3_uint8)drmp3_bs_get_bits(bs, 2); - if (!gr->block_type) - { - return -1; - } - gr->mixed_block_flag = (drmp3_uint8)drmp3_bs_get_bits(bs, 1); - gr->region_count[0] = 7; - gr->region_count[1] = 255; - if (gr->block_type == DRMP3_SHORT_BLOCK_TYPE) - { - scfsi &= 0x0F0F; - if (!gr->mixed_block_flag) - { - gr->region_count[0] = 8; - gr->sfbtab = g_scf_short[sr_idx]; - gr->n_long_sfb = 0; - gr->n_short_sfb = 39; - } else - { - gr->sfbtab = g_scf_mixed[sr_idx]; - gr->n_long_sfb = DRMP3_HDR_TEST_MPEG1(hdr) ? 8 : 6; - gr->n_short_sfb = 30; - } - } - tables = drmp3_bs_get_bits(bs, 10); - tables <<= 5; - gr->subblock_gain[0] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3); - gr->subblock_gain[1] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3); - gr->subblock_gain[2] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3); - } else - { - gr->block_type = 0; - gr->mixed_block_flag = 0; - tables = drmp3_bs_get_bits(bs, 15); - gr->region_count[0] = (drmp3_uint8)drmp3_bs_get_bits(bs, 4); - gr->region_count[1] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3); - gr->region_count[2] = 255; - } - gr->table_select[0] = (drmp3_uint8)(tables >> 10); - gr->table_select[1] = (drmp3_uint8)((tables >> 5) & 31); - gr->table_select[2] = (drmp3_uint8)((tables) & 31); - gr->preflag = (drmp3_uint8)(DRMP3_HDR_TEST_MPEG1(hdr) ? drmp3_bs_get_bits(bs, 1) : (gr->scalefac_compress >= 500)); - gr->scalefac_scale = (drmp3_uint8)drmp3_bs_get_bits(bs, 1); - gr->count1_table = (drmp3_uint8)drmp3_bs_get_bits(bs, 1); - gr->scfsi = (drmp3_uint8)((scfsi >> 12) & 15); - scfsi <<= 4; - gr++; - } while(--gr_count); - - if (part_23_sum + bs->pos > bs->limit + main_data_begin*8) - { - return -1; - } - - return main_data_begin; -} - -static void drmp3_L3_read_scalefactors(drmp3_uint8 *scf, drmp3_uint8 *ist_pos, const drmp3_uint8 *scf_size, const drmp3_uint8 *scf_count, drmp3_bs *bitbuf, int scfsi) -{ - int i, k; - for (i = 0; i < 4 && scf_count[i]; i++, scfsi *= 2) - { - int cnt = scf_count[i]; - if (scfsi & 8) - { - memcpy(scf, ist_pos, cnt); - } else - { - int bits = scf_size[i]; - if (!bits) - { - memset(scf, 0, cnt); - memset(ist_pos, 0, cnt); - } else - { - int max_scf = (scfsi < 0) ? (1 << bits) - 1 : -1; - for (k = 0; k < cnt; k++) - { - int s = drmp3_bs_get_bits(bitbuf, bits); - ist_pos[k] = (drmp3_uint8)(s == max_scf ? -1 : s); - scf[k] = (drmp3_uint8)s; - } - } - } - ist_pos += cnt; - scf += cnt; - } - scf[0] = scf[1] = scf[2] = 0; -} - -static float drmp3_L3_ldexp_q2(float y, int exp_q2) -{ - static const float g_expfrac[4] = { 9.31322575e-10f,7.83145814e-10f,6.58544508e-10f,5.53767716e-10f }; - int e; - do - { - e = DRMP3_MIN(30*4, exp_q2); - y *= g_expfrac[e & 3]*(1 << 30 >> (e >> 2)); - } while ((exp_q2 -= e) > 0); - return y; -} - -static void drmp3_L3_decode_scalefactors(const drmp3_uint8 *hdr, drmp3_uint8 *ist_pos, drmp3_bs *bs, const drmp3_L3_gr_info *gr, float *scf, int ch) -{ - static const drmp3_uint8 g_scf_partitions[3][28] = { - { 6,5,5, 5,6,5,5,5,6,5, 7,3,11,10,0,0, 7, 7, 7,0, 6, 6,6,3, 8, 8,5,0 }, - { 8,9,6,12,6,9,9,9,6,9,12,6,15,18,0,0, 6,15,12,0, 6,12,9,6, 6,18,9,0 }, - { 9,9,6,12,9,9,9,9,9,9,12,6,18,18,0,0,12,12,12,0,12, 9,9,6,15,12,9,0 } - }; - const drmp3_uint8 *scf_partition = g_scf_partitions[!!gr->n_short_sfb + !gr->n_long_sfb]; - drmp3_uint8 scf_size[4], iscf[40]; - int i, scf_shift = gr->scalefac_scale + 1, gain_exp, scfsi = gr->scfsi; - float gain; - - if (DRMP3_HDR_TEST_MPEG1(hdr)) - { - static const drmp3_uint8 g_scfc_decode[16] = { 0,1,2,3, 12,5,6,7, 9,10,11,13, 14,15,18,19 }; - int part = g_scfc_decode[gr->scalefac_compress]; - scf_size[1] = scf_size[0] = (drmp3_uint8)(part >> 2); - scf_size[3] = scf_size[2] = (drmp3_uint8)(part & 3); - } else - { - static const drmp3_uint8 g_mod[6*4] = { 5,5,4,4,5,5,4,1,4,3,1,1,5,6,6,1,4,4,4,1,4,3,1,1 }; - int k, modprod, sfc, ist = DRMP3_HDR_TEST_I_STEREO(hdr) && ch; - sfc = gr->scalefac_compress >> ist; - for (k = ist*3*4; sfc >= 0; sfc -= modprod, k += 4) - { - for (modprod = 1, i = 3; i >= 0; i--) - { - scf_size[i] = (drmp3_uint8)(sfc / modprod % g_mod[k + i]); - modprod *= g_mod[k + i]; - } - } - scf_partition += k; - scfsi = -16; - } - drmp3_L3_read_scalefactors(iscf, ist_pos, scf_size, scf_partition, bs, scfsi); - - if (gr->n_short_sfb) - { - int sh = 3 - scf_shift; - for (i = 0; i < gr->n_short_sfb; i += 3) - { - iscf[gr->n_long_sfb + i + 0] += gr->subblock_gain[0] << sh; - iscf[gr->n_long_sfb + i + 1] += gr->subblock_gain[1] << sh; - iscf[gr->n_long_sfb + i + 2] += gr->subblock_gain[2] << sh; - } - } else if (gr->preflag) - { - static const drmp3_uint8 g_preamp[10] = { 1,1,1,1,2,2,3,3,3,2 }; - for (i = 0; i < 10; i++) - { - iscf[11 + i] += g_preamp[i]; - } - } - - gain_exp = gr->global_gain + DRMP3_BITS_DEQUANTIZER_OUT*4 - 210 - (DRMP3_HDR_IS_MS_STEREO(hdr) ? 2 : 0); - gain = drmp3_L3_ldexp_q2(1 << (DRMP3_MAX_SCFI/4), DRMP3_MAX_SCFI - gain_exp); - for (i = 0; i < (int)(gr->n_long_sfb + gr->n_short_sfb); i++) - { - scf[i] = drmp3_L3_ldexp_q2(gain, iscf[i] << scf_shift); - } -} - -static const float g_drmp3_pow43[129 + 16] = { - 0,-1,-2.519842f,-4.326749f,-6.349604f,-8.549880f,-10.902724f,-13.390518f,-16.000000f,-18.720754f,-21.544347f,-24.463781f,-27.473142f,-30.567351f,-33.741992f,-36.993181f, - 0,1,2.519842f,4.326749f,6.349604f,8.549880f,10.902724f,13.390518f,16.000000f,18.720754f,21.544347f,24.463781f,27.473142f,30.567351f,33.741992f,36.993181f,40.317474f,43.711787f,47.173345f,50.699631f,54.288352f,57.937408f,61.644865f,65.408941f,69.227979f,73.100443f,77.024898f,81.000000f,85.024491f,89.097188f,93.216975f,97.382800f,101.593667f,105.848633f,110.146801f,114.487321f,118.869381f,123.292209f,127.755065f,132.257246f,136.798076f,141.376907f,145.993119f,150.646117f,155.335327f,160.060199f,164.820202f,169.614826f,174.443577f,179.305980f,184.201575f,189.129918f,194.090580f,199.083145f,204.107210f,209.162385f,214.248292f,219.364564f,224.510845f,229.686789f,234.892058f,240.126328f,245.389280f,250.680604f,256.000000f,261.347174f,266.721841f,272.123723f,277.552547f,283.008049f,288.489971f,293.998060f,299.532071f,305.091761f,310.676898f,316.287249f,321.922592f,327.582707f,333.267377f,338.976394f,344.709550f,350.466646f,356.247482f,362.051866f,367.879608f,373.730522f,379.604427f,385.501143f,391.420496f,397.362314f,403.326427f,409.312672f,415.320884f,421.350905f,427.402579f,433.475750f,439.570269f,445.685987f,451.822757f,457.980436f,464.158883f,470.357960f,476.577530f,482.817459f,489.077615f,495.357868f,501.658090f,507.978156f,514.317941f,520.677324f,527.056184f,533.454404f,539.871867f,546.308458f,552.764065f,559.238575f,565.731879f,572.243870f,578.774440f,585.323483f,591.890898f,598.476581f,605.080431f,611.702349f,618.342238f,625.000000f,631.675540f,638.368763f,645.079578f -}; - -static float drmp3_L3_pow_43(int x) -{ - float frac; - int sign, mult = 256; - - if (x < 129) - { - return g_drmp3_pow43[16 + x]; - } - - if (x < 1024) - { - mult = 16; - x <<= 3; - } - - sign = 2*x & 64; - frac = (float)((x & 63) - sign) / ((x & ~63) + sign); - return g_drmp3_pow43[16 + ((x + sign) >> 6)]*(1.f + frac*((4.f/3) + frac*(2.f/9)))*mult; -} - -static void drmp3_L3_huffman(float *dst, drmp3_bs *bs, const drmp3_L3_gr_info *gr_info, const float *scf, int layer3gr_limit) -{ - static const drmp3_int16 tabs[] = { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 785,785,785,785,784,784,784,784,513,513,513,513,513,513,513,513,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256, - -255,1313,1298,1282,785,785,785,785,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,290,288, - -255,1313,1298,1282,769,769,769,769,529,529,529,529,529,529,529,529,528,528,528,528,528,528,528,528,512,512,512,512,512,512,512,512,290,288, - 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static const drmp3_uint8 tab32[] = { 130,162,193,209,44,28,76,140,9,9,9,9,9,9,9,9,190,254,222,238,126,94,157,157,109,61,173,205}; - static const drmp3_uint8 tab33[] = { 252,236,220,204,188,172,156,140,124,108,92,76,60,44,28,12 }; - static const drmp3_int16 tabindex[2*16] = { 0,32,64,98,0,132,180,218,292,364,426,538,648,746,0,1126,1460,1460,1460,1460,1460,1460,1460,1460,1842,1842,1842,1842,1842,1842,1842,1842 }; - static const drmp3_uint8 g_linbits[] = { 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,2,3,4,6,8,10,13,4,5,6,7,8,9,11,13 }; - -#define DRMP3_PEEK_BITS(n) (bs_cache >> (32 - n)) -#define DRMP3_FLUSH_BITS(n) { bs_cache <<= (n); bs_sh += (n); } -#define DRMP3_CHECK_BITS while (bs_sh >= 0) { bs_cache |= (drmp3_uint32)*bs_next_ptr++ << bs_sh; bs_sh -= 8; } -#define DRMP3_BSPOS ((bs_next_ptr - bs->buf)*8 - 24 + bs_sh) - - float one = 0.0f; - int ireg = 0, big_val_cnt = gr_info->big_values; - const drmp3_uint8 *sfb = gr_info->sfbtab; - const drmp3_uint8 *bs_next_ptr = bs->buf + bs->pos/8; - drmp3_uint32 bs_cache = (((bs_next_ptr[0]*256u + bs_next_ptr[1])*256u + bs_next_ptr[2])*256u + bs_next_ptr[3]) << (bs->pos & 7); - int pairs_to_decode, np, bs_sh = (bs->pos & 7) - 8; - bs_next_ptr += 4; - - while (big_val_cnt > 0) - { - int tab_num = gr_info->table_select[ireg]; - int sfb_cnt = gr_info->region_count[ireg++]; - const drmp3_int16 *codebook = tabs + tabindex[tab_num]; - int linbits = g_linbits[tab_num]; - if (linbits) - { - do - { - np = *sfb++ / 2; - pairs_to_decode = DRMP3_MIN(big_val_cnt, np); - one = *scf++; - do - { - int j, w = 5; - int leaf = codebook[DRMP3_PEEK_BITS(w)]; - while (leaf < 0) - { - DRMP3_FLUSH_BITS(w); - w = leaf & 7; - leaf = codebook[DRMP3_PEEK_BITS(w) - (leaf >> 3)]; - } - DRMP3_FLUSH_BITS(leaf >> 8); - - for (j = 0; j < 2; j++, dst++, leaf >>= 4) - { - int lsb = leaf & 0x0F; - if (lsb == 15) - { - lsb += DRMP3_PEEK_BITS(linbits); - DRMP3_FLUSH_BITS(linbits); - DRMP3_CHECK_BITS; - *dst = one*drmp3_L3_pow_43(lsb)*((drmp3_int32)bs_cache < 0 ? -1: 1); - } else - { - *dst = g_drmp3_pow43[16 + lsb - 16*(bs_cache >> 31)]*one; - } - DRMP3_FLUSH_BITS(lsb ? 1 : 0); - } - DRMP3_CHECK_BITS; - } while (--pairs_to_decode); - } while ((big_val_cnt -= np) > 0 && --sfb_cnt >= 0); - } else - { - do - { - np = *sfb++ / 2; - pairs_to_decode = DRMP3_MIN(big_val_cnt, np); - one = *scf++; - do - { - int j, w = 5; - int leaf = codebook[DRMP3_PEEK_BITS(w)]; - while (leaf < 0) - { - DRMP3_FLUSH_BITS(w); - w = leaf & 7; - leaf = codebook[DRMP3_PEEK_BITS(w) - (leaf >> 3)]; - } - DRMP3_FLUSH_BITS(leaf >> 8); - - for (j = 0; j < 2; j++, dst++, leaf >>= 4) - { - int lsb = leaf & 0x0F; - *dst = g_drmp3_pow43[16 + lsb - 16*(bs_cache >> 31)]*one; - DRMP3_FLUSH_BITS(lsb ? 1 : 0); - } - DRMP3_CHECK_BITS; - } while (--pairs_to_decode); - } while ((big_val_cnt -= np) > 0 && --sfb_cnt >= 0); - } - } - - for (np = 1 - big_val_cnt;; dst += 4) - { - const drmp3_uint8 *codebook_count1 = (gr_info->count1_table) ? tab33 : tab32; - int leaf = codebook_count1[DRMP3_PEEK_BITS(4)]; - if (!(leaf & 8)) - { - leaf = codebook_count1[(leaf >> 3) + (bs_cache << 4 >> (32 - (leaf & 3)))]; - } - DRMP3_FLUSH_BITS(leaf & 7); - if (DRMP3_BSPOS > layer3gr_limit) - { - break; - } -#define DRMP3_RELOAD_SCALEFACTOR if (!--np) { np = *sfb++/2; if (!np) break; one = *scf++; } -#define DRMP3_DEQ_COUNT1(s) if (leaf & (128 >> s)) { dst[s] = ((drmp3_int32)bs_cache < 0) ? -one : one; DRMP3_FLUSH_BITS(1) } - DRMP3_RELOAD_SCALEFACTOR; - DRMP3_DEQ_COUNT1(0); - DRMP3_DEQ_COUNT1(1); - DRMP3_RELOAD_SCALEFACTOR; - DRMP3_DEQ_COUNT1(2); - DRMP3_DEQ_COUNT1(3); - DRMP3_CHECK_BITS; - } - - bs->pos = layer3gr_limit; -} - -static void drmp3_L3_midside_stereo(float *left, int n) -{ - int i = 0; - float *right = left + 576; -#if DRMP3_HAVE_SIMD - if (drmp3_have_simd()) for (; i < n - 3; i += 4) - { - drmp3_f4 vl = DRMP3_VLD(left + i); - drmp3_f4 vr = DRMP3_VLD(right + i); - DRMP3_VSTORE(left + i, DRMP3_VADD(vl, vr)); - DRMP3_VSTORE(right + i, DRMP3_VSUB(vl, vr)); - } -#endif - for (; i < n; i++) - { - float a = left[i]; - float b = right[i]; - left[i] = a + b; - right[i] = a - b; - } -} - -static void drmp3_L3_intensity_stereo_band(float *left, int n, float kl, float kr) -{ - int i; - for (i = 0; i < n; i++) - { - left[i + 576] = left[i]*kr; - left[i] = left[i]*kl; - } -} - -static void drmp3_L3_stereo_top_band(const float *right, const drmp3_uint8 *sfb, int nbands, int max_band[3]) -{ - int i, k; - - max_band[0] = max_band[1] = max_band[2] = -1; - - for (i = 0; i < nbands; i++) - { - for (k = 0; k < sfb[i]; k += 2) - { - if (right[k] != 0 || right[k + 1] != 0) - { - max_band[i % 3] = i; - break; - } - } - right += sfb[i]; - } -} - -static void drmp3_L3_stereo_process(float *left, const drmp3_uint8 *ist_pos, const drmp3_uint8 *sfb, const drmp3_uint8 *hdr, int max_band[3], int mpeg2_sh) -{ - static const float g_pan[7*2] = { 0,1,0.21132487f,0.78867513f,0.36602540f,0.63397460f,0.5f,0.5f,0.63397460f,0.36602540f,0.78867513f,0.21132487f,1,0 }; - unsigned i, max_pos = DRMP3_HDR_TEST_MPEG1(hdr) ? 