diff --git a/vall_e.cpp/Makefile b/vall_e.cpp/Makefile index d8e3e11..d47be88 100644 --- a/vall_e.cpp/Makefile +++ b/vall_e.cpp/Makefile @@ -3,7 +3,7 @@ CXX = g++ INCS += -I./include LIBS += -L./libs -LINKS += -lggml -lggml-base -lllama -lencodec +LINKS += -lggml -lggml-base -lllama -lencodec -lespeak-ng FLAGS += -march=native -O3 SRCS := $(shell find ./ -name "*.cpp") diff --git a/vall_e.cpp/README.md b/vall_e.cpp/README.md index 50480a6..1186fa5 100644 --- a/vall_e.cpp/README.md +++ b/vall_e.cpp/README.md @@ -8,7 +8,7 @@ At the moment it's ***very*** work in progress. Populate `./include/` with the `ggml`, `llama.cpp`, and `encodec.cpp` headers. -Populate `./libs/` with the compiled libraries of `llama.cpp` and `encodec.cpp`. +Populate `./libs/` with the compiled libraries of `llama.cpp`, `encodec.cpp`, and `espeak-ng`. Run `make`. @@ -26,10 +26,10 @@ Run `make`. * [x] load the quantized model * [x] orchestrate the required embeddings * [x] juggle the output head / classifier properly -* [ ] phonemize text +* [x] phonemize text * with the help of espeak-ng -* [ ] tokenize phonemes - * the tokenizer is being a huge thorn on actual sequences +* [x] tokenize phonemes + * tokenize with `llama_tokenize` instead of a homebrewed method because the tokenizer is being a huge thorn * [x] load audio from disk * [x] encode audio * [x] sum embeddings for the `prom` and prior `resp`s diff --git a/vall_e.cpp/include/decoder.h b/vall_e.cpp/include/decoder.h new file mode 100644 index 0000000..7f37544 --- /dev/null +++ b/vall_e.cpp/include/decoder.h @@ -0,0 +1,113 @@ +#pragma once + +#include + +#include "ggml.h" +#include "ggml-alloc.h" +#include "ggml-backend.h" + +#include "lstm.h" +#include "utils.h" + + +struct encodec_decoder_block { + // upsampling layers + struct ggml_tensor *us_conv_w; + struct ggml_tensor *us_conv_b; + + // conv1 + struct ggml_tensor *conv_1_w; + struct ggml_tensor *conv_1_b; + + // conv2 + struct ggml_tensor *conv_2_w; + struct ggml_tensor *conv_2_b; + + // shortcut + struct ggml_tensor *conv_sc_w; + struct ggml_tensor *conv_sc_b; +}; + +struct encodec_decoder { + struct ggml_tensor *init_conv_w; + struct ggml_tensor *init_conv_b; + + encodec_lstm lstm; + + struct ggml_tensor *final_conv_w; + struct ggml_tensor *final_conv_b; + + std::vector blocks; +}; + +struct ggml_tensor *encodec_forward_decoder( + const struct encodec_decoder *decoder, struct ggml_context *ctx0, + struct ggml_tensor *quantized_out, const int *ratios, const int kernel_size, const int res_kernel_size, + const int stride) { + + if (!quantized_out) { + fprintf(stderr, "%s: null input tensor\n", __func__); + return NULL; + } + + struct ggml_tensor *inpL = strided_conv_1d( + ctx0, quantized_out, decoder->init_conv_w, decoder->init_conv_b, stride); + + // lstm + { + struct ggml_tensor *cur = inpL; + + const encodec_lstm lstm = decoder->lstm; + + // first lstm layer + char l0_prefix[7] = "dec_l0"; + struct ggml_tensor *hs1 = forward_pass_lstm_unilayer( + ctx0, cur, lstm.l0_ih_w, lstm.l0_hh_w, lstm.l0_ih_b, lstm.l0_hh_b, l0_prefix); + + // second lstm layer + char l1_prefix[7] = "dec_l1"; + struct ggml_tensor *out = forward_pass_lstm_unilayer( + ctx0, hs1, lstm.l1_ih_w, lstm.l1_hh_w, lstm.l1_ih_b, lstm.l1_hh_b, l1_prefix); + + inpL = ggml_add(ctx0, inpL, out); + } + + for (int layer_ix = 0; layer_ix < 4; layer_ix++) { + encodec_decoder_block block = decoder->blocks[layer_ix]; + + // upsampling layers + inpL = ggml_elu(ctx0, inpL); + + inpL = strided_conv_transpose_1d( + ctx0, inpL, block.us_conv_w, block.us_conv_b, ratios[layer_ix]); + + struct ggml_tensor *current = inpL; + + // shortcut + struct ggml_tensor *shortcut = strided_conv_1d( + ctx0, inpL, block.conv_sc_w, block.conv_sc_b, stride); + + // conv1 + current = ggml_elu(ctx0, current); + + current = strided_conv_1d( + ctx0, current, block.conv_1_w, block.conv_1_b, stride); + + // conv2 + current = ggml_elu(ctx0, current); + + current = strided_conv_1d( + ctx0, current, block.conv_2_w, block.conv_2_b, stride); + + // residual connection + inpL = ggml_add(ctx0, current, shortcut); + } + + // final conv + inpL = ggml_elu(ctx0, inpL); + + struct ggml_tensor *decoded_inp = strided_conv_1d( + ctx0, inpL, decoder->final_conv_w, decoder->final_conv_b, stride); + + return decoded_inp; +} diff --git a/vall_e.cpp/include/dr_wav.h b/vall_e.cpp/include/dr_wav.h new file mode 100644 index 0000000..e5fed4d --- /dev/null +++ b/vall_e.cpp/include/dr_wav.h @@ -0,0 +1,6434 @@ +/* +WAV audio loader and writer. Choice of public domain or MIT-0. See license statements at the end of this file. +dr_wav - v0.12.16 - 2020-12-02 + +David Reid - mackron@gmail.com + +GitHub: https://github.com/mackron/dr_libs +*/ + +/* +RELEASE NOTES - VERSION 0.12 +============================ +Version 0.12 includes breaking changes to custom chunk handling. + + +Changes to Chunk Callback +------------------------- +dr_wav supports the ability to fire a callback when a chunk is encounted (except for WAVE and FMT chunks). The callback has been updated to include both the +container (RIFF or Wave64) and the FMT chunk which contains information about the format of the data in the wave file. + +Previously, there was no direct way to determine the container, and therefore no way to discriminate against the different IDs in the chunk header (RIFF and +Wave64 containers encode chunk ID's differently). The `container` parameter can be used to know which ID to use. + +Sometimes it can be useful to know the data format at the time the chunk callback is fired. A pointer to a `drwav_fmt` object is now passed into the chunk +callback which will give you information about the data format. To determine the sample format, use `drwav_fmt_get_format()`. This will return one of the +`DR_WAVE_FORMAT_*` tokens. +*/ + +/* +Introduction +============ +This is a single file library. To use it, do something like the following