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| author | Luca Sas <sas.luca.alex@gmail.com> | 2020-03-06 17:48:44 +0000 |
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| committer | Luca Sas <sas.luca.alex@gmail.com> | 2020-03-06 17:48:44 +0000 |
| commit | 581538a8b371c0a9003dc0f1bf081222b8c4fdd9 (patch) | |
| tree | f5759a699424211d4a66e24365a596072818ab33 /libs/raylib/src/external/dr_mp3.h | |
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Setup the project
Diffstat (limited to 'libs/raylib/src/external/dr_mp3.h')
| -rw-r--r-- | libs/raylib/src/external/dr_mp3.h | 3977 |
1 files changed, 3977 insertions, 0 deletions
diff --git a/libs/raylib/src/external/dr_mp3.h b/libs/raylib/src/external/dr_mp3.h new file mode 100644 index 0000000..0ecd0d3 --- /dev/null +++ b/libs/raylib/src/external/dr_mp3.h @@ -0,0 +1,3977 @@ +/* +MP3 audio decoder. Choice of public domain or MIT-0. See license statements at the end of this file. +dr_mp3 - v0.4.7 - 2019-07-28 + +David Reid - mackron@gmail.com + +Based off minimp3 (https://github.com/lieff/minimp3) which is where the real work was done. See the bottom of this file for +differences between minimp3 and dr_mp3. +*/ + +/* +USAGE +===== +dr_mp3 is a single-file library. To use it, do something like the following in one .c file. + #define DR_MP3_IMPLEMENTATION + #include "dr_mp3.h" + +You can then #include this file in other parts of the program as you would with any other header file. To decode audio data, +do something like the following: + + drmp3 mp3; + if (!drmp3_init_file(&mp3, "MySong.mp3", NULL)) { + // Failed to open file + } + + ... + + drmp3_uint64 framesRead = drmp3_read_pcm_frames_f32(pMP3, framesToRead, pFrames); + +The drmp3 object is transparent so you can get access to the channel count and sample rate like so: + + drmp3_uint32 channels = mp3.channels; + drmp3_uint32 sampleRate = mp3.sampleRate; + +The third parameter of drmp3_init_file() in the example above allows you to control the output channel count and sample rate. It +is a pointer to a drmp3_config object. Setting any of the variables of this object to 0 will cause dr_mp3 to use defaults. + +The example above initializes a decoder from a file, but you can also initialize it from a block of memory and read and seek +callbacks with drmp3_init_memory() and drmp3_init() respectively. + +You do not need to do any annoying memory management when reading PCM frames - this is all managed internally. You can request +any number of PCM frames in each call to drmp3_read_pcm_frames_f32() and it will return as many PCM frames as it can, up to the +requested amount. + +You can also decode an entire file in one go with drmp3_open_and_read_f32(), drmp3_open_memory_and_read_f32() and +drmp3_open_file_and_read_f32(). + + +OPTIONS +======= +#define these options before including this file. + +#define DR_MP3_NO_STDIO + Disable drmp3_init_file(), etc. + +#define DR_MP3_NO_SIMD + Disable SIMD optimizations. +*/ + +#ifndef dr_mp3_h +#define dr_mp3_h + +#ifdef __cplusplus +extern "C" { +#endif + +#include <stddef.h> + +#if defined(_MSC_VER) && _MSC_VER < 1600 +typedef signed char drmp3_int8; +typedef unsigned char drmp3_uint8; +typedef signed short drmp3_int16; +typedef unsigned short drmp3_uint16; +typedef signed int drmp3_int32; +typedef unsigned int drmp3_uint32; +typedef signed __int64 drmp3_int64; +typedef unsigned __int64 drmp3_uint64; +#else +#include <stdint.h> +typedef int8_t drmp3_int8; +typedef uint8_t drmp3_uint8; +typedef int16_t drmp3_int16; +typedef uint16_t drmp3_uint16; +typedef int32_t drmp3_int32; +typedef uint32_t drmp3_uint32; +typedef int64_t drmp3_int64; +typedef uint64_t drmp3_uint64; +#endif +typedef drmp3_uint8 drmp3_bool8; +typedef drmp3_uint32 drmp3_bool32; +#define DRMP3_TRUE 1 +#define DRMP3_FALSE 0 + +#define DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME 1152 +#define DRMP3_MAX_SAMPLES_PER_FRAME (DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME*2) + +#ifdef _MSC_VER +#define DRMP3_INLINE __forceinline +#else +#ifdef __GNUC__ +#define DRMP3_INLINE __inline__ __attribute__((always_inline)) +#else +#define DRMP3_INLINE +#endif +#endif + +/* +Low Level Push API +================== +*/ +typedef struct +{ + int frame_bytes, channels, hz, layer, bitrate_kbps; +} drmp3dec_frame_info; + +typedef struct +{ + float mdct_overlap[2][9*32], qmf_state[15*2*32]; + int reserv, free_format_bytes; + unsigned char header[4], reserv_buf[511]; +} drmp3dec; + +/* Initializes a low level decoder. */ +void drmp3dec_init(drmp3dec *dec); + +/* Reads a frame from a low level decoder. */ +int drmp3dec_decode_frame(drmp3dec *dec, const unsigned char *mp3, int mp3_bytes, void *pcm, drmp3dec_frame_info *info); + +/* Helper for converting between f32 and s16. */ +void drmp3dec_f32_to_s16(const float *in, drmp3_int16 *out, int num_samples); + + + +/* +Main API (Pull API) +=================== +*/ +#ifndef DR_MP3_DEFAULT_CHANNELS +#define DR_MP3_DEFAULT_CHANNELS 2 +#endif +#ifndef DR_MP3_DEFAULT_SAMPLE_RATE +#define DR_MP3_DEFAULT_SAMPLE_RATE 44100 +#endif + +typedef struct drmp3_src drmp3_src; +typedef drmp3_uint64 (* drmp3_src_read_proc)(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, void* pUserData); /* Returns the number of frames that were read. */ + +typedef enum +{ + drmp3_src_algorithm_none, + drmp3_src_algorithm_linear +} drmp3_src_algorithm; + +#define DRMP3_SRC_CACHE_SIZE_IN_FRAMES 512 +typedef struct +{ + drmp3_src* pSRC; + float pCachedFrames[2 * DRMP3_SRC_CACHE_SIZE_IN_FRAMES]; + drmp3_uint32 cachedFrameCount; + drmp3_uint32 iNextFrame; +} drmp3_src_cache; + +typedef struct +{ + drmp3_uint32 sampleRateIn; + drmp3_uint32 sampleRateOut; + drmp3_uint32 channels; + drmp3_src_algorithm algorithm; + drmp3_uint32 cacheSizeInFrames; /* The number of frames to read from the client at a time. */ +} drmp3_src_config; + +struct drmp3_src +{ + drmp3_src_config config; + drmp3_src_read_proc onRead; + void* pUserData; + float bin[256]; + drmp3_src_cache cache; /* <-- For simplifying and optimizing client -> memory reading. */ + union + { + struct + { + double alpha; + drmp3_bool32 isPrevFramesLoaded : 1; + drmp3_bool32 isNextFramesLoaded : 1; + } linear; + } algo; +}; + +typedef enum +{ + drmp3_seek_origin_start, + drmp3_seek_origin_current +} drmp3_seek_origin; + +typedef struct +{ + drmp3_uint64 seekPosInBytes; /* Points to the first byte of an MP3 frame. */ + drmp3_uint64 pcmFrameIndex; /* The index of the PCM frame this seek point targets. */ + drmp3_uint16 mp3FramesToDiscard; /* The number of whole MP3 frames to be discarded before pcmFramesToDiscard. */ + drmp3_uint16 pcmFramesToDiscard; /* The number of leading samples to read and discard. These are discarded after mp3FramesToDiscard. */ +} drmp3_seek_point; + +/* +Callback for when data is read. Return value is the number of bytes actually read. + +pUserData [in] The user data that was passed to drmp3_init(), drmp3_open() and family. +pBufferOut [out] The output buffer. +bytesToRead [in] The number of bytes to read. + +Returns the number of bytes actually read. + +A return value of less than bytesToRead indicates the end of the stream. Do _not_ return from this callback until +either the entire bytesToRead is filled or you have reached the end of the stream. +*/ +typedef size_t (* drmp3_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead); + +/* +Callback for when data needs to be seeked. + +pUserData [in] The user data that was passed to drmp3_init(), drmp3_open() and family. +offset [in] The number of bytes to move, relative to the origin. Will never be negative. +origin [in] The origin of the seek - the current position or the start of the stream. + +Returns whether or not the seek was successful. + +Whether or not it is relative to the beginning or current position is determined by the "origin" parameter which +will be either drmp3_seek_origin_start or drmp3_seek_origin_current. +*/ +typedef drmp3_bool32 (* drmp3_seek_proc)(void* pUserData, int offset, drmp3_seek_origin origin); + +typedef struct +{ + drmp3_uint32 outputChannels; + drmp3_uint32 outputSampleRate; +} drmp3_config; + +typedef struct +{ + drmp3dec decoder; + drmp3dec_frame_info frameInfo; + drmp3_uint32 channels; + drmp3_uint32 sampleRate; + drmp3_read_proc onRead; + drmp3_seek_proc onSeek; + void* pUserData; + drmp3_uint32 mp3FrameChannels; /* The number of channels in the currently loaded MP3 frame. Internal use only. */ + drmp3_uint32 mp3FrameSampleRate; /* The sample rate of the currently loaded MP3 frame. Internal use only. */ + drmp3_uint32 pcmFramesConsumedInMP3Frame; + drmp3_uint32 pcmFramesRemainingInMP3Frame; + drmp3_uint8 pcmFrames[sizeof(float)*DRMP3_MAX_SAMPLES_PER_FRAME]; /* <-- Multipled by sizeof(float) to ensure there's enough room for DR_MP3_FLOAT_OUTPUT. */ + drmp3_uint64 currentPCMFrame; /* The current PCM frame, globally, based on the output sample rate. Mainly used for seeking. */ + drmp3_uint64 streamCursor; /* The current byte the decoder is sitting on in the raw stream. */ + drmp3_src src; + drmp3_seek_point* pSeekPoints; /* NULL by default. Set with drmp3_bind_seek_table(). Memory is owned by the client. dr_mp3 will never attempt to free this pointer. */ + drmp3_uint32 seekPointCount; /* The number of items in pSeekPoints. When set to 0 assumes to no seek table. Defaults to zero. */ + size_t dataSize; + size_t dataCapacity; + drmp3_uint8* pData; + drmp3_bool32 atEnd : 1; + struct + { + const drmp3_uint8* pData; + size_t dataSize; + size_t currentReadPos; + } memory; /* Only used for decoders that were opened against a block of memory. */ +} drmp3; + +/* +Initializes an MP3 decoder. + +onRead [in] The function to call when data needs to be read from the client. +onSeek [in] The function to call when the read position of the client data needs to move. +pUserData [in, optional] A pointer to application defined data that will be passed to onRead and onSeek. + +Returns true if successful; false otherwise. + +Close the loader with drmp3_uninit(). + +See also: drmp3_init_file(), drmp3_init_memory(), drmp3_uninit() +*/ +drmp3_bool32 drmp3_init(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, const drmp3_config* pConfig); + +/* +Initializes an MP3 decoder from a block of memory. + +This does not create a copy of the data. It is up to the application to ensure the buffer remains valid for +the lifetime of the drmp3 object. + +The buffer should contain the contents of the entire MP3 file. +*/ +drmp3_bool32 drmp3_init_memory(drmp3* pMP3, const void* pData, size_t dataSize, const drmp3_config* pConfig); + +#ifndef DR_MP3_NO_STDIO +/* +Initializes an MP3 decoder from a file. + +This holds the internal FILE object until drmp3_uninit() is called. Keep this in mind if you're caching drmp3 +objects because the operating system may restrict the number of file handles an application can have open at +any given time. +*/ +drmp3_bool32 drmp3_init_file(drmp3* pMP3, const char* filePath, const drmp3_config* pConfig); +#endif + +/* +Uninitializes an MP3 decoder. +*/ +void drmp3_uninit(drmp3* pMP3); + +/* +Reads PCM frames as interleaved 32-bit IEEE floating point PCM. + +Note that framesToRead specifies the number of PCM frames to read, _not_ the number of MP3 frames. +*/ +drmp3_uint64 drmp3_read_pcm_frames_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut); + +/* +Reads PCM frames as interleaved signed 16-bit integer PCM. + +Note that framesToRead specifies the number of PCM frames to read, _not_ the number of MP3 frames. +*/ +drmp3_uint64 drmp3_read_pcm_frames_s16(drmp3* pMP3, drmp3_uint64 framesToRead, drmp3_int16* pBufferOut); + +/* +Seeks to a specific frame. + +Note that this is _not_ an MP3 frame, but rather a PCM frame. +*/ +drmp3_bool32 drmp3_seek_to_pcm_frame(drmp3* pMP3, drmp3_uint64 frameIndex); + +/* +Calculates the total number of PCM frames in the MP3 stream. Cannot be used for infinite streams such as internet +radio. Runs in linear time. Returns 0 on error. +*/ +drmp3_uint64 drmp3_get_pcm_frame_count(drmp3* pMP3); + +/* +Calculates the total number of MP3 frames in the MP3 stream. Cannot be used for infinite streams such as internet +radio. Runs in linear time. Returns 0 on error. +*/ +drmp3_uint64 drmp3_get_mp3_frame_count(drmp3* pMP3); + +/* +Calculates the total number of MP3 and PCM frames in the MP3 stream. Cannot be used for infinite streams such as internet +radio. Runs in linear time. Returns 0 on error. + +This is equivalent to calling drmp3_get_mp3_frame_count() and drmp3_get_pcm_frame_count() except that it's more efficient. +*/ +drmp3_bool32 drmp3_get_mp3_and_pcm_frame_count(drmp3* pMP3, drmp3_uint64* pMP3FrameCount, drmp3_uint64* pPCMFrameCount); + +/* +Calculates the seekpoints based on PCM frames. This is slow. + +pSeekpoint count is a pointer to a uint32 containing the seekpoint count. On input it contains the desired count. +On output it contains the actual count. The reason for this design is that the client may request too many +seekpoints, in which case dr_mp3 will return a corrected count. + +Note that seektable seeking is not quite sample exact when the MP3 stream contains inconsistent sample rates. +*/ +drmp3_bool32 drmp3_calculate_seek_points(drmp3* pMP3, drmp3_uint32* pSeekPointCount, drmp3_seek_point* pSeekPoints); + +/* +Binds a seek table to the decoder. + +This does _not_ make a copy of pSeekPoints - it only references it. It is up to the application to ensure this +remains valid while it is bound to the decoder. + +Use drmp3_calculate_seek_points() to calculate the seek points. +*/ +drmp3_bool32 drmp3_bind_seek_table(drmp3* pMP3, drmp3_uint32 seekPointCount, drmp3_seek_point* pSeekPoints); + + +/* +Opens an decodes an entire MP3 stream as a single operation. + +pConfig is both an input and output. On input it contains what you want. On output it contains what you got. + +Free the returned pointer with drmp3_free(). +*/ +float* drmp3_open_and_read_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount); +drmp3_int16* drmp3_open_and_read_s16(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount); + +float* drmp3_open_memory_and_read_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount); +drmp3_int16* drmp3_open_memory_and_read_s16(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount); + +#ifndef DR_MP3_NO_STDIO +float* drmp3_open_file_and_read_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount); +drmp3_int16* drmp3_open_file_and_read_s16(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount); +#endif + +/* +Frees any memory that was allocated by a public drmp3 API. +*/ +void drmp3_free(void* p); + +#ifdef __cplusplus +} +#endif +#endif /* dr_mp3_h */ + + +/************************************************************************************************************************************************************ + ************************************************************************************************************************************************************ + + IMPLEMENTATION + + ************************************************************************************************************************************************************ + ************************************************************************************************************************************************************/ +#ifdef DR_MP3_IMPLEMENTATION +#include <stdlib.h> +#include <string.h> +#include <limits.h> /* For INT_MAX */ + +/* Disable SIMD when compiling with TCC for now. */ +#if defined(__TINYC__) +#define DR_MP3_NO_SIMD +#endif + +#define DRMP3_OFFSET_PTR(p, offset) ((void*)((drmp3_uint8*)(p) + (offset))) + +#define DRMP3_MAX_FREE_FORMAT_FRAME_SIZE 2304 /* more than ISO spec's */ +#ifndef DRMP3_MAX_FRAME_SYNC_MATCHES +#define DRMP3_MAX_FRAME_SYNC_MATCHES 10 +#endif + +#define DRMP3_MAX_L3_FRAME_PAYLOAD_BYTES DRMP3_MAX_FREE_FORMAT_FRAME_SIZE /* MUST be >= 320000/8/32000*1152 = 1440 */ + +#define DRMP3_MAX_BITRESERVOIR_BYTES 511 +#define DRMP3_SHORT_BLOCK_TYPE 2 +#define DRMP3_STOP_BLOCK_TYPE 3 +#define DRMP3_MODE_MONO 3 +#define DRMP3_MODE_JOINT_STEREO 1 +#define DRMP3_HDR_SIZE 4 +#define DRMP3_HDR_IS_MONO(h) (((h[3]) & 0xC0) == 0xC0) +#define DRMP3_HDR_IS_MS_STEREO(h) (((h[3]) & 0xE0) == 0x60) +#define DRMP3_HDR_IS_FREE_FORMAT(h) (((h[2]) & 0xF0) == 0) +#define DRMP3_HDR_IS_CRC(h) (!