diff --git a/code/nel/include/nel/sound/audio_decoder_mp3.h b/code/nel/include/nel/sound/audio_decoder_mp3.h
new file mode 100644
index 000000000..fac2e2693
--- /dev/null
+++ b/code/nel/include/nel/sound/audio_decoder_mp3.h
@@ -0,0 +1,96 @@
+// NeL - MMORPG Framework
+// Copyright (C) 2018 Winch Gate Property Limited
+//
+// This program is free software: you can redistribute it and/or modify
+// it under the terms of the GNU Affero General Public License as
+// published by the Free Software Foundation, either version 3 of the
+// License, or (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+// GNU Affero General Public License for more details.
+//
+// You should have received a copy of the GNU Affero General Public License
+// along with this program. If not, see .
+
+#ifndef NLSOUND_AUDIO_DECODER_MP3_H
+#define NLSOUND_AUDIO_DECODER_MP3_H
+#include
+
+#include
+
+// disable drmp3_init_file()
+#define DR_MP3_NO_STDIO
+#include
+
+namespace NLSOUND {
+
+/**
+ * \brief CAudioDecoderMP3
+ * \date 2019-01-13 12:39GMT
+ * \author Meelis Mägi (Nimetu)
+ * CAudioDecoderMP3
+ * Create trough IAudioDecoder, type "mp3"
+ */
+class CAudioDecoderMP3 : public IAudioDecoder
+{
+protected:
+ NLMISC::IStream *_Stream;
+
+ bool _IsSupported;
+ bool _Loop;
+ bool _IsMusicEnded;
+ sint32 _StreamOffset;
+ sint32 _StreamSize;
+
+ drmp3 _Decoder;
+
+ // set to total pcm frames after getLength() is called
+ uint64 _PCMFrameCount;
+
+public:
+ CAudioDecoderMP3(NLMISC::IStream *stream, bool loop);
+ virtual ~CAudioDecoderMP3();
+
+ inline NLMISC::IStream *getStream() { return _Stream; }
+ inline sint32 getStreamSize() { return _StreamSize; }
+ inline sint32 getStreamOffset() { return _StreamOffset; }
+
+ // Return true if mp3 is valid
+ bool isFormatSupported() const;
+
+ /// Get information on a music file (only ID3v1 tag is read.
+ static bool getInfo(NLMISC::IStream *stream, std::string &artist, std::string &title, float &length);
+
+ /// Get how many bytes the music buffer requires for output minimum.
+ virtual uint32 getRequiredBytes();
+
+ /// Get an amount of bytes between minimum and maximum (can be lower than minimum if at end).
+ virtual uint32 getNextBytes(uint8 *buffer, uint32 minimum, uint32 maximum);
+
+ /// Get the amount of channels (2 is stereo) in output.
+ virtual uint8 getChannels();
+
+ /// Get the samples per second (often 44100) in output.
+ virtual uint getSamplesPerSec();
+
+ /// Get the bits per sample (often 16) in output.
+ virtual uint8 getBitsPerSample();
+
+ /// Get if the music has ended playing (never true if loop).
+ virtual bool isMusicEnded();
+
+ /// Get the total time in seconds.
+ virtual float getLength();
+
+ /// Set looping
+ virtual void setLooping(bool loop);
+
+}; /* class CAudioDecoderMP3 */
+
+} /* namespace NLSOUND */
+
+#endif // NLSOUND_AUDIO_DECODER_MP3_H
+
+/* end of file */
diff --git a/code/nel/include/nel/sound/decoder/dr_mp3.h b/code/nel/include/nel/sound/decoder/dr_mp3.h
new file mode 100644
index 000000000..465438bf5
--- /dev/null
+++ b/code/nel/include/nel/sound/decoder/dr_mp3.h
@@ -0,0 +1,3566 @@
+// MP3 audio decoder. Public domain. See "unlicense" statement at the end of this file.
+// dr_mp3 - v0.4.1 - 2018-12-30
+//
+// 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
+
+#if defined(_MSC_VER) && _MSC_VER < 1600
+typedef signed char drmp3_int8;
+typedef unsigned char drmp3_uint8;
+typedef signed short drmp3_int16;
+typedef unsigned short drmp3_uint16;
+typedef signed int drmp3_int32;
+typedef unsigned int drmp3_uint32;
+typedef signed __int64 drmp3_int64;
+typedef unsigned __int64 drmp3_uint64;
+#else
+#include
+typedef int8_t drmp3_int8;
+typedef uint8_t drmp3_uint8;
+typedef int16_t drmp3_int16;
+typedef uint16_t drmp3_uint16;
+typedef int32_t drmp3_int32;
+typedef uint32_t drmp3_uint32;
+typedef int64_t drmp3_int64;
+typedef uint64_t drmp3_uint64;
+#endif
+typedef drmp3_uint8 drmp3_bool8;
+typedef drmp3_uint32 drmp3_bool32;
+#define DRMP3_TRUE 1
+#define DRMP3_FALSE 0
+
+#define DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME 1152
+#define DRMP3_MAX_SAMPLES_PER_FRAME (DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME*2)
+
+
+// Low Level Push API
+// ==================
+typedef struct
+{
+ int frame_bytes, channels, hz, layer, bitrate_kbps;
+} drmp3dec_frame_info;
+
+typedef struct
+{
+ float mdct_overlap[2][9*32], qmf_state[15*2*32];
+ int reserv, free_format_bytes;
+ unsigned char header[4], reserv_buf[511];
+} drmp3dec;
+
+// Initializes a low level decoder.
+void drmp3dec_init(drmp3dec *dec);
+
+// Reads a frame from a low level decoder.
+int drmp3dec_decode_frame(drmp3dec *dec, const unsigned char *mp3, int mp3_bytes, void *pcm, drmp3dec_frame_info *info);
+
+// Helper for converting between f32 and s16.
+void drmp3dec_f32_to_s16(const float *in, drmp3_int16 *out, int num_samples);
+
+
+
+
+// Main API (Pull API)
+// ===================
+
+typedef struct drmp3_src drmp3_src;
+typedef drmp3_uint64 (* drmp3_src_read_proc)(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, void* pUserData); // Returns the number of frames that were read.
+
+typedef enum
+{
+ drmp3_src_algorithm_none,
+ drmp3_src_algorithm_linear
+} drmp3_src_algorithm;
+
+#define DRMP3_SRC_CACHE_SIZE_IN_FRAMES 512
+typedef struct
+{
+ drmp3_src* pSRC;
+ float pCachedFrames[2 * DRMP3_SRC_CACHE_SIZE_IN_FRAMES];
+ drmp3_uint32 cachedFrameCount;
+ drmp3_uint32 iNextFrame;
+} drmp3_src_cache;
+
+typedef struct
+{
+ drmp3_uint32 sampleRateIn;
+ drmp3_uint32 sampleRateOut;
+ drmp3_uint32 channels;
+ drmp3_src_algorithm algorithm;
+ drmp3_uint32 cacheSizeInFrames; // The number of frames to read from the client at a time.
+} drmp3_src_config;
+
+struct drmp3_src
+{
+ drmp3_src_config config;
+ drmp3_src_read_proc onRead;
+ void* pUserData;
+ float bin[256];
+ drmp3_src_cache cache; // <-- For simplifying and optimizing client -> memory reading.
+ union
+ {
+ struct
+ {
+ 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);
+
+// 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 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);
+float* drmp3_open_memory_and_read_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount);
+#ifndef DR_MP3_NO_STDIO
+float* drmp3_open_file_and_read_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount);
+#endif
+
+// Frees any memory that was allocated by a public drmp3 API.
+void drmp3_free(void* p);
+
+#ifdef __cplusplus
+}
+#endif
+#endif // dr_mp3_h
+
+
+/////////////////////////////////////////////////////
+//
+// IMPLEMENTATION
+//
+/////////////////////////////////////////////////////
+#ifdef DR_MP3_IMPLEMENTATION
+#include
+#include
+#include
+#include // For INT_MAX
+
+// Disable SIMD when compiling with TCC for now.
+#if defined(__TINYC__)
+#define DR_MP3_NO_SIMD
+#endif
+
+#define DRMP3_OFFSET_PTR(p, offset) ((void*)((drmp3_uint8*)(p) + (offset)))
+
+#define DRMP3_MAX_FREE_FORMAT_FRAME_SIZE 2304 /* more than ISO spec's */
+#ifndef DRMP3_MAX_FRAME_SYNC_MATCHES
+#define DRMP3_MAX_FRAME_SYNC_MATCHES 10
+#endif
+
+#define DRMP3_MAX_L3_FRAME_PAYLOAD_BYTES DRMP3_MAX_FREE_FORMAT_FRAME_SIZE /* MUST be >= 320000/8/32000*1152 = 1440 */
+
+#define DRMP3_MAX_BITRESERVOIR_BYTES 511
+#define DRMP3_SHORT_BLOCK_TYPE 2
+#define DRMP3_STOP_BLOCK_TYPE 3
+#define DRMP3_MODE_MONO 3
+#define DRMP3_MODE_JOINT_STEREO 1
+#define DRMP3_HDR_SIZE 4
+#define DRMP3_HDR_IS_MONO(h) (((h[3]) & 0xC0) == 0xC0)
+#define DRMP3_HDR_IS_MS_STEREO(h) (((h[3]) & 0xE0) == 0x60)
+#define DRMP3_HDR_IS_FREE_FORMAT(h) (((h[2]) & 0xF0) == 0)
+#define DRMP3_HDR_IS_CRC(h) (!((h[1]) & 1))
+#define DRMP3_HDR_TEST_PADDING(h) ((h[2]) & 0x2)
+#define DRMP3_HDR_TEST_MPEG1(h) ((h[1]) & 0x8)
+#define DRMP3_HDR_TEST_NOT_MPEG25(h) ((h[1]) & 0x10)
+#define DRMP3_HDR_TEST_I_STEREO(h) ((h[3]) & 0x10)
+#define DRMP3_HDR_TEST_MS_STEREO(h) ((h[3]) & 0x20)
+#define DRMP3_HDR_GET_STEREO_MODE(h) (((h[3]) >> 6) & 3)
+#define DRMP3_HDR_GET_STEREO_MODE_EXT(h) (((h[3]) >> 4) & 3)
+#define DRMP3_HDR_GET_LAYER(h) (((h[1]) >> 1) & 3)
+#define DRMP3_HDR_GET_BITRATE(h) ((h[2]) >> 4)
+#define DRMP3_HDR_GET_SAMPLE_RATE(h) (((h[2]) >> 2) & 3)
+#define DRMP3_HDR_GET_MY_SAMPLE_RATE(h) (DRMP3_HDR_GET_SAMPLE_RATE(h) + (((h[1] >> 3) & 1) + ((h[1] >> 4) & 1))*3)
+#define DRMP3_HDR_IS_FRAME_576(h) ((h[1] & 14) == 2)
+#define DRMP3_HDR_IS_LAYER_1(h) ((h[1] & 6) == 6)
+
+#define DRMP3_BITS_DEQUANTIZER_OUT -1
+#define DRMP3_MAX_SCF (255 + DRMP3_BITS_DEQUANTIZER_OUT*4 - 210)
+#define DRMP3_MAX_SCFI ((DRMP3_MAX_SCF + 3) & ~3)
+
+#define DRMP3_MIN(a, b) ((a) > (b) ? (b) : (a))
+#define DRMP3_MAX(a, b) ((a) < (b) ? (b) : (a))
+
+#if !defined(DR_MP3_NO_SIMD)
+
+#if !defined(DR_MP3_ONLY_SIMD) && (defined(_M_X64) || defined(_M_ARM64) || defined(__x86_64__) || defined(__aarch64__))
+/* x64 always have SSE2, arm64 always have neon, no need for generic code */
+#define DR_MP3_ONLY_SIMD
+#endif
+
+#if (defined(_MSC_VER) && (defined(_M_IX86) || defined(_M_X64))) || ((defined(__i386__) || defined(__x86_64__)) && defined(__SSE2__))
+#if defined(_MSC_VER)
+#include
+#endif
+#include
+#define DRMP3_HAVE_SSE 1
+#define DRMP3_HAVE_SIMD 1
+#define DRMP3_VSTORE _mm_storeu_ps
+#define DRMP3_VLD _mm_loadu_ps
+#define DRMP3_VSET _mm_set1_ps
+#define DRMP3_VADD _mm_add_ps
+#define DRMP3_VSUB _mm_sub_ps
+#define DRMP3_VMUL _mm_mul_ps
+#define DRMP3_VMAC(a, x, y) _mm_add_ps(a, _mm_mul_ps(x, y))
+#define DRMP3_VMSB(a, x, y) _mm_sub_ps(a, _mm_mul_ps(x, y))
+#define DRMP3_VMUL_S(x, s) _mm_mul_ps(x, _mm_set1_ps(s))
+#define DRMP3_VREV(x) _mm_shuffle_ps(x, x, _MM_SHUFFLE(0, 1, 2, 3))
+typedef __m128 drmp3_f4;
+#if defined(_MSC_VER) || defined(DR_MP3_ONLY_SIMD)
+#define drmp3_cpuid __cpuid
+#else
+static __inline__ __attribute__((always_inline)) void drmp3_cpuid(int CPUInfo[], const int InfoType)
+{
+#if defined(__PIC__)
+ __asm__ __volatile__(
+#if defined(__x86_64__)
+ "push %%rbx\n"
+ "cpuid\n"
+ "xchgl %%ebx, %1\n"
+ "pop %%rbx\n"
+#else
+ "xchgl %%ebx, %1\n"
+ "cpuid\n"
+ "xchgl %%ebx, %1\n"
+#endif
+ : "=a" (CPUInfo[0]), "=r" (CPUInfo[1]), "=c" (CPUInfo[2]), "=d" (CPUInfo[3])
+ : "a" (InfoType));
+#else
+ __asm__ __volatile__(
+ "cpuid"
+ : "=a" (CPUInfo[0]), "=b" (CPUInfo[1]), "=c" (CPUInfo[2]), "=d" (CPUInfo[3])
+ : "a" (InfoType));
+#endif
+}
+#endif
+static int drmp3_have_simd()
+{
+#ifdef DR_MP3_ONLY_SIMD
+ return 1;
+#else
+ static int g_have_simd;
+ int CPUInfo[4];
+#ifdef MINIMP3_TEST
+ static int g_counter;
+ if (g_counter++ > 100)
+ return 0;
+#endif
+ if (g_have_simd)
+ goto end;
+ drmp3_cpuid(CPUInfo, 0);
+ if (CPUInfo[0] > 0)
+ {
+ drmp3_cpuid(CPUInfo, 1);
+ g_have_simd = (CPUInfo[3] & (1 << 26)) + 1; /* SSE2 */
+ return g_have_simd - 1;
+ }
+
+end:
+ return g_have_simd - 1;
+#endif
+}
+#elif defined(__ARM_NEON) || defined(__aarch64__)
+#include
+#define DRMP3_HAVE_SIMD 1
+#define DRMP3_VSTORE vst1q_f32
+#define DRMP3_VLD vld1q_f32
+#define DRMP3_VSET vmovq_n_f32
+#define DRMP3_VADD vaddq_f32
+#define DRMP3_VSUB vsubq_f32
+#define DRMP3_VMUL vmulq_f32
+#define DRMP3_VMAC(a, x, y) vmlaq_f32(a, x, y)
+#define DRMP3_VMSB(a, x, y) vmlsq_f32(a, x, y)
+#define DRMP3_VMUL_S(x, s) vmulq_f32(x, vmovq_n_f32(s))
+#define DRMP3_VREV(x) vcombine_f32(vget_high_f32(vrev64q_f32(x)), vget_low_f32(vrev64q_f32(x)))
+typedef float32x4_t drmp3_f4;
+static int drmp3_have_simd()
+{ /* TODO: detect neon for !DR_MP3_ONLY_SIMD */
+ return 1;
+}
+#else
+#define DRMP3_HAVE_SIMD 0
+#ifdef DR_MP3_ONLY_SIMD
+#error DR_MP3_ONLY_SIMD used, but SSE/NEON not enabled
+#endif
+#endif
+
+#else
+
+#define DRMP3_HAVE_SIMD 0
+
+#endif
+
+typedef struct
+{
+ const drmp3_uint8 *buf;
+ int pos, limit;
+} drmp3_bs;
+
+typedef struct
+{
+ float scf[3*64];
+ drmp3_uint8 total_bands, stereo_bands, bitalloc[64], scfcod[64];
+} drmp3_L12_scale_info;
+
+typedef struct
+{
+ drmp3_uint8 tab_offset, code_tab_width, band_count;
+} drmp3_L12_subband_alloc;
+
+typedef struct
+{
+ const drmp3_uint8 *sfbtab;
+ drmp3_uint16 part_23_length, big_values, scalefac_compress;
+ drmp3_uint8 global_gain, block_type, mixed_block_flag, n_long_sfb, n_short_sfb;
+ drmp3_uint8 table_select[3], region_count[3], subblock_gain[3];
+ drmp3_uint8 preflag, scalefac_scale, count1_table, scfsi;
+} drmp3_L3_gr_info;
+
+typedef struct
+{
+ drmp3_bs bs;
+ drmp3_uint8 maindata[DRMP3_MAX_BITRESERVOIR_BYTES + DRMP3_MAX_L3_FRAME_PAYLOAD_BYTES];
+ drmp3_L3_gr_info gr_info[4];
+ float grbuf[2][576], scf[40], syn[18 + 15][2*32];
+ drmp3_uint8 ist_pos[2][39];
+} drmp3dec_scratch;
+
+static void drmp3_bs_init(drmp3_bs *bs, const drmp3_uint8 *data, int bytes)
+{
+ bs->buf = data;
+ bs->pos = 0;
+ bs->limit = bytes*8;
+}
+
+static drmp3_uint32 drmp3_bs_get_bits(drmp3_bs *bs, int n)
+{
+ drmp3_uint32 next, cache = 0, s = bs->pos & 7;
+ int shl = n + s;
+ const drmp3_uint8 *p = bs->buf + (bs->pos >> 3);
+ if ((bs->pos += n) > bs->limit)
+ return 0;
+ next = *p++ & (255 >> s);
+ while ((shl -= 8) > 0)
+ {
+ cache |= next << shl;
+ next = *p++;
+ }
+ return cache | (next >> -shl);
+}
+
+static int drmp3_hdr_valid(const drmp3_uint8 *h)
+{
+ return h[0] == 0xff &&
+ ((h[1] & 0xF0) == 0xf0 || (h[1] & 0xFE) == 0xe2) &&
+ (DRMP3_HDR_GET_LAYER(h) != 0) &&
+ (DRMP3_HDR_GET_BITRATE(h) != 15) &&
+ (DRMP3_HDR_GET_SAMPLE_RATE(h) != 3);
+}
+
+static int drmp3_hdr_compare(const drmp3_uint8 *h1, const drmp3_uint8 *h2)
+{
+ return drmp3_hdr_valid(h2) &&