in one .c file. + + ```c + #define DR_WAV_IMPLEMENTATION + #include "dr_wav.h" + ``` + +You can then #include this file in other parts of the program as you would with any other header file. Do something like the following to read audio data: + + ```c + drwav wav; + if (!drwav_init_file(&wav, "my_song.wav", NULL)) { + // Error opening WAV file. + } + + drwav_int32* pDecodedInterleavedPCMFrames = malloc(wav.totalPCMFrameCount * wav.channels * sizeof(drwav_int32)); + size_t numberOfSamplesActuallyDecoded = drwav_read_pcm_frames_s32(&wav, wav.totalPCMFrameCount, pDecodedInterleavedPCMFrames); + + ... + + drwav_uninit(&wav); + ``` + +If you just want to quickly open and read the audio data in a single operation you can do something like this: + + ```c + unsigned int channels; + unsigned int sampleRate; + drwav_uint64 totalPCMFrameCount; + float* pSampleData = drwav_open_file_and_read_pcm_frames_f32("my_song.wav", &channels, &sampleRate, &totalPCMFrameCount, NULL); + if (pSampleData == NULL) { + // Error opening and reading WAV file. + } + + ... + + drwav_free(pSampleData); + ``` + +The examples above use versions of the API that convert the audio data to a consistent format (32-bit signed PCM, in this case), but you can still output the +audio data in its internal format (see notes below for supported formats): + + ```c + size_t framesRead = drwav_read_pcm_frames(&wav, wav.totalPCMFrameCount, pDecodedInterleavedPCMFrames); + ``` + +You can also read the raw bytes of audio data, which could be useful if dr_wav does not have native support for a particular data format: + + ```c + size_t bytesRead = drwav_read_raw(&wav, bytesToRead, pRawDataBuffer); + ``` + +dr_wav can also be used to output WAV files. This does not currently support compressed formats. To use this, look at `drwav_init_write()`, +`drwav_init_file_write()`, etc. Use `drwav_write_pcm_frames()` to write samples, or `drwav_write_raw()` to write raw data in the "data" chunk. + + ```c + drwav_data_format format; + format.container = drwav_container_riff; // <-- drwav_container_riff = normal WAV files, drwav_container_w64 = Sony Wave64. + format.format = DR_WAVE_FORMAT_PCM; // <-- Any of the DR_WAVE_FORMAT_* codes. + format.channels = 2; + format.sampleRate = 44100; + format.bitsPerSample = 16; + drwav_init_file_write(&wav, "data/recording.wav", &format, NULL); + + ... + + drwav_uint64 framesWritten = drwav_write_pcm_frames(pWav, frameCount, pSamples); + ``` + +dr_wav has seamless support the Sony Wave64 format. The decoder will automatically detect it and it should Just Work without any manual intervention. + + +Build Options +============= +#define these options before including this file. + +#define DR_WAV_NO_CONVERSION_API + Disables conversion APIs such as `drwav_read_pcm_frames_f32()` and `drwav_s16_to_f32()`. + +#define DR_WAV_NO_STDIO + Disables APIs that initialize a decoder from a file such as `drwav_init_file()`, `drwav_init_file_write()`, etc. + + + +Notes +===== +- Samples are always interleaved. +- The default read function does not do any data conversion. Use `drwav_read_pcm_frames_f32()`, `drwav_read_pcm_frames_s32()` and `drwav_read_pcm_frames_s16()` + to read and convert audio data to 32-bit floating point, signed 32-bit integer and signed 16-bit integer samples respectively. Tested and supported internal + formats include the following: + - Unsigned 8-bit PCM + - Signed 12-bit PCM + - Signed 16-bit PCM + - Signed 24-bit PCM + - Signed 32-bit PCM + - IEEE 32-bit floating point + - IEEE 64-bit floating point + - A-law and u-law + - Microsoft ADPCM + - IMA ADPCM (DVI, format code 0x11) +- dr_wav will try to read the WAV file as best it can, even if it's not strictly conformant to the WAV format. +*/ + +#ifndef dr_wav_h +#define dr_wav_h + +#ifdef __cplusplus +extern "C" { +#endif + +#define DRWAV_STRINGIFY(x) #x +#define DRWAV_XSTRINGIFY(x) DRWAV_STRINGIFY(x) + +#define DRWAV_VERSION_MAJOR 0 +#define DRWAV_VERSION_MINOR 12 +#define DRWAV_VERSION_REVISION 16 +#define DRWAV_VERSION_STRING DRWAV_XSTRINGIFY(DRWAV_VERSION_MAJOR) "." DRWAV_XSTRINGIFY(DRWAV_VERSION_MINOR) "." DRWAV_XSTRINGIFY(DRWAV_VERSION_REVISION) + +#include /* For size_t. */ + +/* Sized types. */ +typedef signed char drwav_int8; +typedef unsigned char drwav_uint8; +typedef signed short drwav_int16; +typedef unsigned short drwav_uint16; +typedef signed int drwav_int32; +typedef unsigned int drwav_uint32; +#if defined(_MSC_VER) + typedef signed __int64 drwav_int64; + typedef unsigned __int64 drwav_uint64; +#else + #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic push + #pragma GCC diagnostic ignored "-Wlong-long" + #if defined(__clang__) + #pragma GCC diagnostic ignored "-Wc++11-long-long" + #endif + #endif + typedef signed long long drwav_int64; + typedef unsigned long long drwav_uint64; + #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic pop + #endif +#endif +#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined (_M_IA64) || defined(__aarch64__) || defined(__powerpc64__) + typedef drwav_uint64 drwav_uintptr; +#else + typedef drwav_uint32 drwav_uintptr; +#endif +typedef drwav_uint8 drwav_bool8; +typedef drwav_uint32 drwav_bool32; +#define DRWAV_TRUE 1 +#define DRWAV_FALSE 0 + +#if !defined(DRWAV_API) + #if defined(DRWAV_DLL) + #if defined(_WIN32) + #define DRWAV_DLL_IMPORT __declspec(dllimport) + #define DRWAV_DLL_EXPORT __declspec(dllexport) + #define DRWAV_DLL_PRIVATE static + #else + #if defined(__GNUC__) && __GNUC__ >= 4 + #define DRWAV_DLL_IMPORT __attribute__((visibility("default"))) + #define DRWAV_DLL_EXPORT __attribute__((visibility("default"))) + #define DRWAV_DLL_PRIVATE __attribute__((visibility("hidden"))) + #else + #define DRWAV_DLL_IMPORT + #define DRWAV_DLL_EXPORT + #define DRWAV_DLL_PRIVATE static + #endif + #endif + + #if defined(DR_WAV_IMPLEMENTATION) || defined(DRWAV_IMPLEMENTATION) + #define DRWAV_API DRWAV_DLL_EXPORT + #else + #define DRWAV_API DRWAV_DLL_IMPORT + #endif + #define DRWAV_PRIVATE DRWAV_DLL_PRIVATE + #else + #define DRWAV_API extern + #define DRWAV_PRIVATE static + #endif +#endif + +typedef drwav_int32 drwav_result; +#define DRWAV_SUCCESS 0 +#define DRWAV_ERROR -1 /* A generic error. */ +#define DRWAV_INVALID_ARGS -2 +#define DRWAV_INVALID_OPERATION -3 +#define DRWAV_OUT_OF_MEMORY -4 +#define DRWAV_OUT_OF_RANGE -5 +#define DRWAV_ACCESS_DENIED -6 +#define DRWAV_DOES_NOT_EXIST -7 +#define DRWAV_ALREADY_EXISTS -8 +#define DRWAV_TOO_MANY_OPEN_FILES -9 +#define DRWAV_INVALID_FILE -10 +#define DRWAV_TOO_BIG -11 +#define DRWAV_PATH_TOO_LONG -12 +#define DRWAV_NAME_TOO_LONG -13 +#define DRWAV_NOT_DIRECTORY -14 +#define DRWAV_IS_DIRECTORY -15 +#define DRWAV_DIRECTORY_NOT_EMPTY -16 +#define DRWAV_END_OF_FILE -17 +#define DRWAV_NO_SPACE -18 +#define DRWAV_BUSY -19 +#define DRWAV_IO_ERROR -20 +#define DRWAV_INTERRUPT -21 +#define DRWAV_UNAVAILABLE -22 +#define DRWAV_ALREADY_IN_USE -23 +#define DRWAV_BAD_ADDRESS -24 +#define DRWAV_BAD_SEEK -25 +#define DRWAV_BAD_PIPE -26 +#define DRWAV_DEADLOCK -27 +#define DRWAV_TOO_MANY_LINKS -28 +#define DRWAV_NOT_IMPLEMENTED -29 +#define DRWAV_NO_MESSAGE -30 +#define DRWAV_BAD_MESSAGE -31 +#define DRWAV_NO_DATA_AVAILABLE -32 +#define DRWAV_INVALID_DATA -33 +#define DRWAV_TIMEOUT -34 +#define DRWAV_NO_NETWORK -35 +#define DRWAV_NOT_UNIQUE -36 +#define DRWAV_NOT_SOCKET -37 +#define DRWAV_NO_ADDRESS -38 +#define DRWAV_BAD_PROTOCOL -39 +#define DRWAV_PROTOCOL_UNAVAILABLE -40 +#define DRWAV_PROTOCOL_NOT_SUPPORTED -41 +#define DRWAV_PROTOCOL_FAMILY_NOT_SUPPORTED -42 +#define DRWAV_ADDRESS_FAMILY_NOT_SUPPORTED -43 +#define DRWAV_SOCKET_NOT_SUPPORTED -44 +#define DRWAV_CONNECTION_RESET -45 +#define DRWAV_ALREADY_CONNECTED -46 +#define DRWAV_NOT_CONNECTED -47 +#define DRWAV_CONNECTION_REFUSED -48 +#define DRWAV_NO_HOST -49 +#define DRWAV_IN_PROGRESS -50 +#define DRWAV_CANCELLED -51 +#define DRWAV_MEMORY_ALREADY_MAPPED -52 +#define DRWAV_AT_END -53 + +/* Common data formats. */ +#define DR_WAVE_FORMAT_PCM 0x1 +#define DR_WAVE_FORMAT_ADPCM 0x2 +#define DR_WAVE_FORMAT_IEEE_FLOAT 0x3 +#define DR_WAVE_FORMAT_ALAW 0x6 +#define DR_WAVE_FORMAT_MULAW 0x7 +#define DR_WAVE_FORMAT_DVI_ADPCM 0x11 +#define DR_WAVE_FORMAT_EXTENSIBLE 0xFFFE + +/* Constants. */ +#ifndef DRWAV_MAX_SMPL_LOOPS +#define DRWAV_MAX_SMPL_LOOPS 1 +#endif + +/* Flags to pass into drwav_init_ex(), etc. */ +#define DRWAV_SEQUENTIAL 0x00000001 + +DRWAV_API void drwav_version(drwav_uint32* pMajor, drwav_uint32* pMinor, drwav_uint32* pRevision); +DRWAV_API const char* drwav_version_string(void); + +typedef enum +{ + drwav_seek_origin_start, + drwav_seek_origin_current +} drwav_seek_origin; + +typedef enum +{ + drwav_container_riff, + drwav_container_w64, + drwav_container_rf64 +} drwav_container; + +typedef struct +{ + union + { + drwav_uint8 fourcc[4]; + drwav_uint8 guid[16]; + } id; + + /* The size in bytes of the chunk. */ + drwav_uint64 sizeInBytes; + + /* + RIFF = 2 byte alignment. + W64 = 8 byte alignment. + */ + unsigned int paddingSize; +} drwav_chunk_header; + +typedef struct +{ + /* + The format tag exactly as specified in the wave file's "fmt" chunk. This can be used by applications + that require support for data formats not natively supported by dr_wav. + */ + drwav_uint16 formatTag; + + /* The number of channels making up the audio data. When this is set to 1 it is mono, 2 is stereo, etc. */ + drwav_uint16 channels; + + /* The sample rate. Usually set to something like 44100. */ + drwav_uint32 sampleRate; + + /* Average bytes per second. You probably don't need this, but it's left here for informational purposes. */ + drwav_uint32 avgBytesPerSec; + + /* Block align. This is equal to the number of channels * bytes per sample. */ + drwav_uint16 blockAlign; + + /* Bits per sample. */ + drwav_uint16 bitsPerSample; + + /* The size of the extended data. Only used internally for validation, but left here for informational purposes. */ + drwav_uint16 extendedSize; + + /* + The number of valid bits per sample. When is equal to WAVE_FORMAT_EXTENSIBLE, + is always rounded up to the nearest multiple of 8. This variable contains information about exactly how + many bits are valid per sample. Mainly used for informational purposes. + */ + drwav_uint16 validBitsPerSample; + + /* The channel mask. Not used at the moment. */ + drwav_uint32 channelMask; + + /* The sub-format, exactly as specified by the wave file. */ + drwav_uint8 subFormat[16]; +} drwav_fmt; + +DRWAV_API drwav_uint16 drwav_fmt_get_format(const drwav_fmt* pFMT); + + +/* +Callback for when data is read. Return value is the number of bytes actually read. + +pUserData [in] The user data that was passed to drwav_init() 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 (* drwav_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead); + +/* +Callback for when data is written. Returns value is the number of bytes actually written. + +pUserData [in] The user data that was passed to drwav_init_write() and family. +pData [out] A pointer to the data to write. +bytesToWrite [in] The number of bytes to write. + +Returns the number of bytes actually written. + +If the return value differs from bytesToWrite, it indicates an error. +*/ +typedef size_t (* drwav_write_proc)(void* pUserData, const void* pData, size_t bytesToWrite); + +/* +Callback for when data needs to be seeked. + +pUserData [in] The user data that was passed to drwav_init() 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 drwav_seek_origin_start or +drwav_seek_origin_current. +*/ +typedef drwav_bool32 (* drwav_seek_proc)(void* pUserData, int offset, drwav_seek_origin origin); + +/* +Callback for when drwav_init_ex() finds a chunk. + +pChunkUserData [in] The user data that was passed to the pChunkUserData parameter of drwav_init_ex() and family. +onRead [in] A pointer to the function to call when