((h[1]) & 1)) +#define DRMP3_HDR_TEST_PADDING(h) ((h[2]) & 0x2) +#define DRMP3_HDR_TEST_MPEG1(h) ((h[1]) & 0x8) +#define DRMP3_HDR_TEST_NOT_MPEG25(h) ((h[1]) & 0x10) +#define DRMP3_HDR_TEST_I_STEREO(h) ((h[3]) & 0x10) +#define DRMP3_HDR_TEST_MS_STEREO(h) ((h[3]) & 0x20) +#define DRMP3_HDR_GET_STEREO_MODE(h) (((h[3]) >> 6) & 3) +#define DRMP3_HDR_GET_STEREO_MODE_EXT(h) (((h[3]) >> 4) & 3) +#define DRMP3_HDR_GET_LAYER(h) (((h[1]) >> 1) & 3) +#define DRMP3_HDR_GET_BITRATE(h) ((h[2]) >> 4) +#define DRMP3_HDR_GET_SAMPLE_RATE(h) (((h[2]) >> 2) & 3) +#define DRMP3_HDR_GET_MY_SAMPLE_RATE(h) (DRMP3_HDR_GET_SAMPLE_RATE(h) + (((h[1] >> 3) & 1) + ((h[1] >> 4) & 1))*3) +#define DRMP3_HDR_IS_FRAME_576(h) ((h[1] & 14) == 2) +#define DRMP3_HDR_IS_LAYER_1(h) ((h[1] & 6) == 6) + +#define DRMP3_BITS_DEQUANTIZER_OUT -1 +#define DRMP3_MAX_SCF (255 + DRMP3_BITS_DEQUANTIZER_OUT*4 - 210) +#define DRMP3_MAX_SCFI ((DRMP3_MAX_SCF + 3) & ~3) + +#define DRMP3_MIN(a, b) ((a) > (b) ? (b) : (a)) +#define DRMP3_MAX(a, b) ((a) < (b) ? (b) : (a)) + +#if !defined(DR_MP3_NO_SIMD) + +#if !defined(DR_MP3_ONLY_SIMD) && (defined(_M_X64) || defined(_M_ARM64) || defined(__x86_64__) || defined(__aarch64__)) +/* x64 always have SSE2, arm64 always have neon, no need for generic code */ +#define DR_MP3_ONLY_SIMD +#endif + +#if ((defined(_MSC_VER) && _MSC_VER >= 1400) && (defined(_M_IX86) || defined(_M_X64))) || ((defined(__i386__) || defined(__x86_64__)) && defined(__SSE2__)) +#if defined(_MSC_VER) +#include <intrin.h> +#endif +#include <emmintrin.h> +#define DRMP3_HAVE_SSE 1 +#define DRMP3_HAVE_SIMD 1 +#define DRMP3_VSTORE _mm_storeu_ps +#define DRMP3_VLD _mm_loadu_ps +#define DRMP3_VSET _mm_set1_ps +#define DRMP3_VADD _mm_add_ps +#define DRMP3_VSUB _mm_sub_ps +#define DRMP3_VMUL _mm_mul_ps +#define DRMP3_VMAC(a, x, y) _mm_add_ps(a, _mm_mul_ps(x, y)) +#define DRMP3_VMSB(a, x, y) _mm_sub_ps(a, _mm_mul_ps(x, y)) +#define DRMP3_VMUL_S(x, s) _mm_mul_ps(x, _mm_set1_ps(s)) +#define DRMP3_VREV(x) _mm_shuffle_ps(x, x, _MM_SHUFFLE(0, 1, 2, 3)) +typedef __m128 drmp3_f4; +#if defined(_MSC_VER) || defined(DR_MP3_ONLY_SIMD) +#define drmp3_cpuid __cpuid +#else +static __inline__ __attribute__((always_inline)) void drmp3_cpuid(int CPUInfo[], const int InfoType) +{ +#if defined(__PIC__) + __asm__ __volatile__( +#if defined(__x86_64__) + "push %%rbx\n" + "cpuid\n" + "xchgl %%ebx, %1\n" + "pop %%rbx\n" +#else + "xchgl %%ebx, %1\n" + "cpuid\n" + "xchgl %%ebx, %1\n" +#endif + : "=a" (CPUInfo[0]), "=r" (CPUInfo[1]), "=c" (CPUInfo[2]), "=d" (CPUInfo[3]) + : "a" (InfoType)); +#else + __asm__ __volatile__( + "cpuid" + : "=a" (CPUInfo[0]), "=b" (CPUInfo[1]), "=c" (CPUInfo[2]), "=d" (CPUInfo[3]) + : "a" (InfoType)); +#endif +} +#endif +static int drmp3_have_simd() +{ +#ifdef DR_MP3_ONLY_SIMD + return 1; +#else + static int g_have_simd; + int CPUInfo[4]; +#ifdef MINIMP3_TEST + static int g_counter; + if (g_counter++ > 100) + return 0; +#endif + if (g_have_simd) + goto end; + drmp3_cpuid(CPUInfo, 0); + if (CPUInfo[0] > 0) + { + drmp3_cpuid(CPUInfo, 1); + g_have_simd = (CPUInfo[3] & (1 << 26)) + 1; /* SSE2 */ + return g_have_simd - 1; + } + +end: + return g_have_simd - 1; +#endif +} +#elif defined(__ARM_NEON) || defined(__aarch64__) +#include <arm_neon.h> +#define DRMP3_HAVE_SIMD 1 +#define DRMP3_VSTORE vst1q_f32 +#define DRMP3_VLD vld1q_f32 +#define DRMP3_VSET vmovq_n_f32 +#define DRMP3_VADD vaddq_f32 +#define DRMP3_VSUB vsubq_f32 +#define DRMP3_VMUL vmulq_f32 +#define DRMP3_VMAC(a, x, y) vmlaq_f32(a, x, y) +#define DRMP3_VMSB(a, x, y) vmlsq_f32(a, x, y) +#define DRMP3_VMUL_S(x, s) vmulq_f32(x, vmovq_n_f32(s)) +#define DRMP3_VREV(x) vcombine_f32(vget_high_f32(vrev64q_f32(x)), vget_low_f32(vrev64q_f32(x))) +typedef float32x4_t drmp3_f4; +static int drmp3_have_simd() +{ /* TODO: detect neon for !DR_MP3_ONLY_SIMD */ + return 1; +} +#else +#define DRMP3_HAVE_SIMD 0 +#ifdef DR_MP3_ONLY_SIMD +#error DR_MP3_ONLY_SIMD used, but SSE/NEON not enabled +#endif +#endif + +#else + +#define DRMP3_HAVE_SIMD 0 + +#endif + +typedef struct +{ + const drmp3_uint8 *buf; + int pos, limit; +} drmp3_bs; + +typedef struct +{ + float scf[3*64]; + drmp3_uint8 total_bands, stereo_bands, bitalloc[64], scfcod[64]; +} drmp3_L12_scale_info; + +typedef struct +{ + drmp3_uint8 tab_offset, code_tab_width, band_count; +} drmp3_L12_subband_alloc; + +typedef struct +{ + const drmp3_uint8 *sfbtab; + drmp3_uint16 part_23_length, big_values, scalefac_compress; + drmp3_uint8 global_gain, block_type, mixed_block_flag, n_long_sfb, n_short_sfb; + drmp3_uint8 table_select[3], region_count[3], subblock_gain[3]; + drmp3_uint8 preflag, scalefac_scale, count1_table, scfsi; +} drmp3_L3_gr_info; + +typedef struct +{ + drmp3_bs bs; + drmp3_uint8 maindata[DRMP3_MAX_BITRESERVOIR_BYTES + DRMP3_MAX_L3_FRAME_PAYLOAD_BYTES]; + drmp3_L3_gr_info gr_info[4]; + float grbuf[2][576], scf[40], syn[18 + 15][2*32]; + drmp3_uint8 ist_pos[2][39]; +} drmp3dec_scratch; + +static void drmp3_bs_init(drmp3_bs *bs, const drmp3_uint8 *data, int bytes) +{ + bs->buf = data; + bs->pos = 0; + bs->limit = bytes*8; +} + +static drmp3_uint32 drmp3_bs_get_bits(drmp3_bs *bs, int n) +{ + drmp3_uint32 next, cache = 0, s = bs->pos & 7; + int shl = n + s; + const drmp3_uint8 *p = bs->buf + (bs->pos >> 3); + if ((bs->pos += n) > bs->limit) + return 0; + next = *p++ & (255 >> s); + while ((shl -= 8) > 0) + { + cache |= next << shl; + next = *p++; + } + return cache | (next >> -shl); +} + +static int drmp3_hdr_valid(const drmp3_uint8 *h) +{ + return h[0] == 0xff && + ((h[1] & 0xF0) == 0xf0 || (h[1] & 0xFE) == 0xe2) && + (DRMP3_HDR_GET_LAYER(h) != 0) && + (DRMP3_HDR_GET_BITRATE(h) != 15) && + (DRMP3_HDR_GET_SAMPLE_RATE(h) != 3); +} + +static int drmp3_hdr_compare(const drmp3_uint8 *h1, const drmp3_uint8 *h2) +{ + return drmp3_hdr_valid(h2) && + ((h1[1] ^ h2[1]) & 0xFE) == 0 && + ((h1[2] ^ h2[2]) & 0x0C) == 0 && + !(DRMP3_HDR_IS_FREE_FORMAT(h1) ^ DRMP3_HDR_IS_FREE_FORMAT(h2)); +} + +static unsigned drmp3_hdr_bitrate_kbps(const drmp3_uint8 *h) +{ + static const drmp3_uint8 halfrate[2][3][15] = { + { { 0,4,8,12,16,20,24,28,32,40,48,56,64,72,80 }, { 0,4,8,12,16,20,24,28,32,40,48,56,64,72,80 }, { 0,16,24,28,32,40,48,56,64,72,80,88,96,112,128 } }, + { { 0,16,20,24,28,32,40,48,56,64,80,96,112,128,160 }, { 0,16,24,28,32,40,48,56,64,80,96,112,128,160,192 }, { 0,16,32,48,64,80,96,112,128,144,160,176,192,208,224 } }, + }; + return 2*halfrate[!!DRMP3_HDR_TEST_MPEG1(h)][DRMP3_HDR_GET_LAYER(h) - 1][DRMP3_HDR_GET_BITRATE(h)]; +} + +static unsigned drmp3_hdr_sample_rate_hz(const drmp3_uint8 *h) +{ + static const unsigned g_hz[3] = { 44100, 48000, 32000 }; + return g_hz[DRMP3_HDR_GET_SAMPLE_RATE(h)] >> (int)!DRMP3_HDR_TEST_MPEG1(h) >> (int)!DRMP3_HDR_TEST_NOT_MPEG25(h); +} + +static unsigned drmp3_hdr_frame_samples(const drmp3_uint8 *h) +{ + return DRMP3_HDR_IS_LAYER_1(h) ? 384 : (1152 >> (int)DRMP3_HDR_IS_FRAME_576(h)); +} + +static int drmp3_hdr_frame_bytes(const drmp3_uint8 *h, int free_format_size) +{ + int frame_bytes = drmp3_hdr_frame_samples(h)*drmp3_hdr_bitrate_kbps(h)*125/drmp3_hdr_sample_rate_hz(h); + if (DRMP3_HDR_IS_LAYER_1(h)) + { + frame_bytes &= ~3; /* slot align */ + } + return frame_bytes ? frame_bytes : free_format_size; +} + +static int drmp3_hdr_padding(const drmp3_uint8 *h) +{ + return DRMP3_HDR_TEST_PADDING(h) ? (DRMP3_HDR_IS_LAYER_1(h) ? 4 : 1) : 0; +} + +#ifndef DR_MP3_ONLY_MP3 +static const drmp3_L12_subband_alloc *drmp3_L12_subband_alloc_table(const drmp3_uint8 *hdr, drmp3_L12_scale_info *sci) +{ + const drmp3_L12_subband_alloc *alloc; + int mode = DRMP3_HDR_GET_STEREO_MODE(hdr); + int nbands, stereo_bands = (mode == DRMP3_MODE_MONO) ? 0 : (mode == DRMP3_MODE_JOINT_STEREO) ? (DRMP3_HDR_GET_STEREO_MODE_EXT(hdr) << 2) + 4 : 32; + + if (DRMP3_HDR_IS_LAYER_1(hdr)) + { + static const drmp3_L12_subband_alloc g_alloc_L1[] = { { 76, 4, 32 } }; + alloc = g_alloc_L1; + nbands = 32; + } else if (!DRMP3_HDR_TEST_MPEG1(hdr)) + { + static const drmp3_L12_subband_alloc g_alloc_L2M2[] = { { 60, 4, 4 }, { 44, 3, 7 }, { 44, 2, 19 } }; + alloc = g_alloc_L2M2; + nbands = 30; + } else + { + static const drmp3_L12_subband_alloc g_alloc_L2M1[] = { { 0, 4, 3 }, { 16, 4, 8 }, { 32, 3, 12 }, { 40, 2, 7 } }; + int sample_rate_idx = DRMP3_HDR_GET_SAMPLE_RATE(hdr); + unsigned kbps = drmp3_hdr_bitrate_kbps(hdr) >> (int)(mode != DRMP3_MODE_MONO); + if (!kbps) /* free-format */ + { + kbps = 192; + } + + alloc = g_alloc_L2M1; + nbands = 27; + if (kbps < 56) + { + static const drmp3_L12_subband_alloc g_alloc_L2M1_lowrate[] = { { 44, 4, 2 }, { 44, 3, 10 } }; + alloc = g_alloc_L2M1_lowrate; + nbands = sample_rate_idx == 2 ? 12 : 8; + } else if (kbps >= 96 && sample_rate_idx != 1) + { + nbands = 30; + } + } + + sci->total_bands = (drmp3_uint8)nbands; + sci->stereo_bands = (drmp3_uint8)DRMP3_MIN(stereo_bands, nbands); + + return alloc; +} + +static void drmp3_L12_read_scalefactors(drmp3_bs *bs, drmp3_uint8 *pba, drmp3_uint8 *scfcod, int bands, float *scf) +{ + static const float g_deq_L12[18*3] = { +#define DRMP3_DQ(x) 9.53674316e-07f/x, 7.56931807e-07f/x, 6.00777173e-07f/x + DRMP3_DQ(3),DRMP3_DQ(7),DRMP3_DQ(15),DRMP3_DQ(31),DRMP3_DQ(63),DRMP3_DQ(127),DRMP3_DQ(255),DRMP3_DQ(511),DRMP3_DQ(1023),DRMP3_DQ(2047),DRMP3_DQ(4095),DRMP3_DQ(8191),DRMP3_DQ(16383),DRMP3_DQ(32767),DRMP3_DQ(65535),DRMP3_DQ(3),DRMP3_DQ(5),DRMP3_DQ(9) + }; + int i, m; + for (i = 0; i < bands; i++) + { + float s = 0; + int ba = *pba++; + int mask = ba ? 4 + ((19 >> scfcod[i]) & 3) : 0; + for (m = 4; m; m >>= 1) + { + if (mask & m) + { + int b = drmp3_bs_get_bits(bs, 6); + s = g_deq_L12[ba*3 - 6 + b % 3]*(1 << 21 >> b/3); + } + *scf++ = s; + } + } +} + +static void drmp3_L12_read_scale_info(const drmp3_uint8 *hdr, drmp3_bs *bs, drmp3_L12_scale_info *sci) +{ + static const drmp3_uint8 g_bitalloc_code_tab[] = { + 0,17, 3, 4, 5,6,7, 8,9,10,11,12,13,14,15,16, + 0,17,18, 3,19,4,5, 6,7, 8, 9,10,11,12,13,16, + 0,17,18, 3,19,4,5,16, + 0,17,18,16, + 0,17,18,19, 4,5,6, 7,8, 9,10,11,12,13,14,15, + 0,17,18, 3,19,4,5, 6,7, 8, 9,10,11,12,13,14, + 0, 2, 3, 4, 5,6,7, 8,9,10,11,12,13,14,15,16 + }; + const drmp3_L12_subband_alloc *subband_alloc = drmp3_L12_subband_alloc_table(hdr, sci); + + int i, k = 0, ba_bits = 0; + const drmp3_uint8 *ba_code_tab = g_bitalloc_code_tab; + + for (i = 0; i < sci->total_bands; i++) + { + drmp3_uint8 ba; + if (i == k) + { + k += subband_alloc->band_count; + ba_bits = subband_alloc->code_tab_width; + ba_code_tab = g_bitalloc_code_tab + subband_alloc->tab_offset; + subband_alloc++; + } + ba = ba_code_tab[drmp3_bs_get_bits(bs, ba_bits)]; + sci->bitalloc[2*i] = ba; + if (i < sci->stereo_bands) + { + ba = ba_code_tab[drmp3_bs_get_bits(bs, ba_bits)]; + } + sci->bitalloc[2*i + 1] = sci->stereo_bands ? ba : 0; + } + + for (i = 0; i < 2*sci->total_bands; i++) + { + sci->scfcod[i] = (drmp3_uint8)(sci->bitalloc[i] ? DRMP3_HDR_IS_LAYER_1(hdr) ? 2 : drmp3_bs_get_bits(bs, 2) : 6); + } + + drmp3_L12_read_scalefactors(bs, sci->bitalloc, sci->scfcod, sci->total_bands*2, sci->scf); + + for (i = sci->stereo_bands; i < sci->total_bands; i++) + { + sci->bitalloc[2*i + 1] = 0; + } +} + +static int drmp3_L12_dequantize_granule(float *grbuf, drmp3_bs *bs, drmp3_L12_scale_info *sci, int group_size) +{ + int i, j, k, choff = 576; + for (j = 0; j < 4; j++) + { + float *dst = grbuf + group_size*j; + for (i = 0; i < 2*sci->total_bands; i++) + { + int ba = sci->bitalloc[i]; + if (ba != 0) + { + if (ba < 17) + { + int half = (1 << (ba - 1)) - 1; + for (k = 0; k < group_size; k++) + { + dst[k] = (float)((int)drmp3_bs_get_bits(bs, ba) - half); + } + } else + { + unsigned mod = (2 << (ba - 17)) + 1; /* 3, 5, 9 */ + unsigned code = drmp3_bs_get_bits(bs, mod + 2 - (mod >> 3)); /* 5, 7, 10 */ + for (k = 0; k < group_size; k++, code /= mod) + { + dst[k] = (float)((int)(code % mod - mod/2)); + } + } + } + dst += choff; + choff = 18 - choff; + } + } + return group_size*4; +} + +static void drmp3_L12_apply_scf_384(drmp3_L12_scale_info *sci, const float *scf, float *dst) +{ + int i, k; + memcpy(dst + 576 + sci->stereo_bands*18, dst + sci->stereo_bands*18, (sci->total_bands - sci->stereo_bands)*18*sizeof(float)); + for (i = 0; i < sci->total_bands; i++, dst += 18, scf += 6) + { + for (k = 0; k < 12; k++) + { + dst[k + 0] *= scf[0]; + dst[k + 576] *= scf[3]; + } + } +} +#endif + +static int drmp3_L3_read_side_info(drmp3_bs *bs, drmp3_L3_gr_info *gr, const drmp3_uint8 *hdr) +{ + static const drmp3_uint8 g_scf_long[8][23] = { + { 6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54,0 }, + { 12,12,12,12,12,12,16,20,24,28,32,40,48,56,64,76,90,2,2,2,2,2,0 }, + { 6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54,0 }, + { 6,6,6,6,6,6,8,10,12,14,16,18,22,26,32,38,46,54,62,70,76,36,0 }, + { 6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54,0 }, + { 4,4,4,4,4,4,6,6,8,8,10,12,16,20,24,28,34,42,50,54,76,158,0 }, + { 4,4,4,4,4,4,6,6,6,8,10,12,16,18,22,28,34,40,46,54,54,192,0 }, + { 4,4,4,4,4,4,6,6,8,10,12,16,20,24,30,38,46,56,68,84,102,26,0 } + }; + static const drmp3_uint8 g_scf_short[8][40] = { + { 4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 }, + { 8,8,8,8,8,8,8,8,8,12,12,12,16,16,16,20,20,20,24,24,24,28,28,28,36,36,36,2,2,2,2,2,2,2,2,2,26,26,26,0 }, + { 4,4,4,4,4,4,4,4,4,6,6,6,6,6,6,8,8,8,10,10,10,14,14,14,18,18,18,26,26,26,32,32,32,42,42,42,18,18,18,0 }, + { 4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,32,32,32,44,44,44,12,12,12,0 }, + { 4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 }, + { 4,4,4,4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,22,22,22,30,30,30,56,56,56,0 }, + { 4,4,4,4,4,4,4,4,4,4,4,4,6,6,6,6,6,6,10,10,10,12,12,12,14,14,14,16,16,16,20,20,20,26,26,26,66,66,66,0 }, + { 4,4,4,4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,12,12,12,16,16,16,20,20,20,26,26,26,34,34,34,42,42,42,12,12,12,0 } + }; + static const drmp3_uint8 g_scf_mixed[8][40] = { + { 6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 }, + { 12,12,12,4,4,4,8,8,8,12,12,12,16,16,16,20,20,20,24,24,24,28,28,28,36,36,36,2,2,2,2,2,2,2,2,2,26,26,26,0 }, + { 6,6,6,6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,14,14,14,18,18,18,26,26,26,32,32,32,42,42,42,18,18,18,0 }, + { 6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,32,32,32,44,44,44,12,12,12,0 }, + { 6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 }, + { 4,4,4,4,4,4,6,6,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,22,22,22,30,30,30,56,56,56,0 }, + { 4,4,4,4,4,4,6,6,4,4,4,6,6,6,6,6,6,10,10,10,12,12,12,14,14,14,16,16,16,20,20,20,26,26,26,66,66,66,0 }, + { 4,4,4,4,4,4,6,6,4,4,4,6,6,6,8,8,8,12,12,12,16,16,16,20,20,20,26,26,26,34,34,34,42,42,42,12,12,12,0 } + }; + + unsigned tables, scfsi = 0; + int main_data_begin, part_23_sum = 0; + int gr_count = DRMP3_HDR_IS_MONO(hdr) ? 1 : 2; + int sr_idx = DRMP3_HDR_GET_MY_SAMPLE_RATE(hdr); sr_idx -= (sr_idx != 0); + + if (DRMP3_HDR_TEST_MPEG1(hdr)) + { + gr_count *= 2; + main_data_begin = drmp3_bs_get_bits(bs, 9); + scfsi = drmp3_bs_get_bits(bs, 7 + gr_count); + } else + { + main_data_begin = drmp3_bs_get_bits(bs, 8 + gr_count) >> gr_count; + } + + do + { + if (DRMP3_HDR_IS_MONO(hdr)) + { + scfsi <<= 4; + } + gr->part_23_length = (drmp3_uint16)drmp3_bs_get_bits(bs, 12); + part_23_sum += gr->part_23_length; + gr->big_values = (drmp3_uint16)drmp3_bs_get_bits(bs, 9); + if (gr->big_values > 288) + { + return -1; + } + gr->global_gain = (drmp3_uint8)drmp3_bs_get_bits(bs, 8); + gr->scalefac_compress = (drmp3_uint16)drmp3_bs_get_bits(bs, DRMP3_HDR_TEST_MPEG1(hdr) ? 4 : 9); + gr->sfbtab = g_scf_long[sr_idx]; + gr->n_long_sfb = 22; + gr->n_short_sfb = 0; + if (drmp3_bs_get_bits(bs, 1)) + { + gr->block_type = (drmp3_uint8)drmp3_bs_get_bits(bs, 2); + if (!gr->block_type) + { + return -1; + } + gr->mixed_block_flag = (drmp3_uint8)drmp3_bs_get_bits(bs, 1); + gr->region_count[0] = 7; + gr->region_count[1] = 255; + if (gr->block_type == DRMP3_SHORT_BLOCK_TYPE) + { + scfsi &= 0x0F0F; + if (!gr->mixed_block_flag) + { + gr->region_count[0] = 8; + gr->sfbtab = g_scf_short[sr_idx]; + gr->n_long_sfb = 0; + gr->n_short_sfb = 39; + } else + { + gr->sfbtab = g_scf_mixed[sr_idx]; + gr->n_long_sfb = DRMP3_HDR_TEST_MPEG1(hdr) ? 