// MP3 audio decoder. Public domain. See "unlicense" statement at the end of this file. // dr_mp3 - v0.4.0 - 2018-xx-xx // // 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_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 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_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_decode_f32(), drmp3_open_and_decode_memory_f32() and // drmp3_open_and_decode_file_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 #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 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) // 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) // =================== 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 { float alpha; drmp3_bool32 isPrevFramesLoaded : 1; drmp3_bool32 isNextFramesLoaded : 1; } linear; } algo; }; typedef enum { drmp3_seek_origin_start, drmp3_seek_origin_current } drmp3_seek_origin; // 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_src src; 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); // 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); // 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); float* drmp3_open_memory_and_read_f32(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); #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 #include #include #include // 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) && (defined(_M_IX86) || defined(_M_X64))) || ((defined(__i386__) || defined(__x86_64__)) && defined(__SSE2__)) #if defined(_MSC_VER) #include #endif #include #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 #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 sr_idx = DRMP3_HDR_GET_MY_SAMPLE_RATE(hdr); sr_idx -= (sr_idx != 0); int gr_count = DRMP3_HDR_IS_MONO(hdr) ? 1 : 2; 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, -253,-318,-351,-367,785,785,785,785,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,819,818,547,547,275,275,275,275,561,560,515,546,289,274,288,258, -254,-287,1329,1299,1314,1312,1057,1057,1042,1042,1026,1026,784,784,784,784,529,529,529,529,529,529,529,529,769,769,769,769,768,768,768,768,563,560,306,306,291,259, -252,-413,-477,-542,1298,-575,1041,1041,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,-383,-399,1107,1092,1106,1061,849,849,789,789,1104,1091,773,773,1076,1075,341,340,325,309,834,804,577,577,532,532,516,516,832,818,803,816,561,561,531,531,515,546,289,289,288,258, -252,-429,-493,-559,1057,1057,1042,1042,529,529,529,529,529,529,529,529,784,784,784,784,769,769,769,769,512,512,512,512,512,512,512,512,-382,1077,-415,1106,1061,1104,849,849,789,789,1091,1076,1029,1075,834,834,597,581,340,340,339,324,804,833,532,532,832,772,818,803,817,787,816,771,290,290,290,290,288,258, -253,-349,-414,-447,-463,1329,1299,-479,1314,1312,1057,1057,1042,1042,1026,1026,785,785,785,785,784,784,784,784,769,769,769,769,768,768,768,768,-319,851,821,-335,836,850,805,849,341,340,325,336,533,533,579,579,564,564,773,832,578,548,563,516,321,276,306,291,304,259, -251,-572,-733,-830,-863,-879,1041,1041,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,-511,-527,-543,1396,1351,1381,1366,1395,1335,1380,-559,1334,1138,1138,1063,1063,1350,1392,1031,1031,1062,1062,1364,1363,1120,1120,1333,1348,881,881,881,881,375,374,359,373,343,358,341,325,791,791,1123,1122,-703,1105,1045,-719,865,865,790,790,774,774,1104,1029,338,293,323,308,-799,-815,833,788,772,818,803,816,322,292,307,320,561,531,515,546,289,274,288,258, -251,-525,-605,-685,-765,-831,-846,1298,1057,1057,1312,1282,785,785,785,785,784,784,784,784,769,769,769,