7 : 64; - - for (i = 0; sfb[i]; i++) - { - unsigned ipos = ist_pos[i]; - if ((int)i > max_band[i % 3] && ipos < max_pos) - { - float kl, kr, s = DRMP3_HDR_TEST_MS_STEREO(hdr) ? 1.41421356f : 1; - if (DRMP3_HDR_TEST_MPEG1(hdr)) - { - kl = g_pan[2*ipos]; - kr = g_pan[2*ipos + 1]; - } else - { - kl = 1; - kr = drmp3_L3_ldexp_q2(1, (ipos + 1) >> 1 << mpeg2_sh); - if (ipos & 1) - { - kl = kr; - kr = 1; - } - } - drmp3_L3_intensity_stereo_band(left, sfb[i], kl*s, kr*s); - } else if (DRMP3_HDR_TEST_MS_STEREO(hdr)) - { - drmp3_L3_midside_stereo(left, sfb[i]); - } - left += sfb[i]; - } -} - -static void drmp3_L3_intensity_stereo(float *left, drmp3_uint8 *ist_pos, const drmp3_L3_gr_info *gr, const drmp3_uint8 *hdr) -{ - int max_band[3], n_sfb = gr->n_long_sfb + gr->n_short_sfb; - int i, max_blocks = gr->n_short_sfb ? 3 : 1; - - drmp3_L3_stereo_top_band(left + 576, gr->sfbtab, n_sfb, max_band); - if (gr->n_long_sfb) - { - max_band[0] = max_band[1] = max_band[2] = DRMP3_MAX(DRMP3_MAX(max_band[0], max_band[1]), max_band[2]); - } - for (i = 0; i < max_blocks; i++) - { - int default_pos = DRMP3_HDR_TEST_MPEG1(hdr) ? 3 : 0; - int itop = n_sfb - max_blocks + i; - int prev = itop - max_blocks; - ist_pos[itop] = (drmp3_uint8)(max_band[i] >= prev ? default_pos : ist_pos[prev]); - } - drmp3_L3_stereo_process(left, ist_pos, gr->sfbtab, hdr, max_band, gr[1].scalefac_compress & 1); -} - -static void drmp3_L3_reorder(float *grbuf, float *scratch, const drmp3_uint8 *sfb) -{ - int i, len; - float *src = grbuf, *dst = scratch; - - for (;0 != (len = *sfb); sfb += 3, src += 2*len) - { - for (i = 0; i < len; i++, src++) - { - *dst++ = src[0*len]; - *dst++ = src[1*len]; - *dst++ = src[2*len]; - } - } - memcpy(grbuf, scratch, (dst - scratch)*sizeof(float)); -} - -static void drmp3_L3_antialias(float *grbuf, int nbands) -{ - static const float g_aa[2][8] = { - {0.85749293f,0.88174200f,0.94962865f,0.98331459f,0.99551782f,0.99916056f,0.99989920f,0.99999316f}, - {0.51449576f,0.47173197f,0.31337745f,0.18191320f,0.09457419f,0.04096558f,0.01419856f,0.00369997f} - }; - - for (; nbands > 0; nbands--, grbuf += 18) - { - int i = 0; -#if DRMP3_HAVE_SIMD - if (drmp3_have_simd()) for (; i < 8; i += 4) - { - drmp3_f4 vu = DRMP3_VLD(grbuf + 18 + i); - drmp3_f4 vd = DRMP3_VLD(grbuf + 14 - i); - drmp3_f4 vc0 = DRMP3_VLD(g_aa[0] + i); - drmp3_f4 vc1 = DRMP3_VLD(g_aa[1] + i); - vd = DRMP3_VREV(vd); - DRMP3_VSTORE(grbuf + 18 + i, DRMP3_VSUB(DRMP3_VMUL(vu, vc0), DRMP3_VMUL(vd, vc1))); - vd = DRMP3_VADD(DRMP3_VMUL(vu, vc1), DRMP3_VMUL(vd, vc0)); - DRMP3_VSTORE(grbuf + 14 - i, DRMP3_VREV(vd)); - } -#endif -#ifndef DR_MP3_ONLY_SIMD - for(; i < 8; i++) - { - float u = grbuf[18 + i]; - float d = grbuf[17 - i]; - grbuf[18 + i] = u*g_aa[0][i] - d*g_aa[1][i]; - grbuf[17 - i] = u*g_aa[1][i] + d*g_aa[0][i]; - } -#endif - } -} - -static void drmp3_L3_dct3_9(float *y) -{ - float s0, s1, s2, s3, s4, s5, s6, s7, s8, t0, t2, t4; - - s0 = y[0]; s2 = y[2]; s4 = y[4]; s6 = y[6]; s8 = y[8]; - t0 = s0 + s6*0.5f; - s0 -= s6; - t4 = (s4 + s2)*0.93969262f; - t2 = (s8 + s2)*0.76604444f; - s6 = (s4 - s8)*0.17364818f; - s4 += s8 - s2; - - s2 = s0 - s4*0.5f; - y[4] = s4 + s0; - s8 = t0 - t2 + s6; - s0 = t0 - t4 + t2; - s4 = t0 + t4 - s6; - - s1 = y[1]; s3 = y[3]; s5 = y[5]; s7 = y[7]; - - s3 *= 0.86602540f; - t0 = (s5 + s1)*0.98480775f; - t4 = (s5 - s7)*0.34202014f; - t2 = (s1 + s7)*0.64278761f; - s1 = (s1 - s5 - s7)*0.86602540f; - - s5 = t0 - s3 - t2; - s7 = t4 - s3 - t0; - s3 = t4 + s3 - t2; - - y[0] = s4 - s7; - y[1] = s2 + s1; - y[2] = s0 - s3; - y[3] = s8 + s5; - y[5] = s8 - s5; - y[6] = s0 + s3; - y[7] = s2 - s1; - y[8] = s4 + s7; -} - -static void drmp3_L3_imdct36(float *grbuf, float *overlap, const float *window, int nbands) -{ - int i, j; - static const float g_twid9[18] = { - 0.73727734f,0.79335334f,0.84339145f,0.88701083f,0.92387953f,0.95371695f,0.97629601f,0.99144486f,0.99904822f,0.67559021f,0.60876143f,0.53729961f,0.46174861f,0.38268343f,0.30070580f,0.21643961f,0.13052619f,0.04361938f - }; - - for (j = 0; j < nbands; j++, grbuf += 18, overlap += 9) - { - float co[9], si[9]; - co[0] = -grbuf[0]; - si[0] = grbuf[17]; - for (i = 0; i < 4; i++) - { - si[8 - 2*i] = grbuf[4*i + 1] - grbuf[4*i + 2]; - co[1 + 2*i] = grbuf[4*i + 1] + grbuf[4*i + 2]; - si[7 - 2*i] = grbuf[4*i + 4] - grbuf[4*i + 3]; - co[2 + 2*i] = -(grbuf[4*i + 3] + grbuf[4*i + 4]); - } - drmp3_L3_dct3_9(co); - drmp3_L3_dct3_9(si); - - si[1] = -si[1]; - si[3] = -si[3]; - si[5] = -si[5]; - si[7] = -si[7]; - - i = 0; - -#if DRMP3_HAVE_SIMD - if (drmp3_have_simd()) for (; i < 8; i += 4) - { - drmp3_f4 vovl = DRMP3_VLD(overlap + i); - drmp3_f4 vc = DRMP3_VLD(co + i); - drmp3_f4 vs = DRMP3_VLD(si + i); - drmp3_f4 vr0 = DRMP3_VLD(g_twid9 + i); - drmp3_f4 vr1 = DRMP3_VLD(g_twid9 + 9 + i); - drmp3_f4 vw0 = DRMP3_VLD(window + i); - drmp3_f4 vw1 = DRMP3_VLD(window + 9 + i); - drmp3_f4 vsum = DRMP3_VADD(DRMP3_VMUL(vc, vr1), DRMP3_VMUL(vs, vr0)); - DRMP3_VSTORE(overlap + i, DRMP3_VSUB(DRMP3_VMUL(vc, vr0), DRMP3_VMUL(vs, vr1))); - DRMP3_VSTORE(grbuf + i, DRMP3_VSUB(DRMP3_VMUL(vovl, vw0), DRMP3_VMUL(vsum, vw1))); - vsum = DRMP3_VADD(DRMP3_VMUL(vovl, vw1), DRMP3_VMUL(vsum, vw0)); - DRMP3_VSTORE(grbuf + 14 - i, DRMP3_VREV(vsum)); - } -#endif - for (; i < 9; i++) - { - float ovl = overlap[i]; - float sum = co[i]*g_twid9[9 + i] + si[i]*g_twid9[0 + i]; - overlap[i] = co[i]*g_twid9[0 + i] - si[i]*g_twid9[9 + i]; - grbuf[i] = ovl*window[0 + i] - sum*window[9 + i]; - grbuf[17 - i] = ovl*window[9 + i] + sum*window[0 + i]; - } - } -} - -static void drmp3_L3_idct3(float x0, float x1, float x2, float *dst) -{ - float m1 = x1*0.86602540f; - float a1 = x0 - x2*0.5f; - dst[1] = x0 + x2; - dst[0] = a1 + m1; - dst[2] = a1 - m1; -} - -static void drmp3_L3_imdct12(float *x, float *dst, float *overlap) -{ - static const float g_twid3[6] = { 0.79335334f,0.92387953f,0.99144486f, 0.60876143f,0.38268343f,0.13052619f }; - float co[3], si[3]; - int i; - - drmp3_L3_idct3(-x[0], x[6] + x[3], x[12] + x[9], co); - drmp3_L3_idct3(x[15], x[12] - x[9], x[6] - x[3], si); - si[1] = -si[1]; - - for (i = 0; i < 3; i++) - { - float ovl = overlap[i]; - float sum = co[i]*g_twid3[3 + i] + si[i]*g_twid3[0 + i]; - overlap[i] = co[i]*g_twid3[0 + i] - si[i]*g_twid3[3 + i]; - dst[i] = ovl*g_twid3[2 - i] - sum*g_twid3[5 - i]; - dst[5 - i] = ovl*g_twid3[5 - i] + sum*g_twid3[2 - i]; - } -} - -static void drmp3_L3_imdct_short(float *grbuf, float *overlap, int nbands) -{ - for (;nbands > 0; nbands--, overlap += 9, grbuf += 18) - { - float tmp[18]; - memcpy(tmp, grbuf, sizeof(tmp)); - memcpy(grbuf, overlap, 6*sizeof(float)); - drmp3_L3_imdct12(tmp, grbuf + 6, overlap + 6); - drmp3_L3_imdct12(tmp + 1, grbuf + 12, overlap + 6); - drmp3_L3_imdct12(tmp + 2, overlap, overlap + 6); - } -} - -static void drmp3_L3_change_sign(float *grbuf) -{ - int b, i; - for (b = 0, grbuf += 18; b < 32; b += 2, grbuf += 36) - for (i = 1; i < 18; i += 2) - grbuf[i] = -grbuf[i]; -} - -static void drmp3_L3_imdct_gr(float *grbuf, float *overlap, unsigned block_type, unsigned n_long_bands) -{ - static const float g_mdct_window[2][18] = { - { 0.99904822f,0.99144486f,0.97629601f,0.95371695f,0.92387953f,0.88701083f,0.84339145f,0.79335334f,0.73727734f,0.04361938f,0.13052619f,0.21643961f,0.30070580f,0.38268343f,0.46174861f,0.53729961f,0.60876143f,0.67559021f }, - { 1,1,1,1,1,1,0.99144486f,0.92387953f,0.79335334f,0,0,0,0,0,0,0.13052619f,0.38268343f,0.60876143f } - }; - if (n_long_bands) - { - drmp3_L3_imdct36(grbuf, overlap, g_mdct_window[0], n_long_bands); - grbuf += 18*n_long_bands; - overlap += 9*n_long_bands; - } - if (block_type == DRMP3_SHORT_BLOCK_TYPE) - drmp3_L3_imdct_short(grbuf, overlap, 32 - n_long_bands); - else - drmp3_L3_imdct36(grbuf, overlap, g_mdct_window[block_type == DRMP3_STOP_BLOCK_TYPE], 32 - n_long_bands); -} - -static void drmp3_L3_save_reservoir(drmp3dec *h, drmp3dec_scratch *s) -{ - int pos = (s->bs.pos + 7)/8u; - int remains = s->bs.limit/8u - pos; - if (remains > DRMP3_MAX_BITRESERVOIR_BYTES) - { - pos += remains - DRMP3_MAX_BITRESERVOIR_BYTES; - remains = DRMP3_MAX_BITRESERVOIR_BYTES; - } - if (remains > 0) - { - memmove(h->reserv_buf, s->maindata + pos, remains); - } - h->reserv = remains; -} - -static int drmp3_L3_restore_reservoir(drmp3dec *h, drmp3_bs *bs, drmp3dec_scratch *s, int main_data_begin) -{ - int frame_bytes = (bs->limit - bs->pos)/8; - int bytes_have = DRMP3_MIN(h->reserv, main_data_begin); - memcpy(s->maindata, h->reserv_buf + DRMP3_MAX(0, h->reserv - main_data_begin), DRMP3_MIN(h->reserv, main_data_begin)); - memcpy(s->maindata + bytes_have, bs->buf + bs->pos/8, frame_bytes); - drmp3_bs_init(&s->bs, s->maindata, bytes_have + frame_bytes); - return h->reserv >= main_data_begin; -} - -static void drmp3_L3_decode(drmp3dec *h, drmp3dec_scratch *s, drmp3_L3_gr_info *gr_info, int nch) -{ - int ch; - - for (ch = 0; ch < nch; ch++) - { - int layer3gr_limit = s->bs.pos + gr_info[ch].part_23_length; - drmp3_L3_decode_scalefactors(h->header, s->ist_pos[ch], &s->bs, gr_info + ch, s->scf, ch); - drmp3_L3_huffman(s->grbuf[ch], &s->bs, gr_info + ch, s->scf, layer3gr_limit); - } - - if (DRMP3_HDR_TEST_I_STEREO(h->header)) - { - drmp3_L3_intensity_stereo(s->grbuf[0], s->ist_pos[1], gr_info, h->header); - } else if (DRMP3_HDR_IS_MS_STEREO(h->header)) - { - drmp3_L3_midside_stereo(s->grbuf[0], 576); - } - - for (ch = 0; ch < nch; ch++, gr_info++) - { - int aa_bands = 31; - int n_long_bands = (gr_info->mixed_block_flag ? 2 : 0) << (int)(DRMP3_HDR_GET_MY_SAMPLE_RATE(h->header) == 2); - - if (gr_info->n_short_sfb) - { - aa_bands = n_long_bands - 1; - drmp3_L3_reorder(s->grbuf[ch] + n_long_bands*18, s->syn[0], gr_info->sfbtab + gr_info->n_long_sfb); - } - - drmp3_L3_antialias(s->grbuf[ch], aa_bands); - drmp3_L3_imdct_gr(s->grbuf[ch], h->mdct_overlap[ch], gr_info->block_type, n_long_bands); - drmp3_L3_change_sign(s->grbuf[ch]); - } -} - -static void drmp3d_DCT_II(float *grbuf, int n) -{ - static const float g_sec[24] = { - 10.19000816f,0.50060302f,0.50241929f,3.40760851f,0.50547093f,0.52249861f,2.05778098f,0.51544732f,0.56694406f,1.48416460f,0.53104258f,0.64682180f,1.16943991f,0.55310392f,0.78815460f,0.97256821f,0.58293498f,1.06067765f,0.83934963f,0.62250412f,1.72244716f,0.74453628f,0.67480832f,5.10114861f - }; - int i, k = 0; -#if DRMP3_HAVE_SIMD - if (drmp3_have_simd()) for (; k < n; k += 4) - { - drmp3_f4 t[4][8], *x; - float *y = grbuf + k; - - for (x = t[0], i = 0; i < 8; i++, x++) - { - drmp3_f4 x0 = DRMP3_VLD(&y[i*18]); - drmp3_f4 x1 = DRMP3_VLD(&y[(15 - i)*18]); - drmp3_f4 x2 = DRMP3_VLD(&y[(16 + i)*18]); - drmp3_f4 x3 = DRMP3_VLD(&y[(31 - i)*18]); - drmp3_f4 t0 = DRMP3_VADD(x0, x3); - drmp3_f4 t1 = DRMP3_VADD(x1, x2); - drmp3_f4 t2 = DRMP3_VMUL_S(DRMP3_VSUB(x1, x2), g_sec[3*i + 0]); - drmp3_f4 t3 = DRMP3_VMUL_S(DRMP3_VSUB(x0, x3), g_sec[3*i + 1]); - x[0] = DRMP3_VADD(t0, t1); - x[8] = DRMP3_VMUL_S(DRMP3_VSUB(t0, t1), g_sec[3*i + 2]); - x[16] = DRMP3_VADD(t3, t2); - x[24] = DRMP3_VMUL_S(DRMP3_VSUB(t3, t2), g_sec[3*i + 2]); - } - for (x = t[0], i = 0; i < 4; i++, x += 8) - { - drmp3_f4 x0 = x[0], x1 = x[1], x2 = x[2], x3 = x[3], x4 = x[4], x5 = x[5], x6 = x[6], x7 = x[7], xt; - xt = DRMP3_VSUB(x0, x7); x0 = DRMP3_VADD(x0, x7); - x7 = DRMP3_VSUB(x1, x6); x1 = DRMP3_VADD(x1, x6); - x6 = DRMP3_VSUB(x2, x5); x2 = DRMP3_VADD(x2, x5); - x5 = DRMP3_VSUB(x3, x4); x3 = DRMP3_VADD(x3, x4); - x4 = DRMP3_VSUB(x0, x3); x0 = DRMP3_VADD(x0, x3); - x3 = DRMP3_VSUB(x1, x2); x1 = DRMP3_VADD(x1, x2); - x[0] = DRMP3_VADD(x0, x1); - x[4] = DRMP3_VMUL_S(DRMP3_VSUB(x0, x1), 0.70710677f); - x5 = DRMP3_VADD(x5, x6); - x6 = DRMP3_VMUL_S(DRMP3_VADD(x6, x7), 0.70710677f); - x7 = DRMP3_VADD(x7, xt); - x3 = DRMP3_VMUL_S(DRMP3_VADD(x3, x4), 0.70710677f); - x5 = DRMP3_VSUB(x5, DRMP3_VMUL_S(x7, 0.198912367f)); /* rotate by PI/8 */ - x7 = DRMP3_VADD(x7, DRMP3_VMUL_S(x5, 0.382683432f)); - x5 = DRMP3_VSUB(x5, DRMP3_VMUL_S(x7, 0.198912367f)); - x0 = DRMP3_VSUB(xt, x6); xt = DRMP3_VADD(xt, x6); - x[1] = DRMP3_VMUL_S(DRMP3_VADD(xt, x7), 0.50979561f); - x[2] = DRMP3_VMUL_S(DRMP3_VADD(x4, x3), 0.54119611f); - x[3] = DRMP3_VMUL_S(DRMP3_VSUB(x0, x5), 0.60134488f); - x[5] = DRMP3_VMUL_S(DRMP3_VADD(x0, x5), 0.89997619f); - x[6] = DRMP3_VMUL_S(DRMP3_VSUB(x4, x3), 1.30656302f); - x[7] = DRMP3_VMUL_S(DRMP3_VSUB(xt, x7), 2.56291556f); - } - - if (k > n - 3) - { -#if DRMP3_HAVE_SSE -#define DRMP3_VSAVE2(i, v) _mm_storel_pi((__m64 *)(void*)&y[i*18], v) -#else -#define DRMP3_VSAVE2(i, v) vst1_f32((float32_t *)&y[i*18], vget_low_f32(v)) -#endif - for (i = 0; i < 7; i++, y += 4*18) - { - drmp3_f4 s = DRMP3_VADD(t[3][i], t[3][i + 1]); - DRMP3_VSAVE2(0, t[0][i]); - DRMP3_VSAVE2(1, DRMP3_VADD(t[2][i], s)); - DRMP3_VSAVE2(2, DRMP3_VADD(t[1][i], t[1][i + 1])); - DRMP3_VSAVE2(3, DRMP3_VADD(t[2][1 + i], s)); - } - DRMP3_VSAVE2(0, t[0][7]); - DRMP3_VSAVE2(1, DRMP3_VADD(t[2][7], t[3][7])); - DRMP3_VSAVE2(2, t[1][7]); - DRMP3_VSAVE2(3, t[3][7]); - } else - { -#define DRMP3_VSAVE4(i, v) DRMP3_VSTORE(&y[i*18], v) - for (i = 0; i < 7; i++, y += 4*18) - { - drmp3_f4 s = DRMP3_VADD(t[3][i], t[3][i + 1]); - DRMP3_VSAVE4(0, t[0][i]); - DRMP3_VSAVE4(1, DRMP3_VADD(t[2][i], s)); - DRMP3_VSAVE4(2, DRMP3_VADD(t[1][i], t[1][i + 1])); - DRMP3_VSAVE4(3, DRMP3_VADD(t[2][1 + i], s)); - } - DRMP3_VSAVE4(0, t[0][7]); - DRMP3_VSAVE4(1, DRMP3_VADD(t[2][7], t[3][7])); - DRMP3_VSAVE4(2, t[1][7]); - DRMP3_VSAVE4(3, t[3][7]); - } - } else -#endif -#ifdef DR_MP3_ONLY_SIMD - {} -#else - for (; k < n; k++) - { - float t[4][8], *x, *y = grbuf + k; - - for (x = t[0], i = 0; i < 8; i++, x++) - { - float x0 = y[i*18]; - float x1 = y[(15 - i)*18]; - float x2 = y[(16 + i)*18]; - float x3 = y[(31 - i)*18]; - float t0 = x0 + x3; - float t1 = x1 + x2; - float t2 = (x1 - x2)*g_sec[3*i + 0]; - float t3 = (x0 - x3)*g_sec[3*i + 1]; - x[0] = t0 + t1; - x[8] = (t0 - t1)*g_sec[3*i + 2]; - x[16] = t3 + t2; - x[24] = (t3 - t2)*g_sec[3*i + 2]; - } - for (x = t[0], i = 0; i < 4; i++, x += 8) - { - float x0 = x[0], x1 = x[1], x2 = x[2], x3 = x[3], x4 = x[4], x5 = x[5], x6 = x[6], x7 = x[7], xt; - xt = x0 - x7; x0 += x7; - x7 = x1 - x6; x1 += x6; - x6 = x2 - x5; x2 += x5; - x5 = x3 - x4; x3 += x4; - x4 = x0 - x3; x0 += x3; - x3 = x1 - x2; x1 += x2; - x[0] = x0 + x1; - x[4] = (x0 - x1)*0.70710677f; - x5 = x5 + x6; - x6 = (x6 + x7)*0.70710677f; - x7 = x7 + xt; - x3 = (x3 + x4)*0.70710677f; - x5 -= x7*0.198912367f; /* rotate by PI/8 */ - x7 += x5*0.382683432f; - x5 -= x7*0.198912367f; - x0 = xt - x6; xt += x6; - x[1] = (xt + x7)*0.50979561f; - x[2] = (x4 + x3)*0.54119611f; - x[3] = (x0 - x5)*0.60134488f; - x[5] = (x0 + x5)*0.89997619f; - x[6] = (x4 - x3)*1.30656302f; - x[7] = (xt - x7)*2.56291556f; - - } - for (i = 0; i < 7; i++, y += 4*18) - { - y[0*18] = t[0][i]; - y[1*18] = t[2][i] + t[3][i] + t[3][i + 1]; - y[2*18] = t[1][i] + t[1][i + 1]; - y[3*18] = t[2][i + 1] + t[3][i] + t[3][i + 1]; - } - y[0*18] = t[0][7]; - y[1*18] = t[2][7] + t[3][7]; - y[2*18] = t[1][7]; - y[3*18] = t[3][7]; - } -#endif -} - -#ifndef DR_MP3_FLOAT_OUTPUT -typedef drmp3_int16 drmp3d_sample_t; - -static drmp3_int16 drmp3d_scale_pcm(float sample) -{ - drmp3_int16 s; - if (sample >= 32766.5) return (drmp3_int16) 32767; - if (sample <= -32767.5) return (drmp3_int16)-32768; - s = (drmp3_int16)(sample + .5f); - s -= (s < 0); /* away from zero, to be compliant */ - return (drmp3_int16)s; -} -#else -typedef float drmp3d_sample_t; - -static float drmp3d_scale_pcm(float sample) -{ - return sample*(1.f/32768.f); -} -#endif - -static void drmp3d_synth_pair(drmp3d_sample_t *pcm, int nch, const float *z) -{ - float a; - a = (z[14*64] - z[ 0]) * 29; - a += (z[ 1*64] + z[13*64]) * 213; - a += (z[12*64] - z[ 2*64]) * 459; - a += (z[ 3*64] + z[11*64]) * 2037; - a += (z[10*64] - z[ 4*64]) * 5153; - a += (z[ 5*64] + z[ 9*64]) * 6574; - a += (z[ 8*64] - z[ 6*64]) * 37489; - a += z[ 7*64] * 75038; - pcm[0] = drmp3d_scale_pcm(a); - - z += 2; - a = z[14*64] * 104; - a += z[12*64] * 1567; - a += z[10*64] * 9727; - a += z[ 8*64] * 64019; - a += z[ 6*64] * -9975; - a += z[ 4*64] * -45; - a += z[ 2*64] * 146; - a += z[ 0*64] * -5; - pcm[16*nch] = drmp3d_scale_pcm(a); -} - -static void drmp3d_synth(float *xl, drmp3d_sample_t *dstl, int nch, float *lins) -{ - int i; - float *xr = xl + 576*(nch - 1); - drmp3d_sample_t *dstr = dstl + (nch - 1); - - static const float g_win[] = { - -1,26,-31,208,218,401,-519,2063,2000,4788,-5517,7134,5959,35640,-39336,74992, - -1,24,-35,202,222,347,-581,2080,1952,4425,-5879,7640,5288,33791,-41176,74856, - -1,21,-38,196,225,294,-645,2087,1893,4063,-6237,8092,4561,31947,-43006,74630, - -1,19,-41,190,227,244,-711,2085,1822,3705,-6589,8492,3776,30112,-44821,74313, - -1,17,-45,183,228,197,-779,2075,1739,3351,-6935,8840,2935,28289,-46617,73908, - -1,16,-49,176,228,153,-848,2057,1644,3004,-7271,9139,2037,26482,-48390,73415, - -2,14,-53,169,227,111,-919,2032,1535,2663,-7597,9389,1082,24694,-50137,72835, - -2,13,-58,161,224,72,-991,2001,1414,2330,-7910,9592,70,22929,-51853,72169, - -2,11,-63,154,221,36,-1064,1962,1280,2006,-8209,9750,-998,21189,-53534,71420, - -2,10,-68,147,215,2,-1137,1919,1131,1692,-8491,9863,-2122,19478,-55178,70590, - -3,9,-73,139,208,-29,-1210,1870,970,1388,-8755,9935,-3300,17799,-56778,69679, - -3,8,-79,132,200,-57,-1283,1817,794,1095,-8998,9966,-4533,16155,-58333,68692, - -4,7,-85,125,189,-83,-1356,1759,605,814,-9219,9959,-5818,14548,-59838,67629, - -4,7,-91,117,177,-106,-1428,1698,402,545,-9416,9916,-7154,12980,-61289,66494, - -5,6,-97,111,163,-127,-1498,1634,185,288,-9585,9838,-8540,11455,-62684,65290 - }; - float *zlin = lins + 15*64; - const float *w = g_win; - - zlin[4*15] = xl[18*16]; - zlin[4*15 + 1] = xr[18*16]; - zlin[4*15 + 2] = xl[0]; - zlin[4*15 + 3] = xr[0]; - - zlin[4*31] = xl[1 + 18*16]; - zlin[4*31 + 1] = xr[1 + 18*16]; - zlin[4*31 + 2] = xl[1]; - zlin[4*31 + 3] = xr[1]; - - drmp3d_synth_pair(dstr, nch, lins + 4*15 + 1); - drmp3d_synth_pair(dstr + 32*nch, nch, lins + 4*15 + 64 + 1); - drmp3d_synth_pair(dstl, nch, lins + 4*15); - drmp3d_synth_pair(dstl + 32*nch, nch, lins + 4*15 + 64); - -#if DRMP3_HAVE_SIMD - if (drmp3_have_simd()) for (i = 14; i >= 0; i--) - { -#define DRMP3_VLOAD(k) drmp3_f4 w0 = DRMP3_VSET(*w++); drmp3_f4 w1 = DRMP3_VSET(*w++); drmp3_f4 vz = DRMP3_VLD(&zlin[4*i - 64*k]); drmp3_f4 vy = DRMP3_VLD(&zlin[4*i - 64*(15 - k)]); -#define DRMP3_V0(k) { DRMP3_VLOAD(k) b = DRMP3_VADD(DRMP3_VMUL(vz, w1), DRMP3_VMUL(vy, w0)) ; a = DRMP3_VSUB(DRMP3_VMUL(vz, w0), DRMP3_VMUL(vy, w1)); } -#define DRMP3_V1(k) { DRMP3_VLOAD(k) b = DRMP3_VADD(b, DRMP3_VADD(DRMP3_VMUL(vz, w1), DRMP3_VMUL(vy, w0))); a = DRMP3_VADD(a, DRMP3_VSUB(DRMP3_VMUL(vz, w0), DRMP3_VMUL(vy, w1))); } -#define DRMP3_V2(k) { DRMP3_VLOAD(k) b = DRMP3_VADD(b, DRMP3_VADD(DRMP3_VMUL(vz, w1), DRMP3_VMUL(vy, w0))); a = DRMP3_VADD(a, DRMP3_VSUB(DRMP3_VMUL(vy, w1), DRMP3_VMUL(vz, w0))); } - drmp3_f4 a, b; - zlin[4*i] = xl[18*(31 - i)]; - zlin[4*i + 1] = xr[18*(31 - i)]; - zlin[4*i + 2] = xl[1 + 18*(31 - i)]; - zlin[4*i + 3] = xr[1 + 18*(31 - i)]; - zlin[4*i + 64] = xl[1 + 18*(1 + i)]; - zlin[4*i + 64 + 1] = xr[1 + 18*(1 + i)]; - zlin[4*i - 64 + 2] = xl[18*(1 + i)]; - zlin[4*i - 64 + 3] = xr[18*(1 + i)]; - - DRMP3_V0(0) DRMP3_V2(1) DRMP3_V1(2) DRMP3_V2(3) DRMP3_V1(4) DRMP3_V2(5) DRMP3_V1(6) DRMP3_V2(7) - - { -#ifndef DR_MP3_FLOAT_OUTPUT -#if DRMP3_HAVE_SSE - static const drmp3_f4 g_max = { 32767.0f, 32767.0f, 32767.0f, 32767.0f }; - static const drmp3_f4 g_min = { -32768.0f, -32768.0f, -32768.0f, -32768.0f }; - __m128i pcm8 = _mm_packs_epi32(_mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(a, g_max), g_min)), - _mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(b, g_max), g_min))); - dstr[(15 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 1); - dstr[(17 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 5); - dstl[(15 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 0); - dstl[(17 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 4); - dstr[(47 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 3); - dstr[(49 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 7); - dstl[(47 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 2); - dstl[(49 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 6); -#else - int16x4_t pcma, pcmb; - a = DRMP3_VADD(a, DRMP3_VSET(0.5f)); - b = DRMP3_VADD(b, DRMP3_VSET(0.5f)); - pcma = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(a), vreinterpretq_s32_u32(vcltq_f32(a, DRMP3_VSET(0))))); - pcmb = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(b), vreinterpretq_s32_u32(vcltq_f32(b, DRMP3_VSET(0))))); - vst1_lane_s16(dstr + (15 - i)*nch, pcma, 1); - vst1_lane_s16(dstr + (17 + i)*nch, pcmb, 1); - vst1_lane_s16(dstl + (15 - i)*nch, pcma, 0); - vst1_lane_s16(dstl + (17 + i)*nch, pcmb, 0); - vst1_lane_s16(dstr + (47 - i)*nch, pcma, 3); - vst1_lane_s16(dstr + (49 + i)*nch, pcmb, 3); - vst1_lane_s16(dstl + (47 - i)*nch, pcma, 2); - vst1_lane_s16(dstl + (49 + i)*nch, pcmb, 2); -#endif -#else - static const drmp3_f4 g_scale = { 1.0f/32768.0f, 1.0f/32768.0f, 1.0f/32768.0f, 1.0f/32768.0f }; - a = DRMP3_VMUL(a, g_scale); - b = DRMP3_VMUL(b, g_scale); -#if DRMP3_HAVE_SSE - _mm_store_ss(dstr + (15 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(1, 1, 1, 1))); - _mm_store_ss(dstr + (17 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(1, 1, 1, 1))); - _mm_store_ss(dstl + (15 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(0, 0, 0, 0))); - _mm_store_ss(dstl + (17 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(0, 0, 0, 0))); - _mm_store_ss(dstr + (47 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(3, 3, 3, 3))); - _mm_store_ss(dstr + (49 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(3, 3, 3, 3))); - _mm_store_ss(dstl + (47 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(2, 2, 2, 2))); - _mm_store_ss(dstl + (49 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(2, 2, 2, 2))); -#else - vst1q_lane_f32(dstr + (15 - i)*nch, a, 1); - vst1q_lane_f32(dstr + (17 + i)*nch, b, 1); - vst1q_lane_f32(dstl + (15 - i)*nch, a, 0); - vst1q_lane_f32(dstl + (17 + i)*nch, b, 0); - vst1q_lane_f32(dstr + (47 - i)*nch, a, 3); - vst1q_lane_f32(dstr + (49 + i)*nch, b, 3); - vst1q_lane_f32(dstl + (47 - i)*nch, a, 2); - vst1q_lane_f32(dstl + (49 + i)*nch, b, 2); -#endif -#endif /* DR_MP3_FLOAT_OUTPUT */ - } - } else -#endif -#ifdef DR_MP3_ONLY_SIMD - {} -#else - for (i = 14; i >= 0; i--) - { -#define DRMP3_LOAD(k) float w0 = *w++; float w1 = *w++; float *vz = &zlin[4*i - k*64]; float *vy = &zlin[4*i - (15 - k)*64]; -#define DRMP3_S0(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] = vz[j]*w1 + vy[j]*w0, a[j] = vz[j]*w0 - vy[j]*w1; } -#define DRMP3_S1(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] += vz[j]*w1 + vy[j]*w0, a[j] += vz[j]*w0 - vy[j]*w1; } -#define DRMP3_S2(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] += vz[j]*w1 + vy[j]*w0, a[j] += vy[j]*w1 - vz[j]*w0; } - float a[4], b[4]; - - zlin[4*i] = xl[18*(31 - i)]; - zlin[4*i + 1] = xr[18*(31 - i)]; - zlin[4*i + 2] = xl[1 + 18*(31 - i)]; - zlin[4*i + 3] = xr[1 + 18*(31 - i)]; - zlin[4*(i + 16)] = xl[1 + 18*(1 + i)]; - zlin[4*(i + 16) + 1] = xr[1 + 18*(1 + i)]; - zlin[4*(i - 16) + 2] = xl[18*(1 + i)]; - zlin[4*(i - 16) + 3] = xr[18*(1 + i)]; - - DRMP3_S0(0) DRMP3_S2(1) DRMP3_S1(2) DRMP3_S2(3) DRMP3_S1(4) DRMP3_S2(5) DRMP3_S1(6) DRMP3_S2(7) - - dstr[(15 - i)*nch] = drmp3d_scale_pcm(a[1]); - dstr[(17 + i)*nch] = drmp3d_scale_pcm(b[1]); - dstl[(15 - i)*nch] = drmp3d_scale_pcm(a[0]); - dstl[(17 + i)*nch] = drmp3d_scale_pcm(b[0]); - dstr[(47 - i)*nch] = drmp3d_scale_pcm(a[3]); - dstr[(49 + i)*nch] = drmp3d_scale_pcm(b[3]); - dstl[(47 - i)*nch] = drmp3d_scale_pcm(a[2]); - dstl[(49 + i)*nch] = drmp3d_scale_pcm(b[2]); - } -#endif -} - -static void drmp3d_synth_granule(float *qmf_state, float *grbuf, int nbands, int nch, drmp3d_sample_t *pcm, float *lins) -{ - int i; - for (i = 0; i < nch; i++) - { - drmp3d_DCT_II(grbuf + 576*i, nbands); - } - - memcpy(lins, qmf_state, sizeof(float)*15*64); - - for (i = 0; i < nbands; i += 