8 : 6; + gr->n_short_sfb = 30; + } + } + tables = drmp3_bs_get_bits(bs, 10); + tables <<= 5; + gr->subblock_gain[0] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3); + gr->subblock_gain[1] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3); + gr->subblock_gain[2] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3); + } else + { + gr->block_type = 0; + gr->mixed_block_flag = 0; + tables = drmp3_bs_get_bits(bs, 15); + gr->region_count[0] = (drmp3_uint8)drmp3_bs_get_bits(bs, 4); + gr->region_count[1] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3); + gr->region_count[2] = 255; + } + gr->table_select[0] = (drmp3_uint8)(tables >> 10); + gr->table_select[1] = (drmp3_uint8)((tables >> 5) & 31); + gr->table_select[2] = (drmp3_uint8)((tables) & 31); + gr->preflag = (drmp3_uint8)(DRMP3_HDR_TEST_MPEG1(hdr) ? drmp3_bs_get_bits(bs, 1) : (gr->scalefac_compress >= 500)); + gr->scalefac_scale = (drmp3_uint8)drmp3_bs_get_bits(bs, 1); + gr->count1_table = (drmp3_uint8)drmp3_bs_get_bits(bs, 1); + gr->scfsi = (drmp3_uint8)((scfsi >> 12) & 15); + scfsi <<= 4; + gr++; + } while(--gr_count); + + if (part_23_sum + bs->pos > bs->limit + main_data_begin*8) + { + return -1; + } + + return main_data_begin; +} + +static void drmp3_L3_read_scalefactors(drmp3_uint8 *scf, drmp3_uint8 *ist_pos, const drmp3_uint8 *scf_size, const drmp3_uint8 *scf_count, drmp3_bs *bitbuf, int scfsi) +{ + int i, k; + for (i = 0; i < 4 && scf_count[i]; i++, scfsi *= 2) + { + int cnt = scf_count[i]; + if (scfsi & 8) + { + memcpy(scf, ist_pos, cnt); + } else + { + int bits = scf_size[i]; + if (!bits) + { + memset(scf, 0, cnt); + memset(ist_pos, 0, cnt); + } else + { + int max_scf = (scfsi < 0) ? (1 << bits) - 1 : -1; + for (k = 0; k < cnt; k++) + { + int s = drmp3_bs_get_bits(bitbuf, bits); + ist_pos[k] = (drmp3_uint8)(s == max_scf ? -1 : s); + scf[k] = (drmp3_uint8)s; + } + } + } + ist_pos += cnt; + scf += cnt; + } + scf[0] = scf[1] = scf[2] = 0; +} + +static float drmp3_L3_ldexp_q2(float y, int exp_q2) +{ + static const float g_expfrac[4] = { 9.31322575e-10f,7.83145814e-10f,6.58544508e-10f,5.53767716e-10f }; + int e; + do + { + e = DRMP3_MIN(30*4, exp_q2); + y *= g_expfrac[e & 3]*(1 << 30 >> (e >> 2)); + } while ((exp_q2 -= e) > 0); + return y; +} + +static void drmp3_L3_decode_scalefactors(const drmp3_uint8 *hdr, drmp3_uint8 *ist_pos, drmp3_bs *bs, const drmp3_L3_gr_info *gr, float *scf, int ch) +{ + static const drmp3_uint8 g_scf_partitions[3][28] = { + { 6,5,5, 5,6,5,5,5,6,5, 7,3,11,10,0,0, 7, 7, 7,0, 6, 6,6,3, 8, 8,5,0 }, + { 8,9,6,12,6,9,9,9,6,9,12,6,15,18,0,0, 6,15,12,0, 6,12,9,6, 6,18,9,0 }, + { 9,9,6,12,9,9,9,9,9,9,12,6,18,18,0,0,12,12,12,0,12, 9,9,6,15,12,9,0 } + }; + const drmp3_uint8 *scf_partition = g_scf_partitions[!!gr->n_short_sfb + !gr->n_long_sfb]; + drmp3_uint8 scf_size[4], iscf[40]; + int i, scf_shift = gr->scalefac_scale + 1, gain_exp, scfsi = gr->scfsi; + float gain; + + if (DRMP3_HDR_TEST_MPEG1(hdr)) + { + static const drmp3_uint8 g_scfc_decode[16] = { 0,1,2,3, 12,5,6,7, 9,10,11,13, 14,15,18,19 }; + int part = g_scfc_decode[gr->scalefac_compress]; + scf_size[1] = scf_size[0] = (drmp3_uint8)(part >> 2); + scf_size[3] = scf_size[2] = (drmp3_uint8)(part & 3); + } else + { + static const drmp3_uint8 g_mod[6*4] = { 5,5,4,4,5,5,4,1,4,3,1,1,5,6,6,1,4,4,4,1,4,3,1,1 }; + int k, modprod, sfc, ist = DRMP3_HDR_TEST_I_STEREO(hdr) && ch; + sfc = gr->scalefac_compress >> ist; + for (k = ist*3*4; sfc >= 0; sfc -= modprod, k += 4) + { + for (modprod = 1, i = 3; i >= 0; i--) + { + scf_size[i] = (drmp3_uint8)(sfc / modprod % g_mod[k + i]); + modprod *= g_mod[k + i]; + } + } + scf_partition += k; + scfsi = -16; + } + drmp3_L3_read_scalefactors(iscf, ist_pos, scf_size, scf_partition, bs, scfsi); + + if (gr->n_short_sfb) + { + int sh = 3 - scf_shift; + for (i = 0; i < gr->n_short_sfb; i += 3) + { + iscf[gr->n_long_sfb + i + 0] += gr->subblock_gain[0] << sh; + iscf[gr->n_long_sfb + i + 1] += gr->subblock_gain[1] << sh; + iscf[gr->n_long_sfb + i + 2] += gr->subblock_gain[2] << sh; + } + } else if (gr->preflag) + { + static const drmp3_uint8 g_preamp[10] = { 1,1,1,1,2,2,3,3,3,2 }; + for (i = 0; i < 10; i++) + { + iscf[11 + i] += g_preamp[i]; + } + } + + gain_exp = gr->global_gain + DRMP3_BITS_DEQUANTIZER_OUT*4 - 210 - (DRMP3_HDR_IS_MS_STEREO(hdr) ? 2 : 0); + gain = drmp3_L3_ldexp_q2(1 << (DRMP3_MAX_SCFI/4), DRMP3_MAX_SCFI - gain_exp); + for (i = 0; i < (int)(gr->n_long_sfb + gr->n_short_sfb); i++) + { + scf[i] = drmp3_L3_ldexp_q2(gain, iscf[i] << scf_shift); + } +} + +static const float g_drmp3_pow43[129 + 16] = { + 0,-1,-2.519842f,-4.326749f,-6.349604f,-8.549880f,-10.902724f,-13.390518f,-16.000000f,-18.720754f,-21.544347f,-24.463781f,-27.473142f,-30.567351f,-33.741992f,-36.993181f, + 0,1,2.519842f,4.326749f,6.349604f,8.549880f,10.902724f,13.390518f,16.000000f,18.720754f,21.544347f,24.463781f,27.473142f,30.567351f,33.741992f,36.993181f,40.317474f,43.711787f,47.173345f,50.699631f,54.288352f,57.937408f,61.644865f,65.408941f,69.227979f,73.100443f,77.024898f,81.000000f,85.024491f,89.097188f,93.216975f,97.382800f,101.593667f,105.848633f,110.146801f,114.487321f,118.869381f,123.292209f,127.755065f,132.257246f,136.798076f,141.376907f,145.993119f,150.646117f,155.335327f,160.060199f,164.820202f,169.614826f,174.443577f,179.305980f,184.201575f,189.129918f,194.090580f,199.083145f,204.107210f,209.162385f,214.248292f,219.364564f,224.510845f,229.686789f,234.892058f,240.126328f,245.389280f,250.680604f,256.000000f,261.347174f,266.721841f,272.123723f,277.552547f,283.008049f,288.489971f,293.998060f,299.532071f,305.091761f,310.676898f,316.287249f,321.922592f,327.582707f,333.267377f,338.976394f,344.709550f,350.466646f,356.247482f,362.051866f,367.879608f,373.730522f,379.604427f,385.501143f,391.420496f,397.362314f,403.326427f,409.312672f,415.320884f,421.350905f,427.402579f,433.475750f,439.570269f,445.685987f,451.822757f,457.980436f,464.158883f,470.357960f,476.577530f,482.817459f,489.077615f,495.357868f,501.658090f,507.978156f,514.317941f,520.677324f,527.056184f,533.454404f,539.871867f,546.308458f,552.764065f,559.238575f,565.731879f,572.243870f,578.774440f,585.323483f,591.890898f,598.476581f,605.080431f,611.702349f,618.342238f,625.000000f,631.675540f,638.368763f,645.079578f +}; + +static float drmp3_L3_pow_43(int x) +{ + float frac; + int sign, mult = 256; + + if (x < 129) + { + return g_drmp3_pow43[16 + x]; + } + + if (x < 1024) + { + mult = 16; + x <<= 3; + } + + sign = 2*x & 64; + frac = (float)((x & 63) - sign) / ((x & ~63) + sign); + return g_drmp3_pow43[16 + ((x + sign) >> 6)]*(1.f + frac*((4.f/3) + frac*(2.f/9)))*mult; +} + +static void drmp3_L3_huffman(float *dst, drmp3_bs *bs, const drmp3_L3_gr_info *gr_info, const float *scf, int layer3gr_limit) +{ + static const drmp3_int16 tabs[] = { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 785,785,785,785,784,784,784,784,513,513,513,513,513,513,513,513,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256, + -255,1313,1298,1282,785,785,785,785,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,290,288, + -255,1313,1298,1282,769,769,769,769,529,529,529,529,529,529,529,529,528,528,528,528,528,528,528,528,512,512,512,512,512,512,512,512,290,288, + -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, + 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+ 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+ 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}; + static const drmp3_uint8 tab32[] = { 130,162,193,209,44,28,76,140,9,9,9,9,9,9,9,9,190,254,222,238,126,94,157,157,109,61,173,205}; + static const drmp3_uint8 tab33[] = { 252,236,220,204,188,172,156,140,124,108,92,76,60,44,28,12 }; + static const drmp3_int16 tabindex[2*16] = { 0,32,64,98,0,132,180,218,292,364,426,538,648,746,0,1126,1460,1460,1460,1460,1460,1460,1460,1460,1842,1842,1842,1842,1842,1842,1842,1842 }; + static const drmp3_uint8 g_linbits[] = { 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,2,3,4,6,8,10,13,4,5,6,7,8,9,11,13 }; + +#define DRMP3_PEEK_BITS(n) (bs_cache >> (32 - n)) +#define DRMP3_FLUSH_BITS(n) { bs_cache <<= (n); bs_sh += (n); } +#define DRMP3_CHECK_BITS while (bs_sh >= 0) { bs_cache |= (drmp3_uint32)*bs_next_ptr++ << bs_sh; bs_sh -= 8; } +#define DRMP3_BSPOS ((bs_next_ptr - bs->buf)*8 - 24 + bs_sh) + + float one = 0.0f; + int ireg = 0, big_val_cnt = gr_info->big_values; + const drmp3_uint8 *sfb = gr_info->sfbtab; + const drmp3_uint8 *bs_next_ptr = bs->buf + bs->pos/8; + drmp3_uint32 bs_cache = (((bs_next_ptr[0]*256u + bs_next_ptr[1])*256u + bs_next_ptr[2])*256u + bs_next_ptr[3]) << (bs->pos & 7); + int pairs_to_decode, np, bs_sh = (bs->pos & 7) - 8; + bs_next_ptr += 4; + + while (big_val_cnt > 0) + { + int tab_num = gr_info->table_select[ireg]; + int sfb_cnt = gr_info->region_count[ireg++]; + const drmp3_int16 *codebook = tabs + tabindex[tab_num]; + int linbits = g_linbits[tab_num]; + if (linbits) + { + do + { + np = *sfb++ / 2; + pairs_to_decode = DRMP3_MIN(big_val_cnt, np); + one = *scf++; + do + { + int j, w = 5; + int leaf = codebook[DRMP3_PEEK_BITS(w)]; + while (leaf < 0) + { + DRMP3_FLUSH_BITS(w); + w = leaf & 7; + leaf = codebook[DRMP3_PEEK_BITS(w) - (leaf >> 3)]; + } + DRMP3_FLUSH_BITS(leaf >> 8); + + for (j = 0; j < 2; j++, dst++, leaf >>= 4) + { + int lsb = leaf & 0x0F; + if (lsb == 15) + { + lsb += DRMP3_PEEK_BITS(linbits); + DRMP3_FLUSH_BITS(linbits); + DRMP3_CHECK_BITS; + *dst = one*drmp3_L3_pow_43(lsb)*((drmp3_int32)bs_cache < 0 ? -1: 1); + } else + { + *dst = g_drmp3_pow43[16 + lsb - 16*(bs_cache >> 31)]*one; + } + DRMP3_FLUSH_BITS(lsb ? 1 : 0); + } + DRMP3_CHECK_BITS; + } while (--pairs_to_decode); + } while ((big_val_cnt -= np) > 0 && --sfb_cnt >= 0); + } else + { + do + { + np = *sfb++ / 2; + pairs_to_decode = DRMP3_MIN(big_val_cnt, np); + one = *scf++; + do + { + int j, w = 5; + int leaf = codebook[DRMP3_PEEK_BITS(w)]; + while (leaf < 0) + { + DRMP3_FLUSH_BITS(w); + w = leaf & 7; + leaf = codebook[DRMP3_PEEK_BITS(w) - (leaf >> 3)]; + } + DRMP3_FLUSH_BITS(leaf >> 8); + + for (j = 0; j < 2; j++, dst++, leaf >>= 4) + { + int lsb = leaf & 0x0F; + *dst = g_drmp3_pow43[16 + lsb - 16*(bs_cache >> 31)]*one; + DRMP3_FLUSH_BITS(lsb ? 1 : 0); + } + DRMP3_CHECK_BITS; + } while (--pairs_to_decode); + } while ((big_val_cnt -= np) > 0 && --sfb_cnt >= 0); + } + } + + for (np = 1 - big_val_cnt;; dst += 4) + { + const drmp3_uint8 *codebook_count1 = (gr_info->count1_table) ? tab33 : tab32; + int leaf = codebook_count1[DRMP3_PEEK_BITS(4)]; + if (!(leaf & 8)) + { + leaf = codebook_count1[(leaf >> 3) + (bs_cache << 4 >> (32 - (leaf & 3)))]; + } + DRMP3_FLUSH_BITS(leaf & 7); + if (DRMP3_BSPOS > layer3gr_limit) + { + break; + } +#define DRMP3_RELOAD_SCALEFACTOR if (!--np) { np = *sfb++/2; if (!np) break; one = *scf++; } +#define DRMP3_DEQ_COUNT1(s) if (leaf & (128 >> s)) { dst[s] = ((drmp3_int32)bs_cache < 0) ? -one : one; DRMP3_FLUSH_BITS(1) } + DRMP3_RELOAD_SCALEFACTOR; + DRMP3_DEQ_COUNT1(0); + DRMP3_DEQ_COUNT1(1); + DRMP3_RELOAD_SCALEFACTOR; + DRMP3_DEQ_COUNT1(2); + DRMP3_DEQ_COUNT1(3); + DRMP3_CHECK_BITS; + } + + bs->pos = layer3gr_limit; +} + +static void drmp3_L3_midside_stereo(float *left, int n) +{ + int i = 0; + float *right = left + 576; +#if DRMP3_HAVE_SIMD + if (drmp3_have_simd()) for (; i < n - 3; i += 4) + { + drmp3_f4 vl = DRMP3_VLD(left + i); + drmp3_f4 vr = DRMP3_VLD(right + i); + DRMP3_VSTORE(left + i, DRMP3_VADD(vl, vr)); + DRMP3_VSTORE(right + i, DRMP3_VSUB(vl, vr)); + } +#endif + for (; i < n; i++) + { + float a = left[i]; + float b = right[i]; + left[i] = a + b; + right[i] = a - b; + } +} + +static void drmp3_L3_intensity_stereo_band(float *left, int n, float kl, float kr) +{ + int i; + for (i = 0; i < n; i++) + { + left[i + 576] = left[i]*kr; + left[i] = left[i]*kl; + } +} + +static void drmp3_L3_stereo_top_band(const float *right, const drmp3_uint8 *sfb, int nbands, int max_band[3]) +{ + int i, k; + + max_band[0] = max_band[1] = max_band[2] = -1; + + for (i = 0; i < nbands; i++) + { + for (k = 0; k < sfb[i]; k += 2) + { + if (right[k] != 0 || right[k + 1] != 0) + { + max_band[i % 3] = i; + break; + } + } + right += sfb[i]; + } +} + +static void drmp3_L3_stereo_process(float *left, const drmp3_uint8 *ist_pos, const drmp3_uint8 *sfb, const drmp3_uint8 *hdr, int max_band[3], int mpeg2_sh) +{ + static const float g_pan[7*2] = { 0,1,0.21132487f,0.78867513f,0.36602540f,0.63397460f,0.5f,0.5f,0.63397460f,0.36602540f,0.78867513f,0.21132487f,1,0 }; + unsigned i, max_pos = DRMP3_HDR_TEST_MPEG1(hdr) ? 7 : 64; + + for (i = 0; sfb[i]; i++) + { + unsigned ipos = ist_pos[i]; + if ((int)i > max_band[i % 3] && ipos < max_pos) + { + float kl, kr, s = DRMP3_HDR_TEST_MS_STEREO(hdr) ? 1.41421356f : 1; + if (DRMP3_HDR_TEST_MPEG1(hdr)) + { + kl = g_pan[2*ipos]; + kr = g_pan[2*ipos + 1]; + } else + { + kl = 1; + kr = drmp3_L3_ldexp_q2(1, (ipos + 1) >> 1 << mpeg2_sh); + if (ipos & 1) + { + kl = kr; + kr = 1; + } + } + drmp3_L3_intensity_stereo_band(left, sfb[i], kl*s, kr*s); + } else if (DRMP3_HDR_TEST_MS_STEREO(hdr)) + { + drmp3_L3_midside_stereo(left, sfb[i]); + } + left += sfb[i]; + } +} + +static void drmp3_L3_intensity_stereo(float *left, drmp3_uint8 *ist_pos, const drmp3_L3_gr_info *gr, const drmp3_uint8 *hdr) +{ + int max_band[3], n_sfb = gr->n_long_sfb + gr->n_short_sfb; + int i, max_blocks = gr->n_short_sfb ? 3 : 1; + + drmp3_L3_stereo_top_band(left + 576, gr->sfbtab, n_sfb, max_band); + if (gr->n_long_sfb) + { + max_band[0] = max_band[1] = max_band[2] = DRMP3_MAX(DRMP3_MAX(max_band[0], max_band[1]), max_band[2]); + } + for (i = 0; i < max_blocks; i++) + { + int default_pos = DRMP3_HDR_TEST_MPEG1(hdr) ? 3 : 0; + int itop = n_sfb - max_blocks + i; + int prev = itop - max_blocks; + ist_pos[itop] = (drmp3_uint8)(max_band[i] >= prev ? default_pos : ist_pos[prev]); + } + drmp3_L3_stereo_process(left, ist_pos, gr->sfbtab, hdr, max_band, gr[1].scalefac_compress & 1); +} + +static void drmp3_L3_reorder(float *grbuf, float *scratch, const drmp3_uint8 *sfb) +{ + int i, len; + float *src = grbuf, *dst = scratch; + + for (;0 != (len = *sfb); sfb += 3, src += 2*len) + { + for (i = 0; i < len; i++, src++) + { + *dst++ = src[0*len]; + *dst++ = src[1*len]; + *dst++ = src[2*len]; + } + } + memcpy(grbuf, scratch, (dst - scratch)*sizeof(float)); +} + +static void drmp3_L3_antialias(float *grbuf, int nbands) +{ + static const float g_aa[2][8] = { + {0.85749293f,0.88174200f,0.94962865f,0.98331459f,0.99551782f,0.99916056f,0.99989920f,0.99999316f}, + {0.51449576f,0.47173197f,0.31337745f,0.18191320f,0.09457419f,0.04096558f,0.01419856f,0.00369997f} + }; + + for (; nbands > 0; nbands--, grbuf += 18) + { + int i = 0; +#if DRMP3_HAVE_SIMD + if (drmp3_have_simd()) for (; i < 8; i += 4) + { + drmp3_f4 vu = DRMP3_VLD(grbuf + 18 + i); + drmp3_f4 vd = DRMP3_VLD(grbuf + 14 - i); + drmp3_f4 vc0 = DRMP3_VLD(g_aa[0] + i); + drmp3_f4 vc1 = DRMP3_VLD(g_aa[1] + i); + vd = DRMP3_VREV(vd); + DRMP3_VSTORE(grbuf + 18 + i, DRMP3_VSUB(DRMP3_VMUL(vu, vc0), DRMP3_VMUL(vd, vc1))); + vd = DRMP3_VADD(DRMP3_VMUL(vu, vc1), DRMP3_VMUL(vd, vc0)); + DRMP3_VSTORE(grbuf + 14 - i, DRMP3_VREV(vd)); + } +#endif +#ifndef DR_MP3_ONLY_SIMD + for(; i < 8; i++) + { + float u = grbuf[18 + i]; + float d = grbuf[17 - i]; + grbuf[18 + i] = u*g_aa[0][i] - d*g_aa[1][i]; + grbuf[17 - i] = u*g_aa[1][i] + d*g_aa[0][i]; + } +#endif + } +} + +static void drmp3_L3_dct3_9(float *y) +{ + float s0, s1, s2, s3, s4, s5, s6, s7, s8, t0, t2, t4; + + s0 = y[0]; s2 = y[2]; s4 = y[4]; s6 = y[6]; s8 = y[8]; + t0 = s0 + s6*0.5f; + s0 -= s6; + t4 = (s4 + s2)*0.93969262f; + t2 = (s8 + s2)*0.76604444f; + s6 = (s4 - s8)*0.17364818f; + s4 += s8 - s2; + + s2 = s0 - s4*0.5f; + y[4] = s4 + s0; + s8 = t0 - t2 + s6; + s0 = t0 - t4 + t2; + s4 = t0 + t4 - s6; + + s1 = y[1]; s3 = y[3]; s5 = y[5]; s7 = y[7]; + + s3 *= 0.86602540f; + t0 = (s5 + s1)*0.98480775f; + t4 = (s5 - s7)*0.34202014f; + t2 = (s1 + s7)*0.64278761f; + s1 = (s1 - s5 - s7)*0.86602540f; + + s5 = t0 - s3 - t2; + s7 = t4 - s3 - t0; + s3 = t4 + s3 - t2; + + y[0] = s4 - s7; + y[1] = s2 + s1; + y[2] = s0 - s3; + y[3] = s8 + s5; + y[5] = s8 - s5; + y[6] = s0 + s3; + y[7] = s2 - s1; + y[8] = s4 + s7; +} + +static void drmp3_L3_imdct36(float *grbuf, float *overlap, const float *window, int nbands) +{ + int i, j; + static const float g_twid9[18] = { + 0.73727734f,0.79335334f,0.84339145f,0.88701083f,0.92387953f,0.95371695f,0.97629601f,0.99144486f,0.99904822f,0.67559021f,0.60876143f,0.53729961f,0.46174861f,0.38268343f,0.30070580f,0.21643961f,0.13052619f,0.04361938f + }; + + for (j = 0; j < nbands; j++, grbuf += 18, overlap += 9) + { + float co[9], si[9]; + co[0] = -grbuf[0]; + si[0] = grbuf[17]; + for (i = 0; i < 4; i++) + { + si[8 - 2*i] = grbuf[4*i + 1] - grbuf[4*i + 2]; + co[1 + 2*i] = grbuf[4*i + 1] + grbuf[4*i + 2]; + si[7 - 