reading. +onSeek [in] A pointer to the function to call when seeking. +pReadSeekUserData [in] The user data that was passed to the pReadSeekUserData parameter of drwav_init_ex() and family. +pChunkHeader [in] A pointer to an object containing basic header information about the chunk. Use this to identify the chunk. +container [in] Whether or not the WAV file is a RIFF or Wave64 container. If you're unsure of the difference, assume RIFF. +pFMT [in] A pointer to the object containing the contents of the "fmt" chunk. + +Returns the number of bytes read + seeked. + +To read data from the chunk, call onRead(), passing in pReadSeekUserData as the first parameter. Do the same for seeking with onSeek(). The return value must +be the total number of bytes you have read _plus_ seeked. + +Use the `container` argument to discriminate the fields in `pChunkHeader->id`. If the container is `drwav_container_riff` or `drwav_container_rf64` you should +use `id.fourcc`, otherwise you should use `id.guid`. + +The `pFMT` parameter can be used to determine the data format of the wave file. Use `drwav_fmt_get_format()` to get the sample format, which will be one of the +`DR_WAVE_FORMAT_*` identifiers. + +The read pointer will be sitting on the first byte after the chunk's header. You must not attempt to read beyond the boundary of the chunk. +*/ +typedef drwav_uint64 (* drwav_chunk_proc)(void* pChunkUserData, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pReadSeekUserData, const drwav_chunk_header* pChunkHeader, drwav_container container, const drwav_fmt* pFMT); + +typedef struct +{ + void* pUserData; + void* (* onMalloc)(size_t sz, void* pUserData); + void* (* onRealloc)(void* p, size_t sz, void* pUserData); + void (* onFree)(void* p, void* pUserData); +} drwav_allocation_callbacks; + +/* Structure for internal use. Only used for loaders opened with drwav_init_memory(). */ +typedef struct +{ + const drwav_uint8* data; + size_t dataSize; + size_t currentReadPos; +} drwav__memory_stream; + +/* Structure for internal use. Only used for writers opened with drwav_init_memory_write(). */ +typedef struct +{ + void** ppData; + size_t* pDataSize; + size_t dataSize; + size_t dataCapacity; + size_t currentWritePos; +} drwav__memory_stream_write; + +typedef struct +{ + drwav_container container; /* RIFF, W64. */ + drwav_uint32 format; /* DR_WAVE_FORMAT_* */ + drwav_uint32 channels; + drwav_uint32 sampleRate; + drwav_uint32 bitsPerSample; +} drwav_data_format; + + +/* See the following for details on the 'smpl' chunk: https://sites.google.com/site/musicgapi/technical-documents/wav-file-format#smpl */ +typedef struct +{ + drwav_uint32 cuePointId; + drwav_uint32 type; + drwav_uint32 start; + drwav_uint32 end; + drwav_uint32 fraction; + drwav_uint32 playCount; +} drwav_smpl_loop; + + typedef struct +{ + drwav_uint32 manufacturer; + drwav_uint32 product; + drwav_uint32 samplePeriod; + drwav_uint32 midiUnityNotes; + drwav_uint32 midiPitchFraction; + drwav_uint32 smpteFormat; + drwav_uint32 smpteOffset; + drwav_uint32 numSampleLoops; + drwav_uint32 samplerData; + drwav_smpl_loop loops[DRWAV_MAX_SMPL_LOOPS]; +} drwav_smpl; + +typedef struct +{ + /* A pointer to the function to call when more data is needed. */ + drwav_read_proc onRead; + + /* A pointer to the function to call when data needs to be written. Only used when the drwav object is opened in write mode. */ + drwav_write_proc onWrite; + + /* A pointer to the function to call when the wav file needs to be seeked. */ + drwav_seek_proc onSeek; + + /* The user data to pass to callbacks. */ + void* pUserData; + + /* Allocation callbacks. */ + drwav_allocation_callbacks allocationCallbacks; + + + /* Whether or not the WAV file is formatted as a standard RIFF file or W64. */ + drwav_container container; + + + /* Structure containing format information exactly as specified by the wav file. */ + drwav_fmt fmt; + + /* The sample rate. Will be set to something like 44100. */ + drwav_uint32 sampleRate; + + /* The number of channels. This will be set to 1 for monaural streams, 2 for stereo, etc. */ + drwav_uint16 channels; + + /* The bits per sample. Will be set to something like 16, 24, etc. */ + drwav_uint16 bitsPerSample; + + /* Equal to fmt.formatTag, or the value specified by fmt.subFormat if fmt.formatTag is equal to 65534 (WAVE_FORMAT_EXTENSIBLE). */ + drwav_uint16 translatedFormatTag; + + /* The total number of PCM frames making up the audio data. */ + drwav_uint64 totalPCMFrameCount; + + + /* The size in bytes of the data chunk. */ + drwav_uint64 dataChunkDataSize; + + /* The position in the stream of the first byte of the data chunk. This is used for seeking. */ + drwav_uint64 dataChunkDataPos; + + /* The number of bytes remaining in the data chunk. */ + drwav_uint64 bytesRemaining; + + + /* + Only used in sequential write mode. Keeps track of the desired size of the "data" chunk at the point of initialization time. Always + set to 0 for non-sequential writes and when the drwav object is opened in read mode. Used for validation. + */ + drwav_uint64 dataChunkDataSizeTargetWrite; + + /* Keeps track of whether or not the wav writer was initialized in sequential mode. */ + drwav_bool32 isSequentialWrite; + + + /* smpl chunk. */ + drwav_smpl smpl; + + + /* A hack to avoid a DRWAV_MALLOC() when opening a decoder with drwav_init_memory(). */ + drwav__memory_stream memoryStream; + drwav__memory_stream_write memoryStreamWrite; + + /* Generic data for compressed formats. This data is shared across all block-compressed formats. */ + struct + { + drwav_uint64 iCurrentPCMFrame; /* The index of the next PCM frame that will be read by drwav_read_*(). This is used with "totalPCMFrameCount" to ensure we don't read excess samples at the end of the last block. */ + } compressed; + + /* Microsoft ADPCM specific data. */ + struct + { + drwav_uint32 bytesRemainingInBlock; + drwav_uint16 