2) - { - drmp3d_synth(grbuf + i, pcm + 32*nch*i, nch, lins + i*64); - } -#ifndef DR_MP3_NONSTANDARD_BUT_LOGICAL - if (nch == 1) - { - for (i = 0; i < 15*64; i += 2) - { - qmf_state[i] = lins[nbands*64 + i]; - } - } else -#endif - { - memcpy(qmf_state, lins + nbands*64, sizeof(float)*15*64); - } -} - -static int drmp3d_match_frame(const drmp3_uint8 *hdr, int mp3_bytes, int frame_bytes) -{ - int i, nmatch; - for (i = 0, nmatch = 0; nmatch < DRMP3_MAX_FRAME_SYNC_MATCHES; nmatch++) - { - i += drmp3_hdr_frame_bytes(hdr + i, frame_bytes) + drmp3_hdr_padding(hdr + i); - if (i + DRMP3_HDR_SIZE > mp3_bytes) - return nmatch > 0; - if (!drmp3_hdr_compare(hdr, hdr + i)) - return 0; - } - return 1; -} - -static int drmp3d_find_frame(const drmp3_uint8 *mp3, int mp3_bytes, int *free_format_bytes, int *ptr_frame_bytes) -{ - int i, k; - for (i = 0; i < mp3_bytes - DRMP3_HDR_SIZE; i++, mp3++) - { - if (drmp3_hdr_valid(mp3)) - { - int frame_bytes = drmp3_hdr_frame_bytes(mp3, *free_format_bytes); - int frame_and_padding = frame_bytes + drmp3_hdr_padding(mp3); - - for (k = DRMP3_HDR_SIZE; !frame_bytes && k < DRMP3_MAX_FREE_FORMAT_FRAME_SIZE && i + 2*k < mp3_bytes - DRMP3_HDR_SIZE; k++) - { - if (drmp3_hdr_compare(mp3, mp3 + k)) - { - int fb = k - drmp3_hdr_padding(mp3); - int nextfb = fb + drmp3_hdr_padding(mp3 + k); - if (i + k + nextfb + DRMP3_HDR_SIZE > mp3_bytes || !drmp3_hdr_compare(mp3, mp3 + k + nextfb)) - continue; - frame_and_padding = k; - frame_bytes = fb; - *free_format_bytes = fb; - } - } - - if ((frame_bytes && i + frame_and_padding <= mp3_bytes && - drmp3d_match_frame(mp3, mp3_bytes - i, frame_bytes)) || - (!i && frame_and_padding == mp3_bytes)) - { - *ptr_frame_bytes = frame_and_padding; - return i; - } - *free_format_bytes = 0; - } - } - *ptr_frame_bytes = 0; - return i; -} - -void drmp3dec_init(drmp3dec *dec) -{ - dec->header[0] = 0; -} - -int drmp3dec_decode_frame(drmp3dec *dec, const unsigned char *mp3, int mp3_bytes, void *pcm, drmp3dec_frame_info *info) -{ - int i = 0, igr, frame_size = 0, success = 1; - const drmp3_uint8 *hdr; - drmp3_bs bs_frame[1]; - drmp3dec_scratch scratch; - - if (mp3_bytes > 4 && dec->header[0] == 0xff && drmp3_hdr_compare(dec->header, mp3)) - { - frame_size = drmp3_hdr_frame_bytes(mp3, dec->free_format_bytes) + drmp3_hdr_padding(mp3); - if (frame_size != mp3_bytes && (frame_size + DRMP3_HDR_SIZE > mp3_bytes || !drmp3_hdr_compare(mp3, mp3 + frame_size))) - { - frame_size = 0; - } - } - if (!frame_size) - { - memset(dec, 0, sizeof(drmp3dec)); - i = drmp3d_find_frame(mp3, mp3_bytes, &dec->free_format_bytes, &frame_size); - if (!frame_size || i + frame_size > mp3_bytes) - { - info->frame_bytes = i; - return 0; - } - } - - hdr = mp3 + i; - memcpy(dec->header, hdr, DRMP3_HDR_SIZE); - info->frame_bytes = i + frame_size; - info->channels = DRMP3_HDR_IS_MONO(hdr) ? 1 : 2; - info->hz = drmp3_hdr_sample_rate_hz(hdr); - info->layer = 4 - DRMP3_HDR_GET_LAYER(hdr); - info->bitrate_kbps = drmp3_hdr_bitrate_kbps(hdr); - - drmp3_bs_init(bs_frame, hdr + DRMP3_HDR_SIZE, frame_size - DRMP3_HDR_SIZE); - if (DRMP3_HDR_IS_CRC(hdr)) - { - drmp3_bs_get_bits(bs_frame, 16); - } - - if (info->layer == 3) - { - int main_data_begin = drmp3_L3_read_side_info(bs_frame, scratch.gr_info, hdr); - if (main_data_begin < 0 || bs_frame->pos > bs_frame->limit) - { - drmp3dec_init(dec); - return 0; - } - success = drmp3_L3_restore_reservoir(dec, bs_frame, &scratch, main_data_begin); - if (success && pcm != NULL) - { - for (igr = 0; igr < (DRMP3_HDR_TEST_MPEG1(hdr) ? 2 : 1); igr++, pcm = DRMP3_OFFSET_PTR(pcm, sizeof(drmp3d_sample_t)*576*info->channels)) - { - memset(scratch.grbuf[0], 0, 576*2*sizeof(float)); - drmp3_L3_decode(dec, &scratch, scratch.gr_info + igr*info->channels, info->channels); - drmp3d_synth_granule(dec->qmf_state, scratch.grbuf[0], 18, info->channels, (drmp3d_sample_t*)pcm, scratch.syn[0]); - } - } - drmp3_L3_save_reservoir(dec, &scratch); - } else - { -#ifdef DR_MP3_ONLY_MP3 - return 0; -#else - drmp3_L12_scale_info sci[1]; - - if (pcm == NULL) { - return drmp3_hdr_frame_samples(hdr); - } - - drmp3_L12_read_scale_info(hdr, bs_frame, sci); - - memset(scratch.grbuf[0], 0, 576*2*sizeof(float)); - for (i = 0, igr = 0; igr < 3; igr++) - { - if (12 == (i += drmp3_L12_dequantize_granule(scratch.grbuf[0] + i, bs_frame, sci, info->layer | 1))) - { - i = 0; - drmp3_L12_apply_scf_384(sci, sci->scf + igr, scratch.grbuf[0]); - drmp3d_synth_granule(dec->qmf_state, scratch.grbuf[0], 12, info->channels, (drmp3d_sample_t*)pcm, scratch.syn[0]); - memset(scratch.grbuf[0], 0, 576*2*sizeof(float)); - pcm = DRMP3_OFFSET_PTR(pcm, sizeof(drmp3d_sample_t)*384*info->channels); - } - if (bs_frame->pos > bs_frame->limit) - { - drmp3dec_init(dec); - return 0; - } - } -#endif - } - - return success*drmp3_hdr_frame_samples(dec->header); -} - -void drmp3dec_f32_to_s16(const float *in, drmp3_int16 *out, int num_samples) -{ - if(num_samples > 0) - { - int i = 0; -#if DRMP3_HAVE_SIMD - int aligned_count = num_samples & ~7; - for(; i < aligned_count; i+=8) - { - drmp3_f4 scale = DRMP3_VSET(32768.0f); - drmp3_f4 a = DRMP3_VMUL(DRMP3_VLD(&in[i ]), scale); - drmp3_f4 b = DRMP3_VMUL(DRMP3_VLD(&in[i+4]), scale); -#if DRMP3_HAVE_SSE - drmp3_f4 s16max = DRMP3_VSET( 32767.0f); - drmp3_f4 s16min = DRMP3_VSET(-32768.0f); - __m128i pcm8 = _mm_packs_epi32(_mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(a, s16max), s16min)), - _mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(b, s16max), s16min))); - out[i ] = (drmp3_int16)_mm_extract_epi16(pcm8, 0); - out[i+1] = (drmp3_int16)_mm_extract_epi16(pcm8, 1); - out[i+2] = (drmp3_int16)_mm_extract_epi16(pcm8, 2); - out[i+3] = (drmp3_int16)_mm_extract_epi16(pcm8, 3); - out[i+4] = (drmp3_int16)_mm_extract_epi16(pcm8, 4); - out[i+5] = (drmp3_int16)_mm_extract_epi16(pcm8, 5); - out[i+6] = (drmp3_int16)_mm_extract_epi16(pcm8, 6); - out[i+7] = (drmp3_int16)_mm_extract_epi16(pcm8, 7); -#else - int16x4_t pcma, pcmb; - a = DRMP3_VADD(a, DRMP3_VSET(0.5f)); - b = DRMP3_VADD(b, DRMP3_VSET(0.5f)); - pcma = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(a), vreinterpretq_s32_u32(vcltq_f32(a, DRMP3_VSET(0))))); - pcmb = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(b), vreinterpretq_s32_u32(vcltq_f32(b, DRMP3_VSET(0))))); - vst1_lane_s16(out+i , pcma, 0); - vst1_lane_s16(out+i+1, pcma, 1); - vst1_lane_s16(out+i+2, pcma, 2); - vst1_lane_s16(out+i+3, pcma, 3); - vst1_lane_s16(out+i+4, pcmb, 0); - vst1_lane_s16(out+i+5, pcmb, 1); - vst1_lane_s16(out+i+6, pcmb, 2); - vst1_lane_s16(out+i+7, pcmb, 3); -#endif - } -#endif - for(; i < num_samples; i++) - { - float sample = in[i] * 32768.0f; - if (sample >= 32766.5) - out[i] = (drmp3_int16) 32767; - else if (sample <= -32767.5) - out[i] = (drmp3_int16)-32768; - else - { - short s = (drmp3_int16)(sample + .5f); - s -= (s < 0); /* away from zero, to be compliant */ - out[i] = s; - } - } - } -} - - - -/************************************************************************************************************************************************************ - - Main Public API - - ************************************************************************************************************************************************************/ - -#if defined(SIZE_MAX) - #define DRMP3_SIZE_MAX SIZE_MAX -#else - #if defined(_WIN64) || defined(_LP64) || defined(__LP64__) - #define DRMP3_SIZE_MAX ((drmp3_uint64)0xFFFFFFFFFFFFFFFF) - #else - #define DRMP3_SIZE_MAX 0xFFFFFFFF - #endif -#endif - -/* Options. */ -#ifndef DRMP3_SEEK_LEADING_MP3_FRAMES -#define DRMP3_SEEK_LEADING_MP3_FRAMES 2 -#endif - - -/* Standard library stuff. */ -#ifndef DRMP3_ASSERT -#include <assert.h> -#define DRMP3_ASSERT(expression) assert(expression) -#endif -#ifndef DRMP3_COPY_MEMORY -#define DRMP3_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz)) -#endif -#ifndef DRMP3_ZERO_MEMORY -#define DRMP3_ZERO_MEMORY(p, sz) memset((p), 0, (sz)) -#endif -#define DRMP3_ZERO_OBJECT(p) DRMP3_ZERO_MEMORY((p), sizeof(*(p))) -#ifndef DRMP3_MALLOC -#define DRMP3_MALLOC(sz) malloc((sz)) -#endif -#ifndef DRMP3_REALLOC -#define DRMP3_REALLOC(p, sz) realloc((p), (sz)) -#endif -#ifndef DRMP3_FREE -#define DRMP3_FREE(p) free((p)) -#endif - -#define drmp3_assert DRMP3_ASSERT -#define drmp3_copy_memory DRMP3_COPY_MEMORY -#define drmp3_zero_memory DRMP3_ZERO_MEMORY -#define drmp3_zero_object DRMP3_ZERO_OBJECT -#define drmp3_malloc DRMP3_MALLOC -#define drmp3_realloc DRMP3_REALLOC - -#define drmp3_countof(x) (sizeof(x) / sizeof(x[0])) -#define drmp3_max(x, y) (((x) > (y)) ? (x) : (y)) -#define drmp3_min(x, y) (((x) < (y)) ? (x) : (y)) - -#define DRMP3_DATA_CHUNK_SIZE 16384 /* The size in bytes of each chunk of data to read from the MP3 stream. minimp3 recommends 16K. */ - -static DRMP3_INLINE float drmp3_mix_f32(float x, float y, float a) -{ - return x*(1-a) + y*a; -} - -static void drmp3_blend_f32(float* pOut, float* pInA, float* pInB, float factor, drmp3_uint32 channels) -{ - drmp3_uint32 i; - for (i = 0; i < channels; ++i) { - pOut[i] = drmp3_mix_f32(pInA[i], pInB[i], factor); - } -} - -void drmp3_src_cache_init(drmp3_src* pSRC, drmp3_src_cache* pCache) -{ - drmp3_assert(pSRC != NULL); - drmp3_assert(pCache != NULL); - - pCache->pSRC = pSRC; - pCache->cachedFrameCount = 0; - pCache->iNextFrame = 0; -} - -drmp3_uint64 drmp3_src_cache_read_frames(drmp3_src_cache* pCache, drmp3_uint64 frameCount, float* pFramesOut) -{ - drmp3_uint32 channels; - drmp3_uint64 totalFramesRead = 0; - - drmp3_assert(pCache != NULL); - drmp3_assert(pCache->pSRC != NULL); - drmp3_assert(pCache->pSRC->onRead != NULL); - drmp3_assert(frameCount > 0); - drmp3_assert(pFramesOut != NULL); - - channels = pCache->pSRC->config.channels; - - while (frameCount > 0) { - /* If there's anything in memory go ahead and copy that over first. */ - drmp3_uint32 framesToReadFromClient; - drmp3_uint64 framesRemainingInMemory = pCache->cachedFrameCount - pCache->iNextFrame; - drmp3_uint64 framesToReadFromMemory = frameCount; - if (framesToReadFromMemory > framesRemainingInMemory) { - framesToReadFromMemory = framesRemainingInMemory; - } - - drmp3_copy_memory(pFramesOut, pCache->pCachedFrames + pCache->iNextFrame*channels, (drmp3_uint32)(framesToReadFromMemory * channels * sizeof(float))); - pCache->iNextFrame += (drmp3_uint32)framesToReadFromMemory; - - totalFramesRead += framesToReadFromMemory; - frameCount -= framesToReadFromMemory; - if (frameCount == 0) { - break; - } - - - /* At this point there are still more frames to read from the client, so we'll need to reload the cache with fresh data. */ - drmp3_assert(frameCount > 0); - pFramesOut += framesToReadFromMemory * channels; - - pCache->iNextFrame = 0; - pCache->cachedFrameCount = 0; - - framesToReadFromClient = drmp3_countof(pCache->pCachedFrames) / pCache->pSRC->config.channels; - if (framesToReadFromClient > pCache->pSRC->config.cacheSizeInFrames) { - framesToReadFromClient = pCache->pSRC->config.cacheSizeInFrames; - } - - pCache->cachedFrameCount = (drmp3_uint32)pCache->pSRC->onRead(pCache->pSRC, framesToReadFromClient, pCache->pCachedFrames, pCache->pSRC->pUserData); - - - /* Get out of this loop if nothing was able to be retrieved. */ - if (pCache->cachedFrameCount == 0) { - break; - } - } - - return totalFramesRead; -} - - -drmp3_uint64 drmp3_src_read_frames_passthrough(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, drmp3_bool32 flush); -drmp3_uint64 drmp3_src_read_frames_linear(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, drmp3_bool32 flush); - -drmp3_bool32 drmp3_src_init(const drmp3_src_config* pConfig, drmp3_src_read_proc onRead, void* pUserData, drmp3_src* pSRC) -{ - if (pSRC == NULL) { - return DRMP3_FALSE; - } - - drmp3_zero_object(pSRC); - - if (pConfig == NULL || onRead == NULL) { - return DRMP3_FALSE; - } - - if (pConfig->channels == 0 || pConfig->channels > 2) { - return DRMP3_FALSE; - } - - pSRC->config = *pConfig; - pSRC->onRead = onRead; - pSRC->pUserData = pUserData; - - if (pSRC->config.cacheSizeInFrames > DRMP3_SRC_CACHE_SIZE_IN_FRAMES || pSRC->config.cacheSizeInFrames == 0) { - pSRC->config.cacheSizeInFrames = DRMP3_SRC_CACHE_SIZE_IN_FRAMES; - } - - drmp3_src_cache_init(pSRC, &pSRC->cache); - return DRMP3_TRUE; -} - -drmp3_bool32 