2*i] = grbuf[4*i + 4] - grbuf[4*i + 3]; + co[2 + 2*i] = -(grbuf[4*i + 3] + grbuf[4*i + 4]); + } + drmp3_L3_dct3_9(co); + drmp3_L3_dct3_9(si); + + si[1] = -si[1]; + si[3] = -si[3]; + si[5] = -si[5]; + si[7] = -si[7]; + + i = 0; + +#if DRMP3_HAVE_SIMD + if (drmp3_have_simd()) for (; i < 8; i += 4) + { + drmp3_f4 vovl = DRMP3_VLD(overlap + i); + drmp3_f4 vc = DRMP3_VLD(co + i); + drmp3_f4 vs = DRMP3_VLD(si + i); + drmp3_f4 vr0 = DRMP3_VLD(g_twid9 + i); + drmp3_f4 vr1 = DRMP3_VLD(g_twid9 + 9 + i); + drmp3_f4 vw0 = DRMP3_VLD(window + i); + drmp3_f4 vw1 = DRMP3_VLD(window + 9 + i); + drmp3_f4 vsum = DRMP3_VADD(DRMP3_VMUL(vc, vr1), DRMP3_VMUL(vs, vr0)); + DRMP3_VSTORE(overlap + i, DRMP3_VSUB(DRMP3_VMUL(vc, vr0), DRMP3_VMUL(vs, vr1))); + DRMP3_VSTORE(grbuf + i, DRMP3_VSUB(DRMP3_VMUL(vovl, vw0), DRMP3_VMUL(vsum, vw1))); + vsum = DRMP3_VADD(DRMP3_VMUL(vovl, vw1), DRMP3_VMUL(vsum, vw0)); + DRMP3_VSTORE(grbuf + 14 - i, DRMP3_VREV(vsum)); + } +#endif + for (; i < 9; i++) + { + float ovl = overlap[i]; + float sum = co[i]*g_twid9[9 + i] + si[i]*g_twid9[0 + i]; + overlap[i] = co[i]*g_twid9[0 + i] - si[i]*g_twid9[9 + i]; + grbuf[i] = ovl*window[0 + i] - sum*window[9 + i]; + grbuf[17 - i] = ovl*window[9 + i] + sum*window[0 + i]; + } + } +} + +static void drmp3_L3_idct3(float x0, float x1, float x2, float *dst) +{ + float m1 = x1*0.86602540f; + float a1 = x0 - x2*0.5f; + dst[1] = x0 + x2; + dst[0] = a1 + m1; + dst[2] = a1 - m1; +} + +static void drmp3_L3_imdct12(float *x, float *dst, float *overlap) +{ + static const float g_twid3[6] = { 0.79335334f,0.92387953f,0.99144486f, 0.60876143f,0.38268343f,0.13052619f }; + float co[3], si[3]; + int i; + + drmp3_L3_idct3(-x[0], x[6] + x[3], x[12] + x[9], co); + drmp3_L3_idct3(x[15], x[12] - x[9], x[6] - x[3], si); + si[1] = -si[1]; + + for (i = 0; i < 3; i++) + { + float ovl = overlap[i]; + float sum = co[i]*g_twid3[3 + i] + si[i]*g_twid3[0 + i]; + overlap[i] = co[i]*g_twid3[0 + i] - si[i]*g_twid3[3 + i]; + dst[i] = ovl*g_twid3[2 - i] - sum*g_twid3[5 - i]; + dst[5 - i] = ovl*g_twid3[5 - i] + sum*g_twid3[2 - i]; + } +} + +static void drmp3_L3_imdct_short(float *grbuf, float *overlap, int nbands) +{ + for (;nbands > 0; nbands--, overlap += 9, grbuf += 18) + { + float tmp[18]; + memcpy(tmp, grbuf, sizeof(tmp)); + memcpy(grbuf, overlap, 6*sizeof(float)); + drmp3_L3_imdct12(tmp, grbuf + 6, overlap + 6); + drmp3_L3_imdct12(tmp + 1, grbuf + 12, overlap + 6); + drmp3_L3_imdct12(tmp + 2, overlap, overlap + 6); + } +} + +static void drmp3_L3_change_sign(float *grbuf) +{ + int b, i; + for (b = 0, grbuf += 18; b < 32; b += 2, grbuf += 36) + for (i = 1; i < 18; i += 2) + grbuf[i] = -grbuf[i]; +} + +static void drmp3_L3_imdct_gr(float *grbuf, float *overlap, unsigned block_type, unsigned n_long_bands) +{ + static const float g_mdct_window[2][18] = { + { 0.99904822f,0.99144486f,0.97629601f,0.95371695f,0.92387953f,0.88701083f,0.84339145f,0.79335334f,0.73727734f,0.04361938f,0.13052619f,0.21643961f,0.30070580f,0.38268343f,0.46174861f,0.53729961f,0.60876143f,0.67559021f }, + { 1,1,1,1,1,1,0.99144486f,0.92387953f,0.79335334f,0,0,0,0,0,0,0.13052619f,0.38268343f,0.60876143f } + }; + if (n_long_bands) + { + drmp3_L3_imdct36(grbuf, overlap, g_mdct_window[0], n_long_bands); + grbuf += 18*n_long_bands; + overlap += 9*n_long_bands; + } + if (block_type == DRMP3_SHORT_BLOCK_TYPE) + drmp3_L3_imdct_short(grbuf, overlap, 32 - n_long_bands); + else + drmp3_L3_imdct36(grbuf, overlap, g_mdct_window[block_type == DRMP3_STOP_BLOCK_TYPE], 32 - n_long_bands); +} + +static void drmp3_L3_save_reservoir(drmp3dec *h, drmp3dec_scratch *s) +{ + int pos = (s->bs.pos + 7)/8u; + int remains = s->bs.limit/8u - pos; + if (remains > DRMP3_MAX_BITRESERVOIR_BYTES) + { + pos += remains - DRMP3_MAX_BITRESERVOIR_BYTES; + remains = DRMP3_MAX_BITRESERVOIR_BYTES; + } + if (remains > 0) + { + memmove(h->reserv_buf, s->maindata + pos, remains); + } + h->reserv = remains; +} + +static int drmp3_L3_restore_reservoir(drmp3dec *h, drmp3_bs *bs, drmp3dec_scratch *s, int main_data_begin) +{ + int frame_bytes = (bs->limit - bs->pos)/8; + int bytes_have = DRMP3_MIN(h->reserv, main_data_begin); + memcpy(s->maindata, h->reserv_buf + DRMP3_MAX(0, h->reserv - main_data_begin), DRMP3_MIN(h->reserv, main_data_begin)); + memcpy(s->maindata + bytes_have, bs->buf + bs->pos/8, frame_bytes); + drmp3_bs_init(&s->bs, s->maindata, bytes_have + frame_bytes); + return h->reserv >= main_data_begin; +} + +static void drmp3_L3_decode(drmp3dec *h, drmp3dec_scratch *s, drmp3_L3_gr_info *gr_info, int nch) +{ + int ch; + + for (ch = 0; ch < nch; ch++) + { + int layer3gr_limit = s->bs.pos + gr_info[ch].part_23_length; + drmp3_L3_decode_scalefactors(h->header, s->ist_pos[ch], &s->bs, gr_info + ch, s->scf, ch); + drmp3_L3_huffman(s->grbuf[ch], &s->bs, gr_info + ch, s->scf, layer3gr_limit); + } + + if (DRMP3_HDR_TEST_I_STEREO(h->header)) + { + drmp3_L3_intensity_stereo(s->grbuf[0], s->ist_pos[1], gr_info, h->header); + } else if (DRMP3_HDR_IS_MS_STEREO(h->header)) + { + drmp3_L3_midside_stereo(s->grbuf[0], 576); + } + + for (ch = 0; ch < nch; ch++, gr_info++) + { + int aa_bands = 31; + int n_long_bands = (gr_info->mixed_block_flag ? 2 : 0) << (int)(DRMP3_HDR_GET_MY_SAMPLE_RATE(h->header) == 2); + + if (gr_info->n_short_sfb) + { + aa_bands = n_long_bands - 1; + drmp3_L3_reorder(s->grbuf[ch] + n_long_bands*18, s->syn[0], gr_info->sfbtab + gr_info->n_long_sfb); + } + + drmp3_L3_antialias(s->grbuf[ch], aa_bands); + drmp3_L3_imdct_gr(s->grbuf[ch], h->mdct_overlap[ch], gr_info->block_type, n_long_bands); + drmp3_L3_change_sign(s->grbuf[ch]); + } +} + +static void drmp3d_DCT_II(float *grbuf, int n) +{ + static const float g_sec[24] = { + 10.19000816f,0.50060302f,0.50241929f,3.40760851f,0.50547093f,0.52249861f,2.05778098f,0.51544732f,0.56694406f,1.48416460f,0.53104258f,0.64682180f,1.16943991f,0.55310392f,0.78815460f,0.97256821f,0.58293498f,1.06067765f,0.83934963f,0.62250412f,1.72244716f,0.74453628f,0.67480832f,5.10114861f + }; + int i, k = 0; +#if DRMP3_HAVE_SIMD + if (drmp3_have_simd()) for (; k < n; k += 4) + { + drmp3_f4 t[4][8], *x; + float *y = grbuf + k; + + for (x = t[0], i = 0; i < 8; i++, x++) + { + drmp3_f4 x0 = DRMP3_VLD(&y[i*18]); + drmp3_f4 x1 = DRMP3_VLD(&y[(15 - i)*18]); + drmp3_f4 x2 = DRMP3_VLD(&y[(16 + i)*18]); + drmp3_f4 x3 = DRMP3_VLD(&y[(31 - i)*18]); + drmp3_f4 t0 = DRMP3_VADD(x0, x3); + drmp3_f4 t1 = DRMP3_VADD(x1, x2); + drmp3_f4 t2 = DRMP3_VMUL_S(DRMP3_VSUB(x1, x2), g_sec[3*i + 0]); + drmp3_f4 t3 = DRMP3_VMUL_S(DRMP3_VSUB(x0, x3), g_sec[3*i + 1]); + x[0] = DRMP3_VADD(t0, t1); + x[8] = DRMP3_VMUL_S(DRMP3_VSUB(t0, t1), g_sec[3*i + 2]); + x[16] = DRMP3_VADD(t3, t2); + x[24] = DRMP3_VMUL_S(DRMP3_VSUB(t3, t2), g_sec[3*i + 2]); + } + for (x = t[0], i = 0; i < 4; i++, x += 8) + { + drmp3_f4 x0 = x[0], x1 = x[1], x2 = x[2], x3 = x[3], x4 = x[4], x5 = x[5], x6 = x[6], x7 = x[7], xt; + xt = DRMP3_VSUB(x0, x7); x0 = DRMP3_VADD(x0, x7); + x7 = DRMP3_VSUB(x1, x6); x1 = DRMP3_VADD(x1, x6); + x6 = DRMP3_VSUB(x2, x5); x2 = DRMP3_VADD(x2, x5); + x5 = DRMP3_VSUB(x3, x4); x3 = DRMP3_VADD(x3, x4); + x4 = DRMP3_VSUB(x0, x3); x0 = DRMP3_VADD(x0, x3); + x3 = DRMP3_VSUB(x1, x2); x1 = DRMP3_VADD(x1, x2); + x[0] = DRMP3_VADD(x0, x1); + x[4] = DRMP3_VMUL_S(DRMP3_VSUB(x0, x1), 0.70710677f); + x5 = DRMP3_VADD(x5, x6); + x6 = DRMP3_VMUL_S(DRMP3_VADD(x6, x7), 0.70710677f); + x7 = DRMP3_VADD(x7, xt); + x3 = DRMP3_VMUL_S(DRMP3_VADD(x3, x4), 0.70710677f); + x5 = DRMP3_VSUB(x5, DRMP3_VMUL_S(x7, 0.198912367f)); /* rotate by PI/8 */ + x7 = DRMP3_VADD(x7, DRMP3_VMUL_S(x5, 0.382683432f)); + x5 = DRMP3_VSUB(x5, DRMP3_VMUL_S(x7, 0.198912367f)); + x0 = DRMP3_VSUB(xt, x6); xt = DRMP3_VADD(xt, x6); + x[1] = DRMP3_VMUL_S(DRMP3_VADD(xt, x7), 0.50979561f); + x[2] = DRMP3_VMUL_S(DRMP3_VADD(x4, x3), 0.54119611f); + x[3] = DRMP3_VMUL_S(DRMP3_VSUB(x0, x5), 0.60134488f); + x[5] = DRMP3_VMUL_S(DRMP3_VADD(x0, x5), 0.89997619f); + x[6] = DRMP3_VMUL_S(DRMP3_VSUB(x4, x3), 1.30656302f); + x[7] = DRMP3_VMUL_S(DRMP3_VSUB(xt, x7), 2.56291556f); + } + + if (k > n - 3) + { +#if DRMP3_HAVE_SSE +#define DRMP3_VSAVE2(i, v) _mm_storel_pi((__m64 *)(void*)&y[i*18], v) +#else +#define DRMP3_VSAVE2(i, v) vst1_f32((float32_t *)&y[i*18], vget_low_f32(v)) +#endif + for (i = 0; i < 7; i++, y += 4*18) + { + drmp3_f4 s = DRMP3_VADD(t[3][i], t[3][i + 1]); + DRMP3_VSAVE2(0, t[0][i]); + DRMP3_VSAVE2(1, DRMP3_VADD(t[2][i], s)); + DRMP3_VSAVE2(2, DRMP3_VADD(t[1][i], t[1][i + 1])); + DRMP3_VSAVE2(3, DRMP3_VADD(t[2][1 + i], s)); + } + DRMP3_VSAVE2(0, t[0][7]); + DRMP3_VSAVE2(1, DRMP3_VADD(t[2][7], t[3][7])); + DRMP3_VSAVE2(2, t[1][7]); + DRMP3_VSAVE2(3, t[3][7]); + } else + { +#define DRMP3_VSAVE4(i, v) DRMP3_VSTORE(&y[i*18], v) + for (i = 0; i < 7; i++, y += 4*18) + { + drmp3_f4 s = DRMP3_VADD(t[3][i], t[3][i + 1]); + DRMP3_VSAVE4(0, t[0][i]); + DRMP3_VSAVE4(1, DRMP3_VADD(t[2][i], s)); + DRMP3_VSAVE4(2, DRMP3_VADD(t[1][i], t[1][i + 1])); + DRMP3_VSAVE4(3, DRMP3_VADD(t[2][1 + i], s)); + } + DRMP3_VSAVE4(0, t[0][7]); + DRMP3_VSAVE4(1, DRMP3_VADD(t[2][7], t[3][7])); + DRMP3_VSAVE4(2, t[1][7]); + DRMP3_VSAVE4(3, t[3][7]); + } + } else +#endif +#ifdef DR_MP3_ONLY_SIMD + {} +#else + for (; k < n; k++) + { + float t[4][8], *x, *y = grbuf + k; + + for (x = t[0], i = 0; i < 8; i++, x++) + { + float x0 = y[i*18]; + float x1 = y[(15 - i)*18]; + float x2 = y[(16 + i)*18]; + float x3 = y[(31 - i)*18]; + float t0 = x0 + x3; + float t1 = x1 + x2; + float t2 = (x1 - x2)*g_sec[3*i + 0]; + float t3 = (x0 - x3)*g_sec[3*i + 1]; + x[0] = t0 + t1; + x[8] = (t0 - t1)*g_sec[3*i + 2]; + x[16] = t3 + t2; + x[24] = (t3 - t2)*g_sec[3*i + 2]; + } + for (x = t[0], i = 0; i < 4; i++, x += 8) + { + float x0 = x[0], x1 = x[1], x2 = x[2], x3 = x[3], x4 = x[4], x5 = x[5], x6 = x[6], x7 = x[7], xt; + xt = x0 - x7; x0 += x7; + x7 = x1 - x6; x1 += x6; + x6 = x2 - x5; x2 += x5; + x5 = x3 - x4; x3 += x4; + x4 = x0 - x3; x0 += x3; + x3 = x1 - x2; x1 += x2; + x[0] = x0 + x1; + x[4] = (x0 - x1)*0.70710677f; + x5 = x5 + x6; + x6 = (x6 + x7)*0.70710677f; + x7 = x7 + xt; + x3 = (x3 + x4)*0.70710677f; + x5 -= x7*0.198912367f; /* rotate by PI/8 */ + x7 += x5*0.382683432f; + x5 -= x7*0.198912367f; + x0 = xt - x6; xt += x6; + x[1] = (xt + x7)*0.50979561f; + x[2] = (x4 + x3)*0.54119611f; + x[3] = (x0 - x5)*0.60134488f; + x[5] = (x0 + x5)*0.89997619f; + x[6] = (x4 - x3)*1.30656302f; + x[7] = (xt - x7)*2.56291556f; + + } + for (i = 0; i < 7; i++, y += 4*18) + { + y[0*18] = t[0][i]; + y[1*18] = t[2][i] + t[3][i] + t[3][i + 1]; + y[2*18] = t[1][i] + t[1][i + 1]; + y[3*18] = t[2][i + 1] + t[3][i] + t[3][i + 1]; + } + y[0*18] = t[0][7]; + y[1*18] = t[2][7] + t[3][7]; + y[2*18] = t[1][7]; + y[3*18] = t[3][7]; + } +#endif +} + +#ifndef DR_MP3_FLOAT_OUTPUT +typedef drmp3_int16 drmp3d_sample_t; + +static drmp3_int16 drmp3d_scale_pcm(float sample) +{ + drmp3_int16 s; + if (sample >= 32766.5) return (drmp3_int16) 32767; + if (sample <= -32767.5) return (drmp3_int16)-32768; + s = (drmp3_int16)(sample + .5f); + s -= (s < 0); /* away from zero, to be compliant */ + return (drmp3_int16)s; +} +#else +typedef float drmp3d_sample_t; + +static float drmp3d_scale_pcm(float sample) +{ + return sample*(1.f/32768.f); +} +#endif + +static void drmp3d_synth_pair(drmp3d_sample_t *pcm, int nch, const float *z) +{ + float a; + a = (z[14*64] - z[ 0]) * 29; + a += (z[ 1*64] + z[13*64]) * 213; + a += (z[12*64] - z[ 2*64]) * 459; + a += (z[ 3*64] + z[11*64]) * 2037; + a += (z[10*64] - z[ 4*64]) * 5153; + a += (z[ 5*64] + z[ 9*64]) * 6574; + a += (z[ 8*64] - z[ 6*64]) * 37489; + a += z[ 7*64] * 75038; + pcm[0] = drmp3d_scale_pcm(a); + + z += 2; + a = z[14*64] * 104; + a += z[12*64] * 1567; + a += z[10*64] * 9727; + a += z[ 8*64] * 64019; + a += z[ 6*64] * -9975; + a += z[ 4*64] * -45; + a += z[ 2*64] * 146; + a += z[ 0*64] * -5; + pcm[16*nch] = drmp3d_scale_pcm(a); +} + +static void drmp3d_synth(float *xl, drmp3d_sample_t *dstl, int nch, float *lins) +{ + int i; + float *xr = xl + 576*(nch - 1); + drmp3d_sample_t *dstr = dstl + (nch - 1); + + static const float g_win[] = { + -1,26,-31,208,218,401,-519,2063,2000,4788,-5517,7134,5959,35640,-39336,74992, + -1,24,-35,202,222,347,-581,2080,1952,4425,-5879,7640,5288,33791,-41176,74856, + -1,21,-38,196,225,294,-645,2087,1893,4063,-6237,8092,4561,31947,-43006,74630, + -1,19,-41,190,227,244,-711,2085,1822,3705,-6589,8492,3776,30112,-44821,74313, + -1,17,-45,183,228,197,-779,2075,1739,3351,-6935,8840,2935,28289,-46617,73908, + -1,16,-49,176,228,153,-848,2057,1644,3004,-7271,9139,2037,26482,-48390,73415, + -2,14,-53,169,227,111,-919,2032,1535,2663,-7597,9389,1082,24694,-50137,72835, + -2,13,-58,161,224,72,-991,2001,1414,2330,-7910,9592,70,22929,-51853,72169, + -2,11,-63,154,221,36,-1064,1962,1280,2006,-8209,9750,-998,21189,-53534,71420, + -2,10,-68,147,215,2,-1137,1919,1131,1692,-8491,9863,-2122,19478,-55178,70590, + -3,9,-73,139,208,-29,-1210,1870,970,1388,-8755,9935,-3300,17799,-56778,69679, + -3,8,-79,132,200,-57,-1283,1817,794,1095,-8998,9966,-4533,16155,-58333,68692, + -4,7,-85,125,189,-83,-1356,1759,605,814,-9219,9959,-5818,14548,-59838,67629, + -4,7,-91,117,177,-106,-1428,1698,402,545,-9416,9916,-7154,12980,-61289,66494, + -5,6,-97,111,163,-127,-1498,1634,185,288,-9585,9838,-8540,11455,-62684,65290 + }; + float *zlin = lins + 15*64; + const float *w = g_win; + + zlin[4*15] = xl[18*16]; + zlin[4*15 + 1] = xr[18*16]; + zlin[4*15 + 2] = xl[0]; + zlin[4*15 + 3] = xr[0]; + + zlin[4*31] = xl[1 + 18*16]; + zlin[4*31 + 1] = xr[1 + 18*16]; + zlin[4*31 + 2] = xl[1]; + zlin[4*31 + 3] = xr[1]; + + drmp3d_synth_pair(dstr, nch, lins + 4*15 + 1); + drmp3d_synth_pair(dstr + 32*nch, nch, lins + 4*15 + 64 + 1); + drmp3d_synth_pair(dstl, nch, lins + 4*15); + drmp3d_synth_pair(dstl + 32*nch, nch, lins + 4*15 + 64); + +#if DRMP3_HAVE_SIMD + if (drmp3_have_simd()) for (i = 14; i >= 0; i--) + { +#define DRMP3_VLOAD(k) drmp3_f4 w0 = DRMP3_VSET(*w++); drmp3_f4 w1 = DRMP3_VSET(*w++); drmp3_f4 vz = DRMP3_VLD(&zlin[4*i - 64*k]); drmp3_f4 vy = DRMP3_VLD(&zlin[4*i - 64*(15 - k)]); +#define DRMP3_V0(k) { DRMP3_VLOAD(k) b = DRMP3_VADD(DRMP3_VMUL(vz, w1), DRMP3_VMUL(vy, w0)) ; a = DRMP3_VSUB(DRMP3_VMUL(vz, w0), DRMP3_VMUL(vy, w1)); } +#define DRMP3_V1(k) { DRMP3_VLOAD(k) b = DRMP3_VADD(b, DRMP3_VADD(DRMP3_VMUL(vz, w1), DRMP3_VMUL(vy, w0))); a = DRMP3_VADD(a, DRMP3_VSUB(DRMP3_VMUL(vz, w0), DRMP3_VMUL(vy, w1))); } +#define DRMP3_V2(k) { DRMP3_VLOAD(k) b = DRMP3_VADD(b, DRMP3_VADD(DRMP3_VMUL(vz, w1), DRMP3_VMUL(vy, w0))); a = DRMP3_VADD(a, DRMP3_VSUB(DRMP3_VMUL(vy, w1), DRMP3_VMUL(vz, w0))); } + drmp3_f4 a, b; + zlin[4*i] = xl[18*(31 - i)]; + zlin[4*i + 1] = xr[18*(31 - i)]; + zlin[4*i + 2] = xl[1 + 18*(31 - i)]; + zlin[4*i + 3] = xr[1 + 18*(31 - i)]; + zlin[4*i + 64] = xl[1 + 18*(1 + i)]; + zlin[4*i + 64 + 1] = xr[1 + 18*(1 + i)]; + zlin[4*i - 64 + 2] = xl[18*(1 + i)]; + zlin[4*i - 64 + 3] = xr[18*(1 + i)]; + + DRMP3_V0(0) DRMP3_V2(1) DRMP3_V1(2) DRMP3_V2(3) DRMP3_V1(4) DRMP3_V2(5) DRMP3_V1(6) DRMP3_V2(7) + + { +#ifndef DR_MP3_FLOAT_OUTPUT +#if DRMP3_HAVE_SSE + static const drmp3_f4 g_max = { 32767.0f, 32767.0f, 32767.0f, 32767.0f }; + static const drmp3_f4 g_min = { -32768.0f, -32768.0f, -32768.0f, -32768.0f }; + __m128i pcm8 = _mm_packs_epi32(_mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(a, g_max), g_min)), + _mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(b, g_max), g_min))); + dstr[(15 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 1); + dstr[(17 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 5); + dstl[(15 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 0); + dstl[(17 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 4); + dstr[(47 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 3); + dstr[(49 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 7); + dstl[(47 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 2); + dstl[(49 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 6); +#else + int16x4_t pcma, pcmb; + a = DRMP3_VADD(a, DRMP3_VSET(0.5f)); + b = DRMP3_VADD(b, DRMP3_VSET(0.5f)); + pcma = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(a), vreinterpretq_s32_u32(vcltq_f32(a, DRMP3_VSET(0))))); + pcmb = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(b), vreinterpretq_s32_u32(vcltq_f32(b, DRMP3_VSET(0))))); + vst1_lane_s16(dstr + (15 - i)*nch, pcma, 1); + vst1_lane_s16(dstr + (17 + i)*nch, pcmb, 1); + vst1_lane_s16(dstl + (15 - i)*nch, pcma, 0); + vst1_lane_s16(dstl + (17 + i)*nch, pcmb, 