predictor[2]; + drwav_int32 delta[2]; + drwav_int32 cachedFrames[4]; /* Samples are stored in this cache during decoding. */ + drwav_uint32 cachedFrameCount; + drwav_int32 prevFrames[2][2]; /* The previous 2 samples for each channel (2 channels at most). */ + } msadpcm; + + /* IMA ADPCM specific data. */ + struct + { + drwav_uint32 bytesRemainingInBlock; + drwav_int32 predictor[2]; + drwav_int32 stepIndex[2]; + drwav_int32 cachedFrames[16]; /* Samples are stored in this cache during decoding. */ + drwav_uint32 cachedFrameCount; + } ima; +} drwav; + + +/* +Initializes a pre-allocated drwav object for reading. + +pWav [out] A pointer to the drwav object being initialized. +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. +onChunk [in, optional] The function to call when a chunk is enumerated at initialized time. +pUserData, pReadSeekUserData [in, optional] A pointer to application defined data that will be passed to onRead and onSeek. +pChunkUserData [in, optional] A pointer to application defined data that will be passed to onChunk. +flags [in, optional] A set of flags for controlling how things are loaded. + +Returns true if successful; false otherwise. + +Close the loader with drwav_uninit(). + +This is the lowest level function for initializing a WAV file. You can also use drwav_init_file() and drwav_init_memory() +to open the stream from a file or from a block of memory respectively. + +Possible values for flags: + DRWAV_SEQUENTIAL: Never perform a backwards seek while loading. This disables the chunk callback and will cause this function + to return as soon as the data chunk is found. Any chunks after the data chunk will be ignored. + +drwav_init() is equivalent to "drwav_init_ex(pWav, onRead, onSeek, NULL, pUserData, NULL, 0);". + +The onChunk callback is not called for the WAVE or FMT chunks. The contents of the FMT chunk can be read from pWav->fmt +after the function returns. + +See also: drwav_init_file(), drwav_init_memory(), drwav_uninit() +*/ +DRWAV_API drwav_bool32 drwav_init(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_ex(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_chunk_proc onChunk, void* pReadSeekUserData, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); + +/* +Initializes a pre-allocated drwav object for writing. + +onWrite [in] The function to call when data needs to be written. +onSeek [in] The function to call when the write position needs to move. +pUserData [in, optional] A pointer to application defined data that will be passed to onWrite and onSeek. + +Returns true if successful; false otherwise. + +Close the writer with drwav_uninit(). + +This is the lowest level function for initializing a WAV file. You can also use drwav_init_file_write() and drwav_init_memory_write() +to open the stream from a file or from a block of memory respectively. + +If the total sample count is known, you can use drwav_init_write_sequential(). This avoids the need for dr_wav to perform +a post-processing step for storing the total sample count and the size of the data chunk which requires a backwards seek. + +See also: drwav_init_file_write(), drwav_init_memory_write(), drwav_uninit() +*/ +DRWAV_API drwav_bool32 drwav_init_write(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_write_sequential(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_write_sequential_pcm_frames(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks); + +/* +Utility function to determine the target size of the entire data to be written (including all headers and chunks). + +Returns the target size in bytes. + +Useful if the application needs to know the size to allocate. + +Only writing to the RIFF chunk and one data chunk is currently supported. + +See also: drwav_init_write(), drwav_init_file_write(), drwav_init_memory_write() +*/ +DRWAV_API drwav_uint64 drwav_target_write_size_bytes(const drwav_data_format* pFormat, drwav_uint64 totalSampleCount); + +/* +Uninitializes the given drwav object. + +Use this only for objects initialized with drwav_init*() functions (drwav_init(), drwav_init_ex(), drwav_init_write(), drwav_init_write_sequential()). +*/ +DRWAV_API drwav_result drwav_uninit(drwav* pWav); + + +/* +Reads raw audio data. + +This is the lowest level function for reading audio data. It simply reads the given number of +bytes of the raw internal sample data. + +Consider using drwav_read_pcm_frames_s16(), drwav_read_pcm_frames_s32() or drwav_read_pcm_frames_f32() for +reading sample data in a consistent format. + +pBufferOut can be NULL in which case a seek will be performed. + +Returns the number of bytes actually read. +*/ +DRWAV_API size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut); + +/* +Reads up to the specified number of PCM frames from the WAV file. + +The output data will be in the file's internal format, converted to native-endian byte order. Use +drwav_read_pcm_frames_s16/f32/s32() to read data in a specific format. + +If the return value is less than it means the end of the file has been reached or +you have requested more PCM frames than can possibly fit in the output buffer. + +This function will only work when sample data is of a fixed size and uncompressed. If you are +using a compressed format consider using drwav_read_raw() or drwav_read_pcm_frames_s16/s32/f32(). + +pBufferOut can be NULL in which case a seek will be performed. +*/ +DRWAV_API drwav_uint64 drwav_read_pcm_frames(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_le(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_be(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut); + +/* +Seeks to the given PCM frame. + +Returns true if successful; false otherwise. +*/ +DRWAV_API drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetFrameIndex); + + +/* +Writes raw audio data. + +Returns the number of bytes actually written. If this differs from bytesToWrite, it indicates an error. +*/ +DRWAV_API size_t drwav_write_raw(drwav* pWav, size_t bytesToWrite, const void* pData); + +/* +Writes PCM frames. + +Returns the number of PCM frames written. + +Input samples need to be in native-endian byte order. On big-endian architectures the input data will be converted to +little-endian. Use drwav_write_raw() to write raw audio data without performing any conversion. +*/ +DRWAV_API drwav_uint64 drwav_write_pcm_frames(drwav* pWav, drwav_uint64 framesToWrite, const void* pData); +DRWAV_API drwav_uint64 drwav_write_pcm_frames_le(drwav* pWav, drwav_uint64 framesToWrite, const void* pData); +DRWAV_API drwav_uint64 drwav_write_pcm_frames_be(drwav* pWav, drwav_uint64 framesToWrite, const void* pData); + + +/* Conversion Utilities */ +#ifndef DR_WAV_NO_CONVERSION_API + +/* +Reads a chunk of audio data and converts it to signed 16-bit PCM samples. + +pBufferOut can be NULL in which case a seek will be performed. + +Returns the number of PCM frames actually read. + +If the return value is less than it means the end of the file has been reached. +*/ +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16le(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16be(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut); + +/* Low-level function for converting unsigned 8-bit PCM samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_u8_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 24-bit PCM samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_s24_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 32-bit PCM samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_s32_to_s16(drwav_int16* pOut, const drwav_int32* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 32-bit floating point samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_f32_to_s16(drwav_int16* pOut, const float* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 64-bit floating point samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_f64_to_s16(drwav_int16* pOut, const double* pIn, size_t sampleCount); + +/* Low-level function for converting A-law samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_alaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting u-law samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_mulaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + + +/* +Reads a chunk of audio data and converts it to IEEE 32-bit floating point samples. + +pBufferOut can be NULL in which case a seek will be performed. + +Returns the number of PCM frames actually read. + +If the return value is less than it means the end of the file has been reached. +*/ +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32le(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32be(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut); + +/* Low-level function for converting unsigned 8-bit PCM samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_u8_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 16-bit PCM samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_s16_to_f32(float* pOut, const drwav_int16* pIn, size_t sampleCount); + +/* Low-level function for converting signed 24-bit PCM samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_s24_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 32-bit PCM samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_s32_to_f32(float* pOut, const drwav_int32* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 64-bit floating point samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_f64_to_f32(float* pOut, const double* pIn, size_t sampleCount); + +/* Low-level function for converting A-law samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_alaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting u-law samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_mulaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + + +/* +Reads a chunk of audio data and converts it to signed 32-bit PCM samples. + +pBufferOut can be NULL in which case a seek will be performed. + +Returns the number of PCM frames actually read. + +If the return value is less than it means the end of the file has been reached. +*/ +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32le(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32be(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut); + +/* Low-level function for converting unsigned 8-bit PCM samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_u8_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 16-bit PCM samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_s16_to_s32(drwav_int32* pOut, const drwav_int16* pIn, size_t sampleCount); + +/* Low-level function for converting signed 24-bit PCM samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_s24_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 32-bit floating point samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_f32_to_s32(drwav_int32* pOut, const float* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 64-bit floating point samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_f64_to_s32(drwav_int32* pOut, const double* pIn, size_t sampleCount); + +/* Low-level function for converting A-law samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_alaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting u-law samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_mulaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +#endif /* DR_WAV_NO_CONVERSION_API */ + + +/* High-Level Convenience Helpers */ + +#ifndef DR_WAV_NO_STDIO +/* +Helper for initializing a wave file for reading using stdio. + +This holds the internal FILE object until drwav_uninit() is called. Keep this in mind if you're caching drwav +objects because the operating system may restrict the number of file handles an application can have open at +any given time. +*/ +DRWAV_API drwav_bool32 drwav_init_file(drwav* pWav, const