769,512,512,512,512,512,512,512,512,1399,1398,1383,1367,1382,1396,1351,-511,1381,1366,1139,1139,1079,1079,1124,1124,1364,1349,1363,1333,882,882,882,882,807,807,807,807,1094,1094,1136,1136,373,341,535,535,881,775,867,822,774,-591,324,338,-671,849,550,550,866,864,609,609,293,336,534,534,789,835,773,-751,834,804,308,307,833,788,832,772,562,562,547,547,305,275,560,515,290,290, -252,-397,-477,-557,-622,-653,-719,-735,-750,1329,1299,1314,1057,1057,1042,1042,1312,1282,1024,1024,785,785,785,785,784,784,784,784,769,769,769,769,-383,1127,1141,1111,1126,1140,1095,1110,869,869,883,883,1079,1109,882,882,375,374,807,868,838,881,791,-463,867,822,368,263,852,837,836,-543,610,610,550,550,352,336,534,534,865,774,851,821,850,805,593,533,579,564,773,832,578,578,548,548,577,577,307,276,306,291,516,560,259,259, -250,-2107,-2507,-2764,-2909,-2974,-3007,-3023,1041,1041,1040,1040,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,-767,-1052,-1213,-1277,-1358,-1405,-1469,-1535,-1550,-1582,-1614,-1647,-1662,-1694,-1726,-1759,-1774,-1807,-1822,-1854,-1886,1565,-1919,-1935,-1951,-1967,1731,1730,1580,1717,-1983,1729,1564,-1999,1548,-2015,-2031,1715,1595,-2047,1714,-2063,1610,-2079,1609,-2095,1323,1323,1457,1457,1307,1307,1712,1547,1641,1700,1699,1594,1685,1625,1442,1442,1322,1322,-780,-973,-910,1279,1278,1277,1262,1276,1261,1275,1215,1260,1229,-959,974,974,989,989,-943,735,478,478,495,463,506,414,-1039,1003,958,1017,927,942,987,957,431,476,1272,1167,1228,-1183,1256,-1199,895,895,941,941,1242,1227,1212,1135,1014,1014,490,489,503,487,910,1013,985,925,863,894,970,955,1012,847,-1343,831,755,755,984,909,428,366,754,559,-1391,752,486,457,924,997,698,698,983,893,740,740,908,877,739,739,667,667,953,938,497,287,271,271,683,606,590,712,726,574,302,302,738,736,481,286,526,725,605,711,636,724,696,651,589,681,666,710,364,467,573,695,466,466,301,465,379,379,709,604,665,679,316,316,634,633,436,436,464,269,424,394,452,332,438,363,347,408,393,448,331,422,362,407,392,421,346,406,391,376,375,359,1441,1306,-2367,1290,-2383,1337,-2399,-2415,1426,1321,-2431,1411,1336,-2447,-2463,-2479,1169,1169,1049,1049,1424,1289,1412,1352,1319,-2495,1154,1154,1064,1064,1153,1153,416,390,360,404,403,389,344,374,373,343,358,372,327,357,342,311,356,326,1395,1394,1137,1137,1047,1047,1365,1392,1287,1379,1334,1364,1349,1378,1318,1363,792,792,792,792,1152,1152,1032,1032,1121,1121,1046,1046,1120,1120,1030,1030,-2895,1106,1061,1104,849,849,789,789,1091,1076,1029,1090,1060,1075,833,833,309,324,532,532,832,772,818,803,561,561,531,560,515,546,289,274,288,258, -250,-1179,-1579,-1836,-1996,-2124,-2253,-2333,-2413,-2477,-2542,-2574,-2607,-2622,-2655,1314,1313,1298,1312,1282,785,785,785,785,1040,1040,1025,1025,768,768,768,768,-766,-798,-830,-862,-895,-911,-927,-943,-959,-975,-991,-1007,-1023,-1039,-1055,-1070,1724,1647,-1103,-1119,1631,1767,1662,1738,1708,1723,-1135,1780,1615,1779,1599,1677,1646,1778,1583,-1151,1777,1567,1737,1692,1765,1722,1707,1630,1751,1661,1764,1614,1736,1676,1763,1750,1645,1598,1721,1691,1762,1706,1582,1761,1566,-1167,1749,1629,767,766,751,765,494,494,735,764,719,749,734,763,447,447,748,718,477,506,431,491,446,476,461,505,415,430,475,445,504,399,460,489,414,503,383,474,429,459,502,502,746,752,488,398,501,473,413,472,486,271,480,270,-1439,-1455,1357,-1471,-1487,-1503,1341,1325,-1519,1489,1463,1403,1309,-1535,1372,1448,1418,1476,1356,1462,1387,-1551,1475,1340,1447,1402,1386,-1567,1068,1068,1474,1461,455,380,468,440,395,425,410,454,364,467,466,464,453,269,409,448,268,432,1371,1473,1432,1417,1308,1460,1355,1446,1459,1431,1083,1083,1401,1416,1458,1445,1067