0); + vst1_lane_s16(dstr + (47 - i)*nch, pcma, 3); + vst1_lane_s16(dstr + (49 + i)*nch, pcmb, 3); + vst1_lane_s16(dstl + (47 - i)*nch, pcma, 2); + vst1_lane_s16(dstl + (49 + i)*nch, pcmb, 2); +#endif +#else + static const drmp3_f4 g_scale = { 1.0f/32768.0f, 1.0f/32768.0f, 1.0f/32768.0f, 1.0f/32768.0f }; + a = DRMP3_VMUL(a, g_scale); + b = DRMP3_VMUL(b, g_scale); +#if DRMP3_HAVE_SSE + _mm_store_ss(dstr + (15 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(1, 1, 1, 1))); + _mm_store_ss(dstr + (17 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(1, 1, 1, 1))); + _mm_store_ss(dstl + (15 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(0, 0, 0, 0))); + _mm_store_ss(dstl + (17 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(0, 0, 0, 0))); + _mm_store_ss(dstr + (47 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(3, 3, 3, 3))); + _mm_store_ss(dstr + (49 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(3, 3, 3, 3))); + _mm_store_ss(dstl + (47 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(2, 2, 2, 2))); + _mm_store_ss(dstl + (49 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(2, 2, 2, 2))); +#else + vst1q_lane_f32(dstr + (15 - i)*nch, a, 1); + vst1q_lane_f32(dstr + (17 + i)*nch, b, 1); + vst1q_lane_f32(dstl + (15 - i)*nch, a, 0); + vst1q_lane_f32(dstl + (17 + i)*nch, b, 0); + vst1q_lane_f32(dstr + (47 - i)*nch, a, 3); + vst1q_lane_f32(dstr + (49 + i)*nch, b, 3); + vst1q_lane_f32(dstl + (47 - i)*nch, a, 2); + vst1q_lane_f32(dstl + (49 + i)*nch, b, 2); +#endif +#endif /* DR_MP3_FLOAT_OUTPUT */ + } + } else +#endif +#ifdef DR_MP3_ONLY_SIMD + {} +#else + for (i = 14; i >= 0; i--) + { +#define DRMP3_LOAD(k) float w0 = *w++; float w1 = *w++; float *vz = &zlin[4*i - k*64]; float *vy = &zlin[4*i - (15 - k)*64]; +#define DRMP3_S0(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] = vz[j]*w1 + vy[j]*w0, a[j] = vz[j]*w0 - vy[j]*w1; } +#define DRMP3_S1(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] += vz[j]*w1 + vy[j]*w0, a[j] += vz[j]*w0 - vy[j]*w1; } +#define DRMP3_S2(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] += vz[j]*w1 + vy[j]*w0, a[j] += vy[j]*w1 - vz[j]*w0; } + float a[4], b[4]; + + zlin[4*i] = xl[18*(31 - i)]; + zlin[4*i + 1] = xr[18*(31 - i)]; + zlin[4*i + 2] = xl[1 + 18*(31 - i)]; + zlin[4*i + 3] = xr[1 + 18*(31 - i)]; + zlin[4*(i + 16)] = xl[1 + 18*(1 + i)]; + zlin[4*(i + 16) + 1] = xr[1 + 18*(1 + i)]; + zlin[4*(i - 16) + 2] = xl[18*(1 + i)]; + zlin[4*(i - 16) + 3] = xr[18*(1 + i)]; + + DRMP3_S0(0) DRMP3_S2(1) DRMP3_S1(2) DRMP3_S2(3) DRMP3_S1(4) DRMP3_S2(5) DRMP3_S1(6) DRMP3_S2(7) + + dstr[(15 - i)*nch] = drmp3d_scale_pcm(a[1]); + dstr[(17 + i)*nch] = drmp3d_scale_pcm(b[1]); + dstl[(15 - i)*nch] = drmp3d_scale_pcm(a[0]); + dstl[(17 + i)*nch] = drmp3d_scale_pcm(b[0]); + dstr[(47 - i)*nch] = drmp3d_scale_pcm(a[3]); + dstr[(49 + i)*nch] = drmp3d_scale_pcm(b[3]); + dstl[(47 - i)*nch] = drmp3d_scale_pcm(a[2]); + dstl[(49 + i)*nch] = drmp3d_scale_pcm(b[2]); + } +#endif +} + +static void drmp3d_synth_granule(float *qmf_state, float *grbuf, int nbands, int nch, drmp3d_sample_t *pcm, float *lins) +{ + int i; + for (i = 0; i < nch; i++) + { + drmp3d_DCT_II(grbuf + 576*i, nbands); + } + + memcpy(lins, qmf_state, sizeof(float)*15*64); + + for (i = 0; i < nbands; i += 2) + { + drmp3d_synth(grbuf + i, pcm + 32*nch*i, nch, lins + i*64); + } +#ifndef DR_MP3_NONSTANDARD_BUT_LOGICAL + if (nch == 1) + { + for (i = 0; i < 15*64; i += 2) + { + qmf_state[i] = lins[nbands*64 + i]; + } + } else +#endif + { + memcpy(qmf_state, lins + nbands*64, sizeof(float)*15*64); + } +} + +static int drmp3d_match_frame(const drmp3_uint8 *hdr, int mp3_bytes, int frame_bytes) +{ + int i, nmatch; + for (i = 0, nmatch = 0; nmatch < DRMP3_MAX_FRAME_SYNC_MATCHES; nmatch++) + { + i += drmp3_hdr_frame_bytes(hdr + i, frame_bytes) + drmp3_hdr_padding(hdr + i); + if (i + DRMP3_HDR_SIZE > mp3_bytes) + return nmatch > 0; + if (!drmp3_hdr_compare(hdr, hdr + i)) + return 0; + } + return 1; +} + +static int drmp3d_find_frame(const drmp3_uint8 *mp3, int mp3_bytes, int *free_format_bytes, int *ptr_frame_bytes) +{ + int i, k; + for (i = 0; i < mp3_bytes - DRMP3_HDR_SIZE; i++, mp3++) + { + if (drmp3_hdr_valid(mp3)) + { + int frame_bytes = drmp3_hdr_frame_bytes(mp3, *free_format_bytes); + int frame_and_padding = frame_bytes + drmp3_hdr_padding(mp3); + + for (k = DRMP3_HDR_SIZE; !frame_bytes && k < DRMP3_MAX_FREE_FORMAT_FRAME_SIZE && i + 2*k < mp3_bytes - DRMP3_HDR_SIZE; k++) + { + if (drmp3_hdr_compare(mp3, mp3 + k)) + { + int fb = k - drmp3_hdr_padding(mp3); + int nextfb = fb + drmp3_hdr_padding(mp3 + k); + if (i + k + nextfb + DRMP3_HDR_SIZE > mp3_bytes || !drmp3_hdr_compare(mp3, mp3 + k + nextfb)) + continue; + frame_and_padding = k; + frame_bytes = fb; + *free_format_bytes = fb; + } + } + + if ((frame_bytes && i + frame_and_padding <= mp3_bytes && + drmp3d_match_frame(mp3, mp3_bytes - i, frame_bytes)) || + (!i && frame_and_padding == mp3_bytes)) + { + *ptr_frame_bytes = frame_and_padding; + return i; + } + *free_format_bytes = 0; + } + } + *ptr_frame_bytes = 0; + return i; +} + +void drmp3dec_init(drmp3dec *dec) +{ + dec->header[0] = 0; +} + +int drmp3dec_decode_frame(drmp3dec *dec, const unsigned char *mp3, int mp3_bytes, void *pcm, drmp3dec_frame_info *info) +{ + int i = 0, igr, frame_size = 0, success = 1; + const drmp3_uint8 *hdr; + drmp3_bs bs_frame[1]; + drmp3dec_scratch scratch; + + if (mp3_bytes > 4 && dec->header[0] == 0xff && drmp3_hdr_compare(dec->header, mp3)) + { + frame_size = drmp3_hdr_frame_bytes(mp3, dec->free_format_bytes) + drmp3_hdr_padding(mp3); + if (frame_size != mp3_bytes && (frame_size + DRMP3_HDR_SIZE > mp3_bytes || !drmp3_hdr_compare(mp3, mp3 + frame_size))) + { + frame_size = 0; + } + } + if (!frame_size) + { + memset(dec, 0, sizeof(drmp3dec)); + i = drmp3d_find_frame(mp3, mp3_bytes, &dec->free_format_bytes, &frame_size); + if (!frame_size || i + frame_size > mp3_bytes) + { + info->frame_bytes = i; + return 0; + } + } + + hdr = mp3 + i; + memcpy(dec->header, hdr, DRMP3_HDR_SIZE); + info->frame_bytes = i + frame_size; + info->channels = DRMP3_HDR_IS_MONO(hdr) ? 1 : 2; + info->hz = drmp3_hdr_sample_rate_hz(hdr); + info->layer = 4 - DRMP3_HDR_GET_LAYER(hdr); + info->bitrate_kbps = drmp3_hdr_bitrate_kbps(hdr); + + drmp3_bs_init(bs_frame, hdr + DRMP3_HDR_SIZE, frame_size - DRMP3_HDR_SIZE); + if (DRMP3_HDR_IS_CRC(hdr)) + { + drmp3_bs_get_bits(bs_frame, 16); + } + + if (info->layer == 3) + { + int main_data_begin = drmp3_L3_read_side_info(bs_frame, scratch.gr_info, hdr); + if (main_data_begin < 0 || bs_frame->pos > bs_frame->limit) + { + drmp3dec_init(dec); + return 0; + } + success = drmp3_L3_restore_reservoir(dec, bs_frame, &scratch, main_data_begin); + if (success && pcm != NULL) + { + for (igr = 0; igr < (DRMP3_HDR_TEST_MPEG1(hdr) ? 2 : 1); igr++, pcm = DRMP3_OFFSET_PTR(pcm, sizeof(drmp3d_sample_t)*576*info->channels)) + { + memset(scratch.grbuf[0], 0, 576*2*sizeof(float)); + drmp3_L3_decode(dec, &scratch, scratch.gr_info + igr*info->channels, info->channels); + drmp3d_synth_granule(dec->qmf_state, scratch.grbuf[0], 18, info->channels, (drmp3d_sample_t*)pcm, scratch.syn[0]); + } + } + drmp3_L3_save_reservoir(dec, &scratch); + } else + { +#ifdef DR_MP3_ONLY_MP3 + return 0; +#else + drmp3_L12_scale_info sci[1]; + + if (pcm == NULL) { + return drmp3_hdr_frame_samples(hdr); + } + + drmp3_L12_read_scale_info(hdr, bs_frame, sci); + + memset(scratch.grbuf[0], 0, 576*2*sizeof(float)); + for (i = 0, igr = 0; igr < 3; igr++) + { + if (12 == (i += drmp3_L12_dequantize_granule(scratch.grbuf[0] + i, bs_frame, sci, info->layer | 1))) + { + i = 0; + drmp3_L12_apply_scf_384(sci, sci->scf + igr, scratch.grbuf[0]); + drmp3d_synth_granule(dec->qmf_state, scratch.grbuf[0], 12, info->channels, (drmp3d_sample_t*)pcm, scratch.syn[0]); + memset(scratch.grbuf[0], 0, 576*2*sizeof(float)); + pcm = DRMP3_OFFSET_PTR(pcm, sizeof(drmp3d_sample_t)*384*info->channels); + } + if (bs_frame->pos > bs_frame->limit) + { + drmp3dec_init(dec); + return 0; + } + } +#endif + } + + return success*drmp3_hdr_frame_samples(dec->header); +} + +void drmp3dec_f32_to_s16(const float *in, drmp3_int16 *out, int num_samples) +{ + if(num_samples > 0) + { + int i = 0; +#if DRMP3_HAVE_SIMD + int aligned_count = num_samples & ~7; + for(; i < aligned_count; i+=8) + { + drmp3_f4 scale = DRMP3_VSET(32768.0f); + drmp3_f4 a = DRMP3_VMUL(DRMP3_VLD(&in[i ]), scale); + drmp3_f4 b = DRMP3_VMUL(DRMP3_VLD(&in[i+4]), scale); +#if DRMP3_HAVE_SSE + drmp3_f4 s16max = DRMP3_VSET( 32767.0f); + drmp3_f4 s16min = DRMP3_VSET(-32768.0f); + __m128i pcm8 = _mm_packs_epi32(_mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(a, s16max), s16min)), + _mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(b, s16max), s16min))); + out[i ] = (drmp3_int16)_mm_extract_epi16(pcm8, 0); + out[i+1] = (drmp3_int16)_mm_extract_epi16(pcm8, 1); + out[i+2] = (drmp3_int16)_mm_extract_epi16(pcm8, 2); + out[i+3] = (drmp3_int16)_mm_extract_epi16(pcm8, 3); + out[i+4] = (drmp3_int16)_mm_extract_epi16(pcm8, 4); + out[i+5] = (drmp3_int16)_mm_extract_epi16(pcm8, 5); + out[i+6] = (drmp3_int16)_mm_extract_epi16(pcm8, 6); + out[i+7] = (drmp3_int16)_mm_extract_epi16(pcm8, 7); +#else + int16x4_t pcma, pcmb; + a = DRMP3_VADD(a, DRMP3_VSET(0.5f)); + b = DRMP3_VADD(b, DRMP3_VSET(0.5f)); + pcma = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(a), vreinterpretq_s32_u32(vcltq_f32(a, DRMP3_VSET(0))))); + pcmb = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(b), vreinterpretq_s32_u32(vcltq_f32(b, DRMP3_VSET(0))))); + vst1_lane_s16(out+i , pcma, 0); + vst1_lane_s16(out+i+1, pcma, 1); + vst1_lane_s16(out+i+2, pcma, 2); + vst1_lane_s16(out+i+3, pcma, 3); + vst1_lane_s16(out+i+4, pcmb, 0); + vst1_lane_s16(out+i+5, pcmb, 1); + vst1_lane_s16(out+i+6, pcmb, 2); + vst1_lane_s16(out+i+7, pcmb, 3); +#endif + } +#endif + for(; i < num_samples; i++) + { + float sample = in[i] * 32768.0f; + if (sample >= 32766.5) + out[i] = (drmp3_int16) 32767; + else if (sample <= -32767.5) + out[i] = (drmp3_int16)-32768; + else + { + short s = (drmp3_int16)(sample + .5f); + s -= (s < 0); /* away from zero, to be compliant */ + out[i] = s; + } + } + } +} + + + +/************************************************************************************************************************************************************ + + Main Public API + + ************************************************************************************************************************************************************/ + +#if defined(SIZE_MAX) + #define DRMP3_SIZE_MAX SIZE_MAX +#else + #if defined(_WIN64) || defined(_LP64) || defined(__LP64__) + #define DRMP3_SIZE_MAX ((drmp3_uint64)0xFFFFFFFFFFFFFFFF) + #else + #define DRMP3_SIZE_MAX 0xFFFFFFFF + #endif +#endif + +/* Options. */ +#ifndef DRMP3_SEEK_LEADING_MP3_FRAMES +#define DRMP3_SEEK_LEADING_MP3_FRAMES 2 +#endif + + +/* Standard library stuff. */ +#ifndef DRMP3_ASSERT +#include <assert.h> +#define DRMP3_ASSERT(expression) assert(expression) +#endif +#ifndef DRMP3_COPY_MEMORY +#define DRMP3_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz)) +#endif +#ifndef DRMP3_ZERO_MEMORY +#define DRMP3_ZERO_MEMORY(p, sz) memset((p), 0, (sz)) +#endif +#define DRMP3_ZERO_OBJECT(p) DRMP3_ZERO_MEMORY((p), sizeof(*(p))) +#ifndef DRMP3_MALLOC +#define DRMP3_MALLOC(sz) malloc((sz)) +#endif +#ifndef DRMP3_REALLOC +#define DRMP3_REALLOC(p, sz) realloc((p), (sz)) +#endif +#ifndef DRMP3_FREE +#define DRMP3_FREE(p) free((p)) +#endif + +#define drmp3_assert DRMP3_ASSERT +#define drmp3_copy_memory DRMP3_COPY_MEMORY +#define drmp3_zero_memory DRMP3_ZERO_MEMORY +#define drmp3_zero_object DRMP3_ZERO_OBJECT +#define drmp3_malloc DRMP3_MALLOC +#define drmp3_realloc DRMP3_REALLOC + +#define drmp3_countof(x) (sizeof(x) / sizeof(x[0])) +#define drmp3_max(x, y) (((x) > (y)) ? (x) : (y)) +#define drmp3_min(x, y) (((x) < (y)) ? (x) : (y)) + +#define DRMP3_DATA_CHUNK_SIZE 16384 /* The size in bytes of each chunk of data to read from the MP3 stream. minimp3 recommends 16K. */ + +static DRMP3_INLINE float drmp3_mix_f32(float x, float y, float a) +{ + return x*(1-a) + y*a; +} + +static void drmp3_blend_f32(float* pOut, float* pInA, float* pInB, float factor, drmp3_uint32 channels) +{ + drmp3_uint32 i; + for (i = 0; i < channels; ++i) { + pOut[i] = drmp3_mix_f32(pInA[i], pInB[i], factor); + } +} + +void drmp3_src_cache_init(drmp3_src* pSRC, drmp3_src_cache* pCache) +{ + drmp3_assert(pSRC != NULL); + drmp3_assert(pCache != NULL); + + pCache->pSRC = pSRC; + pCache->cachedFrameCount = 0; + pCache->iNextFrame = 0; +} + +drmp3_uint64 drmp3_src_cache_read_frames(drmp3_src_cache* pCache, drmp3_uint64 frameCount, float* pFramesOut) +{ + drmp3_uint32 channels; + drmp3_uint64 totalFramesRead = 0; + + drmp3_assert(pCache != NULL); + drmp3_assert(pCache->pSRC != NULL); + drmp3_assert(pCache->pSRC->onRead != NULL); + drmp3_assert(frameCount > 0); + drmp3_assert(pFramesOut != NULL); + + channels = pCache->pSRC->config.channels; + + while (frameCount > 0) { + /* If there's anything in memory go ahead and copy that over first. */ + drmp3_uint32 framesToReadFromClient; + drmp3_uint64 framesRemainingInMemory = pCache->cachedFrameCount - pCache->iNextFrame; + drmp3_uint64 framesToReadFromMemory = frameCount; + if (framesToReadFromMemory > framesRemainingInMemory) { + framesToReadFromMemory = framesRemainingInMemory; + } + + drmp3_copy_memory(pFramesOut, pCache->pCachedFrames + pCache->iNextFrame*channels, (drmp3_uint32)(framesToReadFromMemory * channels * sizeof(float))); + pCache->iNextFrame += (drmp3_uint32)framesToReadFromMemory; + + totalFramesRead += framesToReadFromMemory; + frameCount -= framesToReadFromMemory; + if (frameCount == 0) { + break; + } + + + /* At this point there are still more frames to read from the client, so we'll need to reload the cache with fresh data. */ + drmp3_assert(frameCount > 0); + pFramesOut += framesToReadFromMemory * channels; + + pCache->iNextFrame = 0; + pCache->cachedFrameCount = 0; + + framesToReadFromClient = drmp3_countof(pCache->pCachedFrames) / pCache->pSRC->config.channels; + if (framesToReadFromClient > pCache->pSRC->config.cacheSizeInFrames) { + framesToReadFromClient = pCache->pSRC->config.cacheSizeInFrames; + } + + pCache->cachedFrameCount = (drmp3_uint32)pCache->pSRC->onRead(pCache->pSRC, framesToReadFromClient, pCache->pCachedFrames, pCache->pSRC->pUserData); + + + /* Get out of this loop if nothing was able to be retrieved. */ + if (pCache->cachedFrameCount == 0) { + break; + } + } + + return totalFramesRead; +} + + +drmp3_uint64 drmp3_src_read_frames_passthrough(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, drmp3_bool32 flush); +drmp3_uint64 drmp3_src_read_frames_linear(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, drmp3_bool32 flush); + +drmp3_bool32 drmp3_src_init(const drmp3_src_config* pConfig, drmp3_src_read_proc onRead, void* pUserData, drmp3_src* pSRC) +{ + if (pSRC == NULL) { + return DRMP3_FALSE; + } + + drmp3_zero_object(pSRC); + + if (pConfig == NULL || onRead == NULL) { + return DRMP3_FALSE; + } + + if (pConfig->channels == 0 || pConfig->channels > 2) { + return DRMP3_FALSE; + } + + pSRC->config = *pConfig; + pSRC->onRead = onRead; + pSRC->pUserData = pUserData; + + if (pSRC->config.cacheSizeInFrames > DRMP3_SRC_CACHE_SIZE_IN_FRAMES || pSRC->config.cacheSizeInFrames == 0) { + pSRC->config.cacheSizeInFrames = DRMP3_SRC_CACHE_SIZE_IN_FRAMES; + } + + drmp3_src_cache_init(pSRC, &pSRC->cache); + return DRMP3_TRUE; +} + +drmp3_bool32 drmp3_src_set_input_sample_rate(drmp3_src* pSRC, drmp3_uint32 sampleRateIn) +{ + if (pSRC == NULL) { + return DRMP3_FALSE; + } + + /* Must have a sample rate of > 0. */ + if (sampleRateIn == 0) { + return DRMP3_FALSE; + } + + pSRC->config.sampleRateIn = sampleRateIn; + return DRMP3_TRUE; +} + +drmp3_bool32 drmp3_src_set_output_sample_rate(drmp3_src* pSRC, drmp3_uint32 sampleRateOut) +{ + if (pSRC == NULL) { + return DRMP3_FALSE; + } + + /* Must have a sample rate of > 0. */ + if (sampleRateOut == 0) { + return DRMP3_FALSE; + } + + pSRC->config.sampleRateOut = sampleRateOut; + return DRMP3_TRUE; +} + +drmp3_uint64 drmp3_src_read_frames_ex(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, drmp3_bool32 flush) +{ + drmp3_src_algorithm algorithm; + + if (pSRC == NULL || frameCount == 0 || pFramesOut == NULL) { + return 0; + } + + algorithm = pSRC->config.algorithm; + + /* Always use passthrough