char* filename, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_ex(drwav* pWav, const char* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_w(drwav* pWav, const wchar_t* filename, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_ex_w(drwav* pWav, const wchar_t* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); + +/* +Helper for initializing a wave file for writing using stdio. + +This holds the internal FILE object until drwav_uninit() is called. Keep this in mind if you're caching drwav +objects because the operating system may restrict the number of file handles an application can have open at +any given time. +*/ +DRWAV_API drwav_bool32 drwav_init_file_write(drwav* pWav, const char* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_write_sequential(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_write_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks); +#endif /* DR_WAV_NO_STDIO */ + +/* +Helper for initializing a loader from a pre-allocated memory buffer. + +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 drwav object. + +The buffer should contain the contents of the entire wave file, not just the sample data. +*/ +DRWAV_API drwav_bool32 drwav_init_memory(drwav* pWav, const void* data, size_t dataSize, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_memory_ex(drwav* pWav, const void* data, size_t dataSize, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); + +/* +Helper for initializing a writer which outputs data to a memory buffer. + +dr_wav will manage the memory allocations, however it is up to the caller to free the data with drwav_free(). + +The buffer will remain allocated even after drwav_uninit() is called. The buffer should not be considered valid +until after drwav_uninit() has been called. +*/ +DRWAV_API drwav_bool32 drwav_init_memory_write(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_memory_write_sequential(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_memory_write_sequential_pcm_frames(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks); + + +#ifndef DR_WAV_NO_CONVERSION_API +/* +Opens and reads an entire wav file in a single operation. + +The return value is a heap-allocated buffer containing the audio data. Use drwav_free() to free the buffer. +*/ +DRWAV_API drwav_int16* drwav_open_and_read_pcm_frames_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API float* drwav_open_and_read_pcm_frames_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_int32* drwav_open_and_read_pcm_frames_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +#ifndef DR_WAV_NO_STDIO +/* +Opens and decodes an entire wav file in a single operation. + +The return value is a heap-allocated buffer containing the audio data. Use drwav_free() to free the buffer. +*/ +DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +#endif +/* +Opens and decodes an entire wav file from a block of memory in a single operation. + +The return value is a heap-allocated buffer containing the audio data. Use drwav_free() to free the buffer. +*/ +DRWAV_API drwav_int16* drwav_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API float* drwav_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_int32* drwav_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +#endif + +/* Frees data that was allocated internally by dr_wav. */ +DRWAV_API void drwav_free(void* p, const drwav_allocation_callbacks* pAllocationCallbacks); + +/* Converts bytes from a wav stream to a sized type of native endian. */ +DRWAV_API drwav_uint16 drwav_bytes_to_u16(const drwav_uint8* data); +DRWAV_API drwav_int16 drwav_bytes_to_s16(const drwav_uint8* data); +DRWAV_API drwav_uint32 drwav_bytes_to_u32(const drwav_uint8* data); +DRWAV_API drwav_int32 drwav_bytes_to_s32(const drwav_uint8* data); +DRWAV_API drwav_uint64 drwav_bytes_to_u64(const drwav_uint8* data); +DRWAV_API drwav_int64 drwav_bytes_to_s64(const drwav_uint8* data); + +/* Compares a GUID for the purpose of checking the type of a Wave64 chunk. */ +DRWAV_API drwav_bool32 drwav_guid_equal(const drwav_uint8 a[16], const drwav_uint8 b[16]); + +/* Compares a four-character-code for the purpose of checking the type of a RIFF chunk. */ +DRWAV_API drwav_bool32 drwav_fourcc_equal(const drwav_uint8* a, const char* b); + +#ifdef __cplusplus +} +#endif +#endif /* dr_wav_h */ + + +/************************************************************************************************************************************************************ + ************************************************************************************************************************************************************ + + IMPLEMENTATION + + ************************************************************************************************************************************************************ + ************************************************************************************************************************************************************/ +#if defined(DR_WAV_IMPLEMENTATION) || defined(DRWAV_IMPLEMENTATION) +#ifndef dr_wav_c +#define dr_wav_c + +#include +#include /* For memcpy(), memset() */ +#include /* For INT_MAX */ + +#ifndef DR_WAV_NO_STDIO +#include +#include +#endif + +/* Standard library stuff. */ +#ifndef DRWAV_ASSERT +#include +#define DRWAV_ASSERT(expression) assert(expression) +#endif +#ifndef DRWAV_MALLOC +#define DRWAV_MALLOC(sz) malloc((sz)) +#endif +#ifndef DRWAV_REALLOC +#define DRWAV_REALLOC(p, sz) realloc((p), (sz)) +#endif +#ifndef DRWAV_FREE +#define DRWAV_FREE(p) free((p)) +#endif +#ifndef DRWAV_COPY_MEMORY +#define DRWAV_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz)) +#endif +#ifndef DRWAV_ZERO_MEMORY +#define DRWAV_ZERO_MEMORY(p, sz) memset((p), 0, (sz)) +#endif +#ifndef DRWAV_ZERO_OBJECT +#define DRWAV_ZERO_OBJECT(p) DRWAV_ZERO_MEMORY((p), sizeof(*p)) +#endif + +#define drwav_countof(x) (sizeof(x) / sizeof(x[0])) +#define