,1067,1370,1457,1051,1051,1291,1430,1385,1444,1354,1415,1400,1443,1082,1082,1173,1113,1186,1066,1185,1050,-1967,1158,1128,1172,1097,1171,1081,-1983,1157,1112,416,266,375,400,1170,1142,1127,1065,793,793,1169,1033,1156,1096,1141,1111,1155,1080,1126,1140,898,898,808,808,897,897,792,792,1095,1152,1032,1125,1110,1139,1079,1124,882,807,838,881,853,791,-2319,867,368,263,822,852,837,866,806,865,-2399,851,352,262,534,534,821,836,594,594,549,549,593,593,533,533,848,773,579,579,564,578,548,563,276,276,577,576,306,291,516,560,305,305,275,259, -251,-892,-2058,-2620,-2828,-2957,-3023,-3039,1041,1041,1040,1040,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,-511,-527,-543,-559,1530,-575,-591,1528,1527,1407,1526,1391,1023,1023,1023,1023,1525,1375,1268,1268,1103,1103,1087,1087,1039,1039,1523,-604,815,815,815,815,510,495,509,479,508,463,507,447,431,505,415,399,-734,-782,1262,-815,1259,1244,-831,1258,1228,-847,-863,1196,-879,1253,987,987,748,-767,493,493,462,477,414,414,686,669,478,446,461,445,474,429,487,458,412,471,1266,1264,1009,1009,799,799,-1019,-1276,-1452,-1581,-1677,-1757,-1821,-1886,-1933,-1997,1257,1257,1483,1468,1512,1422,1497,1406,1467,1496,1421,1510,1134,1134,1225,1225,1466,1451,1374,1405,1252,1252,1358,1480,1164,1164,1251,1251,1238,1238,1389,1465,-1407,1054,1101,-1423,1207,-1439,830,830,1248,1038,1237,1117,1223,1148,1236,1208,411,426,395,410,379,269,1193,1222,1132,1235,1221,1116,976,976,1192,1162,1177,1220,1131,1191,963,963,-1647,961,780,-1663,558,558,994,993,437,408,393,407,829,978,813,797,947,-1743,721,721,377,392,844,950,828,890,706,706,812,859,796,960,948,843,934,874,571,571,-1919,690,555,689,421,346,539,539,944,779,918,873,932,842,903,888,570,570,931,917,674,674,-2575,1562,-2591,1609,-2607,1654,1322,1322,1441,1441,1696,1546,1683,1593,1669,1624,1426,1426,1321,1321,1639,1680,1425,1425,1305,1305,1545,1668,1608,1623,1667,1592,1638,1666,1320,1320,1652,1607,1409,1409,1304,1304,1288,1288,1664,1637,1395,1395,1335,1335,1622,1636,1394,1394,1319,1319,1606,1621,1392,1392,1137,1137,1137,1137,345,390,360,375,404,373,1047,-2751,-2767,-2783,1062,1121,1046,-2799,1077,-2815,1106,1061,789,789,1105,1104,263,355,310,340,325,354,352,262,339,324,1091,1076,1029,1090,1060,1075,833,833,788,788,1088,1028,818,818,803,803,561,561,531,531,816,771,546,546,289,274,288,258, 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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]; 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 && linbits) { lsb += DRMP3_PEEK_BITS(linbits); DRMP3_FLUSH_BITS(linbits); DRMP3_CHECK_BITS; *dst = one*drmp3_L3_pow_43(lsb)*((int32_t)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); } 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) { if (sample >= 32766.5) return (drmp3_int16) 32767; if (sample <= -32767.5) return (drmp3_int16)-32768; drmp3_int16 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); if (!pcm) { return drmp3_hdr_frame_samples(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) { 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]; 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) { static const drmp3_f4 g_scale = { 32768.0f, 32768.0f, 32768.0f, 32768.0f }; drmp3_f4 a = DRMP3_VMUL(DRMP3_VLD(&in[i ]), g_scale); drmp3_f4 b = DRMP3_VMUL(DRMP3_VLD(&in[i+4]), g_scale); #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))); 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 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 // Standard library stuff. #ifndef DRMP3_ASSERT #include #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 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) { for (drmp3_uint32 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_assert(pCache != NULL); drmp3_assert(pCache->pSRC != NULL); drmp3_assert(pCache->pSRC->onRead != NULL); drmp3_assert(frameCount > 0); drmp3_assert(pFramesOut != NULL); drmp3_uint32 channels = pCache->pSRC->config.channels; drmp3_uint64 totalFramesRead = 0; while (frameCount > 0) { // If there's anything in memory go ahead and copy that over first. 