drmp3_src_set_input_sample_rate(drmp3_src* pSRC, drmp3_uint32 sampleRateIn) -{ - if (pSRC == NULL) { - return DRMP3_FALSE; - } - - /* Must have a sample rate of > 0. */ - if (sampleRateIn == 0) { - return DRMP3_FALSE; - } - - pSRC->config.sampleRateIn = sampleRateIn; - return DRMP3_TRUE; -} - -drmp3_bool32 drmp3_src_set_output_sample_rate(drmp3_src* pSRC, drmp3_uint32 sampleRateOut) -{ - if (pSRC == NULL) { - return DRMP3_FALSE; - } - - /* Must have a sample rate of > 0. */ - if (sampleRateOut == 0) { - return DRMP3_FALSE; - } - - pSRC->config.sampleRateOut = sampleRateOut; - return DRMP3_TRUE; -} - -drmp3_uint64 drmp3_src_read_frames_ex(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, drmp3_bool32 flush) -{ - drmp3_src_algorithm algorithm; - - if (pSRC == NULL || frameCount == 0 || pFramesOut == NULL) { - return 0; - } - - algorithm = pSRC->config.algorithm; - - /* Always use passthrough if the sample rates are the same. */ - if (pSRC->config.sampleRateIn == pSRC->config.sampleRateOut) { - algorithm = drmp3_src_algorithm_none; - } - - /* Could just use a function pointer instead of a switch for this... */ - switch (algorithm) - { - case drmp3_src_algorithm_none: return drmp3_src_read_frames_passthrough(pSRC, frameCount, pFramesOut, flush); - case drmp3_src_algorithm_linear: return drmp3_src_read_frames_linear(pSRC, frameCount, pFramesOut, flush); - default: return 0; - } -} - -drmp3_uint64 drmp3_src_read_frames(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut) -{ - return drmp3_src_read_frames_ex(pSRC, frameCount, pFramesOut, DRMP3_FALSE); -} - -drmp3_uint64 drmp3_src_read_frames_passthrough(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, drmp3_bool32 flush) -{ - drmp3_assert(pSRC != NULL); - drmp3_assert(frameCount > 0); - drmp3_assert(pFramesOut != NULL); - - (void)flush; /* Passthrough need not care about flushing. */ - return pSRC->onRead(pSRC, frameCount, pFramesOut, pSRC->pUserData); -} - -drmp3_uint64 drmp3_src_read_frames_linear(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, drmp3_bool32 flush) -{ - double factor; - drmp3_uint64 totalFramesRead; - - drmp3_assert(pSRC != NULL); - drmp3_assert(frameCount > 0); - drmp3_assert(pFramesOut != NULL); - - /* For linear SRC, the bin is only 2 frames: 1 prior, 1 future. */ - - /* Load the bin if necessary. */ - if (!pSRC->algo.linear.isPrevFramesLoaded) { - drmp3_uint64 framesRead = drmp3_src_cache_read_frames(&pSRC->cache, 1, pSRC->bin); - if (framesRead == 0) { - return 0; - } - pSRC->algo.linear.isPrevFramesLoaded = DRMP3_TRUE; - } - if (!pSRC->algo.linear.isNextFramesLoaded) { - drmp3_uint64 framesRead = drmp3_src_cache_read_frames(&pSRC->cache, 1, pSRC->bin + pSRC->config.channels); - if (framesRead == 0) { - return 0; - } - pSRC->algo.linear.isNextFramesLoaded = DRMP3_TRUE; - } - - factor = (double)pSRC->config.sampleRateIn / pSRC->config.sampleRateOut; - - totalFramesRead = 0; - while (frameCount > 0) { - drmp3_uint32 i; - drmp3_uint32 framesToReadFromClient; - - /* The bin is where the previous and next frames are located. */ - float* pPrevFrame = pSRC->bin; - float* pNextFrame = pSRC->bin + pSRC->config.channels; - - drmp3_blend_f32((float*)pFramesOut, pPrevFrame, pNextFrame, (float)pSRC->algo.linear.alpha, pSRC->config.channels); - - pSRC->algo.linear.alpha += factor; - - /* The new alpha value is how we determine whether or not we need to read fresh frames. */ - framesToReadFromClient = (drmp3_uint32)pSRC->algo.linear.alpha; - pSRC->algo.linear.alpha = pSRC->algo.linear.alpha - framesToReadFromClient; - - for (i = 0; i < framesToReadFromClient; ++i) { - drmp3_uint64 framesRead; - drmp3_uint32 j; - - for (j = 0; j < pSRC->config.channels; ++j) { - pPrevFrame[j] = pNextFrame[j]; - } - - framesRead = drmp3_src_cache_read_frames(&pSRC->cache, 1, pNextFrame); - if (framesRead == 0) { - drmp3_uint32 k; - for (k = 0; k < pSRC->config.channels; ++k) { - pNextFrame[k] = 0; - } - - if (pSRC->algo.linear.isNextFramesLoaded) { - pSRC->algo.linear.isNextFramesLoaded = DRMP3_FALSE; - } else { - if (flush) { - pSRC->algo.linear.isPrevFramesLoaded = DRMP3_FALSE; - } - } - - break; - } - } - - pFramesOut = (drmp3_uint8*)pFramesOut + (1 * pSRC->config.channels * sizeof(float)); - frameCount -= 1; - totalFramesRead += 1; - - /* If there's no frames available we need to get out of this loop. */ - if (!pSRC->algo.linear.isNextFramesLoaded && (!flush || !pSRC->algo.linear.isPrevFramesLoaded)) { - break; - } - } - - return totalFramesRead; -} - - -static size_t drmp3__on_read(drmp3* pMP3, void* pBufferOut, size_t bytesToRead) -{ - size_t bytesRead = pMP3->onRead(pMP3->pUserData, pBufferOut, bytesToRead); - pMP3->streamCursor += bytesRead; - return bytesRead; -} - -static drmp3_bool32 drmp3__on_seek(drmp3* pMP3, int offset, drmp3_seek_origin origin) -{ - drmp3_assert(offset >= 0); - - if (!pMP3->onSeek(pMP3->pUserData, offset, origin)) { - return DRMP3_FALSE; - } - - if (origin == drmp3_seek_origin_start) { - pMP3->streamCursor = (drmp3_uint64)offset; - } else { - pMP3->streamCursor += offset; - } - - return DRMP3_TRUE; -} - -static drmp3_bool32 drmp3__on_seek_64(drmp3* pMP3, drmp3_uint64 offset, drmp3_seek_origin origin) -{ - if (offset <= 0x7FFFFFFF) { - return drmp3__on_seek(pMP3, (int)offset, origin); - } - - - /* Getting here "offset" is too large for a 32-bit integer. We just keep seeking forward until we hit the offset. */ - if (!drmp3__on_seek(pMP3, 0x7FFFFFFF, drmp3_seek_origin_start)) { - return DRMP3_FALSE; - } - - offset -= 0x7FFFFFFF; - while (offset > 0) { - if (offset <= 0x7FFFFFFF) { - if (!drmp3__on_seek(pMP3, (int)offset, drmp3_seek_origin_current)) { - return DRMP3_FALSE; - } - offset = 0; - } else { - if (!drmp3__on_seek(pMP3, 0x7FFFFFFF, drmp3_seek_origin_current)) { - return DRMP3_FALSE; - } - offset -= 0x7FFFFFFF; - } - } - - return DRMP3_TRUE; -} - -static drmp3_uint32 drmp3_decode_next_frame_ex(drmp3* pMP3, drmp3d_sample_t* pPCMFrames, drmp3_bool32 discard); -static drmp3_uint32 drmp3_decode_next_frame(drmp3* pMP3); - -static drmp3_uint64 drmp3_read_src(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, void* pUserData) -{ - drmp3* pMP3 = (drmp3*)pUserData; - float* pFramesOutF = (float*)pFramesOut; - drmp3_uint64 totalFramesRead = 0; - - drmp3_assert(pMP3 != NULL); - drmp3_assert(pMP3->onRead != NULL); - - while (frameCount > 0) { - /* Read from the in-memory buffer first. */ - while (pMP3->pcmFramesRemainingInMP3Frame > 0 && frameCount > 0) { - drmp3d_sample_t* frames = (drmp3d_sample_t*)pMP3->pcmFrames; -#ifndef DR_MP3_FLOAT_OUTPUT - if (pMP3->mp3FrameChannels == 1) { - if (pMP3->channels == 1) { - /* Mono -> Mono. */ - pFramesOutF[0] = frames[pMP3->pcmFramesConsumedInMP3Frame] / 32768.0f; - } else { - /* Mono -> Stereo. */ - pFramesOutF[0] = frames[pMP3->pcmFramesConsumedInMP3Frame] / 32768.0f; - pFramesOutF[1] = frames[pMP3->pcmFramesConsumedInMP3Frame] / 32768.0f; - } - } else { - if (pMP3->channels == 1) { - /* Stereo -> Mono */ - float sample = 0; - sample += frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+0] / 32768.0f; - sample += frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+1] / 32768.0f; - pFramesOutF[0] = sample * 0.5f; - } else { - /* Stereo -> Stereo */ - pFramesOutF[0] = frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+0] / 32768.0f; - pFramesOutF[1] = frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+1] / 32768.0f; - } - } -#else - if (pMP3->mp3FrameChannels == 1) { - if (pMP3->channels == 1) { - /* Mono -> Mono. */ - pFramesOutF[0] = frames[pMP3->pcmFramesConsumedInMP3Frame]; - } else { - /* Mono -> Stereo. */ - pFramesOutF[0] = frames[pMP3->pcmFramesConsumedInMP3Frame]; - pFramesOutF[1] = frames[pMP3->pcmFramesConsumedInMP3Frame]; - } - } else { - if (pMP3->channels == 1) { - /* Stereo -> Mono */ - float sample = 0; - sample += frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+0]; - sample += frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+1]; - pFramesOutF[0] = sample * 0.5f; - } else { - /* Stereo -> Stereo */ - pFramesOutF[0] = frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+0]; - pFramesOutF[1] = frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+1]; - } - } -#endif - - pMP3->pcmFramesConsumedInMP3Frame += 1; - pMP3->pcmFramesRemainingInMP3Frame -= 1; - totalFramesRead += 1; - frameCount -= 1; - pFramesOutF += pSRC->config.channels; - } - - if (frameCount == 0) { - break; - } - - drmp3_assert(pMP3->pcmFramesRemainingInMP3Frame == 0); - - /* - At this point we have exhausted our in-memory buffer so we need to re-fill. Note that the sample rate may have changed - at this point which means we'll also need to update our sample rate conversion pipeline. - */ - if (drmp3_decode_next_frame(pMP3) == 0) { - break; - } - } - - return totalFramesRead; -} - -static drmp3_bool32 drmp3_init_src(drmp3* pMP3) -{ - drmp3_src_config srcConfig; - drmp3_zero_object(&srcConfig); - srcConfig.sampleRateIn = DR_MP3_DEFAULT_SAMPLE_RATE; - srcConfig.sampleRateOut = pMP3->sampleRate; - srcConfig.channels = pMP3->channels; - srcConfig.algorithm = drmp3_src_algorithm_linear; - if (!drmp3_src_init(&srcConfig, drmp3_read_src, pMP3, &pMP3->src)) { - drmp3_uninit(pMP3); - return DRMP3_FALSE; - } - - return DRMP3_TRUE; -} - -static drmp3_uint32 drmp3_decode_next_frame_ex(drmp3* pMP3, drmp3d_sample_t* pPCMFrames, drmp3_bool32 discard) -{ - drmp3_uint32 pcmFramesRead = 0; - - drmp3_assert(pMP3 != NULL); - drmp3_assert(pMP3->onRead != NULL); - - if (pMP3->atEnd) { - return 0; - } - - do { - drmp3dec_frame_info info; - size_t leftoverDataSize; - - /* minimp3 recommends doing data submission in 16K chunks. If we don't have at least 16K bytes available, get more. */ - if (pMP3->dataSize < DRMP3_DATA_CHUNK_SIZE) { - size_t bytesRead; - - if (pMP3->dataCapacity < DRMP3_DATA_CHUNK_SIZE) { - drmp3_uint8* pNewData; - - pMP3->dataCapacity = DRMP3_DATA_CHUNK_SIZE; - pNewData = (drmp3_uint8*)drmp3_realloc(pMP3->pData, pMP3->dataCapacity); - if (pNewData == NULL) { - return 0; /* Out of memory. */ - } - - pMP3->pData = pNewData; - } - - bytesRead = drmp3__on_read(pMP3, pMP3->pData + pMP3->dataSize, (pMP3->dataCapacity - pMP3->dataSize)); - if (bytesRead == 0) { - if (pMP3->dataSize == 0) { - pMP3->atEnd = DRMP3_TRUE; - return 0; /* No data. */ - } - } - - pMP3->dataSize += bytesRead; - } - - if (pMP3->dataSize > INT_MAX) { - pMP3->atEnd = DRMP3_TRUE; - return 0; /* File too big. */ - } - - pcmFramesRead = drmp3dec_decode_frame(&pMP3->decoder, pMP3->pData, (int)pMP3->dataSize, pPCMFrames, &info); /* <-- Safe size_t -> int conversion thanks to the check above. */ - - /* Consume the data. */ - leftoverDataSize = (pMP3->dataSize - (size_t)info.frame_bytes); - if (info.frame_bytes > 0) { - memmove(pMP3->pData, pMP3->pData + info.frame_bytes, leftoverDataSize); - pMP3->dataSize = leftoverDataSize; - } - - /* - pcmFramesRead will be equal to 0 if decoding failed. If it is zero and info.frame_bytes > 0 then we have successfully - decoded the frame. A special case is if we are wanting to discard the frame, in which case we return successfully. - */ - if (pcmFramesRead > 0 || (info.frame_bytes > 0 && discard)) { - pcmFramesRead = drmp3_hdr_frame_samples(pMP3->decoder.header); - pMP3->pcmFramesConsumedInMP3Frame = 0; - pMP3->pcmFramesRemainingInMP3Frame = pcmFramesRead; - pMP3->mp3FrameChannels = info.channels; - pMP3->mp3FrameSampleRate = info.hz; - - /* We need to initialize the resampler if we don't yet have the channel count or sample rate. */ - if (pMP3->channels == 0 || pMP3->sampleRate == 0) { - if (pMP3->channels == 0) { - pMP3->channels = info.channels; - } - if (pMP3->sampleRate == 0) { - pMP3->sampleRate = info.hz; - } - drmp3_init_src(pMP3); - } - - drmp3_src_set_input_sample_rate(&pMP3->src, pMP3->mp3FrameSampleRate); - break; - } else if (info.frame_bytes == 0) { - size_t bytesRead; - - /* Need more data. minimp3 recommends doing data submission in 16K chunks. */ - if (pMP3->dataCapacity == pMP3->dataSize) { - drmp3_uint8* pNewData; - - /* No room. Expand. */ - pMP3->dataCapacity += DRMP3_DATA_CHUNK_SIZE; - pNewData = (drmp3_uint8*)drmp3_realloc(pMP3->pData, pMP3->dataCapacity); - if (pNewData == NULL) { - return 0; /* Out of memory. */ - } - - pMP3->pData = pNewData; - } - - /* Fill in