+ ((h1[1] ^ h2[1]) & 0xFE) == 0 &&
+ ((h1[2] ^ h2[2]) & 0x0C) == 0 &&
+ !(DRMP3_HDR_IS_FREE_FORMAT(h1) ^ DRMP3_HDR_IS_FREE_FORMAT(h2));
+}
+
+static unsigned drmp3_hdr_bitrate_kbps(const drmp3_uint8 *h)
+{
+ static const drmp3_uint8 halfrate[2][3][15] = {
+ { { 0,4,8,12,16,20,24,28,32,40,48,56,64,72,80 }, { 0,4,8,12,16,20,24,28,32,40,48,56,64,72,80 }, { 0,16,24,28,32,40,48,56,64,72,80,88,96,112,128 } },
+ { { 0,16,20,24,28,32,40,48,56,64,80,96,112,128,160 }, { 0,16,24,28,32,40,48,56,64,80,96,112,128,160,192 }, { 0,16,32,48,64,80,96,112,128,144,160,176,192,208,224 } },
+ };
+ return 2*halfrate[!!DRMP3_HDR_TEST_MPEG1(h)][DRMP3_HDR_GET_LAYER(h) - 1][DRMP3_HDR_GET_BITRATE(h)];
+}
+
+static unsigned drmp3_hdr_sample_rate_hz(const drmp3_uint8 *h)
+{
+ static const unsigned g_hz[3] = { 44100, 48000, 32000 };
+ return g_hz[DRMP3_HDR_GET_SAMPLE_RATE(h)] >> (int)!DRMP3_HDR_TEST_MPEG1(h) >> (int)!DRMP3_HDR_TEST_NOT_MPEG25(h);
+}
+
+static unsigned drmp3_hdr_frame_samples(const drmp3_uint8 *h)
+{
+ return DRMP3_HDR_IS_LAYER_1(h) ? 384 : (1152 >> (int)DRMP3_HDR_IS_FRAME_576(h));
+}
+
+static int drmp3_hdr_frame_bytes(const drmp3_uint8 *h, int free_format_size)
+{
+ int frame_bytes = drmp3_hdr_frame_samples(h)*drmp3_hdr_bitrate_kbps(h)*125/drmp3_hdr_sample_rate_hz(h);
+ if (DRMP3_HDR_IS_LAYER_1(h))
+ {
+ frame_bytes &= ~3; /* slot align */
+ }
+ return frame_bytes ? frame_bytes : free_format_size;
+}
+
+static int drmp3_hdr_padding(const drmp3_uint8 *h)
+{
+ return DRMP3_HDR_TEST_PADDING(h) ? (DRMP3_HDR_IS_LAYER_1(h) ? 4 : 1) : 0;
+}
+
+#ifndef DR_MP3_ONLY_MP3
+static const drmp3_L12_subband_alloc *drmp3_L12_subband_alloc_table(const drmp3_uint8 *hdr, drmp3_L12_scale_info *sci)
+{
+ const drmp3_L12_subband_alloc *alloc;
+ int mode = DRMP3_HDR_GET_STEREO_MODE(hdr);
+ int nbands, stereo_bands = (mode == DRMP3_MODE_MONO) ? 0 : (mode == DRMP3_MODE_JOINT_STEREO) ? (DRMP3_HDR_GET_STEREO_MODE_EXT(hdr) << 2) + 4 : 32;
+
+ if (DRMP3_HDR_IS_LAYER_1(hdr))
+ {
+ static const drmp3_L12_subband_alloc g_alloc_L1[] = { { 76, 4, 32 } };
+ alloc = g_alloc_L1;
+ nbands = 32;
+ } else if (!DRMP3_HDR_TEST_MPEG1(hdr))
+ {
+ static const drmp3_L12_subband_alloc g_alloc_L2M2[] = { { 60, 4, 4 }, { 44, 3, 7 }, { 44, 2, 19 } };
+ alloc = g_alloc_L2M2;
+ nbands = 30;
+ } else
+ {
+ static const drmp3_L12_subband_alloc g_alloc_L2M1[] = { { 0, 4, 3 }, { 16, 4, 8 }, { 32, 3, 12 }, { 40, 2, 7 } };
+ int sample_rate_idx = DRMP3_HDR_GET_SAMPLE_RATE(hdr);
+ unsigned kbps = drmp3_hdr_bitrate_kbps(hdr) >> (int)(mode != DRMP3_MODE_MONO);
+ if (!kbps) /* free-format */
+ {
+ kbps = 192;
+ }
+
+ alloc = g_alloc_L2M1;
+ nbands = 27;
+ if (kbps < 56)
+ {
+ static const drmp3_L12_subband_alloc g_alloc_L2M1_lowrate[] = { { 44, 4, 2 }, { 44, 3, 10 } };
+ alloc = g_alloc_L2M1_lowrate;
+ nbands = sample_rate_idx == 2 ? 12 : 8;
+ } else if (kbps >= 96 && sample_rate_idx != 1)
+ {
+ nbands = 30;
+ }
+ }
+
+ sci->total_bands = (drmp3_uint8)nbands;
+ sci->stereo_bands = (drmp3_uint8)DRMP3_MIN(stereo_bands, nbands);
+
+ return alloc;
+}
+
+static void drmp3_L12_read_scalefactors(drmp3_bs *bs, drmp3_uint8 *pba, drmp3_uint8 *scfcod, int bands, float *scf)
+{
+ static const float g_deq_L12[18*3] = {
+#define DRMP3_DQ(x) 9.53674316e-07f/x, 7.56931807e-07f/x, 6.00777173e-07f/x
+ DRMP3_DQ(3),DRMP3_DQ(7),DRMP3_DQ(15),DRMP3_DQ(31),DRMP3_DQ(63),DRMP3_DQ(127),DRMP3_DQ(255),DRMP3_DQ(511),DRMP3_DQ(1023),DRMP3_DQ(2047),DRMP3_DQ(4095),DRMP3_DQ(8191),DRMP3_DQ(16383),DRMP3_DQ(32767),DRMP3_DQ(65535),DRMP3_DQ(3),DRMP3_DQ(5),DRMP3_DQ(9)
+ };
+ int i, m;
+ for (i = 0; i < bands; i++)
+ {
+ float s = 0;
+ int ba = *pba++;
+ int mask = ba ? 4 + ((19 >> scfcod[i]) & 3) : 0;
+ for (m = 4; m; m >>= 1)
+ {
+ if (mask & m)
+ {
+ int b = drmp3_bs_get_bits(bs, 6);
+ s = g_deq_L12[ba*3 - 6 + b % 3]*(1 << 21 >> b/3);
+ }
+ *scf++ = s;
+ }
+ }
+}
+
+static void drmp3_L12_read_scale_info(const drmp3_uint8 *hdr, drmp3_bs *bs, drmp3_L12_scale_info *sci)
+{
+ static const drmp3_uint8 g_bitalloc_code_tab[] = {
+ 0,17, 3, 4, 5,6,7, 8,9,10,11,12,13,14,15,16,
+ 0,17,18, 3,19,4,5, 6,7, 8, 9,10,11,12,13,16,
+ 0,17,18, 3,19,4,5,16,
+ 0,17,18,16,
+ 0,17,18,19, 4,5,6, 7,8, 9,10,11,12,13,14,15,
+ 0,17,18, 3,19,4,5, 6,7, 8, 9,10,11,12,13,14,
+ 0, 2, 3, 4, 5,6,7, 8,9,10,11,12,13,14,15,16
+ };
+ const drmp3_L12_subband_alloc *subband_alloc = drmp3_L12_subband_alloc_table(hdr, sci);
+
+ int i, k = 0, ba_bits = 0;
+ const drmp3_uint8 *ba_code_tab = g_bitalloc_code_tab;
+
+ for (i = 0; i < sci->total_bands; i++)
+ {
+ drmp3_uint8 ba;
+ if (i == k)
+ {
+ k += subband_alloc->band_count;
+ ba_bits = subband_alloc->code_tab_width;
+ ba_code_tab = g_bitalloc_code_tab + subband_alloc->tab_offset;
+ subband_alloc++;
+ }
+ ba = ba_code_tab[drmp3_bs_get_bits(bs, ba_bits)];
+ sci->bitalloc[2*i] = ba;
+ if (i < sci->stereo_bands)
+ {
+ ba = ba_code_tab[drmp3_bs_get_bits(bs, ba_bits)];
+ }
+ sci->bitalloc[2*i + 1] = sci->stereo_bands ? ba : 0;
+ }
+
+ for (i = 0; i < 2*sci->total_bands; i++)
+ {
+ sci->scfcod[i] = (drmp3_uint8)(sci->bitalloc[i] ? DRMP3_HDR_IS_LAYER_1(hdr) ? 2 : drmp3_bs_get_bits(bs, 2) : 6);
+ }
+
+ drmp3_L12_read_scalefactors(bs, sci->bitalloc, sci->scfcod, sci->total_bands*2, sci->scf);
+
+ for (i = sci->stereo_bands; i < sci->total_bands; i++)
+ {
+ sci->bitalloc[2*i + 1] = 0;
+ }
+}
+
+static int drmp3_L12_dequantize_granule(float *grbuf, drmp3_bs *bs, drmp3_L12_scale_info *sci, int group_size)
+{
+ int i, j, k, choff = 576;
+ for (j = 0; j < 4; j++)
+ {
+ float *dst = grbuf + group_size*j;
+ for (i = 0; i < 2*sci->total_bands; i++)
+ {
+ int ba = sci->bitalloc[i];
+ if (ba != 0)
+ {
+ if (ba < 17)
+ {
+ int half = (1 << (ba - 1)) - 1;
+ for (k = 0; k < group_size; k++)
+ {
+ dst[k] = (float)((int)drmp3_bs_get_bits(bs, ba) - half);
+ }
+ } else
+ {
+ unsigned mod = (2 << (ba - 17)) + 1; /* 3, 5, 9 */
+ unsigned code = drmp3_bs_get_bits(bs, mod + 2 - (mod >> 3)); /* 5, 7, 10 */
+ for (k = 0; k < group_size; k++, code /= mod)
+ {
+ dst[k] = (float)((int)(code % mod - mod/2));
+ }
+ }
+ }
+ dst += choff;
+ choff = 18 - choff;
+ }
+ }
+ return group_size*4;
+}
+
+static void drmp3_L12_apply_scf_384(drmp3_L12_scale_info *sci, const float *scf, float *dst)
+{
+ int i, k;
+ memcpy(dst + 576 + sci->stereo_bands*18, dst + sci->stereo_bands*18, (sci->total_bands - sci->stereo_bands)*18*sizeof(float));
+ for (i = 0; i < sci->total_bands; i++, dst += 18, scf += 6)
+ {
+ for (k = 0; k < 12; k++)
+ {
+ dst[k + 0] *= scf[0];
+ dst[k + 576] *= scf[3];
+ }
+ }
+}
+#endif
+
+static int drmp3_L3_read_side_info(drmp3_bs *bs, drmp3_L3_gr_info *gr, const drmp3_uint8 *hdr)
+{
+ static const drmp3_uint8 g_scf_long[8][23] = {
+ { 6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54,0 },
+ { 12,12,12,12,12,12,16,20,24,28,32,40,48,56,64,76,90,2,2,2,2,2,0 },
+ { 6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54,0 },
+ { 6,6,6,6,6,6,8,10,12,14,16,18,22,26,32,38,46,54,62,70,76,36,0 },
+ { 6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54,0 },
+ { 4,4,4,4,4,4,6,6,8,8,10,12,16,20,24,28,34,42,50,54,76,158,0 },
+ { 4,4,4,4,4,4,6,6,6,8,10,12,16,18,22,28,34,40,46,54,54,192,0 },
+ { 4,4,4,4,4,4,6,6,8,10,12,16,20,24,30,38,46,56,68,84,102,26,0 }
+ };
+ static const drmp3_uint8 g_scf_short[8][40] = {
+ { 4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 },
+ { 8,8,8,8,8,8,8,8,8,12,12,12,16,16,16,20,20,20,24,24,24,28,28,28,36,36,36,2,2,2,2,2,2,2,2,2,26,26,26,0 },
+ { 4,4,4,4,4,4,4,4,4,6,6,6,6,6,6,8,8,8,10,10,10,14,14,14,18,18,18,26,26,26,32,32,32,42,42,42,18,18,18,0 },
+ { 4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,32,32,32,44,44,44,12,12,12,0 },
+ { 4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 },
+ { 4,4,4,4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,22,22,22,30,30,30,56,56,56,0 },
+ { 4,4,4,4,4,4,4,4,4,4,4,4,6,6,6,6,6,6,10,10,10,12,12,12,14,14,14,16,16,16,20,20,20,26,26,26,66,66,66,0 },
+ { 4,4,4,4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,12,12,12,16,16,16,20,20,20,26,26,26,34,34,34,42,42,42,12,12,12,0 }
+ };
+ static const drmp3_uint8 g_scf_mixed[8][40] = {
+ { 6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 },
+ { 12,12,12,4,4,4,8,8,8,12,12,12,16,16,16,20,20,20,24,24,24,28,28,28,36,36,36,2,2,2,2,2,2,2,2,2,26,26,26,0 },
+ { 6,6,6,6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,14,14,14,18,18,18,26,26,26,32,32,32,42,42,42,18,18,18,0 },
+ { 6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,32,32,32,44,44,44,12,12,12,0 },
+ { 6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 },
+ { 4,4,4,4,4,4,6,6,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,22,22,22,30,30,30,56,56,56,0 },
+ { 4,4,4,4,4,4,6,6,4,4,4,6,6,6,6,6,6,10,10,10,12,12,12,14,14,14,16,16,16,20,20,20,26,26,26,66,66,66,0 },
+ { 4,4,4,4,4,4,6,6,4,4,4,6,6,6,8,8,8,12,12,12,16,16,16,20,20,20,26,26,26,34,34,34,42,42,42,12,12,12,0 }
+ };
+
+ unsigned tables, scfsi = 0;
+ int main_data_begin, part_23_sum = 0;
+ int sr_idx = DRMP3_HDR_GET_MY_SAMPLE_RATE(hdr); sr_idx -= (sr_idx != 0);
+ int gr_count = DRMP3_HDR_IS_MONO(hdr) ? 1 : 2;
+
+ if (DRMP3_HDR_TEST_MPEG1(hdr))
+ {
+ gr_count *= 2;
+ main_data_begin = drmp3_bs_get_bits(bs, 9);
+ scfsi = drmp3_bs_get_bits(bs, 7 + gr_count);
+ } else
+ {
+ main_data_begin = drmp3_bs_get_bits(bs, 8 + gr_count) >> gr_count;
+ }
+
+ do
+ {
+ if (DRMP3_HDR_IS_MONO(hdr))
+ {
+ scfsi <<= 4;
+ }
+ gr->part_23_length = (drmp3_uint16)drmp3_bs_get_bits(bs, 12);
+ part_23_sum += gr->part_23_length;
+ gr->big_values = (drmp3_uint16)drmp3_bs_get_bits(bs, 9);
+ if (gr->big_values > 288)
+ {
+ return -1;
+ }
+ gr->global_gain = (drmp3_uint8)drmp3_bs_get_bits(bs, 8);
+ gr->scalefac_compress = (drmp3_uint16)drmp3_bs_get_bits(bs, DRMP3_HDR_TEST_MPEG1(hdr) ? 4 : 9);
+ gr->sfbtab = g_scf_long[sr_idx];
+ gr->n_long_sfb = 22;
+ gr->n_short_sfb = 0;
+ if (drmp3_bs_get_bits(bs, 1))
+ {
+ gr->block_type = (drmp3_uint8)drmp3_bs_get_bits(bs, 2);
+ if (!gr->block_type)
+ {
+ return -1;
+ }
+ gr->mixed_block_flag = (drmp3_uint8)drmp3_bs_get_bits(bs, 1);
+ gr->region_count[0] = 7;
+ gr->region_count[1] = 255;
+ if (gr->block_type == DRMP3_SHORT_BLOCK_TYPE)
+ {
+ scfsi &= 0x0F0F;
+ if (!gr->mixed_block_flag)
+ {
+ gr->region_count[0] = 8;
+ gr->sfbtab = g_scf_short[sr_idx];
+ gr->n_long_sfb = 0;
+ gr->n_short_sfb = 39;
+ } else
+ {
+ gr->sfbtab = g_scf_mixed[sr_idx];
+ gr->n_long_sfb = DRMP3_HDR_TEST_MPEG1(hdr) ? 8 : 6;
+ gr->n_short_sfb = 30;
+ }
+ }
+ tables = drmp3_bs_get_bits(bs, 10);
+ tables <<= 5;
+ gr->subblock_gain[0] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3);
+ gr->subblock_gain[1] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3);
+ gr->subblock_gain[2] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3);
+ } else
+ {
+ gr->block_type = 0;
+ gr->mixed_block_flag = 0;
+ tables = drmp3_bs_get_bits(bs, 15);
+ gr->region_count[0] = (drmp3_uint8)drmp3_bs_get_bits(bs, 4);
+ gr->region_count[1] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3);
+ gr->region_count[2] = 255;
+ }
+ gr->table_select[0] = (drmp3_uint8)(tables >> 10);
+ gr->table_select[1] = (drmp3_uint8)((tables >> 5) & 31);
+ gr->table_select[2] = (drmp3_uint8)((tables) & 31);
+ gr->preflag = (drmp3_uint8)(DRMP3_HDR_TEST_MPEG1(hdr) ? drmp3_bs_get_bits(bs, 1) : (gr->scalefac_compress >= 500));
+ gr->scalefac_scale = (drmp3_uint8)drmp3_bs_get_bits(bs, 1);
+ gr->count1_table = (drmp3_uint8)drmp3_bs_get_bits(bs, 1);
+ gr->scfsi = (drmp3_uint8)((scfsi >> 12) & 15);
+ scfsi <<= 4;
+ gr++;
+ } while(--gr_count);
+
+ if (part_23_sum + bs->pos > bs->limit + main_data_begin*8)
+ {
+ return -1;
+ }
+
+ return main_data_begin;
+}
+
+static void drmp3_L3_read_scalefactors(drmp3_uint8 *scf, drmp3_uint8 *ist_pos, const drmp3_uint8 *scf_size, const drmp3_uint8 *scf_count, drmp3_bs *bitbuf, int scfsi)
+{
+ int i, k;
+ for (i = 0; i < 4 && scf_count[i]; i++, scfsi *= 2)
+ {
+ int cnt = scf_count[i];
+ if (scfsi & 8)
+ {
+ memcpy(scf, ist_pos, cnt);
+ } else
+ {
+ int bits = scf_size[i];
+ if (!bits)
+ {
+ memset(scf, 0, cnt);
+ memset(ist_pos, 0, cnt);
+ } else
+ {
+ int max_scf = (scfsi < 0) ? (1 << bits) - 1 : -1;
+ for (k = 0; k < cnt; k++)
+ {
+ int s = drmp3_bs_get_bits(bitbuf, bits);
+ ist_pos[k] = (drmp3_uint8)(s == max_scf ? -1 : s);
+ scf[k] = (drmp3_uint8)s;
+ }
+ }
+ }
+ ist_pos += cnt;
+ scf += cnt;
+ }
+ scf[0] = scf[1] = scf[2] = 0;
+}
+
+static float drmp3_L3_ldexp_q2(float y, int exp_q2)
+{
+ static const float g_expfrac[4] = { 9.31322575e-10f,7.83145814e-10f,6.58544508e-10f,5.53767716e-10f };
+ int e;
+ do
+ {
+ e = DRMP3_MIN(30*4, exp_q2);
+ y *= g_expfrac[e & 3]*(1 << 30 >> (e >> 2));
+ } while ((exp_q2 -= e) > 0);
+ return y;
+}
+
+static void drmp3_L3_decode_scalefactors(const drmp3_uint8 *hdr, drmp3_uint8 *ist_pos, drmp3_bs *bs, const drmp3_L3_gr_info *gr, float *scf, int ch)
+{
+ static const drmp3_uint8 g_scf_partitions[3][28] = {
+ { 6,5,5, 5,6,5,5,5,6,5, 7,3,11,10,0,0, 7, 7, 7,0, 6, 6,6,3, 8, 8,5,0 },
+ { 8,9,6,12,6,9,9,9,6,9,12,6,15,18,0,0, 6,15,12,0, 6,12,9,6, 6,18,9,0 },
+ { 9,9,6,12,9,9,9,9,9,9,12,6,18,18,0,0,12,12,12,0,12, 9,9,6,15,12,9,0 }
+ };
+ const drmp3_uint8 *scf_partition = g_scf_partitions[!!gr->n_short_sfb + !gr->n_long_sfb];
+ drmp3_uint8 scf_size[4], iscf[40];
+ int i, scf_shift = gr->scalefac_scale + 1, gain_exp, scfsi = gr->scfsi;
+ float gain;
+
+ if (DRMP3_HDR_TEST_MPEG1(hdr))
+ {