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; drmp3_uint32 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) { if (pSRC == NULL || frameCount == 0 || pFramesOut == NULL) return 0; drmp3_src_algorithm 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) { 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; } float factor = (float)pSRC->config.sampleRateIn / pSRC->config.sampleRateOut; drmp3_uint64 totalFramesRead = 0; while (frameCount > 0) { // 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, 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. drmp3_uint32 framesToReadFromClient = (drmp3_uint32)pSRC->algo.linear.alpha; pSRC->algo.linear.alpha = pSRC->algo.linear.alpha - framesToReadFromClient; for (drmp3_uint32 i = 0; i < framesToReadFromClient; ++i) { for (drmp3_uint32 j = 0; j < pSRC->config.channels; ++j) { pPrevFrame[j] = pNextFrame[j]; } drmp3_uint64 framesRead = drmp3_src_cache_read_frames(&pSRC->cache, 1, pNextFrame); if (framesRead == 0) { for (drmp3_uint32 j = 0; j < pSRC->config.channels; ++j) { pNextFrame[j] = 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) { return pMP3->onRead(pMP3->pUserData, pBufferOut, bytesToRead); } static drmp3_bool32 drmp3__on_seek(drmp3* pMP3, int offset, drmp3_seek_origin origin) { drmp3_assert(offset >= 0); return pMP3->onSeek(pMP3->pUserData, offset, origin); } static drmp3_uint32 drmp3_decode_next_frame_ex(drmp3* pMP3, drmp3d_sample_t* pPCMFrames) { drmp3_assert(pMP3 != NULL); drmp3_assert(pMP3->onRead != NULL); if (pMP3->atEnd) { return 0; } drmp3_uint32 pcmFramesRead = 0; do { // 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) { if (pMP3->dataCapacity < DRMP3_DATA_CHUNK_SIZE) { pMP3->dataCapacity = DRMP3_DATA_CHUNK_SIZE; drmp3_uint8* pNewData = (drmp3_uint8*)drmp3_realloc(pMP3->pData, pMP3->dataCapacity); if (pNewData == NULL) { return 0; // Out of memory. } pMP3->pData = pNewData; } size_t 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. } drmp3dec_frame_info info; pcmFramesRead = drmp3dec_decode_frame(&pMP3->decoder, pMP3->pData, (int)pMP3->dataSize, pPCMFrames, &info); // <-- Safe size_t -> int conversion thanks to the check above. if (pcmFramesRead != 0) { size_t leftoverDataSize = (pMP3->dataSize - (size_t)info.frame_bytes); for (size_t i = 0; i < leftoverDataSize; ++i) { pMP3->pData[i] = pMP3->pData[i + (size_t)info.frame_bytes]; } pMP3->dataSize = leftoverDataSize; pMP3->pcmFramesConsumedInMP3Frame = 0; pMP3->pcmFramesRemainingInMP3Frame = pcmFramesRead; pMP3->mp3FrameChannels = info.channels; pMP3->mp3FrameSampleRate = info.hz; drmp3_src_set_input_sample_rate(&pMP3->src, pMP3->mp3FrameSampleRate); break; } else { // Need more data. minimp3 recommends doing data submission in 16K chunks. if (pMP3->dataCapacity == pMP3->dataSize) { // No room. Expand. pMP3->dataCapacity += DRMP3_DATA_CHUNK_SIZE; drmp3_uint8* pNewData = (drmp3_uint8*)drmp3_realloc(pMP3->pData, pMP3->dataCapacity); if (pNewData == NULL) { return 0; // Out of memory. } pMP3->pData = pNewData; } // Fill in a chunk. size_t 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); } static