a chunk. */ - bytesRead = drmp3__on_read(pMP3, pMP3->pData + pMP3->dataSize, (pMP3->dataCapacity - pMP3->dataSize)); - if (bytesRead == 0) { - pMP3->atEnd = DRMP3_TRUE; - return 0; /* Error reading more data. */ - } - - pMP3->dataSize += bytesRead; - } - } while (DRMP3_TRUE); - - return pcmFramesRead; -} - -static drmp3_uint32 drmp3_decode_next_frame(drmp3* pMP3) -{ - drmp3_assert(pMP3 != NULL); - return drmp3_decode_next_frame_ex(pMP3, (drmp3d_sample_t*)pMP3->pcmFrames, DRMP3_FALSE); -} - -#if 0 -static drmp3_uint32 drmp3_seek_next_frame(drmp3* pMP3) -{ - drmp3_uint32 pcmFrameCount; - - drmp3_assert(pMP3 != NULL); - - pcmFrameCount = drmp3_decode_next_frame_ex(pMP3, NULL); - if (pcmFrameCount == 0) { - return 0; - } - - /* We have essentially just skipped past the frame, so just set the remaining samples to 0. */ - pMP3->currentPCMFrame += pcmFrameCount; - pMP3->pcmFramesConsumedInMP3Frame = pcmFrameCount; - pMP3->pcmFramesRemainingInMP3Frame = 0; - - return pcmFrameCount; -} -#endif - -drmp3_bool32 drmp3_init_internal(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, const drmp3_config* pConfig) -{ - drmp3_config config; - - drmp3_assert(pMP3 != NULL); - drmp3_assert(onRead != NULL); - - /* This function assumes the output object has already been reset to 0. Do not do that here, otherwise things will break. */ - drmp3dec_init(&pMP3->decoder); - - /* The config can be null in which case we use defaults. */ - if (pConfig != NULL) { - config = *pConfig; - } else { - drmp3_zero_object(&config); - } - - pMP3->channels = config.outputChannels; - - /* Cannot have more than 2 channels. */ - if (pMP3->channels > 2) { - pMP3->channels = 2; - } - - pMP3->sampleRate = config.outputSampleRate; - - pMP3->onRead = onRead; - pMP3->onSeek = onSeek; - pMP3->pUserData = pUserData; - - /* - We need a sample rate converter for converting the sample rate from the MP3 frames to the requested output sample rate. Note that if - we don't yet know the channel count or sample rate we defer this until the first frame is read. - */ - if (pMP3->channels != 0 && pMP3->sampleRate != 0) { - drmp3_init_src(pMP3); - } - - /* Decode the first frame to confirm that it is indeed a valid MP3 stream. */ - if (!drmp3_decode_next_frame(pMP3)) { - drmp3_uninit(pMP3); - return DRMP3_FALSE; /* Not a valid MP3 stream. */ - } - - return DRMP3_TRUE; -} - -drmp3_bool32 drmp3_init(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, const drmp3_config* pConfig) -{ - if (pMP3 == NULL || onRead == NULL) { - return DRMP3_FALSE; - } - - drmp3_zero_object(pMP3); - return drmp3_init_internal(pMP3, onRead, onSeek, pUserData, pConfig); -} - - -static size_t drmp3__on_read_memory(void* pUserData, void* pBufferOut, size_t bytesToRead) -{ - drmp3* pMP3 = (drmp3*)pUserData; - size_t bytesRemaining; - - drmp3_assert(pMP3 != NULL); - drmp3_assert(pMP3->memory.dataSize >= pMP3->memory.currentReadPos); - - bytesRemaining = pMP3->memory.dataSize - pMP3->memory.currentReadPos; - if (bytesToRead > bytesRemaining) { - bytesToRead = bytesRemaining; - } - - if (bytesToRead > 0) { - drmp3_copy_memory(pBufferOut, pMP3->memory.pData + pMP3->memory.currentReadPos, bytesToRead); - pMP3->memory.currentReadPos += bytesToRead; - } - - return bytesToRead; -} - -static drmp3_bool32 drmp3__on_seek_memory(void* pUserData, int byteOffset, drmp3_seek_origin origin) -{ - drmp3* pMP3 = (drmp3*)pUserData; - - drmp3_assert(pMP3 != NULL); - - if (origin == drmp3_seek_origin_current) { - if (byteOffset > 0) { - if (pMP3->memory.currentReadPos + byteOffset > pMP3->memory.dataSize) { - byteOffset = (int)(pMP3->memory.dataSize - pMP3->memory.currentReadPos); /* Trying to seek too far forward. */ - } - } else { - if (pMP3->memory.currentReadPos < (size_t)-byteOffset) { - byteOffset = -(int)pMP3->memory.currentReadPos; /* Trying to seek too far backwards. */ - } - } - - /* This will never underflow thanks to the clamps above. */ - pMP3->memory.currentReadPos += byteOffset; - } else { - if ((drmp3_uint32)byteOffset <= pMP3->memory.dataSize) { - pMP3->memory.currentReadPos = byteOffset; - } else { - pMP3->memory.currentReadPos = pMP3->memory.dataSize; /* Trying to seek too far forward. */ - } - } - - return DRMP3_TRUE; -} - -drmp3_bool32 drmp3_init_memory(drmp3* pMP3, const void* pData, size_t dataSize, const drmp3_config* pConfig) -{ - if (pMP3 == NULL) { - return DRMP3_FALSE; - } - - drmp3_zero_object(pMP3); - - if (pData == NULL || dataSize == 0) { - return DRMP3_FALSE; - } - - pMP3->memory.pData = (const drmp3_uint8*)pData; - pMP3->memory.dataSize = dataSize; - pMP3->memory.currentReadPos = 0; - - return drmp3_init_internal(pMP3, drmp3__on_read_memory, drmp3__on_seek_memory, pMP3, pConfig); -} - - -#ifndef DR_MP3_NO_STDIO -#include <stdio.h> - -static size_t drmp3__on_read_stdio(void* pUserData, void* pBufferOut, size_t bytesToRead) -{ - return fread(pBufferOut, 1, bytesToRead, (FILE*)pUserData); -} - -static drmp3_bool32 drmp3__on_seek_stdio(void* pUserData, int offset, drmp3_seek_origin origin) -{ - return fseek((FILE*)pUserData, offset, (origin == drmp3_seek_origin_current) ? SEEK_CUR : SEEK_SET) == 0; -} - -drmp3_bool32 drmp3_init_file(drmp3* pMP3, const char* filePath, const drmp3_config* pConfig) -{ - FILE* pFile; -#if defined(_MSC_VER) && _MSC_VER >= 1400 - if (fopen_s(&pFile, filePath, "rb") != 0) { - return DRMP3_FALSE; - } -#else - pFile = fopen(filePath, "rb"); - if (pFile == NULL) { - return DRMP3_FALSE; - } -#endif - - return drmp3_init(pMP3, drmp3__on_read_stdio, drmp3__on_seek_stdio, (void*)pFile, pConfig); -} -#endif - -void drmp3_uninit(drmp3* pMP3) -{ - if (pMP3 == NULL) { - return; - } - -#ifndef DR_MP3_NO_STDIO - if (pMP3->onRead == drmp3__on_read_stdio) { - fclose((FILE*)pMP3->pUserData); - } -#endif - - drmp3_free(pMP3->pData); -} - -drmp3_uint64 drmp3_read_pcm_frames_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut) -{ - drmp3_uint64 totalFramesRead = 0; - - if (pMP3 == NULL || pMP3->onRead == NULL) { - return 0; - } - - if (pBufferOut == NULL) { - float temp[4096]; - while (framesToRead > 0) { - drmp3_uint64 framesJustRead; - drmp3_uint64 framesToReadRightNow = sizeof(temp)/sizeof(temp[0]) / pMP3->channels; - if (framesToReadRightNow > framesToRead) { - framesToReadRightNow = framesToRead; - } - - framesJustRead = drmp3_read_pcm_frames_f32(pMP3, framesToReadRightNow, temp); - if (framesJustRead == 0) { - break; - } - - framesToRead -= framesJustRead; - totalFramesRead += framesJustRead; - } - } else { - totalFramesRead = drmp3_src_read_frames_ex(&pMP3->src, framesToRead, pBufferOut, DRMP3_TRUE); - pMP3->currentPCMFrame += totalFramesRead; - } - - return totalFramesRead; -} - -drmp3_uint64 drmp3_read_pcm_frames_s16(drmp3* pMP3, drmp3_uint64 framesToRead, drmp3_int16* pBufferOut) -{ - float tempF32[4096]; - drmp3_uint64 pcmFramesJustRead; - drmp3_uint64 totalPCMFramesRead = 0; - - if (pMP3 == NULL || pMP3->onRead == NULL) { - return 0; - } - - /* Naive implementation: read into a temp f32 buffer, then convert. */ - for (;;) { - drmp3_uint64 pcmFramesToReadThisIteration = (framesToRead - totalPCMFramesRead); - if (pcmFramesToReadThisIteration > drmp3_countof(tempF32)/pMP3->channels) { - pcmFramesToReadThisIteration = drmp3_countof(tempF32)/pMP3->channels; - } - - pcmFramesJustRead = drmp3_read_pcm_frames_f32(pMP3, pcmFramesToReadThisIteration, tempF32); - if (pcmFramesJustRead == 0) { - break; - } - - drmp3dec_f32_to_s16(tempF32, pBufferOut, (int)(pcmFramesJustRead * pMP3->channels)); /* <-- Safe cast since pcmFramesJustRead will be clamped based on the size of tempF32 which is always small. */ - pBufferOut += pcmFramesJustRead * pMP3->channels; - - totalPCMFramesRead += pcmFramesJustRead; - - if (pcmFramesJustRead < pcmFramesToReadThisIteration) { - break; - } - } - - return totalPCMFramesRead; -} - -void drmp3_reset(drmp3* pMP3) -{ - drmp3_assert(pMP3 != NULL); - - pMP3->pcmFramesConsumedInMP3Frame = 0; - pMP3->pcmFramesRemainingInMP3Frame = 0; - pMP3->currentPCMFrame = 0; - pMP3->dataSize = 0; - pMP3->atEnd = DRMP3_FALSE; - pMP3->src.bin[0] = 0; - pMP3->src.bin[1] = 0; - pMP3->src.bin[2] = 0; - pMP3->src.bin[3] = 0; - pMP3->src.cache.cachedFrameCount = 0; - pMP3->src.cache.iNextFrame = 0; - pMP3->src.algo.linear.alpha = 0; - pMP3->src.algo.linear.isNextFramesLoaded = 0; - pMP3->src.algo.linear.isPrevFramesLoaded = 0; - drmp3dec_init(&pMP3->decoder); -} - -drmp3_bool32 drmp3_seek_to_start_of_stream(drmp3* pMP3) -{ - drmp3_assert(pMP3 != NULL); - drmp3_assert(pMP3->onSeek != NULL); - - /* Seek to the start of the stream to begin with. */ - if (!drmp3__on_seek(pMP3, 0, drmp3_seek_origin_start)) { - return DRMP3_FALSE; - } - - /* Clear any cached data. */ - drmp3_reset(pMP3); - return DRMP3_TRUE; -} - -float drmp3_get_cached_pcm_frame_count_from_src(drmp3* pMP3) -{ - return (pMP3->src.cache.cachedFrameCount - pMP3->src.cache.iNextFrame) + (float)pMP3->src.algo.linear.alpha; -} - -float drmp3_get_pcm_frames_remaining_in_mp3_frame(drmp3* pMP3) -{ - float factor = (float)pMP3->src.config.sampleRateOut / (float)pMP3->src.config.sampleRateIn; - float frameCountPreSRC = drmp3_get_cached_pcm_frame_count_from_src(pMP3) + pMP3->pcmFramesRemainingInMP3Frame; - return frameCountPreSRC * factor; -} - -/* -NOTE ON SEEKING -=============== -The seeking code below is a complete mess and is broken for cases when the sample rate changes. The problem -is with the resampling and the crappy resampler used by dr_mp3. What needs to happen is the following: - -1) The resampler needs to be replaced. -2) The resampler has state which needs to be updated whenever an MP3 frame is decoded outside of - drmp3_read_pcm_frames_f32(). The resampler needs an API to "flush" some imaginary input so that it's - state is updated accordingly. -*/ -drmp3_bool32 drmp3_seek_forward_by_pcm_frames__brute_force(drmp3* pMP3, drmp3_uint64 frameOffset) -{ - drmp3_uint64 framesRead; - -#if 0 - /* - MP3 is a bit annoying when it comes to seeking because of the bit reservoir. It basically means that an MP3 frame can possibly - depend on some of the data of prior frames. This means it's not as simple as seeking to the first byte of the MP3 frame that - contains the sample because that MP3 frame will need the data from the previous MP3 frame (which we just seeked past!). To - resolve this we seek past a number of MP3 frames up to a point, and then read-and-discard the remainder. - */ - drmp3_uint64 maxFramesToReadAndDiscard = (drmp3_uint64)(DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME * 3 * ((float)pMP3->src.config.sampleRateOut / (float)pMP3->src.config.sampleRateIn)); - - /* Now get rid of leading whole frames. */ - while (frameOffset > maxFramesToReadAndDiscard) { - float pcmFramesRemainingInCurrentMP3FrameF = drmp3_get_pcm_frames_remaining_in_mp3_frame(pMP3); - drmp3_uint32 pcmFramesRemainingInCurrentMP3Frame = (drmp3_uint32)pcmFramesRemainingInCurrentMP3FrameF; - if (frameOffset > pcmFramesRemainingInCurrentMP3Frame) { - frameOffset -= pcmFramesRemainingInCurrentMP3Frame; - pMP3->currentPCMFrame += pcmFramesRemainingInCurrentMP3Frame; - pMP3->pcmFramesConsumedInMP3Frame += pMP3->pcmFramesRemainingInMP3Frame; - pMP3->pcmFramesRemainingInMP3Frame = 0; - } else { - break; - } - - drmp3_uint32 pcmFrameCount = drmp3_decode_next_frame_ex(pMP3, pMP3->pcmFrames, DRMP3_FALSE); - if (pcmFrameCount == 0) { - break; - } - } - - /* The last step is to read-and-discard any remaining PCM frames to make it sample-exact. */ - framesRead = drmp3_read_pcm_frames_f32(pMP3, frameOffset, NULL); - if (framesRead != frameOffset) { - return DRMP3_FALSE; - } -#else - /* Just using a dumb read-and-discard for now pending updates to the resampler. */ - framesRead = drmp3_read_pcm_frames_f32(pMP3, frameOffset, NULL); - if (framesRead != frameOffset) { - return DRMP3_FALSE; - } -#endif - - return DRMP3_TRUE; -} - -drmp3_bool32 drmp3_seek_to_pcm_frame__brute_force(drmp3* pMP3, drmp3_uint64 frameIndex) -{ - drmp3_assert(pMP3 != NULL); - - if (frameIndex == pMP3->currentPCMFrame) { - return DRMP3_TRUE; - } - - /* - If we're moving foward we just read from where we're at. Otherwise we need to move back to the start of - the stream and read from the beginning. - */ - if (frameIndex < pMP3->currentPCMFrame) { - /* Moving