+ static const drmp3_uint8 g_scfc_decode[16] = { 0,1,2,3, 12,5,6,7, 9,10,11,13, 14,15,18,19 };
+ int part = g_scfc_decode[gr->scalefac_compress];
+ scf_size[1] = scf_size[0] = (drmp3_uint8)(part >> 2);
+ scf_size[3] = scf_size[2] = (drmp3_uint8)(part & 3);
+ } else
+ {
+ static const drmp3_uint8 g_mod[6*4] = { 5,5,4,4,5,5,4,1,4,3,1,1,5,6,6,1,4,4,4,1,4,3,1,1 };
+ int k, modprod, sfc, ist = DRMP3_HDR_TEST_I_STEREO(hdr) && ch;
+ sfc = gr->scalefac_compress >> ist;
+ for (k = ist*3*4; sfc >= 0; sfc -= modprod, k += 4)
+ {
+ for (modprod = 1, i = 3; i >= 0; i--)
+ {
+ scf_size[i] = (drmp3_uint8)(sfc / modprod % g_mod[k + i]);
+ modprod *= g_mod[k + i];
+ }
+ }
+ scf_partition += k;
+ scfsi = -16;
+ }
+ drmp3_L3_read_scalefactors(iscf, ist_pos, scf_size, scf_partition, bs, scfsi);
+
+ if (gr->n_short_sfb)
+ {
+ int sh = 3 - scf_shift;
+ for (i = 0; i < gr->n_short_sfb; i += 3)
+ {
+ iscf[gr->n_long_sfb + i + 0] += gr->subblock_gain[0] << sh;
+ iscf[gr->n_long_sfb + i + 1] += gr->subblock_gain[1] << sh;
+ iscf[gr->n_long_sfb + i + 2] += gr->subblock_gain[2] << sh;
+ }
+ } else if (gr->preflag)
+ {
+ static const drmp3_uint8 g_preamp[10] = { 1,1,1,1,2,2,3,3,3,2 };
+ for (i = 0; i < 10; i++)
+ {
+ iscf[11 + i] += g_preamp[i];
+ }
+ }
+
+ gain_exp = gr->global_gain + DRMP3_BITS_DEQUANTIZER_OUT*4 - 210 - (DRMP3_HDR_IS_MS_STEREO(hdr) ? 2 : 0);
+ gain = drmp3_L3_ldexp_q2(1 << (DRMP3_MAX_SCFI/4), DRMP3_MAX_SCFI - gain_exp);
+ for (i = 0; i < (int)(gr->n_long_sfb + gr->n_short_sfb); i++)
+ {
+ scf[i] = drmp3_L3_ldexp_q2(gain, iscf[i] << scf_shift);
+ }
+}
+
+static const float g_drmp3_pow43[129 + 16] = {
+ 0,-1,-2.519842f,-4.326749f,-6.349604f,-8.549880f,-10.902724f,-13.390518f,-16.000000f,-18.720754f,-21.544347f,-24.463781f,-27.473142f,-30.567351f,-33.741992f,-36.993181f,
+ 0,1,2.519842f,4.326749f,6.349604f,8.549880f,10.902724f,13.390518f,16.000000f,18.720754f,21.544347f,24.463781f,27.473142f,30.567351f,33.741992f,36.993181f,40.317474f,43.711787f,47.173345f,50.699631f,54.288352f,57.937408f,61.644865f,65.408941f,69.227979f,73.100443f,77.024898f,81.000000f,85.024491f,89.097188f,93.216975f,97.382800f,101.593667f,105.848633f,110.146801f,114.487321f,118.869381f,123.292209f,127.755065f,132.257246f,136.798076f,141.376907f,145.993119f,150.646117f,155.335327f,160.060199f,164.820202f,169.614826f,174.443577f,179.305980f,184.201575f,189.129918f,194.090580f,199.083145f,204.107210f,209.162385f,214.248292f,219.364564f,224.510845f,229.686789f,234.892058f,240.126328f,245.389280f,250.680604f,256.000000f,261.347174f,266.721841f,272.123723f,277.552547f,283.008049f,288.489971f,293.998060f,299.532071f,305.091761f,310.676898f,316.287249f,321.922592f,327.582707f,333.267377f,338.976394f,344.709550f,350.466646f,356.247482f,362.051866f,367.879608f,373.730522f,379.604427f,385.501143f,391.420496f,397.362314f,403.326427f,409.312672f,415.320884f,421.350905f,427.402579f,433.475750f,439.570269f,445.685987f,451.822757f,457.980436f,464.158883f,470.357960f,476.577530f,482.817459f,489.077615f,495.357868f,501.658090f,507.978156f,514.317941f,520.677324f,527.056184f,533.454404f,539.871867f,546.308458f,552.764065f,559.238575f,565.731879f,572.243870f,578.774440f,585.323483f,591.890898f,598.476581f,605.080431f,611.702349f,618.342238f,625.000000f,631.675540f,638.368763f,645.079578f
+};
+
+static float drmp3_L3_pow_43(int x)
+{
+ float frac;
+ int sign, mult = 256;
+
+ if (x < 129)
+ {
+ return g_drmp3_pow43[16 + x];
+ }
+
+ if (x < 1024)
+ {
+ mult = 16;
+ x <<= 3;
+ }
+
+ sign = 2*x & 64;
+ frac = (float)((x & 63) - sign) / ((x & ~63) + sign);
+ return g_drmp3_pow43[16 + ((x + sign) >> 6)]*(1.f + frac*((4.f/3) + frac*(2.f/9)))*mult;
+}
+
+static void drmp3_L3_huffman(float *dst, drmp3_bs *bs, const drmp3_L3_gr_info *gr_info, const float *scf, int layer3gr_limit)
+{
+ static const drmp3_int16 tabs[] = { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ 785,785,785,785,784,784,784,784,513,513,513,513,513,513,513,513,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,
+ -255,1313,1298,1282,785,785,785,785,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,290,288,
+ -255,1313,1298,1282,769,769,769,769,529,529,529,529,529,529,529,529,528,528,528,528,528,528,528,528,512,512,512,512,512,512,512,512,290,288,
+ -253,-318,-351,-367,785,785,785,785,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,819,818,547,547,275,275,275,275,561,560,515,546,289,274,288,258,
+ -254,-287,1329,1299,1314,1312,1057,1057,1042,1042,1026,1026,784,784,784,784,529,529,529,529,529,529,529,529,769,769,769,769,768,768,768,768,563,560,306,306,291,259,
+ -252,-413,-477,-542,1298,-575,1041,1041,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,-383,-399,1107,1092,1106,1061,849,849,789,789,1104,1091,773,773,1076,1075,341,340,325,309,834,804,577,577,532,532,516,516,832,818,803,816,561,561,531,531,515,546,289,289,288,258,
+ -252,-429,-493,-559,1057,1057,1042,1042,529,529,529,529,529,529,529,529,784,784,784,784,769,769,769,769,512,512,512,512,512,512,512,512,-382,1077,-415,1106,1061,1104,849,849,789,789,1091,1076,1029,1075,834,834,597,581,340,340,339,324,804,833,532,532,832,772,818,803,817,787,816,771,290,290,290,290,288,258,
+ -253,-349,-414,-447,-463,1329,1299,-479,1314,1312,1057,1057,1042,1042,1026,1026,785,785,785,785,784,784,784,784,769,769,769,769,768,768,768,768,-319,851,821,-335,836,850,805,849,341,340,325,336,533,533,579,579,564,564,773,832,578,548,563,516,321,276,306,291,304,259,
+ -251,-572,-733,-830,-863,-879,1041,1041,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,-511,-527,-543,1396,1351,1381,1366,1395,1335,1380,-559,1334,1138,1138,1063,1063,1350,1392,1031,1031,1062,1062,1364,1363,1120,1120,1333,1348,881,881,881,881,375,374,359,373,343,358,341,325,791,791,1123,1122,-703,1105,1045,-719,865,865,790,790,774,774,1104,1029,338,293,323,308,-799,-815,833,788,772,818,803,816,322,292,307,320,561,531,515,546,289,274,288,258,
+ -251,-525,-605,-685,-765,-831,-846,1298,1057,1057,1312,1282,785,785,785,785,784,784,784,784,769,769,769,769,512,512,512,512,512,512,512,512,1399,1398,1383,1367,1382,1396,1351,-511,1381,1366,1139,1139,1079,1079,1124,1124,1364,1349,1363,1333,882,882,882,882,807,807,807,807,1094,1094,1136,1136,373,341,535,535,881,775,867,822,774,-591,324,338,-671,849,550,550,866,864,609,609,293,336,534,534,789,835,773,-751,834,804,308,307,833,788,832,772,562,562,547,547,305,275,560,515,290,290,
+ -252,-397,-477,-557,-622,-653,-719,-735,-750,1329,1299,1314,1057,1057,1042,1042,1312,1282,1024,1024,785,785,785,785,784,784,784,784,769,769,769,769,-383,1127,1141,1111,1126,1140,1095,1110,869,869,883,883,1079,1109,882,882,375,374,807,868,838,881,791,-463,867,822,368,263,852,837,836,-543,610,610,550,550,352,336,534,534,865,774,851,821,850,805,593,533,579,564,773,832,578,578,548,548,577,577,307,276,306,291,516,560,259,259,
+ -250,-2107,-2507,-2764,-2909,-2974,-3007,-3023,1041,1041,1040,1040,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,-767,-1052,-1213,-1277,-1358,-1405,-1469,-1535,-1550,-1582,-1614,-1647,-1662,-1694,-1726,-1759,-1774,-1807,-1822,-1854,-1886,1565,-1919,-1935,-1951,-1967,1731,1730,1580,1717,-1983,1729,1564,-1999,1548,-2015,-2031,1715,1595,-2047,1714,-2063,1610,-2079,1609,-2095,1323,1323,1457,1457,1307,1307,1712,1547,1641,1700,1699,1594,1685,1625,1442,1442,1322,1322,-780,-973,-910,1279,1278,1277,1262,1276,1261,1275,1215,1260,1229,-959,974,974,989,989,-943,735,478,478,495,463,506,414,-1039,1003,958,1017,927,942,987,957,431,476,1272,1167,1228,-1183,1256,-1199,895,895,941,941,1242,1227,1212,1135,1014,1014,490,489,503,487,910,1013,985,925,863,894,970,955,1012,847,-1343,831,755,755,984,909,428,366,754,559,-1391,752,486,457,924,997,698,698,983,893,740,740,908,877,739,739,667,667,953,938,497,287,271,271,683,606,590,712,726,574,302,302,738,736,481,286,526,725,605,711,636,724,696,651,589,681,666,710,364,467,573,695,466,466,301,465,379,379,709,604,665,679,316,316,634,633,436,436,464,269,424,394,452,332,438,363,347,408,393,448,331,422,362,407,392,421,346,406,391,376,375,359,1441,1306,-2367,1290,-2383,1337,-2399,-2415,1426,1321,-2431,1411,1336,-2447,-2463,-2479,1169,1169,1049,1049,1424,1289,1412,1352,1319,-2495,1154,1154,1064,1064,1153,1153,416,390,360,404,403,389,344,374,373,343,358,372,327,357,342,311,356,326,1395,1394,1137,1137,1047,1047,1365,1392,1287,1379,1334,1364,1349,1378,1318,1363,792,792,792,792,1152,1152,1032,1032,1121,1121,1046,1046,1120,1120,1030,1030,-2895,1106,1061,1104,849,849,789,789,1091,1076,1029,1090,1060,1075,833,833,309,324,532,532,832,772,818,803,561,561,531,560,515,546,289,274,288,258,
+ -250,-1179,-1579,-1836,-1996,-2124,-2253,-2333,-2413,-2477,-2542,-2574,-2607,-2622,-2655,1314,1313,1298,1312,1282,785,785,785,785,1040,1040,1025,1025,768,768,768,768,-766,-798,-830,-862,-895,-911,-927,-943,-959,-975,-991,-1007,-1023,-1039,-1055,-1070,1724,1647,-1103,-1119,1631,1767,1662,1738,1708,1723,-1135,1780,1615,1779,1599,1677,1646,1778,1583,-1151,1777,1567,1737,1692,1765,1722,1707,1630,1751,1661,1764,1614,1736,1676,1763,1750,1645,1598,1721,1691,1762,1706,1582,1761,1566,-1167,1749,1629,767,766,751,765,494,494,735,764,719,749,734,763,447,447,748,718,477,506,431,491,446,476,461,505,415,430,475,445,504,399,460,489,414,503,383,474,429,459,502,502,746,752,488,398,501,473,413,472,486,271,480,270,-1439,-1455,1357,-1471,-1487,-1503,1341,1325,-1519,1489,1463,1403,1309,-1535,1372,1448,1418,1476,1356,1462,1387,-1551,1475,1340,1447,1402,1386,-1567,1068,1068,1474,1461,455,380,468,440,395,425,410,454,364,467,466,464,453,269,409,448,268,432,1371,1473,1432,1417,1308,1460,1355,1446,1459,1431,1083,1083,1401,1416,1458,1445,1067,1067,1370,1457,1051,1051,1291,1430,1385,1444,1354,1415,1400,1443,1082,1082,1173,1113,1186,1066,1185,1050,-1967,1158,1128,1172,1097,1171,1081,-1983,1157,1112,416,266,375,400,1170,1142,1127,1065,793,793,1169,1033,1156,1096,1141,1111,1155,1080,1126,1140,898,898,808,808,897,897,792,792,1095,1152,1032,1125,1110,1139,1079,1124,882,807,838,881,853,791,-2319,867,368,263,822,852,837,866,806,865,-2399,851,352,262,534,534,821,836,594,594,549,549,593,593,533,533,848,773,579,579,564,578,548,563,276,276,577,576,306,291,516,560,305,305,275,259,
+ -251,-892,-2058,-2620,-2828,-2957,-3023,-3039,1041,1041,1040,1040,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,-511,-527,-543,-559,1530,-575,-591,1528,1527,1407,1526,1391,1023,1023,1023,1023,1525,1375,1268,1268,1103,1103,1087,1087,1039,1039,1523,-604,815,815,815,815,510,495,509,479,508,463,507,447,431,505,415,399,-734,-782,1262,-815,1259,1244,-831,1258,1228,-847,-863,1196,-879,1253,987,987,748,-767,493,493,462,477,414,414,686,669,478,446,461,445,474,429,487,458,412,471,1266,1264,1009,1009,799,799,-1019,-1276,-1452,-1581,-1677,-1757,-1821,-1886,-1933,-1997,1257,1257,1483,1468,1512,1422,1497,1406,1467,1496,1421,1510,1134,1134,1225,1225,1466,1451,1374,1405,1252,1252,1358,1480,1164,1164,1251,1251,1238,1238,1389,1465,-1407,1054,1101,-1423,1207,-1439,830,830,1248,1038,1237,1117,1223,1148,1236,1208,411,426,395,410,379,269,1193,1222,1132,1235,1221,1116,976,976,1192,1162,1177,1220,1131,1191,963,963,-1647,961,780,-1663,558,558,994,993,437,408,393,407,829,978,813,797,947,-1743,721,721,377,392,844,950,828,890,706,706,812,859,796,960,948,843,934,874,571,571,-1919,690,555,689,421,346,539,539,944,779,918,873,932,842,903,888,570,570,931,917,674,674,-2575,1562,-2591,1609,-2607,1654,1322,1322,1441,1441,1696,1546,1683,1593,1669,1624,1426,1426,1321,1321,1639,1680,1425,1425,1305,1305,1545,1668,1608,1623,1667,1592,1638,1666,1320,1320,1652,1607,1409,1409,1304,1304,1288,1288,1664,1637,1395,1395,1335,1335,1622,1636,1394,1394,1319,1319,1606,1621,1392,1392,1137,1137,1137,1137,345,390,360,375,404,373,1047,-2751,-2767,-2783,1062,1121,1046,-2799,1077,-2815,1106,1061,789,789,1105,1104,263,355,310,340,325,354,352,262,339,324,1091,1076,1029,1090,1060,1075,833,833,788,788,1088,1028,818,818,803,803,561,561,531,531,816,771,546,546,289,274,288,258,
+ -253,-317,-381,-446,-478,-509,1279,1279,-811,-1179,-1451,-1756,-1900,-2028,-2189,-2253,-2333,-2414,-2445,-2511,-2526,1313,1298,-2559,1041,1041,1040,1040,1025,1025,1024,1024,1022,1007,1021,991,1020,975,1019,959,687,687,1018,1017,671,671,655,655,1016,1015,639,639,758,758,623,623,757,607,756,591,755,575,754,559,543,543,1009,783,-575,-621,-685,-749,496,-590,750,749,734,748,974,989,1003,958,988,973,1002,942,987,957,972,1001,926,986,941,971,956,1000,910,985,925,999,894,970,-1071,-1087,-1102,1390,-1135,1436,1509,1451,1374,-1151,1405,1358,1480,1420,-1167,1507,1494,1389,1342,1465,1435,1450,1326,1505,1310,1493,1373,1479,1404,1492,1464,1419,428,443,472,397,736,526,464,464,486,457,442,471,484,482,1357,1449,1434,1478,1388,1491,1341,1490,1325,1489,1463,1403,1309,1477,1372,1448,1418,1433,1476,1356,1462,1387,-1439,1475,1340,1447,1402,1474,1324,1461,1371,1473,269,448,1432,1417,1308,1460,-1711,1459,-1727,1441,1099,1099,1446,1386,1431,1401,-1743,1289,1083,1083,1160,1160,1458,1445,1067,1067,1370,1457,1307,1430,1129,1129,1098,1098,268,432,267,416,266,400,-1887,1144,1187,1082,1173,1113,1186,1066,1050,1158,1128,1143,1172,1097,1171,1081,420,391,1157,1112,1170,1142,1127,1065,1169,1049,1156,1096,1141,1111,1155,1080,1126,1154,1064,1153,1140,1095,1048,-2159,1125,1110,1137,-2175,823,823,1139,1138,807,807,384,264,368,263,868,838,853,791,867,822,852,837,866,806,865,790,-2319,851,821,836,352,262,850,805,849,-2399,533,533,835,820,336,261,578,548,563,577,532,532,832,772,562,562,547,547,305,275,560,515,290,290,288,258 };