drmp3_uint32 drmp3_seek_next_frame(drmp3* pMP3) { drmp3_assert(pMP3 != NULL); drmp3_uint32 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; } static drmp3_uint64 drmp3_read_src(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, void* pUserData) { drmp3* pMP3 = (drmp3*)pUserData; drmp3_assert(pMP3 != NULL); drmp3_assert(pMP3->onRead != NULL); float* pFramesOutF = (float*)pFramesOut; drmp3_uint64 totalFramesRead = 0; 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; } drmp3_bool32 drmp3_init_internal(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, const drmp3_config* pConfig) { 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. drmp3_config config; if (pConfig != NULL) { config = *pConfig; } else { drmp3_zero_object(&config); } pMP3->channels = config.outputChannels; if (pMP3->channels == 0) { pMP3->channels = DR_MP3_DEFAULT_CHANNELS; } // Cannot have more than 2 channels. if (pMP3->channels > 2) { pMP3->channels = 2; } pMP3->sampleRate = config.outputSampleRate; if (pMP3->sampleRate == 0) { pMP3->sampleRate = DR_MP3_DEFAULT_SAMPLE_RATE; } 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. 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; } // 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; drmp3_assert(pMP3 != NULL); drmp3_assert(pMP3->memory.dataSize >= pMP3->memory.currentReadPos); size_t 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 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) { if (pMP3 == NULL || pMP3->onRead == NULL) { return 0; } drmp3_uint64 totalFramesRead = 0; if (pBufferOut == NULL) { float temp[4096]; while (framesToRead > 0) { drmp3_uint64 framesToReadRightNow = sizeof(temp)/sizeof(temp[0]) / pMP3->channels; if (framesToReadRightNow > framesToRead) { framesToReadRightNow = framesToRead; } drmp3_uint64 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_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. pMP3->pcmFramesConsumedInMP3Frame = 0; pMP3->pcmFramesRemainingInMP3Frame = 0; pMP3->currentPCMFrame = 0; pMP3->dataSize = 0; pMP3->atEnd = DRMP3_FALSE; 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. drmp3_uint64 framesToReadAndDiscard; if (frameIndex >= pMP3->currentPCMFrame) { // Moving foward. framesToReadAndDiscard = frameIndex - pMP3->currentPCMFrame; } else { // Moving backward. Move to the start of the stream and then move forward. framesToReadAndDiscard = frameIndex; if (!drmp3_seek_to_start_of_stream(pMP3)) { return DRMP3_FALSE; } } // 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_MAX_PCM_FRAMES_PER_MP3_FRAME * 3; // First get rid of anything that's still sitting in the buffer. if (framesToReadAndDiscard > maxFramesToReadAndDiscard && framesToReadAndDiscard > pMP3->pcmFramesRemainingInMP3Frame) { framesToReadAndDiscard -= pMP3->pcmFramesRemainingInMP3Frame; pMP3->currentPCMFrame += pMP3->pcmFramesRemainingInMP3Frame; pMP3->pcmFramesConsumedInMP3Frame += pMP3->pcmFramesRemainingInMP3Frame; pMP3->pcmFramesRemainingInMP3Frame = 0; } // Now get rid of leading whole frames. while (framesToReadAndDiscard > maxFramesToReadAndDiscard) { drmp3_uint32 pcmFramesSeeked = drmp3_seek_next_frame(pMP3); if (pcmFramesSeeked == 0) { break; } framesToReadAndDiscard -= pcmFramesSeeked; } // The last step is to read-and-discard any remaining PCM frames to make it sample-exact. drmp3_uint64 framesRead = drmp3_read_pcm_frames_f32(pMP3, framesToReadAndDiscard, NULL); if (framesRead != framesToReadAndDiscard) { return DRMP3_FALSE; } return DRMP3_TRUE; } drmp3_bool32 drmp3_seek_to_pcm_frame(drmp3* pMP3, drmp3_uint64 