backward. Move to the start of the stream and then move forward. */ - if (!drmp3_seek_to_start_of_stream(pMP3)) { - return DRMP3_FALSE; - } - } - - drmp3_assert(frameIndex >= pMP3->currentPCMFrame); - return drmp3_seek_forward_by_pcm_frames__brute_force(pMP3, (frameIndex - pMP3->currentPCMFrame)); -} - -drmp3_bool32 drmp3_find_closest_seek_point(drmp3* pMP3, drmp3_uint64 frameIndex, drmp3_uint32* pSeekPointIndex) -{ - drmp3_uint32 iSeekPoint; - - drmp3_assert(pSeekPointIndex != NULL); - - *pSeekPointIndex = 0; - - if (frameIndex < pMP3->pSeekPoints[0].pcmFrameIndex) { - return DRMP3_FALSE; - } - - /* Linear search for simplicity to begin with while I'm getting this thing working. Once it's all working change this to a binary search. */ - for (iSeekPoint = 0; iSeekPoint < pMP3->seekPointCount; ++iSeekPoint) { - if (pMP3->pSeekPoints[iSeekPoint].pcmFrameIndex > frameIndex) { - break; /* Found it. */ - } - - *pSeekPointIndex = iSeekPoint; - } - - return DRMP3_TRUE; -} - -drmp3_bool32 drmp3_seek_to_pcm_frame__seek_table(drmp3* pMP3, drmp3_uint64 frameIndex) -{ - drmp3_seek_point seekPoint; - drmp3_uint32 priorSeekPointIndex; - drmp3_uint16 iMP3Frame; - drmp3_uint64 leftoverFrames; - - drmp3_assert(pMP3 != NULL); - drmp3_assert(pMP3->pSeekPoints != NULL); - drmp3_assert(pMP3->seekPointCount > 0); - - /* If there is no prior seekpoint it means the target PCM frame comes before the first seek point. Just assume a seekpoint at the start of the file in this case. */ - if (drmp3_find_closest_seek_point(pMP3, frameIndex, &priorSeekPointIndex)) { - seekPoint = pMP3->pSeekPoints[priorSeekPointIndex]; - } else { - seekPoint.seekPosInBytes = 0; - seekPoint.pcmFrameIndex = 0; - seekPoint.mp3FramesToDiscard = 0; - seekPoint.pcmFramesToDiscard = 0; - } - - /* First thing to do is seek to the first byte of the relevant MP3 frame. */ - if (!drmp3__on_seek_64(pMP3, seekPoint.seekPosInBytes, drmp3_seek_origin_start)) { - return DRMP3_FALSE; /* Failed to seek. */ - } - - /* Clear any cached data. */ - drmp3_reset(pMP3); - - /* Whole MP3 frames need to be discarded first. */ - for (iMP3Frame = 0; iMP3Frame < seekPoint.mp3FramesToDiscard; ++iMP3Frame) { - drmp3_uint32 pcmFramesReadPreSRC; - drmp3d_sample_t* pPCMFrames; - - /* Pass in non-null for the last frame because we want to ensure the sample rate converter is preloaded correctly. */ - pPCMFrames = NULL; - if (iMP3Frame == seekPoint.mp3FramesToDiscard-1) { - pPCMFrames = (drmp3d_sample_t*)pMP3->pcmFrames; - } - - /* We first need to decode the next frame, and then we need to flush the resampler. */ - pcmFramesReadPreSRC = drmp3_decode_next_frame_ex(pMP3, pPCMFrames, DRMP3_TRUE); - if (pcmFramesReadPreSRC == 0) { - return DRMP3_FALSE; - } - } - - /* We seeked to an MP3 frame in the raw stream so we need to make sure the current PCM frame is set correctly. */ - pMP3->currentPCMFrame = seekPoint.pcmFrameIndex - seekPoint.pcmFramesToDiscard; - - /* - Update resampler. This is wrong. Need to instead update it on a per MP3 frame basis. Also broken for cases when - the sample rate is being reduced in my testing. Should work fine when the input and output sample rate is the same - or a clean multiple. - */ - pMP3->src.algo.linear.alpha = (drmp3_int64)pMP3->currentPCMFrame * ((double)pMP3->src.config.sampleRateIn / pMP3->src.config.sampleRateOut); /* <-- Cast to int64 is required for VC6. */ - pMP3->src.algo.linear.alpha = pMP3->src.algo.linear.alpha - (drmp3_uint32)(pMP3->src.algo.linear.alpha); - if (pMP3->src.algo.linear.alpha > 0) { - pMP3->src.algo.linear.isPrevFramesLoaded = 1; - } - - /* - Now at this point we can follow the same process as the brute force technique where we just skip over unnecessary MP3 frames and then - read-and-discard at least 2 whole MP3 frames. - */ - leftoverFrames = frameIndex - pMP3->currentPCMFrame; - return drmp3_seek_forward_by_pcm_frames__brute_force(pMP3, leftoverFrames); -} - -drmp3_bool32 drmp3_seek_to_pcm_frame(drmp3* pMP3, drmp3_uint64 frameIndex) -{ - if (pMP3 == NULL || pMP3->onSeek == NULL) { - return DRMP3_FALSE; - } - - if (frameIndex == 0) { - return drmp3_seek_to_start_of_stream(pMP3); - } - - /* Use the seek table if we have one. */ - if (pMP3->pSeekPoints != NULL && pMP3->seekPointCount > 0) { - return drmp3_seek_to_pcm_frame__seek_table(pMP3, frameIndex); - } else { - return drmp3_seek_to_pcm_frame__brute_force(pMP3, frameIndex); - } -} - -drmp3_bool32 drmp3_get_mp3_and_pcm_frame_count(drmp3* pMP3, drmp3_uint64* pMP3FrameCount, drmp3_uint64* pPCMFrameCount) -{ - drmp3_uint64 currentPCMFrame; - drmp3_uint64 totalPCMFrameCount; - drmp3_uint64 totalMP3FrameCount; - float totalPCMFrameCountFractionalPart; - - if (pMP3 == NULL) { - return DRMP3_FALSE; - } - - /* - The way this works is we move back to the start of the stream, iterate over each MP3 frame and calculate the frame count based - on our output sample rate, the seek back to the PCM frame we were sitting on before calling this function. - */ - - /* The stream must support seeking for this to work. */ - if (pMP3->onSeek == NULL) { - return DRMP3_FALSE; - } - - /* We'll need to seek back to where we were, so grab the PCM frame we're currently sitting on so we can restore later. */ - currentPCMFrame = pMP3->currentPCMFrame; - - if (!drmp3_seek_to_start_of_stream(pMP3)) { - return DRMP3_FALSE; - } - - totalPCMFrameCount = 0; - totalMP3FrameCount = 0; - - totalPCMFrameCountFractionalPart = 0; /* <-- With resampling there will be a fractional part to each MP3 frame that we need to accumulate. */ - for (;;) { - drmp3_uint32 pcmFramesInCurrentMP3FrameIn; - float srcRatio; - float pcmFramesInCurrentMP3FrameOutF; - drmp3_uint32 pcmFramesInCurrentMP3FrameOut; - - pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL, DRMP3_FALSE); - if (pcmFramesInCurrentMP3FrameIn == 0) { - break; - } - - srcRatio = (float)pMP3->mp3FrameSampleRate / (float)pMP3->sampleRate; - drmp3_assert(srcRatio > 0); - - pcmFramesInCurrentMP3FrameOutF = totalPCMFrameCountFractionalPart + (pcmFramesInCurrentMP3FrameIn / srcRatio); - pcmFramesInCurrentMP3FrameOut = (drmp3_uint32)pcmFramesInCurrentMP3FrameOutF; - totalPCMFrameCountFractionalPart = pcmFramesInCurrentMP3FrameOutF - pcmFramesInCurrentMP3FrameOut; - totalPCMFrameCount += pcmFramesInCurrentMP3FrameOut; - totalMP3FrameCount += 1; - } - - /* Finally, we need to seek back to where we were. */ - if (!drmp3_seek_to_start_of_stream(pMP3)) { - return DRMP3_FALSE; - } - - if (!drmp3_seek_to_pcm_frame(pMP3, currentPCMFrame)) { - return DRMP3_FALSE; - } - - if (pMP3FrameCount != NULL) { - *pMP3FrameCount = totalMP3FrameCount; - } - if (pPCMFrameCount != NULL) { - *pPCMFrameCount = totalPCMFrameCount; - } - - return DRMP3_TRUE; -} - -drmp3_uint64 drmp3_get_pcm_frame_count(drmp3* pMP3) -{ - drmp3_uint64 totalPCMFrameCount; - if (!drmp3_get_mp3_and_pcm_frame_count(pMP3, NULL, &totalPCMFrameCount)) { - return 0; - } - - return totalPCMFrameCount; -} - -drmp3_uint64 drmp3_get_mp3_frame_count(drmp3* pMP3) -{ - drmp3_uint64 totalMP3FrameCount; - if (!drmp3_get_mp3_and_pcm_frame_count(pMP3, &totalMP3FrameCount, NULL)) { - return 0; - } - - return totalMP3FrameCount; -} - -void drmp3__accumulate_running_pcm_frame_count(drmp3* pMP3, drmp3_uint32 pcmFrameCountIn, drmp3_uint64* pRunningPCMFrameCount, float* pRunningPCMFrameCountFractionalPart) -{ - float srcRatio; - float pcmFrameCountOutF; - drmp3_uint32 pcmFrameCountOut; - - srcRatio = (float)pMP3->mp3FrameSampleRate / (float)pMP3->sampleRate; - drmp3_assert(srcRatio > 0); - - pcmFrameCountOutF = *pRunningPCMFrameCountFractionalPart + (pcmFrameCountIn / srcRatio); - pcmFrameCountOut = (drmp3_uint32)pcmFrameCountOutF; - *pRunningPCMFrameCountFractionalPart = pcmFrameCountOutF - pcmFrameCountOut; - *pRunningPCMFrameCount += pcmFrameCountOut; -} - -typedef struct -{ - drmp3_uint64 bytePos; - drmp3_uint64 pcmFrameIndex; /* <-- After sample rate conversion. */ -} drmp3__seeking_mp3_frame_info; - -drmp3_bool32 drmp3_calculate_seek_points(drmp3* pMP3, drmp3_uint32* pSeekPointCount, drmp3_seek_point* pSeekPoints) -{ - drmp3_uint32 seekPointCount; - drmp3_uint64 currentPCMFrame; - drmp3_uint64 totalMP3FrameCount; - drmp3_uint64 totalPCMFrameCount; - - if (pMP3 == NULL || pSeekPointCount == NULL || pSeekPoints == NULL) { - return DRMP3_FALSE; /* Invalid args. */ - } - - seekPointCount = *pSeekPointCount; - if (seekPointCount == 0) { - return DRMP3_FALSE; /* The client has requested no seek points. Consider this to be invalid arguments since the client has probably not intended this. */ - } - - /* We'll need to seek back to the current sample after calculating the seekpoints so we need to go ahead and grab the current location at the top. */ - currentPCMFrame = pMP3->currentPCMFrame; - - /* We never do more than the total number of MP3 frames and we limit it to 32-bits. */ - if (!drmp3_get_mp3_and_pcm_frame_count(pMP3, &totalMP3FrameCount, &totalPCMFrameCount)) { - return DRMP3_FALSE; - } - - /* If there's less than DRMP3_SEEK_LEADING_MP3_FRAMES+1 frames we just report 1 seek point which will be the very start of the stream. */ - if (totalMP3FrameCount < DRMP3_SEEK_LEADING_MP3_FRAMES+1) { - seekPointCount = 1; - pSeekPoints[0].seekPosInBytes = 0; - pSeekPoints[0].pcmFrameIndex = 0; - pSeekPoints[0].mp3FramesToDiscard = 0; - pSeekPoints[0].pcmFramesToDiscard = 0; - } else { - drmp3_uint64 pcmFramesBetweenSeekPoints; - drmp3__seeking_mp3_frame_info mp3FrameInfo[DRMP3_SEEK_LEADING_MP3_FRAMES+1]; - drmp3_uint64 runningPCMFrameCount = 0; - float runningPCMFrameCountFractionalPart = 0; - drmp3_uint64 nextTargetPCMFrame; - drmp3_uint32 iMP3Frame; - drmp3_uint32 iSeekPoint; - - if (seekPointCount > totalMP3FrameCount-1) { - seekPointCount = (drmp3_uint32)totalMP3FrameCount-1; - } - - pcmFramesBetweenSeekPoints = totalPCMFrameCount / (seekPointCount+1); - - /* - Here is where we actually calculate the seek points. We need to start by moving the start of the stream. We then enumerate over each - MP3 frame. - */ - if (!drmp3_seek_to_start_of_stream(pMP3)) { - return DRMP3_FALSE; - } - - /* - We need to cache the byte positions of the previous MP3 frames. As a new MP3 frame is iterated, we cycle the byte positions in this - array. The value in the first item in this array is the byte position that will be reported in the next seek point. - */ - - /* We need to initialize the array of MP3 byte positions for the leading MP3 frames. */ - for (iMP3Frame = 0; iMP3Frame < DRMP3_SEEK_LEADING_MP3_FRAMES+1; ++iMP3Frame) { - drmp3_uint32 pcmFramesInCurrentMP3FrameIn; - - /* The byte position of the next frame will be the stream's cursor position, minus whatever is sitting in the buffer. */ - drmp3_assert(pMP3->streamCursor >= pMP3->dataSize); - mp3FrameInfo[iMP3Frame].bytePos = pMP3->streamCursor - pMP3->dataSize; - mp3FrameInfo[iMP3Frame].pcmFrameIndex = runningPCMFrameCount; - - /* We need to get information about this frame so we can know how many samples it contained. */ - pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL, DRMP3_FALSE); - if (pcmFramesInCurrentMP3FrameIn == 0) { - return DRMP3_FALSE; /* This should never happen. */ - } - - drmp3__accumulate_running_pcm_frame_count(pMP3, pcmFramesInCurrentMP3FrameIn, &runningPCMFrameCount, &runningPCMFrameCountFractionalPart); - } - - /* - At this point we will have extracted the byte positions of the leading MP3 frames. We can now start iterating over each seek point and - calculate them. - */ - nextTargetPCMFrame = 0; - for (iSeekPoint = 0; iSeekPoint < seekPointCount; ++iSeekPoint) { - nextTargetPCMFrame += pcmFramesBetweenSeekPoints; - - for (;;) { - if (nextTargetPCMFrame < runningPCMFrameCount) { - /* The next seek point is in the current MP3 frame. */ - pSeekPoints[iSeekPoint].seekPosInBytes = mp3FrameInfo[0].bytePos; - pSeekPoints[iSeekPoint].pcmFrameIndex = nextTargetPCMFrame; - pSeekPoints[iSeekPoint].mp3FramesToDiscard = DRMP3_SEEK_LEADING_MP3_FRAMES; - pSeekPoints[iSeekPoint].pcmFramesToDiscard = (drmp3_uint16)(nextTargetPCMFrame - mp3FrameInfo[DRMP3_SEEK_LEADING_MP3_FRAMES-1].pcmFrameIndex); - break; - } else { - size_t i; - drmp3_uint32 pcmFramesInCurrentMP3FrameIn; - - /* - The next seek point is not in the current MP3 frame, so continue on to the next one. The first thing