+ static const drmp3_uint8 tab32[] = { 130,162,193,209,44,28,76,140,9,9,9,9,9,9,9,9,190,254,222,238,126,94,157,157,109,61,173,205};
+ static const drmp3_uint8 tab33[] = { 252,236,220,204,188,172,156,140,124,108,92,76,60,44,28,12 };
+ static const drmp3_int16 tabindex[2*16] = { 0,32,64,98,0,132,180,218,292,364,426,538,648,746,0,1126,1460,1460,1460,1460,1460,1460,1460,1460,1842,1842,1842,1842,1842,1842,1842,1842 };
+ static const drmp3_uint8 g_linbits[] = { 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,2,3,4,6,8,10,13,4,5,6,7,8,9,11,13 };
+
+#define DRMP3_PEEK_BITS(n) (bs_cache >> (32 - n))
+#define DRMP3_FLUSH_BITS(n) { bs_cache <<= (n); bs_sh += (n); }
+#define DRMP3_CHECK_BITS while (bs_sh >= 0) { bs_cache |= (drmp3_uint32)*bs_next_ptr++ << bs_sh; bs_sh -= 8; }
+#define DRMP3_BSPOS ((bs_next_ptr - bs->buf)*8 - 24 + bs_sh)
+
+ float one = 0.0f;
+ int ireg = 0, big_val_cnt = gr_info->big_values;
+ const drmp3_uint8 *sfb = gr_info->sfbtab;
+ const drmp3_uint8 *bs_next_ptr = bs->buf + bs->pos/8;
+ drmp3_uint32 bs_cache = (((bs_next_ptr[0]*256u + bs_next_ptr[1])*256u + bs_next_ptr[2])*256u + bs_next_ptr[3]) << (bs->pos & 7);
+ int pairs_to_decode, np, bs_sh = (bs->pos & 7) - 8;
+ bs_next_ptr += 4;
+
+ while (big_val_cnt > 0)
+ {
+ int tab_num = gr_info->table_select[ireg];
+ int sfb_cnt = gr_info->region_count[ireg++];
+ const drmp3_int16 *codebook = tabs + tabindex[tab_num];
+ int linbits = g_linbits[tab_num];
+ do
+ {
+ np = *sfb++ / 2;
+ pairs_to_decode = DRMP3_MIN(big_val_cnt, np);
+ one = *scf++;
+ do
+ {
+ int j, w = 5;
+ int leaf = codebook[DRMP3_PEEK_BITS(w)];
+ while (leaf < 0)
+ {
+ DRMP3_FLUSH_BITS(w);
+ w = leaf & 7;
+ leaf = codebook[DRMP3_PEEK_BITS(w) - (leaf >> 3)];
+ }
+ DRMP3_FLUSH_BITS(leaf >> 8);
+
+ for (j = 0; j < 2; j++, dst++, leaf >>= 4)
+ {
+ int lsb = leaf & 0x0F;
+ if (lsb == 15 && linbits)
+ {
+ lsb += DRMP3_PEEK_BITS(linbits);
+ DRMP3_FLUSH_BITS(linbits);
+ DRMP3_CHECK_BITS;
+ *dst = one*drmp3_L3_pow_43(lsb)*((int32_t)bs_cache < 0 ? -1: 1);
+ } else
+ {
+ *dst = g_drmp3_pow43[16 + lsb - 16*(bs_cache >> 31)]*one;
+ }
+ DRMP3_FLUSH_BITS(lsb ? 1 : 0);
+ }
+ DRMP3_CHECK_BITS;
+ } while (--pairs_to_decode);
+ } while ((big_val_cnt -= np) > 0 && --sfb_cnt >= 0);
+ }
+
+ for (np = 1 - big_val_cnt;; dst += 4)
+ {
+ const drmp3_uint8 *codebook_count1 = (gr_info->count1_table) ? tab33 : tab32;
+ int leaf = codebook_count1[DRMP3_PEEK_BITS(4)];
+ if (!(leaf & 8))
+ {
+ leaf = codebook_count1[(leaf >> 3) + (bs_cache << 4 >> (32 - (leaf & 3)))];
+ }
+ DRMP3_FLUSH_BITS(leaf & 7);
+ if (DRMP3_BSPOS > layer3gr_limit)
+ {
+ break;
+ }
+#define DRMP3_RELOAD_SCALEFACTOR if (!--np) { np = *sfb++/2; if (!np) break; one = *scf++; }
+#define DRMP3_DEQ_COUNT1(s) if (leaf & (128 >> s)) { dst[s] = ((drmp3_int32)bs_cache < 0) ? -one : one; DRMP3_FLUSH_BITS(1) }
+ DRMP3_RELOAD_SCALEFACTOR;
+ DRMP3_DEQ_COUNT1(0);
+ DRMP3_DEQ_COUNT1(1);
+ DRMP3_RELOAD_SCALEFACTOR;
+ DRMP3_DEQ_COUNT1(2);
+ DRMP3_DEQ_COUNT1(3);
+ DRMP3_CHECK_BITS;
+ }
+
+ bs->pos = layer3gr_limit;
+}
+
+static void drmp3_L3_midside_stereo(float *left, int n)
+{
+ int i = 0;
+ float *right = left + 576;
+#if DRMP3_HAVE_SIMD
+ if (drmp3_have_simd()) for (; i < n - 3; i += 4)
+ {
+ drmp3_f4 vl = DRMP3_VLD(left + i);
+ drmp3_f4 vr = DRMP3_VLD(right + i);
+ DRMP3_VSTORE(left + i, DRMP3_VADD(vl, vr));
+ DRMP3_VSTORE(right + i, DRMP3_VSUB(vl, vr));
+ }
+#endif
+ for (; i < n; i++)
+ {
+ float a = left[i];
+ float b = right[i];
+ left[i] = a + b;
+ right[i] = a - b;
+ }
+}
+
+static void drmp3_L3_intensity_stereo_band(float *left, int n, float kl, float kr)
+{
+ int i;
+ for (i = 0; i < n; i++)
+ {
+ left[i + 576] = left[i]*kr;
+ left[i] = left[i]*kl;
+ }
+}
+
+static void drmp3_L3_stereo_top_band(const float *right, const drmp3_uint8 *sfb, int nbands, int max_band[3])
+{
+ int i, k;
+
+ max_band[0] = max_band[1] = max_band[2] = -1;
+
+ for (i = 0; i < nbands; i++)
+ {
+ for (k = 0; k < sfb[i]; k += 2)
+ {
+ if (right[k] != 0 || right[k + 1] != 0)
+ {
+ max_band[i % 3] = i;
+ break;
+ }
+ }
+ right += sfb[i];
+ }
+}
+
+static void drmp3_L3_stereo_process(float *left, const drmp3_uint8 *ist_pos, const drmp3_uint8 *sfb, const drmp3_uint8 *hdr, int max_band[3], int mpeg2_sh)
+{
+ static const float g_pan[7*2] = { 0,1,0.21132487f,0.78867513f,0.36602540f,0.63397460f,0.5f,0.5f,0.63397460f,0.36602540f,0.78867513f,0.21132487f,1,0 };
+ unsigned i, max_pos = DRMP3_HDR_TEST_MPEG1(hdr) ? 7 : 64;
+
+ for (i = 0; sfb[i]; i++)
+ {
+ unsigned ipos = ist_pos[i];
+ if ((int)i > max_band[i % 3] && ipos < max_pos)
+ {
+ float kl, kr, s = DRMP3_HDR_TEST_MS_STEREO(hdr) ? 1.41421356f : 1;
+ if (DRMP3_HDR_TEST_MPEG1(hdr))
+ {
+ kl = g_pan[2*ipos];
+ kr = g_pan[2*ipos + 1];
+ } else
+ {
+ kl = 1;
+ kr = drmp3_L3_ldexp_q2(1, (ipos + 1) >> 1 << mpeg2_sh);
+ if (ipos & 1)
+ {
+ kl = kr;
+ kr = 1;
+ }
+ }
+ drmp3_L3_intensity_stereo_band(left, sfb[i], kl*s, kr*s);
+ } else if (DRMP3_HDR_TEST_MS_STEREO(hdr))
+ {
+ drmp3_L3_midside_stereo(left, sfb[i]);
+ }
+ left += sfb[i];
+ }
+}
+
+static void drmp3_L3_intensity_stereo(float *left, drmp3_uint8 *ist_pos, const drmp3_L3_gr_info *gr, const drmp3_uint8 *hdr)
+{
+ int max_band[3], n_sfb = gr->n_long_sfb + gr->n_short_sfb;
+ int i, max_blocks = gr->n_short_sfb ? 3 : 1;
+
+ drmp3_L3_stereo_top_band(left + 576, gr->sfbtab, n_sfb, max_band);
+ if (gr->n_long_sfb)
+ {
+ max_band[0] = max_band[1] = max_band[2] = DRMP3_MAX(DRMP3_MAX(max_band[0], max_band[1]), max_band[2]);
+ }
+ for (i = 0; i < max_blocks; i++)
+ {
+ int default_pos = DRMP3_HDR_TEST_MPEG1(hdr) ? 3 : 0;
+ int itop = n_sfb - max_blocks + i;
+ int prev = itop - max_blocks;
+ ist_pos[itop] = (drmp3_uint8)(max_band[i] >= prev ? default_pos : ist_pos[prev]);
+ }
+ drmp3_L3_stereo_process(left, ist_pos, gr->sfbtab, hdr, max_band, gr[1].scalefac_compress & 1);
+}
+
+static void drmp3_L3_reorder(float *grbuf, float *scratch, const drmp3_uint8 *sfb)
+{
+ int i, len;
+ float *src = grbuf, *dst = scratch;
+
+ for (;0 != (len = *sfb); sfb += 3, src += 2*len)
+ {
+ for (i = 0; i < len; i++, src++)
+ {
+ *dst++ = src[0*len];
+ *dst++ = src[1*len];
+ *dst++ = src[2*len];
+ }
+ }
+ memcpy(grbuf, scratch, (dst - scratch)*sizeof(float));
+}
+
+static void drmp3_L3_antialias(float *grbuf, int nbands)
+{
+ static const float g_aa[2][8] = {
+ {0.85749293f,0.88174200f,0.94962865f,0.98331459f,0.99551782f,0.99916056f,0.99989920f,0.99999316f},
+ {0.51449576f,0.47173197f,0.31337745f,0.18191320f,0.09457419f,0.04096558f,0.01419856f,0.00369997f}
+ };
+
+ for (; nbands > 0; nbands--, grbuf += 18)
+ {
+ int i = 0;
+#if DRMP3_HAVE_SIMD
+ if (drmp3_have_simd()) for (; i < 8; i += 4)
+ {
+ drmp3_f4 vu = DRMP3_VLD(grbuf + 18 + i);
+ drmp3_f4 vd = DRMP3_VLD(grbuf + 14 - i);
+ drmp3_f4 vc0 = DRMP3_VLD(g_aa[0] + i);
+ drmp3_f4 vc1 = DRMP3_VLD(g_aa[1] + i);
+ vd = DRMP3_VREV(vd);
+ DRMP3_VSTORE(grbuf + 18 + i, DRMP3_VSUB(DRMP3_VMUL(vu, vc0), DRMP3_VMUL(vd, vc1)));
+ vd = DRMP3_VADD(DRMP3_VMUL(vu, vc1), DRMP3_VMUL(vd, vc0));
+ DRMP3_VSTORE(grbuf + 14 - i, DRMP3_VREV(vd));
+ }
+#endif
+#ifndef DR_MP3_ONLY_SIMD
+ for(; i < 8; i++)
+ {
+ float u = grbuf[18 + i];
+ float d = grbuf[17 - i];
+ grbuf[18 + i] = u*g_aa[0][i] - d*g_aa[1][i];
+ grbuf[17 - i] = u*g_aa[1][i] + d*g_aa[0][i];
+ }
+#endif
+ }
+}
+
+static void drmp3_L3_dct3_9(float *y)
+{
+ float s0, s1, s2, s3, s4, s5, s6, s7, s8, t0, t2, t4;
+
+ s0 = y[0]; s2 = y[2]; s4 = y[4]; s6 = y[6]; s8 = y[8];
+ t0 = s0 + s6*0.5f;
+ s0 -= s6;
+ t4 = (s4 + s2)*0.93969262f;
+ t2 = (s8 + s2)*0.76604444f;
+ s6 = (s4 - s8)*0.17364818f;
+ s4 += s8 - s2;
+
+ s2 = s0 - s4*0.5f;
+ y[4] = s4 + s0;
+ s8 = t0 - t2 + s6;
+ s0 = t0 - t4 + t2;
+ s4 = t0 + t4 - s6;
+
+ s1 = y[1]; s3 = y[3]; s5 = y[5]; s7 = y[7];
+
+ s3 *= 0.86602540f;
+ t0 = (s5 + s1)*0.98480775f;
+ t4 = (s5 - s7)*0.34202014f;
+ t2 = (s1 + s7)*0.64278761f;
+ s1 = (s1 - s5 - s7)*0.86602540f;
+
+ s5 = t0 - s3 - t2;
+ s7 = t4 - s3 - t0;
+ s3 = t4 + s3 - t2;
+
+ y[0] = s4 - s7;
+ y[1] = s2 + s1;
+ y[2] = s0 - s3;
+ y[3] = s8 + s5;
+ y[5] = s8 - s5;
+ y[6] = s0 + s3;
+ y[7] = s2 - s1;
+ y[8] = s4 + s7;
+}
+
+static void drmp3_L3_imdct36(float *grbuf, float *overlap, const float *window, int nbands)
+{
+ int i, j;
+ static const float g_twid9[18] = {
+ 0.73727734f,0.79335334f,0.84339145f,0.88701083f,0.92387953f,0.95371695f,0.97629601f,0.99144486f,0.99904822f,0.67559021f,0.60876143f,0.53729961f,0.46174861f,0.38268343f,0.30070580f,0.21643961f,0.13052619f,0.04361938f
+ };
+
+ for (j = 0; j < nbands; j++, grbuf += 18, overlap += 9)
+ {
+ float co[9], si[9];
+ co[0] = -grbuf[0];
+ si[0] = grbuf[17];
+ for (i = 0; i < 4; i++)
+ {
+ si[8 - 2*i] = grbuf[4*i + 1] - grbuf[4*i + 2];
+ co[1 + 2*i] = grbuf[4*i + 1] + grbuf[4*i + 2];
+ si[7 - 2*i] = grbuf[4*i + 4] - grbuf[4*i + 3];
+ co[2 + 2*i] = -(grbuf[4*i + 3] + grbuf[4*i + 4]);
+ }
+ drmp3_L3_dct3_9(co);
+ drmp3_L3_dct3_9(si);
+
+ si[1] = -si[1];
+ si[3] = -si[3];
+ si[5] = -si[5];
+ si[7] = -si[7];
+
+ i = 0;
+
+#if DRMP3_HAVE_SIMD
+ if (drmp3_have_simd()) for (; i < 8; i += 4)
+ {
+ drmp3_f4 vovl = DRMP3_VLD(overlap + i);
+ drmp3_f4 vc = DRMP3_VLD(co + i);
+ drmp3_f4 vs = DRMP3_VLD(si + i);
+ drmp3_f4 vr0 = DRMP3_VLD(g_twid9 + i);
+ drmp3_f4 vr1 = DRMP3_VLD(g_twid9 + 9 + i);
+ drmp3_f4 vw0 = DRMP3_VLD(window + i);
+ drmp3_f4 vw1 = DRMP3_VLD(window + 9 + i);
+ drmp3_f4 vsum = DRMP3_VADD(DRMP3_VMUL(vc, vr1), DRMP3_VMUL(vs, vr0));
+ DRMP3_VSTORE(overlap + i, DRMP3_VSUB(DRMP3_VMUL(vc, vr0), DRMP3_VMUL(vs, vr1)));
+ DRMP3_VSTORE(grbuf + i, DRMP3_VSUB(DRMP3_VMUL(vovl, vw0), DRMP3_VMUL(vsum, vw1)));
+ vsum = DRMP3_VADD(DRMP3_VMUL(vovl, vw1), DRMP3_VMUL(vsum, vw0));
+ DRMP3_VSTORE(grbuf + 14 - i, DRMP3_VREV(vsum));
+ }
+#endif
+ for (; i < 9; i++)
+ {
+ float ovl = overlap[i];
+ float sum = co[i]*g_twid9[9 + i] + si[i]*g_twid9[0 + i];
+ overlap[i] = co[i]*g_twid9[0 + i] - si[i]*g_twid9[9 + i];
+ grbuf[i] = ovl*window[0 + i] - sum*window[9 + i];
+ grbuf[17 - i] = ovl*window[9 + i] + sum*window[0 + i];
+ }
+ }
+}
+
+static void drmp3_L3_idct3(float x0, float x1, float x2, float *dst)
+{
+ float m1 = x1*0.86602540f;
+ float a1 = x0 - x2*0.5f;
+ dst[1] = x0 + x2;
+ dst[0] = a1 + m1;
+ dst[2] = a1 - m1;
+}
+
+static void drmp3_L3_imdct12(float *x, float *dst, float *overlap)
+{
+ static const float g_twid3[6] = { 0.79335334f,0.92387953f,0.99144486f, 0.60876143f,0.38268343f,0.13052619f };
+ float co[3], si[3];
+ int i;
+
+ drmp3_L3_idct3(-x[0], x[6] + x[3], x[12] + x[9], co);
+ drmp3_L3_idct3(x[15], x[12] - x[9], x[6] - x[3], si);
+ si[1] = -si[1];
+
+ for (i = 0; i < 3; i++)
+ {
+ float ovl = overlap[i];
+ float sum = co[i]*g_twid3[3 + i] + si[i]*g_twid3[0 + i];
+ overlap[i] = co[i]*g_twid3[0 + i] - si[i]*g_twid3[3 + i];
+ dst[i] = ovl*g_twid3[2 - i] - sum*g_twid3[5 - i];
+ dst[5 - i] = ovl*g_twid3[5 - i] + sum*g_twid3[2 - i];
+ }
+}
+
+static void drmp3_L3_imdct_short(float *grbuf, float *overlap, int nbands)
+{
+ for (;nbands > 0; nbands--, overlap += 9, grbuf += 18)
+ {
+ float tmp[18];
+ memcpy(tmp, grbuf, sizeof(tmp));
+ memcpy(grbuf, overlap, 6*sizeof(float));
+ drmp3_L3_imdct12(tmp, grbuf + 6, overlap + 6);
+ drmp3_L3_imdct12(tmp + 1, grbuf + 12, overlap + 6);
+ drmp3_L3_imdct12(tmp + 2, overlap, overlap + 6);
+ }
+}
+
+static void drmp3_L3_change_sign(float *grbuf)
+{
+ int b, i;
+ for (b = 0, grbuf += 18; b < 32; b += 2, grbuf += 36)
+ for (i = 1; i < 18; i += 2)
+ grbuf[i] = -grbuf[i];
+}
+
+static void drmp3_L3_imdct_gr(float *grbuf, float *overlap, unsigned block_type, unsigned n_long_bands)
+{
+ static const float g_mdct_window[2][18] = {
+ { 0.99904822f,0.99144486f,0.97629601f,0.95371695f,0.92387953f,0.88701083f,0.84339145f,0.79335334f,0.73727734f,0.04361938f,0.13052619f,0.21643961f,0.30070580f,0.38268343f,0.46174861f,0.53729961f,0.60876143f,0.67559021f },
+ { 1,1,1,1,1,1,0.99144486f,0.92387953f,0.79335334f,0,0,0,0,0,0,0.13052619f,0.38268343f,0.60876143f }
+ };
+ if (n_long_bands)
+ {
+ drmp3_L3_imdct36(grbuf, overlap, g_mdct_window[0], n_long_bands);
+ grbuf += 18*n_long_bands;
+ overlap += 9*n_long_bands;
+ }
+ if (block_type == DRMP3_SHORT_BLOCK_TYPE)
+ drmp3_L3_imdct_short(grbuf, overlap, 32 - n_long_bands);
+ else
+ drmp3_L3_imdct36(grbuf, overlap, g_mdct_window[block_type == DRMP3_STOP_BLOCK_TYPE], 32 - n_long_bands);
+}
+
+static void drmp3_L3_save_reservoir(drmp3dec *h, drmp3dec_scratch *s)
+{
+ int pos = (s->bs.pos + 7)/8u;
+ int remains = s->bs.limit/8u - pos;
+ if (remains > DRMP3_MAX_BITRESERVOIR_BYTES)
+ {
+ pos += remains - DRMP3_MAX_BITRESERVOIR_BYTES;
+ remains = DRMP3_MAX_BITRESERVOIR_BYTES;
+ }
+ if (remains > 0)
+ {
+ memmove(h->reserv_buf, s->maindata + pos, remains);
+ }
+ h->reserv = remains;
+}
+
+static int drmp3_L3_restore_reservoir(drmp3dec *h, drmp3_bs *bs, drmp3dec_scratch *s, int main_data_begin)
+{
+ int frame_bytes = (bs->limit - bs->pos)/8;
+ int bytes_have = DRMP3_MIN(h->reserv, main_data_begin);
+ memcpy(s->maindata, h->reserv_buf + DRMP3_MAX(0, h->reserv - main_data_begin), DRMP3_MIN(h->reserv, main_data_begin));
+ memcpy(s->maindata + bytes_have, bs->buf + bs->pos/8, frame_bytes);
+ drmp3_bs_init(&s->bs, s->maindata, bytes_have + frame_bytes);
+ return h->reserv >= main_data_begin;
+}
+
+static void drmp3_L3_decode(drmp3dec *h, drmp3dec_scratch *s, drmp3_L3_gr_info *gr_info, int nch)
+{
+ int ch;