if the sample rates are the same. */ + if (pSRC->config.sampleRateIn == pSRC->config.sampleRateOut) { + algorithm = drmp3_src_algorithm_none; + } + + /* Could just use a function pointer instead of a switch for this... */ + switch (algorithm) + { + case drmp3_src_algorithm_none: return drmp3_src_read_frames_passthrough(pSRC, frameCount, pFramesOut, flush); + case drmp3_src_algorithm_linear: return drmp3_src_read_frames_linear(pSRC, frameCount, pFramesOut, flush); + default: return 0; + } +} + +drmp3_uint64 drmp3_src_read_frames(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut) +{ + return drmp3_src_read_frames_ex(pSRC, frameCount, pFramesOut, DRMP3_FALSE); +} + +drmp3_uint64 drmp3_src_read_frames_passthrough(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, drmp3_bool32 flush) +{ + drmp3_assert(pSRC != NULL); + drmp3_assert(frameCount > 0); + drmp3_assert(pFramesOut != NULL); + + (void)flush; /* Passthrough need not care about flushing. */ + return pSRC->onRead(pSRC, frameCount, pFramesOut, pSRC->pUserData); +} + +drmp3_uint64 drmp3_src_read_frames_linear(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, drmp3_bool32 flush) +{ + double factor; + drmp3_uint64 totalFramesRead; + + drmp3_assert(pSRC != NULL); + drmp3_assert(frameCount > 0); + drmp3_assert(pFramesOut != NULL); + + /* For linear SRC, the bin is only 2 frames: 1 prior, 1 future. */ + + /* Load the bin if necessary. */ + if (!pSRC->algo.linear.isPrevFramesLoaded) { + drmp3_uint64 framesRead = drmp3_src_cache_read_frames(&pSRC->cache, 1, pSRC->bin); + if (framesRead == 0) { + return 0; + } + pSRC->algo.linear.isPrevFramesLoaded = DRMP3_TRUE; + } + if (!pSRC->algo.linear.isNextFramesLoaded) { + drmp3_uint64 framesRead = drmp3_src_cache_read_frames(&pSRC->cache, 1, pSRC->bin + pSRC->config.channels); + if (framesRead == 0) { + return 0; + } + pSRC->algo.linear.isNextFramesLoaded = DRMP3_TRUE; + } + + factor = (double)pSRC->config.sampleRateIn / pSRC->config.sampleRateOut; + + totalFramesRead = 0; + while (frameCount > 0) { + drmp3_uint32 i; + drmp3_uint32 framesToReadFromClient; + + /* The bin is where the previous and next frames are located. */ + float* pPrevFrame = pSRC->bin; + float* pNextFrame = pSRC->bin + pSRC->config.channels; + + drmp3_blend_f32((float*)pFramesOut, pPrevFrame, pNextFrame, (float)pSRC->algo.linear.alpha, pSRC->config.channels); + + pSRC->algo.linear.alpha += factor; + + /* The new alpha value is how we determine whether or not we need to read fresh frames. */ + framesToReadFromClient = (drmp3_uint32)pSRC->algo.linear.alpha; + pSRC->algo.linear.alpha = pSRC->algo.linear.alpha - framesToReadFromClient; + + for (i = 0; i < framesToReadFromClient; ++i) { + drmp3_uint64 framesRead; + drmp3_uint32 j; + + for (j = 0; j < pSRC->config.channels; ++j) { + pPrevFrame[j] = pNextFrame[j]; + } + + framesRead = drmp3_src_cache_read_frames(&pSRC->cache, 1, pNextFrame); + if (framesRead == 0) { + drmp3_uint32 k; + for (k = 0; k < pSRC->config.channels; ++k) { + pNextFrame[k] = 0; + } + + if (pSRC->algo.linear.isNextFramesLoaded) { + pSRC->algo.linear.isNextFramesLoaded = DRMP3_FALSE; + } else { + if (flush) { + pSRC->algo.linear.isPrevFramesLoaded = DRMP3_FALSE; + } + } + + break; + } + } + + pFramesOut = (drmp3_uint8*)pFramesOut + (1 * pSRC->config.channels * sizeof(float)); + frameCount -= 1; + totalFramesRead += 1; + + /* If there's no frames available we need to get out of this loop. */ + if (!pSRC->algo.linear.isNextFramesLoaded && (!flush || !pSRC->algo.linear.isPrevFramesLoaded)) { + break; + } + } + + return totalFramesRead; +} + + +static size_t drmp3__on_read(drmp3* pMP3, void* pBufferOut, size_t bytesToRead) +{ + size_t bytesRead = pMP3->onRead(pMP3->pUserData, pBufferOut, bytesToRead); + pMP3->streamCursor += bytesRead; + return bytesRead; +} + +static drmp3_bool32 drmp3__on_seek(drmp3* pMP3, int offset, drmp3_seek_origin origin) +{ + drmp3_assert(offset >= 0); + + if (!pMP3->onSeek(pMP3->pUserData, offset, origin)) { + return DRMP3_FALSE; + } + + if (origin == drmp3_seek_origin_start) { + pMP3->streamCursor = (drmp3_uint64)offset; + } else { + pMP3->streamCursor += offset; + } + + return DRMP3_TRUE; +} + +static drmp3_bool32 drmp3__on_seek_64(drmp3* pMP3, drmp3_uint64 offset, drmp3_seek_origin origin) +{ + if (offset <= 0x7FFFFFFF) { + return drmp3__on_seek(pMP3, (int)offset, origin); + } + + + /* Getting here "offset" is too large for a 32-bit integer. We just keep seeking forward until we hit the offset. */ + if (!drmp3__on_seek(pMP3, 0x7FFFFFFF, drmp3_seek_origin_start)) { + return DRMP3_FALSE; + } + + offset -= 0x7FFFFFFF; + while (offset > 0) { + if (offset <= 0x7FFFFFFF) { + if (!drmp3__on_seek(pMP3, (int)offset, drmp3_seek_origin_current)) { + return DRMP3_FALSE; + } + offset = 0; + } else { + if (!drmp3__on_seek(pMP3, 0x7FFFFFFF, drmp3_seek_origin_current)) { + return DRMP3_FALSE; + } + offset -= 0x7FFFFFFF; + } + } + + return DRMP3_TRUE; +} + +static drmp3_uint32 drmp3_decode_next_frame_ex(drmp3* pMP3, drmp3d_sample_t* pPCMFrames, drmp3_bool32 discard); +static drmp3_uint32 drmp3_decode_next_frame(drmp3* pMP3); + +static drmp3_uint64 drmp3_read_src(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, void* pUserData) +{ + drmp3* pMP3 = (drmp3*)pUserData; + float* pFramesOutF = (float*)pFramesOut; + drmp3_uint64 totalFramesRead = 0; + + drmp3_assert(pMP3 != NULL); + drmp3_assert(pMP3->onRead != NULL); + + while (frameCount > 0) { + /* Read from the in-memory buffer first. */ + while (pMP3->pcmFramesRemainingInMP3Frame > 0 && frameCount > 0) { + drmp3d_sample_t* frames = (drmp3d_sample_t*)pMP3->pcmFrames; +#ifndef DR_MP3_FLOAT_OUTPUT + if (pMP3->mp3FrameChannels == 1) { + if (pMP3->channels == 1) { + /* Mono -> Mono. */ + pFramesOutF[0] = frames[pMP3->pcmFramesConsumedInMP3Frame] / 32768.0f; + } else { + /* Mono -> Stereo. */ + pFramesOutF[0] = frames[pMP3->pcmFramesConsumedInMP3Frame] / 32768.0f; + pFramesOutF[1] = frames[pMP3->pcmFramesConsumedInMP3Frame] / 32768.0f; + } + } else { + if (pMP3->channels == 1) { + /* Stereo -> Mono */ + float sample = 0; + sample += frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+0] / 32768.0f; + sample += frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+1] / 32768.0f; + pFramesOutF[0] = sample * 0.5f; + } else { + /* Stereo -> Stereo */ + pFramesOutF[0] = frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+0] / 32768.0f; + pFramesOutF[1] = frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+1] / 32768.0f; + } + } +#else + if (pMP3->mp3FrameChannels == 1) { + if (pMP3->channels == 1) { + /* Mono -> Mono. */ + pFramesOutF[0] = frames[pMP3->pcmFramesConsumedInMP3Frame]; + } else { + /* Mono -> Stereo. */ + pFramesOutF[0] = frames[pMP3->pcmFramesConsumedInMP3Frame]; + pFramesOutF[1] = frames[pMP3->pcmFramesConsumedInMP3Frame]; + } + } else { + if (pMP3->channels == 1) { + /* Stereo -> Mono */ + float sample = 0; + sample += frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+0]; + sample += frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+1]; + pFramesOutF[0] = sample * 0.5f; + } else { + /* Stereo -> Stereo */ + pFramesOutF[0] = frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+0]; + pFramesOutF[1] = frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+1]; + } + } +#endif + + pMP3->pcmFramesConsumedInMP3Frame += 1; + pMP3->pcmFramesRemainingInMP3Frame -= 1; + totalFramesRead += 1; + frameCount -= 1; + pFramesOutF += pSRC->config.channels; + } + + if (frameCount == 0) { + break; + } + + drmp3_assert(pMP3->pcmFramesRemainingInMP3Frame == 0); + + /* + At this point we have exhausted our in-memory buffer so we need to re-fill. Note that the sample rate may have changed + at this point which means we'll also need to update our sample rate conversion pipeline. + */ + if (drmp3_decode_next_frame(pMP3) == 0) { + break; + } + } + + return totalFramesRead; +} + +static drmp3_bool32 drmp3_init_src(drmp3* pMP3) +{ + drmp3_src_config srcConfig; + drmp3_zero_object(&srcConfig); + srcConfig.sampleRateIn = DR_MP3_DEFAULT_SAMPLE_RATE; + srcConfig.sampleRateOut = pMP3->sampleRate; + srcConfig.channels = pMP3->channels; + srcConfig.algorithm = drmp3_src_algorithm_linear; + if (!drmp3_src_init(&srcConfig, drmp3_read_src, pMP3, &pMP3->src)) { + drmp3_uninit(pMP3); + return DRMP3_FALSE; + } + + return DRMP3_TRUE; +} + +static drmp3_uint32 drmp3_decode_next_frame_ex(drmp3* pMP3, drmp3d_sample_t* pPCMFrames, drmp3_bool32 discard) +{ + drmp3_uint32 pcmFramesRead = 0; + + drmp3_assert(pMP3 != NULL); + drmp3_assert(pMP3->onRead != NULL); + + if (pMP3->atEnd) { + return 0; + } + + do { + drmp3dec_frame_info info; + size_t leftoverDataSize; + + /* minimp3 recommends doing data submission in 16K chunks. If we don't have at least 16K bytes available, get more. */ + if (pMP3->dataSize < DRMP3_DATA_CHUNK_SIZE) { + size_t bytesRead; + + if (pMP3->dataCapacity < DRMP3_DATA_CHUNK_SIZE) { + drmp3_uint8* pNewData; + + pMP3->dataCapacity = DRMP3_DATA_CHUNK_SIZE; + pNewData = (drmp3_uint8*)drmp3_realloc(pMP3->pData, pMP3->dataCapacity); + if (pNewData == NULL) { + return 0; /* Out of memory. */ + } + + pMP3->pData = pNewData; + } + + bytesRead = drmp3__on_read(pMP3, pMP3->pData + pMP3->dataSize, (pMP3->dataCapacity - pMP3->dataSize)); + if (bytesRead == 0) { + if (pMP3->dataSize == 0) { + pMP3->atEnd = DRMP3_TRUE; + return 0; /* No data. */ + } + } + + pMP3->dataSize += bytesRead; + } + + if (pMP3->dataSize > INT_MAX) { + pMP3->atEnd = DRMP3_TRUE; + return 0; /* File too big. */ + } + + pcmFramesRead = drmp3dec_decode_frame(&pMP3->decoder, pMP3->pData, (int)pMP3->dataSize, pPCMFrames, &info); /* <-- Safe size_t -> int conversion thanks to the check above. */ + + /* Consume the data. */ + leftoverDataSize = (pMP3->dataSize - (size_t)info.frame_bytes); + if (info.frame_bytes > 0) { + memmove(pMP3->pData, pMP3->pData + info.frame_bytes, leftoverDataSize); + pMP3->dataSize = leftoverDataSize; + } + + /* + pcmFramesRead will be equal to 0 if decoding failed. If it is zero and info.frame_bytes > 0 then we have successfully + decoded the frame. A special case is if we are wanting to discard the frame, in which case we return successfully. + */ + if (pcmFramesRead > 0 || (info.frame_bytes > 0 && discard)) { + pcmFramesRead = drmp3_hdr_frame_samples(pMP3->decoder.header); + pMP3->pcmFramesConsumedInMP3Frame = 0; + pMP3->pcmFramesRemainingInMP3Frame = pcmFramesRead; + pMP3->mp3FrameChannels = info.channels; + pMP3->mp3FrameSampleRate = info.hz; + + /* We need to initialize the resampler if we don't yet have the channel count or sample rate. */ + if (pMP3->channels == 0 || pMP3->sampleRate == 0) { + if (pMP3->channels == 0) { + pMP3->channels = info.channels; + } + if (pMP3->sampleRate == 0) { + pMP3->sampleRate = info.hz; + } + drmp3_init_src(pMP3); + } + + drmp3_src_set_input_sample_rate(&pMP3->src, pMP3->mp3FrameSampleRate); + break; + } else if (info.frame_bytes == 0) { + size_t bytesRead; + + /* Need more data. minimp3 recommends doing data submission in 16K chunks. */ + if (pMP3->dataCapacity == pMP3->dataSize) { + drmp3_uint8* pNewData; + + /* No room. Expand. */ + pMP3->dataCapacity += DRMP3_DATA_CHUNK_SIZE; + pNewData = (drmp3_uint8*)drmp3_realloc(pMP3->pData, pMP3->dataCapacity); + if (pNewData == NULL) { + return 0; /* Out of memory. */ + } + + pMP3->pData = pNewData; + } + + /* Fill in a chunk. */ + bytesRead = drmp3__on_read(pMP3, pMP3->pData + pMP3->dataSize, (pMP3->dataCapacity - pMP3->dataSize)); + if (bytesRead == 0) { + pMP3->atEnd = DRMP3_TRUE; + return 0; /* Error reading more data. */ + } + + pMP3->dataSize += bytesRead; + } + } while (DRMP3_TRUE); + + return pcmFramesRead; +} + +static drmp3_uint32 drmp3_decode_next_frame(drmp3* pMP3) +{ + drmp3_assert(pMP3 != NULL); + return drmp3_decode_next_frame_ex(pMP3, (drmp3d_sample_t*)pMP3->pcmFrames, DRMP3_FALSE); +} + +#if 0 +static drmp3_uint32 drmp3_seek_next_frame(drmp3* pMP3) +{ + drmp3_uint32 pcmFrameCount; + + drmp3_assert(pMP3 != NULL); + + pcmFrameCount = drmp3_decode_next_frame_ex(pMP3, NULL); + if (pcmFrameCount == 0) { + return 0; + } + + /* We have essentially just skipped past the frame, so just set the remaining samples to 0. */ + pMP3->currentPCMFrame += pcmFrameCount; + pMP3->pcmFramesConsumedInMP3Frame = pcmFrameCount; + pMP3->pcmFramesRemainingInMP3Frame = 0; + + return pcmFrameCount; +} +#endif + +drmp3_bool32 drmp3_init_internal(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, const drmp3_config* pConfig) +{ + drmp3_config config; + + drmp3_assert(pMP3 != NULL); + drmp3_assert(onRead != NULL); + + /* This function assumes the output object has already been reset to 0. Do not do that here, otherwise things will break. */ + drmp3dec_init(&pMP3->decoder); + + /* The config can be null in which case we use defaults. */ + if (pConfig != NULL) { + config = *pConfig; + } else { + drmp3_zero_object(&config); + } + + pMP3->channels = config.outputChannels; + + /* Cannot have more than 2 channels. */ + if (pMP3->channels > 2) { + pMP3->channels = 2; + } + + pMP3->sampleRate = config.outputSampleRate; + + pMP3->onRead = onRead; + pMP3->onSeek = onSeek; + pMP3->pUserData = pUserData; + + /* + We need a sample rate converter for converting the sample rate from the MP3 frames to the requested output sample rate. Note that if + we don't yet know the channel count or sample rate we defer this until the first frame is read. + */ + if (pMP3->channels != 0 && pMP3->sampleRate != 0) { + drmp3_init_src(pMP3); + } + + /* Decode the first frame to confirm that it is indeed a valid MP3 stream. */ + if (!drmp3_decode_next_frame(pMP3)) { + drmp3_uninit(pMP3); + return DRMP3_FALSE; /* Not a valid MP3 stream. */ + } + + return DRMP3_TRUE; +} + +drmp3_bool32 drmp3_init(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, const drmp3_config* pConfig) +{ + if (pMP3 == NULL || onRead == NULL) { + return DRMP3_FALSE; + } + + drmp3_zero_object(pMP3); + return drmp3_init_internal(pMP3, onRead, onSeek, pUserData, pConfig); +} + + +static size_t drmp3__on_read_memory(void* pUserData, void* pBufferOut, size_t bytesToRead) +{ + drmp3* pMP3 = (drmp3*)pUserData; + size_t bytesRemaining; + + drmp3_assert(pMP3 != NULL); + drmp3_assert(pMP3->memory.dataSize >= pMP3->memory.currentReadPos); + + bytesRemaining = pMP3->memory.dataSize - pMP3->memory.currentReadPos; + if (bytesToRead > bytesRemaining) { + bytesToRead = bytesRemaining; + } + + if (bytesToRead > 0) { + drmp3_copy_memory(pBufferOut, pMP3->memory.pData + pMP3->memory.currentReadPos, bytesToRead); + pMP3->memory.currentReadPos += bytesToRead; + } + + return bytesToRead; +} + +static drmp3_bool32 drmp3__on_seek_memory(void* pUserData, int byteOffset, drmp3_seek_origin origin) +{ + drmp3* pMP3 = (drmp3*)pUserData; + + drmp3_assert(pMP3 != NULL); + + if (origin == drmp3_seek_origin_current) { + if (byteOffset > 0) { + if (pMP3->memory.currentReadPos + byteOffset > pMP3->memory.dataSize) { + byteOffset = (int)(pMP3->memory.dataSize - pMP3->memory.currentReadPos); /* Trying to seek too far forward. */ + } + } else { + if (pMP3->memory.currentReadPos < (size_t)-byteOffset) { + byteOffset = -(int)pMP3->memory.currentReadPos; /* Trying to seek too far backwards. */ + } + } + + /* This will never underflow thanks to the clamps above. */ + pMP3->memory.currentReadPos += byteOffset; + } else { + if ((drmp3_uint32)byteOffset <= pMP3->memory.dataSize) { + pMP3->memory.currentReadPos = byteOffset; + } else { + pMP3->memory.currentReadPos = pMP3->memory.dataSize; /* Trying to seek too far forward. */ + } + } + + return DRMP3_TRUE; +} + +drmp3_bool32 drmp3_init_memory(drmp3* pMP3, const void* pData, size_t dataSize, const drmp3_config* pConfig) +{ + if (pMP3 == NULL) { + return DRMP3_FALSE; + } + + drmp3_zero_object(pMP3); + + if (pData == NULL || dataSize == 0) { + return DRMP3_FALSE; + } + + pMP3->memory.pData = (const drmp3_uint8*)pData; + pMP3->memory.dataSize = dataSize; + pMP3->memory.currentReadPos = 0; + + return drmp3_init_internal(pMP3, drmp3__on_read_memory, drmp3__on_seek_memory, pMP3, pConfig); +} + + +#ifndef DR_MP3_NO_STDIO +#include <stdio.h> + +static size_t drmp3__on_read_stdio(void* pUserData, void* pBufferOut, size_t bytesToRead) +{ + return fread(pBufferOut, 1, bytesToRead, (FILE*)pUserData); +} + +static drmp3_bool32 drmp3__on_seek_stdio(void* pUserData, int offset, drmp3_seek_origin origin) +{ + return