to do is cycle the cached - MP3 frame info. - */ - for (i = 0; i < drmp3_countof(mp3FrameInfo)-1; ++i) { - mp3FrameInfo[i] = mp3FrameInfo[i+1]; - } - - /* Cache previous MP3 frame info. */ - mp3FrameInfo[drmp3_countof(mp3FrameInfo)-1].bytePos = pMP3->streamCursor - pMP3->dataSize; - mp3FrameInfo[drmp3_countof(mp3FrameInfo)-1].pcmFrameIndex = runningPCMFrameCount; - - /* - Go to the next MP3 frame. This shouldn't ever fail, but just in case it does we just set the seek point and break. If it happens, it - should only ever do it for the last seek point. - */ - pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL, DRMP3_TRUE); - if (pcmFramesInCurrentMP3FrameIn == 0) { - pSeekPoints[iSeekPoint].seekPosInBytes = mp3FrameInfo[0].bytePos; - pSeekPoints[iSeekPoint].pcmFrameIndex = nextTargetPCMFrame; - pSeekPoints[iSeekPoint].mp3FramesToDiscard = DRMP3_SEEK_LEADING_MP3_FRAMES; - pSeekPoints[iSeekPoint].pcmFramesToDiscard = (drmp3_uint16)(nextTargetPCMFrame - mp3FrameInfo[DRMP3_SEEK_LEADING_MP3_FRAMES-1].pcmFrameIndex); - break; - } - - drmp3__accumulate_running_pcm_frame_count(pMP3, pcmFramesInCurrentMP3FrameIn, &runningPCMFrameCount, &runningPCMFrameCountFractionalPart); - } - } - } - - /* Finally, we need to seek back to where we were. */ - if (!drmp3_seek_to_start_of_stream(pMP3)) { - return DRMP3_FALSE; - } - if (!drmp3_seek_to_pcm_frame(pMP3, currentPCMFrame)) { - return DRMP3_FALSE; - } - } - - *pSeekPointCount = seekPointCount; - return DRMP3_TRUE; -} - -drmp3_bool32 drmp3_bind_seek_table(drmp3* pMP3, drmp3_uint32 seekPointCount, drmp3_seek_point* pSeekPoints) -{ - if (pMP3 == NULL) { - return DRMP3_FALSE; - } - - if (seekPointCount == 0 || pSeekPoints == NULL) { - /* Unbinding. */ - pMP3->seekPointCount = 0; - pMP3->pSeekPoints = NULL; - } else { - /* Binding. */ - pMP3->seekPointCount = seekPointCount; - pMP3->pSeekPoints = pSeekPoints; - } - - return DRMP3_TRUE; -} - - -float* drmp3__full_read_and_close_f32(drmp3* pMP3, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) -{ - drmp3_uint64 totalFramesRead = 0; - drmp3_uint64 framesCapacity = 0; - float* pFrames = NULL; - float temp[4096]; - - drmp3_assert(pMP3 != NULL); - - for (;;) { - drmp3_uint64 framesToReadRightNow = drmp3_countof(temp) / pMP3->channels; - drmp3_uint64 framesJustRead = drmp3_read_pcm_frames_f32(pMP3, framesToReadRightNow, temp); - if (framesJustRead == 0) { - break; - } - - /* Reallocate the output buffer if there's not enough room. */ - if (framesCapacity < totalFramesRead + framesJustRead) { - drmp3_uint64 newFramesBufferSize; - float* pNewFrames; - - framesCapacity *= 2; - if (framesCapacity < totalFramesRead + framesJustRead) { - framesCapacity = totalFramesRead + framesJustRead; - } - - newFramesBufferSize = framesCapacity*pMP3->channels*sizeof(float); - if (newFramesBufferSize > DRMP3_SIZE_MAX) { - break; - } - - pNewFrames = (float*)drmp3_realloc(pFrames, (size_t)newFramesBufferSize); - if (pNewFrames == NULL) { - drmp3_free(pFrames); - break; - } - - pFrames = pNewFrames; - } - - drmp3_copy_memory(pFrames + totalFramesRead*pMP3->channels, temp, (size_t)(framesJustRead*pMP3->channels*sizeof(float))); - totalFramesRead += framesJustRead; - - /* If the number of frames we asked for is less that what we actually read it means we've reached the end. */ - if (framesJustRead != framesToReadRightNow) { - break; - } - } - - if (pConfig != NULL) { - pConfig->outputChannels = pMP3->channels; - pConfig->outputSampleRate = pMP3->sampleRate; - } - - drmp3_uninit(pMP3); - - if (pTotalFrameCount) { - *pTotalFrameCount = totalFramesRead; - } - - return pFrames; -} - -drmp3_int16* drmp3__full_read_and_close_s16(drmp3* pMP3, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) -{ - drmp3_uint64 totalFramesRead = 0; - drmp3_uint64 framesCapacity = 0; - drmp3_int16* pFrames = NULL; - drmp3_int16 temp[4096]; - - drmp3_assert(pMP3 != NULL); - - for (;;) { - drmp3_uint64 framesToReadRightNow = drmp3_countof(temp) / pMP3->channels; - drmp3_uint64 framesJustRead = drmp3_read_pcm_frames_s16(pMP3, framesToReadRightNow, temp); - if (framesJustRead == 0) { - break; - } - - /* Reallocate the output buffer if there's not enough room. */ - if (framesCapacity < totalFramesRead + framesJustRead) { - drmp3_uint64 newFramesBufferSize; - drmp3_int16* pNewFrames; - - framesCapacity *= 2; - if (framesCapacity < totalFramesRead + framesJustRead) { - framesCapacity = totalFramesRead + framesJustRead; - } - - newFramesBufferSize = framesCapacity*pMP3->channels*sizeof(drmp3_int16); - if (newFramesBufferSize > DRMP3_SIZE_MAX) { - break; - } - - pNewFrames = (drmp3_int16*)drmp3_realloc(pFrames, (size_t)newFramesBufferSize); - if (pNewFrames == NULL) { - drmp3_free(pFrames); - break; - } - - pFrames = pNewFrames; - } - - drmp3_copy_memory(pFrames + totalFramesRead*pMP3->channels, temp, (size_t)(framesJustRead*pMP3->channels*sizeof(drmp3_int16))); - totalFramesRead += framesJustRead; - - /* If the number of frames we asked for is less that what we actually read it means we've reached the end. */ - if (framesJustRead != framesToReadRightNow) { - break; - } - } - - if (pConfig != NULL) { - pConfig->outputChannels = pMP3->channels; - pConfig->outputSampleRate = pMP3->sampleRate; - } - - drmp3_uninit(pMP3); - - if (pTotalFrameCount) { - *pTotalFrameCount = totalFramesRead; - } - - return pFrames; -} - - -float* drmp3_open_and_read_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) -{ - drmp3 mp3; - if (!drmp3_init(&mp3, onRead, onSeek, pUserData, pConfig)) { - return NULL; - } - - return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount); -} - -drmp3_int16* drmp3_open_and_read_s16(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) -{ - drmp3 mp3; - if (!drmp3_init(&mp3, onRead, onSeek, pUserData, pConfig)) { - return NULL; - } - - return drmp3__full_read_and_close_s16(&mp3, pConfig, pTotalFrameCount); -} - - -float* drmp3_open_memory_and_read_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) -{ - drmp3 mp3; - if (!drmp3_init_memory(&mp3, pData, dataSize, pConfig)) { - return NULL; - } - - return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount); -} - -drmp3_int16* drmp3_open_memory_and_read_s16(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) -{ - drmp3 mp3; - if (!drmp3_init_memory(&mp3, pData, dataSize, pConfig)) { - return NULL; - } - - return drmp3__full_read_and_close_s16(&mp3, pConfig, pTotalFrameCount); -} - - -#ifndef DR_MP3_NO_STDIO -float* drmp3_open_file_and_read_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) -{ - drmp3 mp3; - if (!drmp3_init_file(&mp3, filePath, pConfig)) { - return NULL; - } - - return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount); -} - -drmp3_int16* drmp3_open_file_and_read_s16(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) -{ - drmp3 mp3; - if (!drmp3_init_file(&mp3, filePath, pConfig)) { - return NULL; - } - - return drmp3__full_read_and_close_s16(&mp3, pConfig, pTotalFrameCount); -} -#endif - -void drmp3_free(void* p) -{ - DRMP3_FREE(p); -} - -#endif /*DR_MP3_IMPLEMENTATION*/ - -/* -DIFFERENCES BETWEEN minimp3 AND dr_mp3 -====================================== -- First, keep in mind that minimp3 (https://github.com/lieff/minimp3) is where all the real work was done. All of the - code relating to the actual decoding remains mostly unmodified, apart from some namespacing changes. -- dr_mp3 adds a pulling style API which allows you to deliver raw data via callbacks. So, rather than pushing data - to the decoder, the decoder _pulls_ data from your callbacks. -- In addition to callbacks, a decoder can be initialized from a block of memory and a file. -- The dr_mp3 pull API reads PCM frames rather than whole MP3 frames. -- dr_mp3 adds convenience APIs for opening and decoding entire files in one go. -- dr_mp3 is fully namespaced, including the implementation section, which is more suitable when compiling projects - as a single translation unit (aka unity builds). At the time of writing this, a unity build is not possible when - using minimp3 in conjunction with stb_vorbis. dr_mp3 addresses this. -*/ - -/* -REVISION HISTORY -================ -v0.4.7 - 2019-07-28 - - Fix a compiler error. - -v0.4.6 - 2019-06-14 - - Fix a compiler error. - -v0.4.5 - 2019-06-06 - - Bring up to date with minimp3. - -v0.4.4 - 2019-05-06 - - Fixes to the VC6 build. - -v0.4.3 - 2019-05-05 - - Use the channel count and/or sample rate of the first MP3 frame instead of DR_MP3_DEFAULT_CHANNELS and - DR_MP3_DEFAULT_SAMPLE_RATE when they are set to 0. To use the old behaviour, just set the relevant property to - DR_MP3_DEFAULT_CHANNELS or DR_MP3_DEFAULT_SAMPLE_RATE. - - Add s16 reading APIs - - drmp3_read_pcm_frames_s16 - - drmp3_open_memory_and_read_s16 - - drmp3_open_and_read_s16 - - drmp3_open_file_and_read_s16 - - Add drmp3_get_mp3_and_pcm_frame_count() to the public header section. - - Add support for C89. - - Change license to choice of public domain or MIT-0. - -v0.4.2 - 2019-02-21 - - Fix a warning. - -v0.4.1 - 2018-12-30 - - Fix a warning. - -v0.4.0 - 2018-12-16 - - API CHANGE: Rename some APIs: - - drmp3_read_f32 -> to drmp3_read_pcm_frames_f32 - - drmp3_seek_to_frame -> drmp3_seek_to_pcm_frame - - drmp3_open_and_decode_f32 -> drmp3_open_and_read_f32 - - drmp3_open_and_decode_memory_f32 -> drmp3_open_memory_and_read_f32 - - drmp3_open_and_decode_file_f32 -> drmp3_open_file_and_read_f32 - - Add drmp3_get_pcm_frame_count(). - - Add drmp3_get_mp3_frame_count(). - - Improve seeking performance. - -v0.3.2 - 2018-09-11 - - Fix a couple of memory leaks. - - Bring up to date with minimp3. - -v0.3.1 - 2018-08-25 - - Fix C++ build. - -v0.3.0 - 2018-08-25 - - Bring up to date with minimp3. This has a minor API change: the "pcm" parameter of drmp3dec_decode_frame() has - been changed from short* to void* because it can now output both s16 and f32 samples, depending on whether or - not the DR_MP3_FLOAT_OUTPUT option is set. - -v0.2.11 - 2018-08-08 - - Fix a bug where the last part of a file is not read. - -v0.2.10 - 2018-08-07 - - Improve 64-bit detection. - -v0.2.9 - 2018-08-05 - - Fix C++ build on older versions of GCC. - - Bring up to date with minimp3. - -v0.2.8 - 2018-08-02 - - Fix compilation errors with older versions of GCC. - -v0.2.7 - 2018-07-13 - - Bring up to date with minimp3. - -v0.2.6 - 2018-07-12 - - Bring up to date with minimp3. - -v0.2.5 - 2018-06-22 - - Bring up to date with minimp3. - -v0.2.4 - 2018-05-12 - - Bring up to date with minimp3. - -v0.2.3 - 2018-04-29 - - Fix TCC build. - -v0.2.2 - 2018-04-28 - - Fix bug when opening a decoder from memory. - -v0.2.1 - 2018-04-27 - - Efficiency improvements when the decoder reaches the end of the stream. - -v0.2 - 2018-04-21 - - Bring up to date with minimp3. - - Start using major.minor.revision versioning. - -v0.1d - 2018-03-30 - - Bring up to date with minimp3. - -v0.1c - 2018-03-11 - - Fix C++ build error. - -v0.1b - 2018-03-07 - - Bring up to date with minimp3. - -v0.1a - 2018-02-28 - - Fix compilation error on GCC/Clang. - - Fix some warnings. - -v0.1 - 2018-02-xx - - Initial versioned release. -*/ - -/* -This software is available as a choice of the following licenses. Choose -whichever you prefer. - -=============================================================================== -ALTERNATIVE 1 - 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, -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 -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 -WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - -For more information, please refer to <http://unlicense.org/> - -=============================================================================== -ALTERNATIVE 2 - MIT No Attribution -=============================================================================== -Copyright 2018 David Reid - -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. - -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. -*/ - -/* - https://github.com/lieff/minimp3 - To the extent possible under law, the author(s) have dedicated all copyright and related and neighboring rights to this software to the public domain worldwide. - This software is distributed without any warranty. - See <http://creativecommons.org/publicdomain/zero/1.0/>. -*/ |