+
+ for (ch = 0; ch < nch; ch++)
+ {
+ int layer3gr_limit = s->bs.pos + gr_info[ch].part_23_length;
+ drmp3_L3_decode_scalefactors(h->header, s->ist_pos[ch], &s->bs, gr_info + ch, s->scf, ch);
+ drmp3_L3_huffman(s->grbuf[ch], &s->bs, gr_info + ch, s->scf, layer3gr_limit);
+ }
+
+ if (DRMP3_HDR_TEST_I_STEREO(h->header))
+ {
+ drmp3_L3_intensity_stereo(s->grbuf[0], s->ist_pos[1], gr_info, h->header);
+ } else if (DRMP3_HDR_IS_MS_STEREO(h->header))
+ {
+ drmp3_L3_midside_stereo(s->grbuf[0], 576);
+ }
+
+ for (ch = 0; ch < nch; ch++, gr_info++)
+ {
+ int aa_bands = 31;
+ int n_long_bands = (gr_info->mixed_block_flag ? 2 : 0) << (int)(DRMP3_HDR_GET_MY_SAMPLE_RATE(h->header) == 2);
+
+ if (gr_info->n_short_sfb)
+ {
+ aa_bands = n_long_bands - 1;
+ drmp3_L3_reorder(s->grbuf[ch] + n_long_bands*18, s->syn[0], gr_info->sfbtab + gr_info->n_long_sfb);
+ }
+
+ drmp3_L3_antialias(s->grbuf[ch], aa_bands);
+ drmp3_L3_imdct_gr(s->grbuf[ch], h->mdct_overlap[ch], gr_info->block_type, n_long_bands);
+ drmp3_L3_change_sign(s->grbuf[ch]);
+ }
+}
+
+static void drmp3d_DCT_II(float *grbuf, int n)
+{
+ static const float g_sec[24] = {
+ 10.19000816f,0.50060302f,0.50241929f,3.40760851f,0.50547093f,0.52249861f,2.05778098f,0.51544732f,0.56694406f,1.48416460f,0.53104258f,0.64682180f,1.16943991f,0.55310392f,0.78815460f,0.97256821f,0.58293498f,1.06067765f,0.83934963f,0.62250412f,1.72244716f,0.74453628f,0.67480832f,5.10114861f
+ };
+ int i, k = 0;
+#if DRMP3_HAVE_SIMD
+ if (drmp3_have_simd()) for (; k < n; k += 4)
+ {
+ drmp3_f4 t[4][8], *x;
+ float *y = grbuf + k;
+
+ for (x = t[0], i = 0; i < 8; i++, x++)
+ {
+ drmp3_f4 x0 = DRMP3_VLD(&y[i*18]);
+ drmp3_f4 x1 = DRMP3_VLD(&y[(15 - i)*18]);
+ drmp3_f4 x2 = DRMP3_VLD(&y[(16 + i)*18]);
+ drmp3_f4 x3 = DRMP3_VLD(&y[(31 - i)*18]);
+ drmp3_f4 t0 = DRMP3_VADD(x0, x3);
+ drmp3_f4 t1 = DRMP3_VADD(x1, x2);
+ drmp3_f4 t2 = DRMP3_VMUL_S(DRMP3_VSUB(x1, x2), g_sec[3*i + 0]);
+ drmp3_f4 t3 = DRMP3_VMUL_S(DRMP3_VSUB(x0, x3), g_sec[3*i + 1]);
+ x[0] = DRMP3_VADD(t0, t1);
+ x[8] = DRMP3_VMUL_S(DRMP3_VSUB(t0, t1), g_sec[3*i + 2]);
+ x[16] = DRMP3_VADD(t3, t2);
+ x[24] = DRMP3_VMUL_S(DRMP3_VSUB(t3, t2), g_sec[3*i + 2]);
+ }
+ for (x = t[0], i = 0; i < 4; i++, x += 8)
+ {
+ drmp3_f4 x0 = x[0], x1 = x[1], x2 = x[2], x3 = x[3], x4 = x[4], x5 = x[5], x6 = x[6], x7 = x[7], xt;
+ xt = DRMP3_VSUB(x0, x7); x0 = DRMP3_VADD(x0, x7);
+ x7 = DRMP3_VSUB(x1, x6); x1 = DRMP3_VADD(x1, x6);
+ x6 = DRMP3_VSUB(x2, x5); x2 = DRMP3_VADD(x2, x5);
+ x5 = DRMP3_VSUB(x3, x4); x3 = DRMP3_VADD(x3, x4);
+ x4 = DRMP3_VSUB(x0, x3); x0 = DRMP3_VADD(x0, x3);
+ x3 = DRMP3_VSUB(x1, x2); x1 = DRMP3_VADD(x1, x2);
+ x[0] = DRMP3_VADD(x0, x1);
+ x[4] = DRMP3_VMUL_S(DRMP3_VSUB(x0, x1), 0.70710677f);
+ x5 = DRMP3_VADD(x5, x6);
+ x6 = DRMP3_VMUL_S(DRMP3_VADD(x6, x7), 0.70710677f);
+ x7 = DRMP3_VADD(x7, xt);
+ x3 = DRMP3_VMUL_S(DRMP3_VADD(x3, x4), 0.70710677f);
+ x5 = DRMP3_VSUB(x5, DRMP3_VMUL_S(x7, 0.198912367f)); /* rotate by PI/8 */
+ x7 = DRMP3_VADD(x7, DRMP3_VMUL_S(x5, 0.382683432f));
+ x5 = DRMP3_VSUB(x5, DRMP3_VMUL_S(x7, 0.198912367f));
+ x0 = DRMP3_VSUB(xt, x6); xt = DRMP3_VADD(xt, x6);
+ x[1] = DRMP3_VMUL_S(DRMP3_VADD(xt, x7), 0.50979561f);
+ x[2] = DRMP3_VMUL_S(DRMP3_VADD(x4, x3), 0.54119611f);
+ x[3] = DRMP3_VMUL_S(DRMP3_VSUB(x0, x5), 0.60134488f);
+ x[5] = DRMP3_VMUL_S(DRMP3_VADD(x0, x5), 0.89997619f);
+ x[6] = DRMP3_VMUL_S(DRMP3_VSUB(x4, x3), 1.30656302f);
+ x[7] = DRMP3_VMUL_S(DRMP3_VSUB(xt, x7), 2.56291556f);
+ }
+
+ if (k > n - 3)
+ {
+#if DRMP3_HAVE_SSE
+#define DRMP3_VSAVE2(i, v) _mm_storel_pi((__m64 *)(void*)&y[i*18], v)
+#else
+#define DRMP3_VSAVE2(i, v) vst1_f32((float32_t *)&y[i*18], vget_low_f32(v))
+#endif
+ for (i = 0; i < 7; i++, y += 4*18)
+ {
+ drmp3_f4 s = DRMP3_VADD(t[3][i], t[3][i + 1]);
+ DRMP3_VSAVE2(0, t[0][i]);
+ DRMP3_VSAVE2(1, DRMP3_VADD(t[2][i], s));
+ DRMP3_VSAVE2(2, DRMP3_VADD(t[1][i], t[1][i + 1]));
+ DRMP3_VSAVE2(3, DRMP3_VADD(t[2][1 + i], s));
+ }
+ DRMP3_VSAVE2(0, t[0][7]);
+ DRMP3_VSAVE2(1, DRMP3_VADD(t[2][7], t[3][7]));
+ DRMP3_VSAVE2(2, t[1][7]);
+ DRMP3_VSAVE2(3, t[3][7]);
+ } else
+ {
+#define DRMP3_VSAVE4(i, v) DRMP3_VSTORE(&y[i*18], v)
+ for (i = 0; i < 7; i++, y += 4*18)
+ {
+ drmp3_f4 s = DRMP3_VADD(t[3][i], t[3][i + 1]);
+ DRMP3_VSAVE4(0, t[0][i]);
+ DRMP3_VSAVE4(1, DRMP3_VADD(t[2][i], s));
+ DRMP3_VSAVE4(2, DRMP3_VADD(t[1][i], t[1][i + 1]));
+ DRMP3_VSAVE4(3, DRMP3_VADD(t[2][1 + i], s));
+ }
+ DRMP3_VSAVE4(0, t[0][7]);
+ DRMP3_VSAVE4(1, DRMP3_VADD(t[2][7], t[3][7]));
+ DRMP3_VSAVE4(2, t[1][7]);
+ DRMP3_VSAVE4(3, t[3][7]);
+ }
+ } else
+#endif
+#ifdef DR_MP3_ONLY_SIMD
+ {}
+#else
+ for (; k < n; k++)
+ {
+ float t[4][8], *x, *y = grbuf + k;
+
+ for (x = t[0], i = 0; i < 8; i++, x++)
+ {
+ float x0 = y[i*18];
+ float x1 = y[(15 - i)*18];
+ float x2 = y[(16 + i)*18];
+ float x3 = y[(31 - i)*18];
+ float t0 = x0 + x3;
+ float t1 = x1 + x2;
+ float t2 = (x1 - x2)*g_sec[3*i + 0];
+ float t3 = (x0 - x3)*g_sec[3*i + 1];
+ x[0] = t0 + t1;
+ x[8] = (t0 - t1)*g_sec[3*i + 2];
+ x[16] = t3 + t2;
+ x[24] = (t3 - t2)*g_sec[3*i + 2];
+ }
+ for (x = t[0], i = 0; i < 4; i++, x += 8)
+ {
+ float x0 = x[0], x1 = x[1], x2 = x[2], x3 = x[3], x4 = x[4], x5 = x[5], x6 = x[6], x7 = x[7], xt;
+ xt = x0 - x7; x0 += x7;
+ x7 = x1 - x6; x1 += x6;
+ x6 = x2 - x5; x2 += x5;
+ x5 = x3 - x4; x3 += x4;
+ x4 = x0 - x3; x0 += x3;
+ x3 = x1 - x2; x1 += x2;
+ x[0] = x0 + x1;
+ x[4] = (x0 - x1)*0.70710677f;
+ x5 = x5 + x6;
+ x6 = (x6 + x7)*0.70710677f;
+ x7 = x7 + xt;
+ x3 = (x3 + x4)*0.70710677f;
+ x5 -= x7*0.198912367f; /* rotate by PI/8 */
+ x7 += x5*0.382683432f;
+ x5 -= x7*0.198912367f;
+ x0 = xt - x6; xt += x6;
+ x[1] = (xt + x7)*0.50979561f;
+ x[2] = (x4 + x3)*0.54119611f;
+ x[3] = (x0 - x5)*0.60134488f;
+ x[5] = (x0 + x5)*0.89997619f;
+ x[6] = (x4 - x3)*1.30656302f;
+ x[7] = (xt - x7)*2.56291556f;
+
+ }
+ for (i = 0; i < 7; i++, y += 4*18)
+ {
+ y[0*18] = t[0][i];
+ y[1*18] = t[2][i] + t[3][i] + t[3][i + 1];
+ y[2*18] = t[1][i] + t[1][i + 1];
+ y[3*18] = t[2][i + 1] + t[3][i] + t[3][i + 1];
+ }
+ y[0*18] = t[0][7];
+ y[1*18] = t[2][7] + t[3][7];
+ y[2*18] = t[1][7];
+ y[3*18] = t[3][7];
+ }
+#endif
+}
+
+#ifndef DR_MP3_FLOAT_OUTPUT
+typedef drmp3_int16 drmp3d_sample_t;
+
+static drmp3_int16 drmp3d_scale_pcm(float sample)
+{
+ if (sample >= 32766.5) return (drmp3_int16) 32767;
+ if (sample <= -32767.5) return (drmp3_int16)-32768;
+ drmp3_int16 s = (drmp3_int16)(sample + .5f);
+ s -= (s < 0); /* away from zero, to be compliant */
+ return (drmp3_int16)s;
+}
+#else
+typedef float drmp3d_sample_t;
+
+static float drmp3d_scale_pcm(float sample)
+{
+ return sample*(1.f/32768.f);
+}
+#endif
+
+static void drmp3d_synth_pair(drmp3d_sample_t *pcm, int nch, const float *z)
+{
+ float a;
+ a = (z[14*64] - z[ 0]) * 29;
+ a += (z[ 1*64] + z[13*64]) * 213;
+ a += (z[12*64] - z[ 2*64]) * 459;
+ a += (z[ 3*64] + z[11*64]) * 2037;
+ a += (z[10*64] - z[ 4*64]) * 5153;
+ a += (z[ 5*64] + z[ 9*64]) * 6574;
+ a += (z[ 8*64] - z[ 6*64]) * 37489;
+ a += z[ 7*64] * 75038;
+ pcm[0] = drmp3d_scale_pcm(a);
+
+ z += 2;
+ a = z[14*64] * 104;
+ a += z[12*64] * 1567;
+ a += z[10*64] * 9727;
+ a += z[ 8*64] * 64019;
+ a += z[ 6*64] * -9975;
+ a += z[ 4*64] * -45;
+ a += z[ 2*64] * 146;
+ a += z[ 0*64] * -5;
+ pcm[16*nch] = drmp3d_scale_pcm(a);
+}
+
+static void drmp3d_synth(float *xl, drmp3d_sample_t *dstl, int nch, float *lins)
+{
+ int i;
+ float *xr = xl + 576*(nch - 1);
+ drmp3d_sample_t *dstr = dstl + (nch - 1);
+
+ static const float g_win[] = {
+ -1,26,-31,208,218,401,-519,2063,2000,4788,-5517,7134,5959,35640,-39336,74992,
+ -1,24,-35,202,222,347,-581,2080,1952,4425,-5879,7640,5288,33791,-41176,74856,
+ -1,21,-38,196,225,294,-645,2087,1893,4063,-6237,8092,4561,31947,-43006,74630,
+ -1,19,-41,190,227,244,-711,2085,1822,3705,-6589,8492,3776,30112,-44821,74313,
+ -1,17,-45,183,228,197,-779,2075,1739,3351,-6935,8840,2935,28289,-46617,73908,
+ -1,16,-49,176,228,153,-848,2057,1644,3004,-7271,9139,2037,26482,-48390,73415,
+ -2,14,-53,169,227,111,-919,2032,1535,2663,-7597,9389,1082,24694,-50137,72835,
+ -2,13,-58,161,224,72,-991,2001,1414,2330,-7910,9592,70,22929,-51853,72169,
+ -2,11,-63,154,221,36,-1064,1962,1280,2006,-8209,9750,-998,21189,-53534,71420,
+ -2,10,-68,147,215,2,-1137,1919,1131,1692,-8491,9863,-2122,19478,-55178,70590,
+ -3,9,-73,139,208,-29,-1210,1870,970,1388,-8755,9935,-3300,17799,-56778,69679,
+ -3,8,-79,132,200,-57,-1283,1817,794,1095,-8998,9966,-4533,16155,-58333,68692,
+ -4,7,-85,125,189,-83,-1356,1759,605,814,-9219,9959,-5818,14548,-59838,67629,
+ -4,7,-91,117,177,-106,-1428,1698,402,545,-9416,9916,-7154,12980,-61289,66494,
+ -5,6,-97,111,163,-127,-1498,1634,185,288,-9585,9838,-8540,11455,-62684,65290
+ };
+ float *zlin = lins + 15*64;
+ const float *w = g_win;
+
+ zlin[4*15] = xl[18*16];
+ zlin[4*15 + 1] = xr[18*16];
+ zlin[4*15 + 2] = xl[0];
+ zlin[4*15 + 3] = xr[0];
+
+ zlin[4*31] = xl[1 + 18*16];
+ zlin[4*31 + 1] = xr[1 + 18*16];
+ zlin[4*31 + 2] = xl[1];
+ zlin[4*31 + 3] = xr[1];
+
+ drmp3d_synth_pair(dstr, nch, lins + 4*15 + 1);
+ drmp3d_synth_pair(dstr + 32*nch, nch, lins + 4*15 + 64 + 1);
+ drmp3d_synth_pair(dstl, nch, lins + 4*15);
+ drmp3d_synth_pair(dstl + 32*nch, nch, lins + 4*15 + 64);
+
+#if DRMP3_HAVE_SIMD
+ if (drmp3_have_simd()) for (i = 14; i >= 0; i--)
+ {
+#define DRMP3_VLOAD(k) drmp3_f4 w0 = DRMP3_VSET(*w++); drmp3_f4 w1 = DRMP3_VSET(*w++); drmp3_f4 vz = DRMP3_VLD(&zlin[4*i - 64*k]); drmp3_f4 vy = DRMP3_VLD(&zlin[4*i - 64*(15 - k)]);
+#define DRMP3_V0(k) { DRMP3_VLOAD(k) b = DRMP3_VADD(DRMP3_VMUL(vz, w1), DRMP3_VMUL(vy, w0)) ; a = DRMP3_VSUB(DRMP3_VMUL(vz, w0), DRMP3_VMUL(vy, w1)); }
+#define DRMP3_V1(k) { DRMP3_VLOAD(k) b = DRMP3_VADD(b, DRMP3_VADD(DRMP3_VMUL(vz, w1), DRMP3_VMUL(vy, w0))); a = DRMP3_VADD(a, DRMP3_VSUB(DRMP3_VMUL(vz, w0), DRMP3_VMUL(vy, w1))); }
+#define DRMP3_V2(k) { DRMP3_VLOAD(k) b = DRMP3_VADD(b, DRMP3_VADD(DRMP3_VMUL(vz, w1), DRMP3_VMUL(vy, w0))); a = DRMP3_VADD(a, DRMP3_VSUB(DRMP3_VMUL(vy, w1), DRMP3_VMUL(vz, w0))); }
+ drmp3_f4 a, b;
+ zlin[4*i] = xl[18*(31 - i)];
+ zlin[4*i + 1] = xr[18*(31 - i)];
+ zlin[4*i + 2] = xl[1 + 18*(31 - i)];
+ zlin[4*i + 3] = xr[1 + 18*(31 - i)];
+ zlin[4*i + 64] = xl[1 + 18*(1 + i)];
+ zlin[4*i + 64 + 1] = xr[1 + 18*(1 + i)];
+ zlin[4*i - 64 + 2] = xl[18*(1 + i)];
+ zlin[4*i - 64 + 3] = xr[18*(1 + i)];
+
+ DRMP3_V0(0) DRMP3_V2(1) DRMP3_V1(2) DRMP3_V2(3) DRMP3_V1(4) DRMP3_V2(5) DRMP3_V1(6) DRMP3_V2(7)
+
+ {
+#ifndef DR_MP3_FLOAT_OUTPUT
+#if DRMP3_HAVE_SSE
+ static const drmp3_f4 g_max = { 32767.0f, 32767.0f, 32767.0f, 32767.0f };
+ static const drmp3_f4 g_min = { -32768.0f, -32768.0f, -32768.0f, -32768.0f };
+ __m128i pcm8 = _mm_packs_epi32(_mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(a, g_max), g_min)),
+ _mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(b, g_max), g_min)));
+ dstr[(15 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 1);
+ dstr[(17 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 5);
+ dstl[(15 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 0);
+ dstl[(17 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 4);
+ dstr[(47 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 3);
+ dstr[(49 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 7);
+ dstl[(47 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 2);
+ dstl[(49 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 6);
+#else
+ int16x4_t pcma, pcmb;
+ a = DRMP3_VADD(a, DRMP3_VSET(0.5f));
+ b = DRMP3_VADD(b, DRMP3_VSET(0.5f));
+ pcma = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(a), vreinterpretq_s32_u32(vcltq_f32(a, DRMP3_VSET(0)))));
+ pcmb = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(b), vreinterpretq_s32_u32(vcltq_f32(b, DRMP3_VSET(0)))));
+ vst1_lane_s16(dstr + (15 - i)*nch, pcma, 1);
+ vst1_lane_s16(dstr + (17 + i)*nch, pcmb, 1);
+ vst1_lane_s16(dstl + (15 - i)*nch, pcma, 0);
+ vst1_lane_s16(dstl + (17 + i)*nch, pcmb, 0);
+ vst1_lane_s16(dstr + (47 - i)*nch, pcma, 3);
+ vst1_lane_s16(dstr + (49 + i)*nch, pcmb, 3);
+ vst1_lane_s16(dstl + (47 - i)*nch, pcma, 2);
+ vst1_lane_s16(dstl + (49 + i)*nch, pcmb, 2);
+#endif
+#else
+ static const drmp3_f4 g_scale = { 1.0f/32768.0f, 1.0f/32768.0f, 1.0f/32768.0f, 1.0f/32768.0f };
+ a = DRMP3_VMUL(a, g_scale);
+ b = DRMP3_VMUL(b, g_scale);
+#if DRMP3_HAVE_SSE
+ _mm_store_ss(dstr + (15 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(1, 1, 1, 1)));
+ _mm_store_ss(dstr + (17 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(1, 1, 1, 1)));
+ _mm_store_ss(dstl + (15 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(0, 0, 0, 0)));
+ _mm_store_ss(dstl + (17 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(0, 0, 0, 0)));
+ _mm_store_ss(dstr + (47 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(3, 3, 3, 3)));
+ _mm_store_ss(dstr + (49 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(3, 3, 3, 3)));
+ _mm_store_ss(dstl + (47 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(2, 2, 2, 2)));
+ _mm_store_ss(dstl + (49 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(2, 2, 2, 2)));
+#else
+ vst1q_lane_f32(dstr + (15 - i)*nch, a, 1);
+ vst1q_lane_f32(dstr + (17 + i)*nch, b, 1);
+ vst1q_lane_f32(dstl + (15 - i)*nch, a, 0);
+ vst1q_lane_f32(dstl + (17 + i)*nch, b, 0);
+ vst1q_lane_f32(dstr + (47 - i)*nch, a, 3);
+ vst1q_lane_f32(dstr + (49 + i)*nch, b, 3);