frameIndex) { if (pMP3 == NULL || pMP3->onSeek == NULL) { return DRMP3_FALSE; } // We currently only support brute force seeking. return drmp3_seek_to_pcm_frame__brute_force(pMP3, frameIndex); } drmp3_uint64 drmp3_get_pcm_frame_count(drmp3* pMP3) { if (pMP3 == NULL) { return 0; } // 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 0; } // 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. drmp3_uint64 currentPCMFrame = pMP3->currentPCMFrame; if (!drmp3_seek_to_start_of_stream(pMP3)) { return 0; } drmp3_uint64 totalPCMFrameCount = 0; float totalPCMFrameCountFractionalPart = 0; // <-- With resampling there will be a fractional part to each MP3 frame that we need to accumulate. for (;;) { drmp3_uint32 pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL); // <-- Passing in NULL here will prevent decoding of the MP3 frame which should save time. if (pcmFramesInCurrentMP3FrameIn == 0) { break; } float srcRatio = (float)pMP3->mp3FrameSampleRate / (float)pMP3->sampleRate; drmp3_assert(srcRatio > 0); float pcmFramesInCurrentMP3FrameOutF = totalPCMFrameCountFractionalPart + (pcmFramesInCurrentMP3FrameIn / srcRatio); drmp3_uint32 pcmFramesInCurrentMP3FrameOut = (drmp3_uint32)pcmFramesInCurrentMP3FrameOutF; totalPCMFrameCountFractionalPart = pcmFramesInCurrentMP3FrameOutF - pcmFramesInCurrentMP3FrameOut; totalPCMFrameCount += pcmFramesInCurrentMP3FrameOut; } // Finally, we need to seek back to where we were. if (!drmp3_seek_to_start_of_stream(pMP3)) { return 0; } if (!drmp3_seek_to_pcm_frame(pMP3, currentPCMFrame)) { return 0; } return totalPCMFrameCount; } drmp3_uint64 drmp3_get_mp3_frame_count(drmp3* pMP3) { if (pMP3 == NULL) { return 0; } // This works the same way as drmp3_get_pcm_frame_count() - move to the start, count MP3 frames, move back to the previous position. // The stream must support seeking for this to work. if (pMP3->onSeek == NULL) { return 0; } // 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. drmp3_uint64 currentPCMFrame = pMP3->currentPCMFrame; if (!drmp3_seek_to_start_of_stream(pMP3)) { return 0; } drmp3_uint64 totalMP3FrameCount = 0; for (;;) { drmp3_uint32 pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL); if (pcmFramesInCurrentMP3FrameIn == 0) { break; } totalMP3FrameCount += 1; } // Finally, we need to seek back to where we were. if (!drmp3_seek_to_start_of_stream(pMP3)) { return 0; } if (!drmp3_seek_to_pcm_frame(pMP3, currentPCMFrame)) { return 0; } return totalMP3FrameCount; } float* drmp3__full_read_and_close_f32(drmp3* pMP3, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) { drmp3_assert(pMP3 != NULL); drmp3_uint64 totalFramesRead = 0; drmp3_uint64 framesCapacity = 0; float* pFrames = NULL; float temp[4096]; 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) { framesCapacity *= 2; if (framesCapacity < totalFramesRead + framesJustRead) { framesCapacity = totalFramesRead + framesJustRead; } drmp3_uint64 newFramesBufferSize = framesCapacity*pMP3->channels*sizeof(float); if (newFramesBufferSize > DRMP3_SIZE_MAX) { break; } float* 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; } 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); } 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); } #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); } #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 // ================ // // v0.4.0 - 2018-xx-xx // - 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 is free and unencumbered software released into the public domain. 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