fseek((FILE*)pUserData, offset, (origin == drmp3_seek_origin_current) ? SEEK_CUR : SEEK_SET) == 0; +} + +drmp3_bool32 drmp3_init_file(drmp3* pMP3, const char* filePath, const drmp3_config* pConfig) +{ + FILE* pFile; +#if defined(_MSC_VER) && _MSC_VER >= 1400 + if (fopen_s(&pFile, filePath, "rb") != 0) { + return DRMP3_FALSE; + } +#else + pFile = fopen(filePath, "rb"); + if (pFile == NULL) { + return DRMP3_FALSE; + } +#endif + + return drmp3_init(pMP3, drmp3__on_read_stdio, drmp3__on_seek_stdio, (void*)pFile, pConfig); +} +#endif + +void drmp3_uninit(drmp3* pMP3) +{ + if (pMP3 == NULL) { + return; + } + +#ifndef DR_MP3_NO_STDIO + if (pMP3->onRead == drmp3__on_read_stdio) { + fclose((FILE*)pMP3->pUserData); + } +#endif + + drmp3_free(pMP3->pData); +} + +drmp3_uint64 drmp3_read_pcm_frames_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut) +{ + drmp3_uint64 totalFramesRead = 0; + + if (pMP3 == NULL || pMP3->onRead == NULL) { + return 0; + } + + if (pBufferOut == NULL) { + float temp[4096]; + while (framesToRead > 0) { + drmp3_uint64 framesJustRead; + drmp3_uint64 framesToReadRightNow = sizeof(temp)/sizeof(temp[0]) / pMP3->channels; + if (framesToReadRightNow > framesToRead) { + framesToReadRightNow = framesToRead; + } + + framesJustRead = drmp3_read_pcm_frames_f32(pMP3, framesToReadRightNow, temp); + if (framesJustRead == 0) { + break; + } + + framesToRead -= framesJustRead; + totalFramesRead += framesJustRead; + } + } else { + totalFramesRead = drmp3_src_read_frames_ex(&pMP3->src, framesToRead, pBufferOut, DRMP3_TRUE); + pMP3->currentPCMFrame += totalFramesRead; + } + + return totalFramesRead; +} + +drmp3_uint64 drmp3_read_pcm_frames_s16(drmp3* pMP3, drmp3_uint64 framesToRead, drmp3_int16* pBufferOut) +{ + float tempF32[4096]; + drmp3_uint64 pcmFramesJustRead; + drmp3_uint64 totalPCMFramesRead = 0; + + if (pMP3 == NULL || pMP3->onRead == NULL) { + return 0; + } + + /* Naive implementation: read into a temp f32 buffer, then convert. */ + for (;;) { + drmp3_uint64 pcmFramesToReadThisIteration = (framesToRead - totalPCMFramesRead); + if (pcmFramesToReadThisIteration > drmp3_countof(tempF32)/pMP3->channels) { + pcmFramesToReadThisIteration = drmp3_countof(tempF32)/pMP3->channels; + } + + pcmFramesJustRead = drmp3_read_pcm_frames_f32(pMP3, pcmFramesToReadThisIteration, tempF32); + if (pcmFramesJustRead == 0) { + break; + } + + drmp3dec_f32_to_s16(tempF32, pBufferOut, (int)(pcmFramesJustRead * pMP3->channels)); /* <-- Safe cast since pcmFramesJustRead will be clamped based on the size of tempF32 which is always small. */ + pBufferOut += pcmFramesJustRead * pMP3->channels; + + totalPCMFramesRead += pcmFramesJustRead; + + if (pcmFramesJustRead < pcmFramesToReadThisIteration) { + break; + } + } + + return totalPCMFramesRead; +} + +void drmp3_reset(drmp3* pMP3) +{ + drmp3_assert(pMP3 != NULL); + + pMP3->pcmFramesConsumedInMP3Frame = 0; + pMP3->pcmFramesRemainingInMP3Frame = 0; + pMP3->currentPCMFrame = 0; + pMP3->dataSize = 0; + pMP3->atEnd = DRMP3_FALSE; + pMP3->src.bin[0] = 0; + pMP3->src.bin[1] = 0; + pMP3->src.bin[2] = 0; + pMP3->src.bin[3] = 0; + pMP3->src.cache.cachedFrameCount = 0; + pMP3->src.cache.iNextFrame = 0; + pMP3->src.algo.linear.alpha = 0; + pMP3->src.algo.linear.isNextFramesLoaded = 0; + pMP3->src.algo.linear.isPrevFramesLoaded = 0; + drmp3dec_init(&pMP3->decoder); +} + +drmp3_bool32 drmp3_seek_to_start_of_stream(drmp3* pMP3) +{ + drmp3_assert(pMP3 != NULL); + drmp3_assert(pMP3->onSeek != NULL); + + /* Seek to the start of the stream to begin with. */ + if (!drmp3__on_seek(pMP3, 0, drmp3_seek_origin_start)) { + return DRMP3_FALSE; + } + + /* Clear any cached data. */ + drmp3_reset(pMP3); + return DRMP3_TRUE; +} + +float drmp3_get_cached_pcm_frame_count_from_src(drmp3* pMP3) +{ + return (pMP3->src.cache.cachedFrameCount - pMP3->src.cache.iNextFrame) + (float)pMP3->src.algo.linear.alpha; +} + +float drmp3_get_pcm_frames_remaining_in_mp3_frame(drmp3* pMP3) +{ + float factor = (float)pMP3->src.config.sampleRateOut / (float)pMP3->src.config.sampleRateIn; + float frameCountPreSRC = drmp3_get_cached_pcm_frame_count_from_src(pMP3) + pMP3->pcmFramesRemainingInMP3Frame; + return frameCountPreSRC * factor; +} + +/* +NOTE ON SEEKING +=============== +The seeking code below is a complete mess and is broken for cases when the sample rate changes. The problem +is with the resampling and the crappy resampler used by dr_mp3. What needs to happen is the following: + +1) The resampler needs to be replaced. +2) The resampler has state which needs to be updated whenever an MP3 frame is decoded outside of + drmp3_read_pcm_frames_f32(). The resampler needs an API to "flush" some imaginary input so that it's + state is updated accordingly. +*/ +drmp3_bool32 drmp3_seek_forward_by_pcm_frames__brute_force(drmp3* pMP3, drmp3_uint64 frameOffset) +{ + drmp3_uint64 framesRead; + +#if 0 + /* + MP3 is a bit annoying when it comes to seeking because of the bit reservoir. It basically means that an MP3 frame can possibly + depend on some of the data of prior frames. This means it's not as simple as seeking to the first byte of the MP3 frame that + contains the sample because that MP3 frame will need the data from the previous MP3 frame (which we just seeked past!). To + resolve this we seek past a number of MP3 frames up to a point, and then read-and-discard the remainder. + */ + drmp3_uint64 maxFramesToReadAndDiscard = (drmp3_uint64)(DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME * 3 * ((float)pMP3->src.config.sampleRateOut / (float)pMP3->src.config.sampleRateIn)); + + /* Now get rid of leading whole frames. */ + while (frameOffset > maxFramesToReadAndDiscard) { + float pcmFramesRemainingInCurrentMP3FrameF = drmp3_get_pcm_frames_remaining_in_mp3_frame(pMP3); + drmp3_uint32 pcmFramesRemainingInCurrentMP3Frame = (drmp3_uint32)pcmFramesRemainingInCurrentMP3FrameF; + if (frameOffset > pcmFramesRemainingInCurrentMP3Frame) { + frameOffset -= pcmFramesRemainingInCurrentMP3Frame; + pMP3->currentPCMFrame += pcmFramesRemainingInCurrentMP3Frame; + pMP3->pcmFramesConsumedInMP3Frame += pMP3->pcmFramesRemainingInMP3Frame; + pMP3->pcmFramesRemainingInMP3Frame = 0; + } else { + break; + } + + drmp3_uint32 pcmFrameCount = drmp3_decode_next_frame_ex(pMP3, pMP3->pcmFrames, DRMP3_FALSE); + if (pcmFrameCount == 0) { + break; + } + } + + /* The last step is to read-and-discard any remaining PCM frames to make it sample-exact. */ + framesRead = drmp3_read_pcm_frames_f32(pMP3, frameOffset, NULL); + if (framesRead != frameOffset) { + return DRMP3_FALSE; + } +#else + /* Just using a dumb read-and-discard for now pending updates to the resampler. */ + framesRead = drmp3_read_pcm_frames_f32(pMP3, frameOffset, NULL); + if (framesRead != frameOffset) { + return DRMP3_FALSE; + } +#endif + + return DRMP3_TRUE; +} + +drmp3_bool32 drmp3_seek_to_pcm_frame__brute_force(drmp3* pMP3, drmp3_uint64 frameIndex) +{ + drmp3_assert(pMP3 != NULL); + + if (frameIndex == pMP3->currentPCMFrame) { + return DRMP3_TRUE; + } + + /* + If we're moving foward we just read from where we're at. Otherwise we need to move back to the start of + the stream and read from the beginning. + */ + if (frameIndex < pMP3->currentPCMFrame) { + /* Moving backward. Move to the start of the stream and then move forward. */ + if (!drmp3_seek_to_start_of_stream(pMP3)) { + return DRMP3_FALSE; + } + } + + drmp3_assert(frameIndex >= pMP3->currentPCMFrame); + return drmp3_seek_forward_by_pcm_frames__brute_force(pMP3, (frameIndex - pMP3->currentPCMFrame)); +} + +drmp3_bool32 drmp3_find_closest_seek_point(drmp3* pMP3, drmp3_uint64 frameIndex, drmp3_uint32* pSeekPointIndex) +{ + drmp3_uint32 iSeekPoint; + + drmp3_assert(pSeekPointIndex != NULL); + + *pSeekPointIndex = 0; + + if (frameIndex < pMP3->pSeekPoints[0].pcmFrameIndex) { + return DRMP3_FALSE; + } + + /* Linear search for simplicity to begin with while I'm getting this thing working. Once it's all working change this to a binary search. */ + for (iSeekPoint = 0; iSeekPoint < pMP3->seekPointCount; ++iSeekPoint) { + if (pMP3->pSeekPoints[iSeekPoint].pcmFrameIndex > frameIndex) { + break; /* Found it. */ + } + + *pSeekPointIndex = iSeekPoint; + } + + return DRMP3_TRUE; +} + +drmp3_bool32 drmp3_seek_to_pcm_frame__seek_table(drmp3* pMP3, drmp3_uint64 frameIndex) +{ + drmp3_seek_point seekPoint; + drmp3_uint32 priorSeekPointIndex; + drmp3_uint16 iMP3Frame; + drmp3_uint64 leftoverFrames; + + drmp3_assert(pMP3 != NULL); + drmp3_assert(pMP3->pSeekPoints != NULL); + drmp3_assert(pMP3->seekPointCount > 0); + + /* If there is no prior seekpoint it means the target PCM frame comes before the first seek point. Just assume a seekpoint at the start of the file in this case. */ + if (drmp3_find_closest_seek_point(pMP3, frameIndex, &priorSeekPointIndex)) { + seekPoint = pMP3->pSeekPoints[priorSeekPointIndex]; + } else { + seekPoint.seekPosInBytes = 0; + seekPoint.pcmFrameIndex = 0; + seekPoint.mp3FramesToDiscard = 0; + seekPoint.pcmFramesToDiscard = 0; + } + + /* First thing to do is seek to the first byte of the relevant MP3 frame. */ + if (!drmp3__on_seek_64(pMP3, seekPoint.seekPosInBytes, drmp3_seek_origin_start)) { + return DRMP3_FALSE; /* Failed to seek. */ + } + + /* Clear any cached data. */ + drmp3_reset(pMP3); + + /* Whole MP3 frames need to be discarded first. */ + for (iMP3Frame = 0; iMP3Frame < seekPoint.mp3FramesToDiscard; ++iMP3Frame) { + drmp3_uint32 pcmFramesReadPreSRC; + drmp3d_sample_t* pPCMFrames; + + /* Pass in non-null for the last frame because we want to ensure the sample rate converter is preloaded correctly. */ + pPCMFrames = NULL; + if (iMP3Frame == seekPoint.mp3FramesToDiscard-1) { + pPCMFrames = (drmp3d_sample_t*)pMP3->pcmFrames; + } + + /* We first need to decode the next frame, and then we need to flush the resampler. */ + pcmFramesReadPreSRC = drmp3_decode_next_frame_ex(pMP3, pPCMFrames, DRMP3_TRUE); + if (pcmFramesReadPreSRC == 0) { + return DRMP3_FALSE; + } + } + + /* We seeked to an MP3 frame in the raw stream so we need to make sure the current PCM frame is set correctly. */ + pMP3->currentPCMFrame = seekPoint.pcmFrameIndex - seekPoint.pcmFramesToDiscard; + + /* + Update resampler. This is wrong. Need to instead update it on a per MP3 frame basis. Also broken for cases when + the sample rate is being reduced in my testing. Should work fine when the input and output sample rate is the same + or a clean multiple. + */ + pMP3->src.algo.linear.alpha = (drmp3_int64)pMP3->currentPCMFrame * ((double)pMP3->src.config.sampleRateIn / pMP3->src.config.sampleRateOut); /* <-- Cast to int64 is required for VC6. */ + pMP3->src.algo.linear.alpha = pMP3->src.algo.linear.alpha - (drmp3_uint32)(pMP3->src.algo.linear.alpha); + if (pMP3->src.algo.linear.alpha > 0) { + pMP3->src.algo.linear.isPrevFramesLoaded = 1; + } + + /* + Now at this point we can follow the same process as the brute force technique where we just skip over unnecessary MP3 frames and then + read-and-discard at least 2 whole MP3 frames. + */ + leftoverFrames = frameIndex - pMP3->currentPCMFrame; + return drmp3_seek_forward_by_pcm_frames__brute_force(pMP3, leftoverFrames); +} + +drmp3_bool32 drmp3_seek_to_pcm_frame(drmp3* pMP3, drmp3_uint64 frameIndex) +{ + if (pMP3 == NULL || pMP3->onSeek == NULL) { + return DRMP3_FALSE; + } + + if (frameIndex == 0) { + return drmp3_seek_to_start_of_stream(pMP3); + } + + /* Use the seek table if we have one. */ + if (pMP3->pSeekPoints != NULL && pMP3->seekPointCount > 0) { + return drmp3_seek_to_pcm_frame__seek_table(pMP3, frameIndex); + } else { + return drmp3_seek_to_pcm_frame__brute_force(pMP3, frameIndex); + } +} + +drmp3_bool32 drmp3_get_mp3_and_pcm_frame_count(drmp3* pMP3, drmp3_uint64* pMP3FrameCount, drmp3_uint64* pPCMFrameCount) +{ + drmp3_uint64 currentPCMFrame; + drmp3_uint64 totalPCMFrameCount; + drmp3_uint64 totalMP3FrameCount; + float totalPCMFrameCountFractionalPart; + + if (pMP3 == NULL) { + return DRMP3_FALSE; + } + + /* + The way this works is we move back to the start of the stream, iterate over each MP3 frame and calculate the frame count based + on our output sample rate, the seek back to the PCM frame we were sitting on before calling this function. + */ + + /* The stream must support seeking for this to work. */ + if (pMP3->onSeek == NULL) { + return DRMP3_FALSE; + } + + /* We'll need to seek back to where we were, so grab the PCM frame we're currently sitting on so we can restore later. */ + currentPCMFrame = pMP3->currentPCMFrame; + + if (!drmp3_seek_to_start_of_stream(pMP3)) { + return DRMP3_FALSE; + } + + totalPCMFrameCount = 0; + totalMP3FrameCount = 0; + + totalPCMFrameCountFractionalPart = 0; /* <-- With resampling there will be a fractional part to each MP3 frame that we need to accumulate. */ + for (;;) { + drmp3_uint32 pcmFramesInCurrentMP3FrameIn; + float srcRatio; + float pcmFramesInCurrentMP3FrameOutF; + drmp3_uint32 pcmFramesInCurrentMP3FrameOut; + + pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL, DRMP3_FALSE); + if (pcmFramesInCurrentMP3FrameIn == 0) { + break; + } + + srcRatio = (float)pMP3->mp3FrameSampleRate / (float)pMP3->sampleRate; + drmp3_assert(srcRatio > 0); + + pcmFramesInCurrentMP3FrameOutF = totalPCMFrameCountFractionalPart + (pcmFramesInCurrentMP3FrameIn / srcRatio); + pcmFramesInCurrentMP3FrameOut = (drmp3_uint32)pcmFramesInCurrentMP3FrameOutF; + totalPCMFrameCountFractionalPart = pcmFramesInCurrentMP3FrameOutF - pcmFramesInCurrentMP3FrameOut; + totalPCMFrameCount += pcmFramesInCurrentMP3FrameOut; + totalMP3FrameCount += 1; + } + + /* Finally, we need to seek back to where we were. */ + if (!drmp3_seek_to_start_of_stream(pMP3)) { + return DRMP3_FALSE; + } + + if (!drmp3_seek_to_pcm_frame(pMP3, currentPCMFrame)) { + return DRMP3_FALSE; + } + + if (pMP3FrameCount != NULL) { + *pMP3FrameCount = totalMP3FrameCount; + } + if (pPCMFrameCount != NULL) { + *pPCMFrameCount = totalPCMFrameCount; + } + + return DRMP3_TRUE; +} + +drmp3_uint64 drmp3_get_pcm_frame_count(drmp3* pMP3) +{ + drmp3_uint64 totalPCMFrameCount; + if (!drmp3_get_mp3_and_pcm_frame_count(pMP3, NULL, &totalPCMFrameCount)) { + return 0; + } + + return totalPCMFrameCount; +} + +drmp3_uint64 drmp3_get_mp3_frame_count(drmp3* pMP3) +{ + drmp3_uint64 totalMP3FrameCount; + if (!drmp3_get_mp3_and_pcm_frame_count(pMP3, &totalMP3FrameCount, NULL)) { + return 0; + } + + return totalMP3FrameCount; +} + +void drmp3__accumulate_running_pcm_frame_count(drmp3* pMP3, drmp3_uint32 pcmFrameCountIn, drmp3_uint64* pRunningPCMFrameCount, float* pRunningPCMFrameCountFractionalPart) +{ + float srcRatio; + float pcmFrameCountOutF; + drmp3_uint32 pcmFrameCountOut; + + srcRatio = (float)pMP3->mp3FrameSampleRate / (float)pMP3->sampleRate; + drmp3_assert(srcRatio > 0); + + pcmFrameCountOutF = *pRunningPCMFrameCountFractionalPart + (pcmFrameCountIn / srcRatio); + pcmFrameCountOut = (drmp3_uint32)pcmFrameCountOutF; + *pRunningPCMFrameCountFractionalPart = pcmFrameCountOutF - pcmFrameCountOut; + *pRunningPCMFrameCount += pcmFrameCountOut; +} + +typedef struct +{ + drmp3_uint64 bytePos; + drmp3_uint64 pcmFrameIndex; /* <-- After sample rate conversion. */ +} drmp3__seeking_mp3_frame_info; + +drmp3_bool32 drmp3_calculate_seek_points(drmp3* pMP3, drmp3_uint32* pSeekPointCount, drmp3_seek_point* pSeekPoints) +{ + drmp3_uint32 seekPointCount; + drmp3_uint64 currentPCMFrame; + drmp3_uint64 totalMP3FrameCount; + drmp3_uint64 totalPCMFrameCount; + + if (pMP3 == NULL || pSeekPointCount == NULL || pSeekPoints == NULL) { + return DRMP3_FALSE; /* Invalid args. */ + } + + seekPointCount = *pSeekPointCount; + if (seekPointCount == 0) { + return DRMP3_FALSE; /* The client has requested no seek points. Consider this to be invalid arguments since the client has probably not intended this. */ + } + + /* We'll need to seek back to the current sample after calculating the seekpoints so we need to go ahead and grab the current location at the top. */ + currentPCMFrame = pMP3->currentPCMFrame; + + /* We never do more than the total number of MP3 frames and we limit it to 32-bits. */ + if (!drmp3_get_mp3_and_pcm_frame_count(pMP3, &totalMP3FrameCount, &totalPCMFrameCount)) { + return DRMP3_FALSE; + } + + /* If there's less than DRMP3_SEEK_LEADING_MP3_FRAMES+1 frames we just report 1 seek point which will be the very start of the stream. */ + if (totalMP3FrameCount < DRMP3_SEEK_LEADING_MP3_FRAMES+1) { + seekPointCount = 1; + pSeekPoints[0].seekPosInBytes = 0; + pSeekPoints[0].pcmFrameIndex = 0; + pSeekPoints[0].mp3FramesToDiscard = 0; + pSeekPoints[0].pcmFramesToDiscard = 0; + } else { + drmp3_uint64 pcmFramesBetweenSeekPoints; + drmp3__seeking_mp3_frame_info mp3FrameInfo[DRMP3_SEEK_LEADING_MP3_FRAMES+1]; + drmp3_uint64 runningPCMFrameCount = 0; + float runningPCMFrameCountFractionalPart = 0; + drmp3_uint64 nextTargetPCMFrame; + drmp3_uint32 iMP3Frame; + drmp3_uint32 iSeekPoint; + + if (seekPointCount > totalMP3FrameCount-1) { + seekPointCount = (drmp3_uint32)totalMP3FrameCount-1; + } + + pcmFramesBetweenSeekPoints = totalPCMFrameCount / (seekPointCount+1); + + /* + Here is where we actually calculate the seek points. We need to start by moving the start of the stream. We then enumerate over each + MP3 frame. + */ + if (!drmp3_seek_to_start_of_stream(pMP3)) { + return DRMP3_FALSE; + } + + /* + We need to cache the byte positions of the previous MP3 frames. As a new MP3 frame is iterated, we cycle the byte positions in this + array. The value in the first item in this array is the byte position that will be reported in the next seek point. + */ + + /* We need to initialize the array of MP3 byte positions for the leading MP3 frames. */ + for (iMP3Frame = 0; iMP3Frame < DRMP3_SEEK_LEADING_MP3_FRAMES+1; ++iMP3Frame) { + drmp3_uint32 pcmFramesInCurrentMP3FrameIn; + + /* The byte position of the next frame will be the stream's cursor position, minus whatever is sitting in the buffer. */ + drmp3_assert(pMP3->streamCursor >= pMP3->dataSize); + mp3FrameInfo[iMP3Frame].bytePos = pMP3->streamCursor - pMP3->dataSize; + mp3FrameInfo[iMP3Frame].pcmFrameIndex = runningPCMFrameCount; + + /* We need to get information about this frame so we can know how many samples it contained. */ + pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL, DRMP3_FALSE); + if (pcmFramesInCurrentMP3FrameIn == 0) { + return DRMP3_FALSE; /* This should never happen. */ + } + + drmp3__accumulate_running_pcm_frame_count(pMP3, pcmFramesInCurrentMP3FrameIn, &runningPCMFrameCount, &runningPCMFrameCountFractionalPart); + } + + /* + At this point we will have extracted the byte positions of the leading MP3 frames. We can now start iterating over each seek point and + calculate them. + */ + nextTargetPCMFrame = 0; + for (iSeekPoint = 0; iSeekPoint < seekPointCount; ++iSeekPoint) { + nextTargetPCMFrame += pcmFramesBetweenSeekPoints; + + for (;;) { + if (nextTargetPCMFrame < runningPCMFrameCount) { + /* The next seek point is in the current MP3 frame. */ + pSeekPoints[iSeekPoint].seekPosInBytes = mp3FrameInfo[0].bytePos; + pSeekPoints[iSeekPoint].pcmFrameIndex = nextTargetPCMFrame; + pSeekPoints[iSeekPoint].mp3FramesToDiscard = DRMP3_SEEK_LEADING_MP3_FRAMES; + pSeekPoints[iSeekPoint].pcmFramesToDiscard = (drmp3_uint16)(nextTargetPCMFrame - mp3FrameInfo[DRMP3_SEEK_LEADING_MP3_FRAMES-1].pcmFrameIndex); + break; + } else { + size_t i; + drmp3_uint32 pcmFramesInCurrentMP3FrameIn; + + /* + The next seek point is not in the current MP3 frame, so continue on to the next one. The first thing to do is cycle the cached + MP3 frame info. + */ + for (i = 0; i < drmp3_countof(mp3FrameInfo)-1; ++i) { + mp3FrameInfo[i] = mp3FrameInfo[i+1]; + } + + /* Cache previous MP3 frame info. */ + mp3FrameInfo[drmp3_countof(mp3FrameInfo)-1].bytePos = pMP3->streamCursor - pMP3->dataSize; + mp3FrameInfo[drmp3_countof(mp3FrameInfo)-1].pcmFrameIndex = runningPCMFrameCount; + + /* + Go to the next MP3 frame. This shouldn't ever fail, but just in case it does we just set the seek point and break. If it happens, it + should only ever do it for the last seek point. + */ + pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL, DRMP3_TRUE); + if (pcmFramesInCurrentMP3FrameIn == 0) { + pSeekPoints[iSeekPoint].seekPosInBytes = mp3FrameInfo[0].bytePos; + pSeekPoints[iSeekPoint].pcmFrameIndex = nextTargetPCMFrame; + pSeekPoints[iSeekPoint].mp3FramesToDiscard = DRMP3_SEEK_LEADING_MP3_FRAMES; + pSeekPoints[iSeekPoint].pcmFramesToDiscard = (drmp3_uint16)(nextTargetPCMFrame - mp3FrameInfo[DRMP3_SEEK_LEADING_MP3_FRAMES-1].pcmFrameIndex); + break; + } + + drmp3__accumulate_running_pcm_frame_count(pMP3, pcmFramesInCurrentMP3FrameIn, &runningPCMFrameCount, &runningPCMFrameCountFractionalPart); + } + } + } + + /* Finally, we need to seek back to where we were. */ + if (!drmp3_seek_to_start_of_stream(pMP3)) { + return DRMP3_FALSE; + } + if (!drmp3_seek_to_pcm_frame(pMP3, currentPCMFrame)) { + return DRMP3_FALSE; + } + } + + *pSeekPointCount = seekPointCount; + return DRMP3_TRUE; +} + +drmp3_bool32 drmp3_bind_seek_table(drmp3* pMP3, drmp3_uint32 seekPointCount, drmp3_seek_point* pSeekPoints) +{ + if (pMP3 == NULL) { + return DRMP3_FALSE; + } + + if (seekPointCount == 0 || pSeekPoints == NULL) { + /* Unbinding. */ + pMP3->seekPointCount = 0; + pMP3->pSeekPoints = NULL; + } else { + /* Binding. */ + pMP3->seekPointCount = seekPointCount; + pMP3->pSeekPoints = pSeekPoints; + } + + return DRMP3_TRUE; +} + + +float* drmp3__full_read_and_close_f32(drmp3* pMP3, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) +{ + drmp3_uint64 totalFramesRead = 0; + drmp3_uint64 framesCapacity = 0; + float* pFrames = NULL; + float temp[4096]; + + drmp3_assert(pMP3 != NULL); + + for (;;) { + drmp3_uint64 framesToReadRightNow = drmp3_countof(temp) / pMP3->channels; + drmp3_uint64 framesJustRead = drmp3_read_pcm_frames_f32(pMP3, framesToReadRightNow, temp); + if (framesJustRead == 0) { + break; + } + + /* Reallocate the output buffer if there's not enough room. */ + if (framesCapacity < totalFramesRead + framesJustRead) { + drmp3_uint64 newFramesBufferSize; + float* pNewFrames; + + framesCapacity *= 2; + if (framesCapacity < totalFramesRead + framesJustRead) { + framesCapacity = totalFramesRead + framesJustRead; + } + + newFramesBufferSize = framesCapacity*pMP3->channels*sizeof(float); + if (newFramesBufferSize > DRMP3_SIZE_MAX) { + break; + } + + pNewFrames = (float*)drmp3_realloc(pFrames, (size_t)newFramesBufferSize); + if (pNewFrames == NULL) { + drmp3_free(pFrames); + break; + } + + pFrames = pNewFrames; + } + + drmp3_copy_memory(pFrames + totalFramesRead*pMP3->channels, temp, (size_t)(framesJustRead*pMP3->channels*sizeof(float))); + totalFramesRead += framesJustRead; + + /* If the number of frames we asked for is less that what we actually read it means we've reached the end. */ + if (framesJustRead != framesToReadRightNow) { + break; + } + } + + if (pConfig != NULL) { + pConfig->outputChannels = pMP3->channels; + pConfig->outputSampleRate = pMP3->sampleRate; + } + + drmp3_uninit(pMP3); + + if (pTotalFrameCount) { + *pTotalFrameCount = totalFramesRead; + } + + return pFrames; +} + +drmp3_int16* drmp3__full_read_and_close_s16(drmp3* pMP3, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) +{ + drmp3_uint64 totalFramesRead = 0; + drmp3_uint64 framesCapacity = 0; + drmp3_int16* pFrames = NULL; + drmp3_int16 temp[4096]; + + drmp3_assert(pMP3 != NULL); + + for (;;) { + drmp3_uint64 framesToReadRightNow = drmp3_countof(temp) / pMP3->channels; + drmp3_uint64 framesJustRead = drmp3_read_pcm_frames_s16(pMP3, framesToReadRightNow, temp); + if (framesJustRead == 0) { + break; + } + + /* Reallocate the output buffer if there's not enough room. */ + if (framesCapacity < totalFramesRead + framesJustRead) { + drmp3_uint64 newFramesBufferSize; + drmp3_int16* pNewFrames; + + framesCapacity *= 2; + if (framesCapacity < totalFramesRead + framesJustRead) { + framesCapacity = totalFramesRead + framesJustRead; + } + + newFramesBufferSize = framesCapacity*pMP3->channels*sizeof(drmp3_int16); + if (newFramesBufferSize > DRMP3_SIZE_MAX) { + break; + } + + pNewFrames = (drmp3_int16*)drmp3_realloc(pFrames, (size_t)newFramesBufferSize); + if (pNewFrames == NULL) { + drmp3_free(pFrames); + break; + } + + pFrames = pNewFrames; + } + + drmp3_copy_memory(pFrames + totalFramesRead*pMP3->channels, temp, (size_t)(framesJustRead*pMP3->channels*sizeof(drmp3_int16))); + totalFramesRead += framesJustRead; + + /* If the number of frames we asked for is less that what we actually read it means we've reached the end. */ + if (framesJustRead != framesToReadRightNow) { + break; + } + } + + if (pConfig != NULL) { + pConfig->outputChannels = pMP3->channels; + pConfig->outputSampleRate = pMP3->sampleRate; + } + + drmp3_uninit(pMP3); + + if (pTotalFrameCount) { + *pTotalFrameCount = totalFramesRead; + } + + return pFrames; +} + + +float* drmp3_open_and_read_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) +{ + drmp3 mp3; + if (!drmp3_init(&mp3, onRead, onSeek, pUserData, pConfig)) { + return NULL; + } + + return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount); +} + +drmp3_int16* drmp3_open_and_read_s16(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) +{ + drmp3 mp3; + if (!drmp3_init(&mp3, onRead, onSeek, pUserData, pConfig)) { + return NULL; + } + + return drmp3__full_read_and_close_s16(&mp3, pConfig, pTotalFrameCount); +} + + +float* drmp3_open_memory_and_read_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) +{ + drmp3 mp3; + if (!drmp3_init_memory(&mp3, pData, dataSize, pConfig)) { + return NULL; + } + + return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount); +} + +drmp3_int16* drmp3_open_memory_and_read_s16(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) +{ + drmp3 mp3; + if (!drmp3_init_memory(&mp3, pData, dataSize, pConfig)) { + return NULL; + } + + return drmp3__full_read_and_close_s16(&mp3, pConfig, pTotalFrameCount); +} + + +#ifndef DR_MP3_NO_STDIO +float* drmp3_open_file_and_read_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) +{ + drmp3 mp3; + if (!drmp3_init_file(&mp3, filePath, pConfig)) { + return NULL; + } + + return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount); +} + +drmp3_int16* drmp3_open_file_and_read_s16(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) +{ + drmp3 mp3; + if (!drmp3_init_file(&mp3, filePath, pConfig)) { + return NULL; + } + + return drmp3__full_read_and_close_s16(&mp3, pConfig, pTotalFrameCount); +} +#endif + +void drmp3_free(void* p) +{ + DRMP3_FREE(p); +} + +#endif /*DR_MP3_IMPLEMENTATION*/ + +/* +DIFFERENCES BETWEEN minimp3 AND dr_mp3 +====================================== +- First, keep in mind that minimp3 (https://github.com/lieff/minimp3) is where all the real work was done. All of the + code relating to the actual decoding remains mostly unmodified, apart from some namespacing changes. +- dr_mp3 adds a pulling style API which allows you to deliver raw data via callbacks. So, rather than pushing data + to the decoder, the decoder _pulls_ data from your callbacks. +- In addition to callbacks, a decoder can be initialized from a block of memory and a file. +- The dr_mp3 pull API reads PCM frames rather than whole MP3 frames. +- dr_mp3 adds convenience APIs for opening and decoding entire files in one go. +- dr_mp3 is fully namespaced, including the implementation section, which is more suitable when compiling projects + as a single translation unit (aka unity builds). At the time of writing this, a unity build is not possible when + using minimp3 in conjunction with stb_vorbis. dr_mp3 addresses this. +*/ + +/* +REVISION HISTORY +================ +v0.4.7 - 2019-07-28 + - Fix a compiler error. + +v0.4.6 - 2019-06-14 + - Fix a compiler error. + +v0.4.5 - 2019-06-06 + - Bring up to date with minimp3. + +v0.4.4 - 2019-05-06 + - Fixes to the VC6 build. + +v0.4.3 - 2019-05-05 + - Use the channel count and/or sample rate of the first MP3 frame instead of DR_MP3_DEFAULT_CHANNELS and + DR_MP3_DEFAULT_SAMPLE_RATE when they are set to 0. To use the old behaviour, just set the relevant property to + DR_MP3_DEFAULT_CHANNELS or DR_MP3_DEFAULT_SAMPLE_RATE. + - Add s16 reading APIs + - drmp3_read_pcm_frames_s16 + - drmp3_open_memory_and_read_s16 + - drmp3_open_and_read_s16 + - drmp3_open_file_and_read_s16 + - Add drmp3_get_mp3_and_pcm_frame_count() to the public header section. + - Add support for C89. + - Change license to choice of public domain or MIT-0. + +v0.4.2 - 2019-02-21 + - Fix a warning. + +v0.4.1 - 2018-12-30 + - Fix a warning. + +v0.4.0 - 2018-12-16 + - API CHANGE: Rename some APIs: + - drmp3_read_f32 -> to drmp3_read_pcm_frames_f32 + - drmp3_seek_to_frame -> drmp3_seek_to_pcm_frame + - drmp3_open_and_decode_f32 -> drmp3_open_and_read_f32 + - drmp3_open_and_decode_memory_f32 -> drmp3_open_memory_and_read_f32 + - drmp3_open_and_decode_file_f32 -> drmp3_open_file_and_read_f32 + - Add drmp3_get_pcm_frame_count(). + - Add drmp3_get_mp3_frame_count(). + - Improve seeking performance. + +v0.3.2 - 2018-09-11 + - Fix a couple of memory leaks. + - Bring up to date with minimp3. + +v0.3.1 - 2018-08-25 + - Fix C++ build. + +v0.3.0 - 2018-08-25 + - Bring up to date with minimp3. This has a minor API change: the "pcm" parameter of drmp3dec_decode_frame() has + been changed from short* to void* because it can now output both s16 and f32 samples, depending on whether or + not the DR_MP3_FLOAT_OUTPUT option is set. + +v0.2.11 - 2018-08-08 + - Fix a bug where the last part of a file is not read. + +v0.2.10 - 2018-08-07 + - Improve 64-bit detection. + +v0.2.9 - 2018-08-05 + - Fix C++ build on older versions of GCC. + - Bring up to date with minimp3. + +v0.2.8 - 2018-08-02 + - Fix compilation errors with older versions of GCC. + +v0.2.7 - 2018-07-13 + - Bring up to date with minimp3. + +v0.2.6 - 2018-07-12 + - Bring up to date with minimp3. + +v0.2.5 - 2018-06-22 + - Bring up to date with minimp3. + +v0.2.4 - 2018-05-12 + - Bring up to date with minimp3. + +v0.2.3 - 2018-04-29 + - Fix TCC build. + +v0.2.2 - 2018-04-28 + - Fix bug when opening a decoder from memory. + +v0.2.1 - 2018-04-27 + - Efficiency improvements when the decoder reaches the end of the stream. + +v0.2 - 2018-04-21 + - Bring up to date with minimp3. + - Start using major.minor.revision versioning. + +v0.1d - 2018-03-30 + - Bring up to date with minimp3. + +v0.1c - 2018-03-11 + - Fix C++ build error. + +v0.1b - 2018-03-07 + - Bring up to date with minimp3. + +v0.1a - 2018-02-28 + - Fix compilation error on GCC/Clang. + - Fix some warnings. + +v0.1 - 2018-02-xx + - Initial versioned release. +*/ + +/* +This software is available as a choice of the following licenses. Choose +whichever you prefer. + +=============================================================================== +ALTERNATIVE 1 - Public Domain (www.unlicense.org) +=============================================================================== +This is free and unencumbered software released into the public domain. + +Anyone is free to copy, modify, publish, use, compile, sell, or distribute this +software, either in source code form or as a compiled binary, for any purpose, +commercial or non-commercial, and by any means. + +In jurisdictions that recognize copyright laws, the author or authors of this +software dedicate any and all copyright interest in the software to the public +domain. We make this dedication for the benefit of the public at large and to +the detriment of our heirs and successors. We intend this dedication to be an +overt act of relinquishment in perpetuity of all present and future rights to +this software under copyright law. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN +ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION +WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + +For more information, please refer to <http://unlicense.org/> + +=============================================================================== +ALTERNATIVE 2 - MIT No Attribution +=============================================================================== +Copyright 2018 David Reid + +Permission is hereby granted, free of charge, to any person obtaining a copy of +this software and associated documentation files (the "Software"), to deal in +the Software without restriction, including without limitation the rights to +use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies +of the Software, and to permit persons to whom the Software is furnished to do +so. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. +*/ + +/* + https://github.com/lieff/minimp3 + To the extent possible under law, the author(s) have dedicated all copyright and related and neighboring rights to this software to the public domain worldwide. + This software is distributed without any warranty. + See <http://creativecommons.org/publicdomain/zero/1.0/>. +*/ |