+ vst1q_lane_f32(dstl + (47 - i)*nch, a, 2);
+ vst1q_lane_f32(dstl + (49 + i)*nch, b, 2);
+#endif
+#endif /* DR_MP3_FLOAT_OUTPUT */
+ }
+ } else
+#endif
+#ifdef DR_MP3_ONLY_SIMD
+ {}
+#else
+ for (i = 14; i >= 0; i--)
+ {
+#define DRMP3_LOAD(k) float w0 = *w++; float w1 = *w++; float *vz = &zlin[4*i - k*64]; float *vy = &zlin[4*i - (15 - k)*64];
+#define DRMP3_S0(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] = vz[j]*w1 + vy[j]*w0, a[j] = vz[j]*w0 - vy[j]*w1; }
+#define DRMP3_S1(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] += vz[j]*w1 + vy[j]*w0, a[j] += vz[j]*w0 - vy[j]*w1; }
+#define DRMP3_S2(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] += vz[j]*w1 + vy[j]*w0, a[j] += vy[j]*w1 - vz[j]*w0; }
+ float a[4], b[4];
+
+ zlin[4*i] = xl[18*(31 - i)];
+ zlin[4*i + 1] = xr[18*(31 - i)];
+ zlin[4*i + 2] = xl[1 + 18*(31 - i)];
+ zlin[4*i + 3] = xr[1 + 18*(31 - i)];
+ zlin[4*(i + 16)] = xl[1 + 18*(1 + i)];
+ zlin[4*(i + 16) + 1] = xr[1 + 18*(1 + i)];
+ zlin[4*(i - 16) + 2] = xl[18*(1 + i)];
+ zlin[4*(i - 16) + 3] = xr[18*(1 + i)];
+
+ DRMP3_S0(0) DRMP3_S2(1) DRMP3_S1(2) DRMP3_S2(3) DRMP3_S1(4) DRMP3_S2(5) DRMP3_S1(6) DRMP3_S2(7)
+
+ dstr[(15 - i)*nch] = drmp3d_scale_pcm(a[1]);
+ dstr[(17 + i)*nch] = drmp3d_scale_pcm(b[1]);
+ dstl[(15 - i)*nch] = drmp3d_scale_pcm(a[0]);
+ dstl[(17 + i)*nch] = drmp3d_scale_pcm(b[0]);
+ dstr[(47 - i)*nch] = drmp3d_scale_pcm(a[3]);
+ dstr[(49 + i)*nch] = drmp3d_scale_pcm(b[3]);
+ dstl[(47 - i)*nch] = drmp3d_scale_pcm(a[2]);
+ dstl[(49 + i)*nch] = drmp3d_scale_pcm(b[2]);
+ }
+#endif
+}
+
+static void drmp3d_synth_granule(float *qmf_state, float *grbuf, int nbands, int nch, drmp3d_sample_t *pcm, float *lins)
+{
+ int i;
+ for (i = 0; i < nch; i++)
+ {
+ drmp3d_DCT_II(grbuf + 576*i, nbands);
+ }
+
+ memcpy(lins, qmf_state, sizeof(float)*15*64);
+
+ for (i = 0; i < nbands; i += 2)
+ {
+ drmp3d_synth(grbuf + i, pcm + 32*nch*i, nch, lins + i*64);
+ }
+#ifndef DR_MP3_NONSTANDARD_BUT_LOGICAL
+ if (nch == 1)
+ {
+ for (i = 0; i < 15*64; i += 2)
+ {
+ qmf_state[i] = lins[nbands*64 + i];
+ }
+ } else
+#endif
+ {
+ memcpy(qmf_state, lins + nbands*64, sizeof(float)*15*64);
+ }
+}
+
+static int drmp3d_match_frame(const drmp3_uint8 *hdr, int mp3_bytes, int frame_bytes)
+{
+ int i, nmatch;
+ for (i = 0, nmatch = 0; nmatch < DRMP3_MAX_FRAME_SYNC_MATCHES; nmatch++)
+ {
+ i += drmp3_hdr_frame_bytes(hdr + i, frame_bytes) + drmp3_hdr_padding(hdr + i);
+ if (i + DRMP3_HDR_SIZE > mp3_bytes)
+ return nmatch > 0;
+ if (!drmp3_hdr_compare(hdr, hdr + i))
+ return 0;
+ }
+ return 1;
+}
+
+static int drmp3d_find_frame(const drmp3_uint8 *mp3, int mp3_bytes, int *free_format_bytes, int *ptr_frame_bytes)
+{
+ int i, k;
+ for (i = 0; i < mp3_bytes - DRMP3_HDR_SIZE; i++, mp3++)
+ {
+ if (drmp3_hdr_valid(mp3))
+ {
+ int frame_bytes = drmp3_hdr_frame_bytes(mp3, *free_format_bytes);
+ int frame_and_padding = frame_bytes + drmp3_hdr_padding(mp3);
+
+ for (k = DRMP3_HDR_SIZE; !frame_bytes && k < DRMP3_MAX_FREE_FORMAT_FRAME_SIZE && i + 2*k < mp3_bytes - DRMP3_HDR_SIZE; k++)
+ {
+ if (drmp3_hdr_compare(mp3, mp3 + k))
+ {
+ int fb = k - drmp3_hdr_padding(mp3);
+ int nextfb = fb + drmp3_hdr_padding(mp3 + k);
+ if (i + k + nextfb + DRMP3_HDR_SIZE > mp3_bytes || !drmp3_hdr_compare(mp3, mp3 + k + nextfb))
+ continue;
+ frame_and_padding = k;
+ frame_bytes = fb;
+ *free_format_bytes = fb;
+ }
+ }
+
+ if ((frame_bytes && i + frame_and_padding <= mp3_bytes &&
+ drmp3d_match_frame(mp3, mp3_bytes - i, frame_bytes)) ||
+ (!i && frame_and_padding == mp3_bytes))
+ {
+ *ptr_frame_bytes = frame_and_padding;
+ return i;
+ }
+ *free_format_bytes = 0;
+ }
+ }
+ *ptr_frame_bytes = 0;
+ return i;
+}
+
+void drmp3dec_init(drmp3dec *dec)
+{
+ dec->header[0] = 0;
+}
+
+int drmp3dec_decode_frame(drmp3dec *dec, const unsigned char *mp3, int mp3_bytes, void *pcm, drmp3dec_frame_info *info)
+{
+ int i = 0, igr, frame_size = 0, success = 1;
+ const drmp3_uint8 *hdr;
+ drmp3_bs bs_frame[1];
+ drmp3dec_scratch scratch;
+
+ if (mp3_bytes > 4 && dec->header[0] == 0xff && drmp3_hdr_compare(dec->header, mp3))
+ {
+ frame_size = drmp3_hdr_frame_bytes(mp3, dec->free_format_bytes) + drmp3_hdr_padding(mp3);
+ if (frame_size != mp3_bytes && (frame_size + DRMP3_HDR_SIZE > mp3_bytes || !drmp3_hdr_compare(mp3, mp3 + frame_size)))
+ {
+ frame_size = 0;
+ }
+ }
+ if (!frame_size)
+ {
+ memset(dec, 0, sizeof(drmp3dec));
+ i = drmp3d_find_frame(mp3, mp3_bytes, &dec->free_format_bytes, &frame_size);
+ if (!frame_size || i + frame_size > mp3_bytes)
+ {
+ info->frame_bytes = i;
+ return 0;
+ }
+ }
+
+ hdr = mp3 + i;
+ memcpy(dec->header, hdr, DRMP3_HDR_SIZE);
+ info->frame_bytes = i + frame_size;
+ info->channels = DRMP3_HDR_IS_MONO(hdr) ? 1 : 2;
+ info->hz = drmp3_hdr_sample_rate_hz(hdr);
+ info->layer = 4 - DRMP3_HDR_GET_LAYER(hdr);
+ info->bitrate_kbps = drmp3_hdr_bitrate_kbps(hdr);
+
+ 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
+ if (pcm == NULL) {
+ return drmp3_hdr_frame_samples(hdr);
+ }
+
+ drmp3_L12_scale_info sci[1];
+ drmp3_L12_read_scale_info(hdr, bs_frame, sci);
+
+ memset(scratch.grbuf[0], 0, 576*2*sizeof(float));
+ for (i = 0, igr = 0; igr < 3; igr++)
+ {
+ if (12 == (i += drmp3_L12_dequantize_granule(scratch.grbuf[0] + i, bs_frame, sci, info->layer | 1)))
+ {
+ i = 0;
+ drmp3_L12_apply_scf_384(sci, sci->scf + igr, scratch.grbuf[0]);
+ drmp3d_synth_granule(dec->qmf_state, scratch.grbuf[0], 12, info->channels, (drmp3d_sample_t*)pcm, scratch.syn[0]);
+ memset(scratch.grbuf[0], 0, 576*2*sizeof(float));
+ pcm = DRMP3_OFFSET_PTR(pcm, sizeof(drmp3d_sample_t)*384*info->channels);
+ }
+ if (bs_frame->pos > bs_frame->limit)
+ {
+ drmp3dec_init(dec);
+ return 0;
+ }
+ }
+#endif
+ }
+
+ return success*drmp3_hdr_frame_samples(dec->header);
+}
+
+void drmp3dec_f32_to_s16(const float *in, drmp3_int16 *out, int num_samples)
+{
+ if(num_samples > 0)
+ {
+ int i = 0;
+#if DRMP3_HAVE_SIMD
+ int aligned_count = num_samples & ~7;
+ for(; i < aligned_count; i+=8)
+ {
+ static const drmp3_f4 g_scale = { 32768.0f, 32768.0f, 32768.0f, 32768.0f };
+ drmp3_f4 a = DRMP3_VMUL(DRMP3_VLD(&in[i ]), g_scale);
+ drmp3_f4 b = DRMP3_VMUL(DRMP3_VLD(&in[i+4]), g_scale);
+#if DRMP3_HAVE_SSE
+ static const drmp3_f4 g_max = { 32767.0f, 32767.0f, 32767.0f, 32767.0f };
+ static const drmp3_f4 g_min = { -32768.0f, -32768.0f, -32768.0f, -32768.0f };
+ __m128i pcm8 = _mm_packs_epi32(_mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(a, g_max), g_min)),
+ _mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(b, g_max), g_min)));
+ out[i ] = (drmp3_int16)_mm_extract_epi16(pcm8, 0);
+ out[i+1] = (drmp3_int16)_mm_extract_epi16(pcm8, 1);
+ out[i+2] = (drmp3_int16)_mm_extract_epi16(pcm8, 2);
+ out[i+3] = (drmp3_int16)_mm_extract_epi16(pcm8, 3);
+ out[i+4] = (drmp3_int16)_mm_extract_epi16(pcm8, 4);
+ out[i+5] = (drmp3_int16)_mm_extract_epi16(pcm8, 5);
+ out[i+6] = (drmp3_int16)_mm_extract_epi16(pcm8, 6);
+ out[i+7] = (drmp3_int16)_mm_extract_epi16(pcm8, 7);
+#else
+ int16x4_t pcma, pcmb;
+ a = DRMP3_VADD(a, DRMP3_VSET(0.5f));
+ b = DRMP3_VADD(b, DRMP3_VSET(0.5f));
+ pcma = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(a), vreinterpretq_s32_u32(vcltq_f32(a, DRMP3_VSET(0)))));
+ pcmb = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(b), vreinterpretq_s32_u32(vcltq_f32(b, DRMP3_VSET(0)))));
+ vst1_lane_s16(out+i , pcma, 0);
+ vst1_lane_s16(out+i+1, pcma, 1);
+ vst1_lane_s16(out+i+2, pcma, 2);
+ vst1_lane_s16(out+i+3, pcma, 3);
+ vst1_lane_s16(out+i+4, pcmb, 0);
+ vst1_lane_s16(out+i+5, pcmb, 1);
+ vst1_lane_s16(out+i+6, pcmb, 2);
+ vst1_lane_s16(out+i+7, pcmb, 3);
+#endif
+ }
+#endif
+ for(; i < num_samples; i++)
+ {
+ float sample = in[i] * 32768.0f;
+ if (sample >= 32766.5)
+ out[i] = (drmp3_int16) 32767;
+ else if (sample <= -32767.5)
+ out[i] = (drmp3_int16)-32768;
+ else
+ {
+ short s = (drmp3_int16)(sample + .5f);
+ s -= (s < 0); /* away from zero, to be compliant */
+ out[i] = s;
+ }
+ }
+ }
+}
+
+
+
+///////////////////////////////////////////////////////////////////////////////
+//
+// Main Public API
+//
+///////////////////////////////////////////////////////////////////////////////
+
+#if defined(SIZE_MAX)
+ #define DRMP3_SIZE_MAX SIZE_MAX
+#else
+ #if defined(_WIN64) || defined(_LP64) || defined(__LP64__)
+ #define DRMP3_SIZE_MAX ((drmp3_uint64)0xFFFFFFFFFFFFFFFF)
+ #else
+ #define DRMP3_SIZE_MAX 0xFFFFFFFF
+ #endif
+#endif
+
+// Options.
+#ifndef DR_MP3_DEFAULT_CHANNELS
+#define DR_MP3_DEFAULT_CHANNELS 2
+#endif
+#ifndef DR_MP3_DEFAULT_SAMPLE_RATE
+#define DR_MP3_DEFAULT_SAMPLE_RATE 44100
+#endif
+#ifndef DRMP3_SEEK_LEADING_MP3_FRAMES
+#define DRMP3_SEEK_LEADING_MP3_FRAMES 2
+#endif
+
+
+// Standard library stuff.
+#ifndef DRMP3_ASSERT
+#include
+#define DRMP3_ASSERT(expression) assert(expression)
+#endif
+#ifndef DRMP3_COPY_MEMORY
+#define DRMP3_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz))
+#endif
+#ifndef DRMP3_ZERO_MEMORY
+#define DRMP3_ZERO_MEMORY(p, sz) memset((p), 0, (sz))
+#endif
+#define DRMP3_ZERO_OBJECT(p) DRMP3_ZERO_MEMORY((p), sizeof(*(p)))
+#ifndef DRMP3_MALLOC
+#define DRMP3_MALLOC(sz) malloc((sz))
+#endif
+#ifndef DRMP3_REALLOC
+#define DRMP3_REALLOC(p, sz) realloc((p), (sz))
+#endif
+#ifndef DRMP3_FREE
+#define DRMP3_FREE(p) free((p))
+#endif
+
+#define drmp3_assert DRMP3_ASSERT
+#define drmp3_copy_memory DRMP3_COPY_MEMORY
+#define drmp3_zero_memory DRMP3_ZERO_MEMORY
+#define drmp3_zero_object DRMP3_ZERO_OBJECT
+#define drmp3_malloc DRMP3_MALLOC
+#define drmp3_realloc DRMP3_REALLOC
+
+#define drmp3_countof(x) (sizeof(x) / sizeof(x[0]))
+#define drmp3_max(x, y) (((x) > (y)) ? (x) : (y))
+#define drmp3_min(x, y) (((x) < (y)) ? (x) : (y))
+
+#define DRMP3_DATA_CHUNK_SIZE 16384 // The size in bytes of each chunk of data to read from the MP3 stream. minimp3 recommends 16K.
+
+static inline float drmp3_mix_f32(float x, float y, float a)
+{
+ return x*(1-a) + y*a;
+}
+
+static void drmp3_blend_f32(float* pOut, float* pInA, float* pInB, float factor, drmp3_uint32 channels)
+{
+ for (drmp3_uint32 i = 0; i < channels; ++i) {
+ pOut[i] = drmp3_mix_f32(pInA[i], pInB[i], factor);
+ }
+}
+
+void drmp3_src_cache_init(drmp3_src* pSRC, drmp3_src_cache* pCache)
+{
+ drmp3_assert(pSRC != NULL);
+ drmp3_assert(pCache != NULL);
+
+ pCache->pSRC = pSRC;
+ pCache->cachedFrameCount = 0;
+ pCache->iNextFrame = 0;
+}
+
+drmp3_uint64 drmp3_src_cache_read_frames(drmp3_src_cache* pCache, drmp3_uint64 frameCount, float* pFramesOut)
+{
+ drmp3_assert(pCache != NULL);
+ drmp3_assert(pCache->pSRC != NULL);
+ drmp3_assert(pCache->pSRC->onRead != NULL);
+ drmp3_assert(frameCount > 0);
+ drmp3_assert(pFramesOut != NULL);
+
+ drmp3_uint32 channels = pCache->pSRC->config.channels;
+
+ drmp3_uint64 totalFramesRead = 0;
+ while (frameCount > 0) {
+ // If there's anything in memory go ahead and copy that over first.
+ drmp3_uint64 framesRemainingInMemory = pCache->cachedFrameCount - pCache->iNextFrame;
+ drmp3_uint64 framesToReadFromMemory = frameCount;
+ if (framesToReadFromMemory > framesRemainingInMemory) {
+ framesToReadFromMemory = framesRemainingInMemory;
+ }
+
+ drmp3_copy_memory(pFramesOut, pCache->pCachedFrames + pCache->iNextFrame*channels, (drmp3_uint32)(framesToReadFromMemory * channels * sizeof(float)));
+ pCache->iNextFrame += (drmp3_uint32)framesToReadFromMemory;
+
+ totalFramesRead += framesToReadFromMemory;
+ frameCount -= framesToReadFromMemory;
+ if (frameCount == 0) {
+ break;
+ }
+
+
+ // At this point there are still more frames to read from the client, so we'll need to reload the cache with fresh data.
+ drmp3_assert(frameCount > 0);
+ pFramesOut += framesToReadFromMemory * channels;
+
+ pCache->iNextFrame = 0;
+ pCache->cachedFrameCount = 0;
+
+ drmp3_uint32 framesToReadFromClient = drmp3_countof(pCache->pCachedFrames) / pCache->pSRC->config.channels;
+ if (framesToReadFromClient > pCache->pSRC->config.cacheSizeInFrames) {
+ framesToReadFromClient = pCache->pSRC->config.cacheSizeInFrames;
+ }
+
+ pCache->cachedFrameCount = (drmp3_uint32)pCache->pSRC->onRead(pCache->pSRC, framesToReadFromClient, pCache->pCachedFrames, pCache->pSRC->pUserData);
+
+
+ // Get out of this loop if nothing was able to be retrieved.
+ if (pCache->cachedFrameCount == 0) {
+ break;
+ }
+ }
+
+ return totalFramesRead;
+}
+
+
+drmp3_uint64 drmp3_src_read_frames_passthrough(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, drmp3_bool32 flush);
+drmp3_uint64 drmp3_src_read_frames_linear(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, drmp3_bool32 flush);
+
+drmp3_bool32 drmp3_src_init(const drmp3_src_config* pConfig, drmp3_src_read_proc onRead, void* pUserData, drmp3_src* pSRC)
+{
+ if (pSRC == NULL) return DRMP3_FALSE;
+ drmp3_zero_object(pSRC);
+
+ if (pConfig == NULL || onRead == NULL) return DRMP3_FALSE;
+ if (pConfig->channels == 0 || pConfig->channels > 2) return DRMP3_FALSE;
+
+ pSRC->config = *pConfig;
+ pSRC->onRead = onRead;
+ pSRC->pUserData = pUserData;
+
+ if (pSRC->config.cacheSizeInFrames > DRMP3_SRC_CACHE_SIZE_IN_FRAMES || pSRC->config.cacheSizeInFrames == 0) {
+ pSRC->config.cacheSizeInFrames = DRMP3_SRC_CACHE_SIZE_IN_FRAMES;
+ }
+
+ drmp3_src_cache_init(pSRC, &pSRC->cache);
+ return DRMP3_TRUE;
+}
+
+drmp3_bool32 drmp3_src_set_input_sample_rate(drmp3_src* pSRC, drmp3_uint32 sampleRateIn)
+{
+ if (pSRC == NULL) return DRMP3_FALSE;
+
+ // Must have a sample rate of > 0.
+ if (sampleRateIn == 0) {
+ return DRMP3_FALSE;
+ }
+
+ pSRC->config.sampleRateIn = sampleRateIn;
+ return DRMP3_TRUE;
+}
+
+drmp3_bool32 drmp3_src_set_output_sample_rate(drmp3_src* pSRC, drmp3_uint32 sampleRateOut)
+{
+ if (pSRC == NULL) return DRMP3_FALSE;
+
+ // Must have a sample rate of > 0.
+ if (sampleRateOut == 0) {
+ return DRMP3_FALSE;
+ }
+
+ pSRC->config.sampleRateOut = sampleRateOut;
+ return DRMP3_TRUE;
+}
+
+drmp3_uint64 drmp3_src_read_frames_ex(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, drmp3_bool32 flush)
+{
+ if (pSRC == NULL || frameCount == 0 || pFramesOut == NULL) return 0;
+
+ drmp3_src_algorithm algorithm = pSRC->config.algorithm;
+
+ // Always use passthrough if the sample rates are the same.
+ if (pSRC->config.sampleRateIn == pSRC->config.sampleRateOut) {
+ algorithm = drmp3_src_algorithm_none;
+ }
+
+ // Could just use a function pointer instead of a switch for this...
+ switch (algorithm)
+ {
+ case drmp3_src_algorithm_none: return drmp3_src_read_frames_passthrough(pSRC, frameCount, pFramesOut, flush);
+ case drmp3_src_algorithm_linear: return drmp3_src_read_frames_linear(pSRC, frameCount, pFramesOut, flush);
+ default: return 0;
+ }
+}
+
+drmp3_uint64 drmp3_src_read_frames(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut)
+{
+ return drmp3_src_read_frames_ex(pSRC, frameCount, pFramesOut, DRMP3_FALSE);
+}
+
+drmp3_uint64 drmp3_src_read_frames_passthrough(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, drmp3_bool32 flush)
+{
+ drmp3_assert(pSRC != NULL);
+ drmp3_assert(frameCount > 0);
+ drmp3_assert(pFramesOut != NULL);
+
+ (void)flush; // Passthrough need not care about flushing.
+ return pSRC->onRead(pSRC, frameCount, pFramesOut, pSRC->pUserData);
+}
+
+drmp3_uint64 drmp3_src_read_frames_linear(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, drmp3_bool32 flush)
+{
+ drmp3_assert(pSRC != NULL);
+ drmp3_assert(frameCount > 0);
+ drmp3_assert(pFramesOut != NULL);
+
+ // For linear SRC, the bin is only 2 frames: 1 prior, 1 future.
+
+ // Load the bin if necessary.
+ if (!pSRC->algo.linear.isPrevFramesLoaded) {
+ drmp3_uint64 framesRead = drmp3_src_cache_read_frames(&pSRC->cache, 1, pSRC->bin);
+ if (framesRead == 0) {
+ return 0;
+ }
+ pSRC->algo.linear.isPrevFramesLoaded = DRMP3_TRUE;
+ }
+ if (!pSRC->algo.linear.isNextFramesLoaded) {
+ drmp3_uint64 framesRead = drmp3_src_cache_read_frames(&pSRC->cache, 1, pSRC->bin + pSRC->config.channels);
+ if (framesRead == 0) {
+ return 0;
+ }
+ pSRC->algo.linear.isNextFramesLoaded = DRMP3_TRUE;
+ }
+
+ double factor = (double)pSRC->config.sampleRateIn / pSRC->config.sampleRateOut;
+
+ drmp3_uint64 totalFramesRead = 0;
+ while (frameCount > 0) {
+ // The bin is where the previous and next frames are located.
+ float* pPrevFrame = pSRC->bin;
+ float* pNextFrame = pSRC->bin + pSRC->config.channels;
+
+ drmp3_blend_f32((float*)pFramesOut, pPrevFrame, pNextFrame, (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.
+ drmp3_uint32 framesToReadFromClient = (drmp3_uint32)pSRC->algo.linear.alpha;
+ pSRC->algo.linear.alpha = pSRC->algo.linear.alpha - framesToReadFromClient;
+
+ for (drmp3_uint32 i = 0; i < framesToReadFromClient; ++i) {
+ for (drmp3_uint32 j = 0; j < pSRC->config.channels; ++j) {
+ pPrevFrame[j] = pNextFrame[j];
+ }
+
+ drmp3_uint64 framesRead = drmp3_src_cache_read_frames(&pSRC->cache, 1, pNextFrame);
+ if (framesRead == 0) {
+ for (drmp3_uint32 j = 0; j < pSRC->config.channels; ++j) {
+ pNextFrame[j] = 0;
+ }
+
+ if (pSRC->algo.linear.isNextFramesLoaded) {
+ pSRC->algo.linear.isNextFramesLoaded = DRMP3_FALSE;
+ } else {
+ if (flush) {
+ pSRC->algo.linear.isPrevFramesLoaded = DRMP3_FALSE;
+ }
+ }
+
+ break;
+ }
+ }
+
+ pFramesOut = (drmp3_uint8*)pFramesOut + (1 * pSRC->config.channels * sizeof(float));
+ frameCount -= 1;
+ totalFramesRead += 1;
+
+ // If there's no frames available we need to get out of this loop.
+ if (!pSRC->algo.linear.isNextFramesLoaded && (!flush || !pSRC->algo.linear.isPrevFramesLoaded)) {
+ break;
+ }
+ }
+
+ return totalFramesRead;
+}
+
+
+static size_t drmp3__on_read(drmp3* pMP3, void* pBufferOut, size_t bytesToRead)
+{
+ 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)
+{
+ drmp3_assert(pMP3 != NULL);
+ drmp3_assert(pMP3->onRead != NULL);
+
+ if (pMP3->atEnd) {
+ return 0;
+ }
+
+ drmp3_uint32 pcmFramesRead = 0;
+ do {
+ // minimp3 recommends doing data submission in 16K chunks. If we don't have at least 16K bytes available, get more.
+ if (pMP3->dataSize < DRMP3_DATA_CHUNK_SIZE) {
+ if (pMP3->dataCapacity < DRMP3_DATA_CHUNK_SIZE) {
+ pMP3->dataCapacity = DRMP3_DATA_CHUNK_SIZE;
+ drmp3_uint8* pNewData = (drmp3_uint8*)drmp3_realloc(pMP3->pData, pMP3->dataCapacity);
+ if (pNewData == NULL) {
+ return 0; // Out of memory.
+ }
+
+ pMP3->pData = pNewData;
+ }
+
+ size_t bytesRead = drmp3__on_read(pMP3, pMP3->pData + pMP3->dataSize, (pMP3->dataCapacity - pMP3->dataSize));
+ if (bytesRead == 0) {
+ if (pMP3->dataSize == 0) {
+ pMP3->atEnd = DRMP3_TRUE;
+ return 0; // No data.
+ }
+ }
+
+ pMP3->dataSize += bytesRead;
+ }
+
+ if (pMP3->dataSize > INT_MAX) {
+ pMP3->atEnd = DRMP3_TRUE;
+ return 0; // File too big.
+ }
+
+ drmp3dec_frame_info info;
+ pcmFramesRead = drmp3dec_decode_frame(&pMP3->decoder, pMP3->pData, (int)pMP3->dataSize, pPCMFrames, &info); // <-- Safe size_t -> int conversion thanks to the check above.
+
+ // Consume the data.
+ size_t 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;
+ drmp3_src_set_input_sample_rate(&pMP3->src, pMP3->mp3FrameSampleRate);
+ break;
+ } else if (info.frame_bytes == 0) {
+ // Need more data. minimp3 recommends doing data submission in 16K chunks.
+ if (pMP3->dataCapacity == pMP3->dataSize) {
+ // No room. Expand.
+ pMP3->dataCapacity += DRMP3_DATA_CHUNK_SIZE;
+ drmp3_uint8* pNewData = (drmp3_uint8*)drmp3_realloc(pMP3->pData, pMP3->dataCapacity);
+ if (pNewData == NULL) {
+ return 0; // Out of memory.
+ }
+
+ pMP3->pData = pNewData;
+ }
+
+ // Fill in a chunk.
+ size_t bytesRead = drmp3__on_read(pMP3, pMP3->pData + pMP3->dataSize, (pMP3->dataCapacity - pMP3->dataSize));
+ if (bytesRead == 0) {
+ pMP3->atEnd = DRMP3_TRUE;
+ return 0; // Error reading more data.
+ }
+
+ pMP3->dataSize += bytesRead;
+ }
+ } while (DRMP3_TRUE);
+
+ return pcmFramesRead;
+}
+
+static drmp3_uint32 drmp3_decode_next_frame(drmp3* pMP3)
+{
+ drmp3_assert(pMP3 != NULL);
+ return drmp3_decode_next_frame_ex(pMP3, (drmp3d_sample_t*)pMP3->pcmFrames, DRMP3_FALSE);
+}
+
+#if 0
+static drmp3_uint32 drmp3_seek_next_frame(drmp3* pMP3)
+{
+ drmp3_assert(pMP3 != NULL);
+
+ drmp3_uint32 pcmFrameCount = drmp3_decode_next_frame_ex(pMP3, NULL);
+ if (pcmFrameCount == 0) {
+ return 0;
+ }
+
+ // We have essentially just skipped past the frame, so just set the remaining samples to 0.
+ pMP3->currentPCMFrame += pcmFrameCount;
+ pMP3->pcmFramesConsumedInMP3Frame = pcmFrameCount;
+ pMP3->pcmFramesRemainingInMP3Frame = 0;
+
+ return pcmFrameCount;
+}
+#endif
+
+static drmp3_uint64 drmp3_read_src(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, void* pUserData)
+{
+ drmp3* pMP3 = (drmp3*)pUserData;
+ drmp3_assert(pMP3 != NULL);
+ drmp3_assert(pMP3->onRead != NULL);
+
+ float* pFramesOutF = (float*)pFramesOut;
+ drmp3_uint64 totalFramesRead = 0;
+
+ while (frameCount > 0) {
+ // Read from the in-memory buffer first.
+ while (pMP3->pcmFramesRemainingInMP3Frame > 0 && frameCount > 0) {
+ drmp3d_sample_t* frames = (drmp3d_sample_t*)pMP3->pcmFrames;
+#ifndef DR_MP3_FLOAT_OUTPUT
+ if (pMP3->mp3FrameChannels == 1) {
+ if (pMP3->channels == 1) {
+ // Mono -> Mono.
+ pFramesOutF[0] = frames[pMP3->pcmFramesConsumedInMP3Frame] / 32768.0f;
+ } else {
+ // Mono -> Stereo.
+ pFramesOutF[0] = frames[pMP3->pcmFramesConsumedInMP3Frame] / 32768.0f;
+ pFramesOutF[1] = frames[pMP3->pcmFramesConsumedInMP3Frame] / 32768.0f;
+ }
+ } else {
+ if (pMP3->channels == 1) {
+ // Stereo -> Mono
+ float sample = 0;
+ sample += frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+0] / 32768.0f;
+ sample += frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+1] / 32768.0f;
+ pFramesOutF[0] = sample * 0.5f;
+ } else {
+ // Stereo -> Stereo
+ pFramesOutF[0] = frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+0] / 32768.0f;
+ pFramesOutF[1] = frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+1] / 32768.0f;
+ }
+ }
+#else
+ if (pMP3->mp3FrameChannels == 1) {
+ if (pMP3->channels == 1) {
+ // Mono -> Mono.
+ pFramesOutF[0] = frames[pMP3->pcmFramesConsumedInMP3Frame];
+ } else {
+ // Mono -> Stereo.
+ pFramesOutF[0] = frames[pMP3->pcmFramesConsumedInMP3Frame];
+ pFramesOutF[1] = frames[pMP3->pcmFramesConsumedInMP3Frame];
+ }
+ } else {
+ if (pMP3->channels == 1) {
+ // Stereo -> Mono
+ float sample = 0;
+ sample += frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+0];
+ sample += frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+1];
+ pFramesOutF[0] = sample * 0.5f;
+ } else {
+ // Stereo -> Stereo
+ pFramesOutF[0] = frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+0];
+ pFramesOutF[1] = frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+1];
+ }
+ }
+#endif
+
+ pMP3->pcmFramesConsumedInMP3Frame += 1;
+ pMP3->pcmFramesRemainingInMP3Frame -= 1;
+ totalFramesRead += 1;
+ frameCount -= 1;
+ pFramesOutF += pSRC->config.channels;
+ }
+
+ if (frameCount == 0) {
+ break;
+ }
+
+ drmp3_assert(pMP3->pcmFramesRemainingInMP3Frame == 0);
+
+ // At this point we have exhausted our in-memory buffer so we need to re-fill. Note that the sample rate may have changed
+ // at this point which means we'll also need to update our sample rate conversion pipeline.
+ if (drmp3_decode_next_frame(pMP3) == 0) {
+ break;
+ }
+ }
+
+ return totalFramesRead;
+}
+
+drmp3_bool32 drmp3_init_internal(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, const drmp3_config* pConfig)
+{
+ drmp3_assert(pMP3 != NULL);
+ drmp3_assert(onRead != NULL);
+
+ // This function assumes the output object has already been reset to 0. Do not do that here, otherwise things will break.
+ drmp3dec_init(&pMP3->decoder);
+
+ // The config can be null in which case we use defaults.
+ drmp3_config config;
+ if (pConfig != NULL) {
+ config = *pConfig;
+ } else {
+ drmp3_zero_object(&config);
+ }
+
+ pMP3->channels = config.outputChannels;
+ if (pMP3->channels == 0) {
+ pMP3->channels = DR_MP3_DEFAULT_CHANNELS;
+ }
+
+ // Cannot have more than 2 channels.
+ if (pMP3->channels > 2) {
+ pMP3->channels = 2;
+ }
+
+ pMP3->sampleRate = config.outputSampleRate;
+ if (pMP3->sampleRate == 0) {
+ pMP3->sampleRate = DR_MP3_DEFAULT_SAMPLE_RATE;
+ }
+
+ pMP3->onRead = onRead;
+ pMP3->onSeek = onSeek;
+ pMP3->pUserData = pUserData;
+
+ // We need a sample rate converter for converting the sample rate from the MP3 frames to the requested output sample rate.
+ drmp3_src_config srcConfig;
+ drmp3_zero_object(&srcConfig);
+ srcConfig.sampleRateIn = DR_MP3_DEFAULT_SAMPLE_RATE;
+ srcConfig.sampleRateOut = pMP3->sampleRate;
+ srcConfig.channels = pMP3->channels;
+ srcConfig.algorithm = drmp3_src_algorithm_linear;
+ if (!drmp3_src_init(&srcConfig, drmp3_read_src, pMP3, &pMP3->src)) {
+ drmp3_uninit(pMP3);
+ return DRMP3_FALSE;
+ }
+
+ // Decode the first frame to confirm that it is indeed a valid MP3 stream.
+ if (!drmp3_decode_next_frame(pMP3)) {
+ drmp3_uninit(pMP3);
+ return DRMP3_FALSE; // Not a valid MP3 stream.
+ }
+
+ return DRMP3_TRUE;
+}
+
+drmp3_bool32 drmp3_init(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, const drmp3_config* pConfig)
+{
+ if (pMP3 == NULL || onRead == NULL) {
+ return DRMP3_FALSE;
+ }
+
+ drmp3_zero_object(pMP3);
+ return drmp3_init_internal(pMP3, onRead, onSeek, pUserData, pConfig);
+}
+
+
+static size_t drmp3__on_read_memory(void* pUserData, void* pBufferOut, size_t bytesToRead)
+{
+ drmp3* pMP3 = (drmp3*)pUserData;
+ drmp3_assert(pMP3 != NULL);
+ drmp3_assert(pMP3->memory.dataSize >= pMP3->memory.currentReadPos);
+
+ size_t bytesRemaining = pMP3->memory.dataSize - pMP3->memory.currentReadPos;
+ if (bytesToRead > bytesRemaining) {
+ bytesToRead = bytesRemaining;
+ }
+
+ if (bytesToRead > 0) {
+ drmp3_copy_memory(pBufferOut, pMP3->memory.pData + pMP3->memory.currentReadPos, bytesToRead);
+ pMP3->memory.currentReadPos += bytesToRead;
+ }
+
+ return bytesToRead;
+}
+
+static drmp3_bool32 drmp3__on_seek_memory(void* pUserData, int byteOffset, drmp3_seek_origin origin)
+{
+ drmp3* pMP3 = (drmp3*)pUserData;
+ drmp3_assert(pMP3 != NULL);
+
+ if (origin == drmp3_seek_origin_current) {
+ if (byteOffset > 0) {
+ if (pMP3->memory.currentReadPos + byteOffset > pMP3->memory.dataSize) {
+ byteOffset = (int)(pMP3->memory.dataSize - pMP3->memory.currentReadPos); // Trying to seek too far forward.
+ }
+ } else {
+ if (pMP3->memory.currentReadPos < (size_t)-byteOffset) {
+ byteOffset = -(int)pMP3->memory.currentReadPos; // Trying to seek too far backwards.
+ }
+ }
+
+ // This will never underflow thanks to the clamps above.
+ pMP3->memory.currentReadPos += byteOffset;
+ } else {
+ if ((drmp3_uint32)byteOffset <= pMP3->memory.dataSize) {
+ pMP3->memory.currentReadPos = byteOffset;
+ } else {
+ pMP3->memory.currentReadPos = pMP3->memory.dataSize; // Trying to seek too far forward.
+ }
+ }
+
+ return DRMP3_TRUE;
+}
+
+drmp3_bool32 drmp3_init_memory(drmp3* pMP3, const void* pData, size_t dataSize, const drmp3_config* pConfig)
+{
+ if (pMP3 == NULL) {
+ return DRMP3_FALSE;
+ }
+
+ drmp3_zero_object(pMP3);
+
+ if (pData == NULL || dataSize == 0) {
+ return DRMP3_FALSE;
+ }
+
+ pMP3->memory.pData = (const drmp3_uint8*)pData;
+ pMP3->memory.dataSize = dataSize;
+ pMP3->memory.currentReadPos = 0;
+
+ return drmp3_init_internal(pMP3, drmp3__on_read_memory, drmp3__on_seek_memory, pMP3, pConfig);
+}
+
+
+#ifndef DR_MP3_NO_STDIO
+#include
+
+static size_t drmp3__on_read_stdio(void* pUserData, void* pBufferOut, size_t bytesToRead)
+{
+ return fread(pBufferOut, 1, bytesToRead, (FILE*)pUserData);
+}
+
+static drmp3_bool32 drmp3__on_seek_stdio(void* pUserData, int offset, drmp3_seek_origin origin)
+{
+ return fseek((FILE*)pUserData, offset, (origin == drmp3_seek_origin_current) ? SEEK_CUR : SEEK_SET) == 0;
+}
+
+drmp3_bool32 drmp3_init_file(drmp3* pMP3, const char* filePath, const drmp3_config* pConfig)
+{
+ FILE* pFile;
+#if defined(_MSC_VER) && _MSC_VER >= 1400
+ if (fopen_s(&pFile, filePath, "rb") != 0) {
+ return DRMP3_FALSE;
+ }
+#else
+ pFile = fopen(filePath, "rb");
+ if (pFile == NULL) {
+ return DRMP3_FALSE;
+ }
+#endif
+
+ return drmp3_init(pMP3, drmp3__on_read_stdio, drmp3__on_seek_stdio, (void*)pFile, pConfig);
+}
+#endif
+
+void drmp3_uninit(drmp3* pMP3)
+{
+ if (pMP3 == NULL) {
+ return;
+ }
+
+#ifndef DR_MP3_NO_STDIO
+ if (pMP3->onRead == drmp3__on_read_stdio) {
+ fclose((FILE*)pMP3->pUserData);
+ }
+#endif
+
+ drmp3_free(pMP3->pData);
+}
+
+drmp3_uint64 drmp3_read_pcm_frames_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut)
+{
+ if (pMP3 == NULL || pMP3->onRead == NULL) {
+ return 0;
+ }
+
+ drmp3_uint64 totalFramesRead = 0;
+
+ if (pBufferOut == NULL) {
+ float temp[4096];
+ while (framesToRead > 0) {
+ drmp3_uint64 framesToReadRightNow = sizeof(temp)/sizeof(temp[0]) / pMP3->channels;
+ if (framesToReadRightNow > framesToRead) {
+ framesToReadRightNow = framesToRead;
+ }
+
+ drmp3_uint64 framesJustRead = drmp3_read_pcm_frames_f32(pMP3, framesToReadRightNow, temp);
+ if (framesJustRead == 0) {
+ break;
+ }
+
+ framesToRead -= framesJustRead;
+ totalFramesRead += framesJustRead;
+ }
+ } else {
+ totalFramesRead = drmp3_src_read_frames_ex(&pMP3->src, framesToRead, pBufferOut, DRMP3_TRUE);
+ pMP3->currentPCMFrame += totalFramesRead;
+ }
+
+ return totalFramesRead;
+}
+
+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;
+ //drmp3_zero_object(&pMP3->decoder);
+ 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)
+{
+#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.
+ drmp3_uint64 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.
+ drmp3_uint64 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.
+ //drmp3_uint64 framesToReadAndDiscard;
+ 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_assert(pSeekPointIndex != NULL);
+
+ 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 (drmp3_uint32 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_assert(pMP3 != NULL);
+ drmp3_assert(pMP3->pSeekPoints != NULL);
+ drmp3_assert(pMP3->seekPointCount > 0);
+
+ drmp3_seek_point seekPoint;
+
+ // 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.
+ drmp3_uint32 priorSeekPointIndex;
+ 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 (drmp3_uint16 iMP3Frame = 0; iMP3Frame < seekPoint.mp3FramesToDiscard; ++iMP3Frame) {
+ // Pass in non-null for the last frame because we want to ensure the sample rate converter is preloaded correctly.
+ drmp3d_sample_t* 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.
+ drmp3_uint32 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 = pMP3->currentPCMFrame * ((double)pMP3->src.config.sampleRateIn / pMP3->src.config.sampleRateOut);
+ 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.
+ drmp3_uint64 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)
+{
+ 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.
+ drmp3_uint64 currentPCMFrame = pMP3->currentPCMFrame;
+
+ if (!drmp3_seek_to_start_of_stream(pMP3)) {
+ return DRMP3_FALSE;
+ }
+
+ drmp3_uint64 totalPCMFrameCount = 0;
+ drmp3_uint64 totalMP3FrameCount = 0;
+
+ float totalPCMFrameCountFractionalPart = 0; // <-- With resampling there will be a fractional part to each MP3 frame that we need to accumulate.
+ for (;;) {
+ drmp3_uint32 pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL, DRMP3_FALSE);
+ if (pcmFramesInCurrentMP3FrameIn == 0) {
+ break;
+ }
+
+ float srcRatio = (float)pMP3->mp3FrameSampleRate / (float)pMP3->sampleRate;
+ drmp3_assert(srcRatio > 0);
+
+ float pcmFramesInCurrentMP3FrameOutF = totalPCMFrameCountFractionalPart + (pcmFramesInCurrentMP3FrameIn / srcRatio);
+ drmp3_uint32 pcmFramesInCurrentMP3FrameOut = (drmp3_uint32)pcmFramesInCurrentMP3FrameOutF;
+ totalPCMFrameCountFractionalPart = pcmFramesInCurrentMP3FrameOutF - pcmFramesInCurrentMP3FrameOut;
+ totalPCMFrameCount += pcmFramesInCurrentMP3FrameOut;
+ 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)pMP3->mp3FrameSampleRate / (float)pMP3->sampleRate;
+ drmp3_assert(srcRatio > 0);
+
+ float pcmFrameCountOutF = *pRunningPCMFrameCountFractionalPart + (pcmFrameCountIn / srcRatio);
+ drmp3_uint32 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)
+{
+ if (pMP3 == NULL || pSeekPointCount == NULL || pSeekPoints == NULL) {
+ return DRMP3_FALSE; // Invalid args.
+ }
+
+ drmp3_uint32 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.
+ drmp3_uint64 currentPCMFrame = pMP3->currentPCMFrame;
+
+ // We never do more than the total number of MP3 frames and we limit it to 32-bits.
+ drmp3_uint64 totalMP3FrameCount;
+ drmp3_uint64 totalPCMFrameCount;
+ 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 {
+ if (seekPointCount > totalMP3FrameCount-1) {
+ seekPointCount = (drmp3_uint32)totalMP3FrameCount-1;
+ }
+
+ drmp3_uint64 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.
+ drmp3__seeking_mp3_frame_info mp3FrameInfo[DRMP3_SEEK_LEADING_MP3_FRAMES+1];
+
+ drmp3_uint64 runningPCMFrameCount = 0;
+ float runningPCMFrameCountFractionalPart = 0;
+
+ // We need to initialize the array of MP3 byte positions for the leading MP3 frames.
+ for (int iMP3Frame = 0; iMP3Frame < DRMP3_SEEK_LEADING_MP3_FRAMES+1; ++iMP3Frame) {
+ // 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.
+ drmp3_uint32 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.
+ drmp3_uint64 nextTargetPCMFrame = 0;
+ for (drmp3_uint32 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 {
+ // 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 (size_t 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.
+ drmp3_uint32 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_assert(pMP3 != NULL);
+
+ drmp3_uint64 totalFramesRead = 0;
+ drmp3_uint64 framesCapacity = 0;
+ float* pFrames = NULL;
+
+ float temp[4096];
+ for (;;) {
+ drmp3_uint64 framesToReadRightNow = drmp3_countof(temp) / pMP3->channels;
+ drmp3_uint64 framesJustRead = drmp3_read_pcm_frames_f32(pMP3, framesToReadRightNow, temp);
+ if (framesJustRead == 0) {
+ break;
+ }
+
+ // Reallocate the output buffer if there's not enough room.
+ if (framesCapacity < totalFramesRead + framesJustRead) {
+ framesCapacity *= 2;
+ if (framesCapacity < totalFramesRead + framesJustRead) {
+ framesCapacity = totalFramesRead + framesJustRead;
+ }
+
+ drmp3_uint64 newFramesBufferSize = framesCapacity*pMP3->channels*sizeof(float);
+ if (newFramesBufferSize > DRMP3_SIZE_MAX) {
+ break;
+ }
+
+ float* pNewFrames = (float*)drmp3_realloc(pFrames, (size_t)newFramesBufferSize);
+ if (pNewFrames == NULL) {
+ drmp3_free(pFrames);
+ break;
+ }
+
+ pFrames = pNewFrames;
+ }
+
+ drmp3_copy_memory(pFrames + totalFramesRead*pMP3->channels, temp, (size_t)(framesJustRead*pMP3->channels*sizeof(float)));
+ totalFramesRead += framesJustRead;
+
+ // If the number of frames we asked for is less that what we actually read it means we've reached the end.
+ if (framesJustRead != framesToReadRightNow) {
+ break;
+ }
+ }
+
+ if (pConfig != NULL) {
+ pConfig->outputChannels = pMP3->channels;
+ pConfig->outputSampleRate = pMP3->sampleRate;
+ }
+
+ drmp3_uninit(pMP3);
+
+ if (pTotalFrameCount) *pTotalFrameCount = totalFramesRead;
+ return pFrames;
+}
+
+float* drmp3_open_and_read_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount)
+{
+ drmp3 mp3;
+ if (!drmp3_init(&mp3, onRead, onSeek, pUserData, pConfig)) {
+ return NULL;
+ }
+
+ return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount);
+}
+
+float* drmp3_open_memory_and_read_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount)
+{
+ drmp3 mp3;
+ if (!drmp3_init_memory(&mp3, pData, dataSize, pConfig)) {
+ return NULL;
+ }
+
+ return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount);
+}
+
+#ifndef DR_MP3_NO_STDIO
+float* drmp3_open_file_and_read_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount)
+{
+ drmp3 mp3;
+ if (!drmp3_init_file(&mp3, filePath, pConfig)) {
+ return NULL;
+ }
+
+ return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount);
+}
+#endif
+
+void drmp3_free(void* p)
+{
+ DRMP3_FREE(p);
+}
+
+#endif /*DR_MP3_IMPLEMENTATION*/
+
+
+// DIFFERENCES BETWEEN minimp3 AND dr_mp3
+// ======================================
+// - First, keep in mind that minimp3 (https://github.com/lieff/minimp3) is where all the real work was done. All of the
+// code relating to the actual decoding remains mostly unmodified, apart from some namespacing changes.
+// - dr_mp3 adds a pulling style API which allows you to deliver raw data via callbacks. So, rather than pushing data
+// to the decoder, the decoder _pulls_ data from your callbacks.
+// - In addition to callbacks, a decoder can be initialized from a block of memory and a file.
+// - The dr_mp3 pull API reads PCM frames rather than whole MP3 frames.
+// - dr_mp3 adds convenience APIs for opening and decoding entire files in one go.
+// - dr_mp3 is fully namespaced, including the implementation section, which is more suitable when compiling projects
+// as a single translation unit (aka unity builds). At the time of writing this, a unity build is not possible when
+// using minimp3 in conjunction with stb_vorbis. dr_mp3 addresses this.
+
+
+// REVISION HISTORY
+// ================
+//
+// v0.4.1 - 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 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
+*/
+
+/*
+ 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 .
+*/
diff --git a/code/nel/src/sound/CMakeLists.txt b/code/nel/src/sound/CMakeLists.txt
index 7b9eebc9f..d2c47387e 100644
--- a/code/nel/src/sound/CMakeLists.txt
+++ b/code/nel/src/sound/CMakeLists.txt
@@ -62,6 +62,7 @@ FILE(GLOB STREAM
FILE(GLOB STREAM_FILE
audio_decoder.cpp ../../include/nel/sound/audio_decoder.h
audio_decoder_vorbis.cpp ../../include/nel/sound/audio_decoder_vorbis.h
+ audio_decoder_mp3.cpp ../../include/nel/sound/audio_decoder_mp3.h
audio_decoder_ffmpeg.cpp ../../include/nel/sound/audio_decoder_ffmpeg.h
stream_file_sound.cpp ../../include/nel/sound/stream_file_sound.h
stream_file_source.cpp ../../include/nel/sound/stream_file_source.h
diff --git a/code/nel/src/sound/audio_decoder.cpp b/code/nel/src/sound/audio_decoder.cpp
index f0eb80efd..d849ed770 100644
--- a/code/nel/src/sound/audio_decoder.cpp
+++ b/code/nel/src/sound/audio_decoder.cpp
@@ -36,6 +36,7 @@
// Project includes
#include
+#include
#ifdef FFMPEG_ENABLED
#include
@@ -102,6 +103,10 @@ IAudioDecoder *IAudioDecoder::createAudioDecoder(const std::string &type, NLMISC
{
return new CAudioDecoderVorbis(stream, loop);
}
+ else if (type_lower == "mp3")
+ {
+ return new CAudioDecoderMP3(stream, loop);
+ }
else
{
nlwarning("Music file type unknown: '%s'", type_lower.c_str());
@@ -139,6 +144,16 @@ bool IAudioDecoder::getInfo(const std::string &filepath, std::string &artist, st
nlwarning("Unable to open: '%s'", filepath.c_str());
}
+ else if (type_lower == "mp3")
+ {
+ CIFile ifile;
+ ifile.setCacheFileOnOpen(false);
+ ifile.allowBNPCacheFileOnOpen(false);
+ if (ifile.open(lookup))
+ return CAudioDecoderMP3::getInfo(&ifile, artist, title, length);
+
+ nlwarning("Unable to open: '%s'", filepath.c_str());
+ }
else
{
nlwarning("Music file type unknown: '%s'", type_lower.c_str());
@@ -157,6 +172,10 @@ void IAudioDecoder::getMusicExtensions(std::vector &extensions)
{
extensions.push_back("ogg");
}
+ if (std::find(extensions.begin(), extensions.end(), "mp3") == extensions.end())
+ {
+ extensions.push_back("mp3");
+ }
#ifdef FFMPEG_ENABLED
extensions.push_back("mp3");
extensions.push_back("flac");
diff --git a/code/nel/src/sound/audio_decoder_mp3.cpp b/code/nel/src/sound/audio_decoder_mp3.cpp
new file mode 100644
index 000000000..bef0aad71
--- /dev/null
+++ b/code/nel/src/sound/audio_decoder_mp3.cpp
@@ -0,0 +1,221 @@
+// NeL - MMORPG Framework
+// Copyright (C) 2018 Winch Gate Property Limited
+//
+// This program is free software: you can redistribute it and/or modify
+// it under the terms of the GNU Affero General Public License as
+// published by the Free Software Foundation, either version 3 of the
+// License, or (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+// GNU Affero General Public License for more details.
+//
+// You should have received a copy of the GNU Affero General Public License
+// along with this program. If not, see .
+
+
+#include "stdsound.h"
+
+#include
+
+#define DR_MP3_IMPLEMENTATION
+#include
+
+using namespace std;
+using namespace NLMISC;
+using namespace NLSOUND;
+
+namespace NLSOUND {
+
+// callback for drmp3
+static size_t drmp3_read(void* pUserData, void* pBufferOut, size_t bytesToRead)
+{
+ NLSOUND::CAudioDecoderMP3 *decoder = static_cast(pUserData);
+ NLMISC::IStream *stream = decoder->getStream();
+ nlassert(stream->isReading());
+
+ uint32 available = decoder->getStreamSize() - stream->getPos();
+ if (available == 0)
+ return 0;
+
+ if (bytesToRead > available)
+ bytesToRead = available;
+
+ stream->serialBuffer((uint8 *)pBufferOut, bytesToRead);
+ return bytesToRead;
+}
+
+// callback for drmp3
+static drmp3_bool32 drmp3_seek(void* pUserData, int offset, drmp3_seek_origin origin)
+{
+ NLSOUND::CAudioDecoderMP3 *decoder = static_cast(pUserData);
+ NLMISC::IStream *stream = decoder->getStream();
+ nlassert(stream->isReading());
+
+ NLMISC::IStream::TSeekOrigin seekOrigin;
+ if (origin == drmp3_seek_origin_start)
+ seekOrigin = NLMISC::IStream::begin;
+ else if (origin == drmp3_seek_origin_current)
+ seekOrigin = NLMISC::IStream::current;
+ else
+ return false;
+
+ stream->seek((sint32) offset, seekOrigin);
+ return true;
+}
+
+// these should always be 44100Hz/16bit/2ch
+#define MP3_SAMPLE_RATE 44100
+#define MP3_BITS_PER_SAMPLE 16
+#define MP3_CHANNELS 2
+
+CAudioDecoderMP3::CAudioDecoderMP3(NLMISC::IStream *stream, bool loop)
+: IAudioDecoder(),
+ _Stream(stream), _Loop(loop), _IsMusicEnded(false), _StreamSize(0), _IsSupported(false), _PCMFrameCount(0)
+{
+ _StreamOffset = stream->getPos();
+ stream->seek(0, NLMISC::IStream::end);
+ _StreamSize = stream->getPos();
+ stream->seek(_StreamOffset, NLMISC::IStream::begin);
+
+ drmp3_config config;
+ config.outputChannels = MP3_CHANNELS;
+ config.outputSampleRate = MP3_SAMPLE_RATE;
+
+ _IsSupported = drmp3_init(&_Decoder, &drmp3_read, &drmp3_seek, this, &config);
+ if (!_IsSupported)
+ {
+ nlwarning("MP3: Decoder failed to read stream");
+ }
+}
+
+CAudioDecoderMP3::~CAudioDecoderMP3()
+{
+ drmp3_uninit(&_Decoder);
+}
+
+bool CAudioDecoderMP3::isFormatSupported() const
+{
+ return _IsSupported;
+}
+
+/// Get information on a music file.
+bool CAudioDecoderMP3::getInfo(NLMISC::IStream *stream, std::string &artist, std::string &title, float &length)
+{
+ CAudioDecoderMP3 mp3(stream, false);
+ if (!mp3.isFormatSupported())
+ {
+ title.clear();
+ artist.clear();
+ length = 0.f;
+
+ return false;
+ }
+ length = mp3.getLength();
+
+ // ID3v1
+ stream->seek(-128, NLMISC::IStream::end);
+ {
+ uint8 buf[128];
+ stream->serialBuffer(buf, 128);
+
+ if(buf[0] == 'T' && buf[1] == 'A' && buf[2] == 'G')
+ {
+ uint i;
+ for(i = 0; i < 30; ++i) if (buf[3+i] == '\0') break;
+ artist.assign((char *)&buf[3], i);
+
+ for(i = 0; i < 30; ++i) if (buf[33+i] == '\0') break;
+ title.assign((char *)&buf[33], i);
+ }
+ }
+
+ return true;
+}
+
+uint32 CAudioDecoderMP3::getRequiredBytes()
+{
+ return 0; // no minimum requirement of bytes to buffer out
+}
+
+uint32 CAudioDecoderMP3::getNextBytes(uint8 *buffer, uint32 minimum, uint32 maximum)
+{
+ if (_IsMusicEnded) return 0;
+ nlassert(minimum <= maximum); // can't have this..
+
+ // TODO: CStreamFileSource::play() will stall when there is no frames on warmup
+ // supported can be set false if there is an issue creating converter
+ if (!_IsSupported)
+ {
+ _IsMusicEnded = true;
+ return 1;
+ }
+
+ sint16 *pFrameBufferOut = (sint16 *)buffer;
+ uint32 bytesPerFrame = MP3_BITS_PER_SAMPLE / 8 * _Decoder.channels;
+
+ uint32 totalFramesRead = 0;
+ uint32 framesToRead = minimum / bytesPerFrame;
+ while(framesToRead > 0)
+ {
+ float tempBuffer[4096];
+ uint64 tempFrames = drmp3_countof(tempBuffer) / _Decoder.channels;
+
+ if (tempFrames > framesToRead)
+ tempFrames = framesToRead;
+
+ tempFrames = drmp3_read_pcm_frames_f32(&_Decoder, tempFrames, tempBuffer);
+ if (tempFrames == 0)
+ break;
+
+ drmp3dec_f32_to_s16(tempBuffer, pFrameBufferOut, tempFrames * _Decoder.channels);
+ pFrameBufferOut += tempFrames * _Decoder.channels;
+
+ framesToRead -= tempFrames;
+ totalFramesRead += tempFrames;
+ }
+
+ _IsMusicEnded = (framesToRead > 0);
+ return totalFramesRead * bytesPerFrame;
+}
+
+uint8 CAudioDecoderMP3::getChannels()
+{
+ return _Decoder.channels;
+}
+
+uint CAudioDecoderMP3::getSamplesPerSec()
+{
+ return _Decoder.sampleRate;
+}
+
+uint8 CAudioDecoderMP3::getBitsPerSample()
+{
+ return MP3_BITS_PER_SAMPLE;
+}
+
+bool CAudioDecoderMP3::isMusicEnded()
+{
+ return _IsMusicEnded;
+}
+
+float CAudioDecoderMP3::getLength()
+{
+ // cached because drmp3_get_pcm_frame_count is reading full file
+ if (_PCMFrameCount == 0)
+ {
+ _PCMFrameCount = drmp3_get_pcm_frame_count(&_Decoder);
+ }
+
+ return _PCMFrameCount / (float) _Decoder.sampleRate;
+}
+
+void CAudioDecoderMP3::setLooping(bool loop)
+{
+ _Loop = loop;
+}
+
+} /* namespace NLSOUND */
+
+/* end of file */