drwav_align(x, a) ((((x) + (a) - 1) / (a)) * (a)) +#define drwav_min(a, b) (((a) < (b)) ? (a) : (b)) +#define drwav_max(a, b) (((a) > (b)) ? (a) : (b)) +#define drwav_clamp(x, lo, hi) (drwav_max((lo), drwav_min((hi), (x)))) + +#define DRWAV_MAX_SIMD_VECTOR_SIZE 64 /* 64 for AVX-512 in the future. */ + +/* CPU architecture. */ +#if defined(__x86_64__) || defined(_M_X64) + #define DRWAV_X64 +#elif defined(__i386) || defined(_M_IX86) + #define DRWAV_X86 +#elif defined(__arm__) || defined(_M_ARM) + #define DRWAV_ARM +#endif + +#ifdef _MSC_VER + #define DRWAV_INLINE __forceinline +#elif defined(__GNUC__) + /* + I've had a bug report where GCC is emitting warnings about functions possibly not being inlineable. This warning happens when + the __attribute__((always_inline)) attribute is defined without an "inline" statement. I think therefore there must be some + case where "__inline__" is not always defined, thus the compiler emitting these warnings. When using -std=c89 or -ansi on the + command line, we cannot use the "inline" keyword and instead need to use "__inline__". In an attempt to work around this issue + I am using "__inline__" only when we're compiling in strict ANSI mode. + */ + #if defined(__STRICT_ANSI__) + #define DRWAV_INLINE __inline__ __attribute__((always_inline)) + #else + #define DRWAV_INLINE inline __attribute__((always_inline)) + #endif +#elif defined(__WATCOMC__) + #define DRWAV_INLINE __inline +#else + #define DRWAV_INLINE +#endif + +#if defined(SIZE_MAX) + #define DRWAV_SIZE_MAX SIZE_MAX +#else + #if defined(_WIN64) || defined(_LP64) || defined(__LP64__) + #define DRWAV_SIZE_MAX ((drwav_uint64)0xFFFFFFFFFFFFFFFF) + #else + #define DRWAV_SIZE_MAX 0xFFFFFFFF + #endif +#endif + +#if defined(_MSC_VER) && _MSC_VER >= 1400 + #define DRWAV_HAS_BYTESWAP16_INTRINSIC + #define DRWAV_HAS_BYTESWAP32_INTRINSIC + #define DRWAV_HAS_BYTESWAP64_INTRINSIC +#elif defined(__clang__) + #if defined(__has_builtin) + #if __has_builtin(__builtin_bswap16) + #define DRWAV_HAS_BYTESWAP16_INTRINSIC + #endif + #if __has_builtin(__builtin_bswap32) + #define DRWAV_HAS_BYTESWAP32_INTRINSIC + #endif + #if __has_builtin(__builtin_bswap64) + #define DRWAV_HAS_BYTESWAP64_INTRINSIC + #endif + #endif +#elif defined(__GNUC__) + #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 3)) + #define DRWAV_HAS_BYTESWAP32_INTRINSIC + #define DRWAV_HAS_BYTESWAP64_INTRINSIC + #endif + #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8)) + #define DRWAV_HAS_BYTESWAP16_INTRINSIC + #endif +#endif + +DRWAV_API void drwav_version(drwav_uint32* pMajor, drwav_uint32* pMinor, drwav_uint32* pRevision) +{ + if (pMajor) { + *pMajor = DRWAV_VERSION_MAJOR; + } + + if (pMinor) { + *pMinor = DRWAV_VERSION_MINOR; + } + + if (pRevision) { + *pRevision = DRWAV_VERSION_REVISION; + } +} + +DRWAV_API const char* drwav_version_string(void) +{ + return DRWAV_VERSION_STRING; +} + +/* +These limits are used for basic validation when initializing the decoder. If you exceed these limits, first of all: what on Earth are +you doing?! (Let me know, I'd be curious!) Second, you can adjust these by #define-ing them before the dr_wav implementation. +*/ +#ifndef DRWAV_MAX_SAMPLE_RATE +#define DRWAV_MAX_SAMPLE_RATE 384000 +#endif +#ifndef DRWAV_MAX_CHANNELS +#define DRWAV_MAX_CHANNELS 256 +#endif +#ifndef DRWAV_MAX_BITS_PER_SAMPLE +#define DRWAV_MAX_BITS_PER_SAMPLE 64 +#endif + +static const drwav_uint8 drwavGUID_W64_RIFF[16] = {0x72,0x69,0x66,0x66, 0x2E,0x91, 0xCF,0x11, 0xA5,0xD6, 0x28,0xDB,0x04,0xC1,0x00,0x00}; /* 66666972-912E-11CF-A5D6-28DB04C10000 */ +static const drwav_uint8 drwavGUID_W64_WAVE[16] = {0x77,0x61,0x76,0x65, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 65766177-ACF3-11D3-8CD1-00C04F8EDB8A */ +/*static const drwav_uint8 drwavGUID_W64_JUNK[16] = {0x6A,0x75,0x6E,0x6B, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A};*/ /* 6B6E756A-ACF3-11D3-8CD1-00C04F8EDB8A */ +static const drwav_uint8 drwavGUID_W64_FMT [16] = {0x66,0x6D,0x74,0x20, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 20746D66-ACF3-11D3-8CD1-00C04F8EDB8A */ +static const drwav_uint8 drwavGUID_W64_FACT[16] = {0x66,0x61,0x63,0x74, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 74636166-ACF3-11D3-8CD1-00C04F8EDB8A */ +static const drwav_uint8 drwavGUID_W64_DATA[16] = {0x64,0x61,0x74,0x61, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 61746164-ACF3-11D3-8CD1-00C04F8EDB8A */ +static const drwav_uint8 drwavGUID_W64_SMPL[16] = {0x73,0x6D,0x70,0x6C, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 6C706D73-ACF3-11D3-8CD1-00C04F8EDB8A */ + +static DRWAV_INLINE drwav_bool32 drwav__guid_equal(const drwav_uint8 a[16], const drwav_uint8 b[16]) +{ + int i; + for (i = 0; i < 16; i += 1) { + if (a[i] != b[i]) { + return DRWAV_FALSE; + } + } + + return DRWAV_TRUE; +} + +static DRWAV_INLINE drwav_bool32 drwav__fourcc_equal(const drwav_uint8* a, const char* b) +{ + return + a[0] == b[0] && + a[1] == b[1] && + a[2] == b[2] && + a[3] == b[3]; +} + + + +static DRWAV_INLINE int drwav__is_little_endian(void) +{ +#if defined(DRWAV_X86) || defined(DRWAV_X64) + return DRWAV_TRUE; +#elif defined(__BYTE_ORDER) && defined(__LITTLE_ENDIAN) && __BYTE_ORDER == __LITTLE_ENDIAN + return DRWAV_TRUE; +#else + int n = 1; + return (*(char*)&n) == 1; +#endif +} + +static DRWAV_INLINE drwav_uint16 drwav__bytes_to_u16(const drwav_uint8* data) +{ + return (data[0] << 0) | (data[1] << 8); +} + +static DRWAV_INLINE drwav_int16 drwav__bytes_to_s16(const drwav_uint8* data) +{ + return (short)drwav__bytes_to_u16(data); +} + +static DRWAV_INLINE drwav_uint32 drwav__bytes_to_u32(const drwav_uint8* data) +{ + return (data[0] << 0) | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); +} + +static DRWAV_INLINE drwav_int32 drwav__bytes_to_s32(const drwav_uint8* data) +{ + return (drwav_int32)drwav__bytes_to_u32(data); +} + +static DRWAV_INLINE drwav_uint64 drwav__bytes_to_u64(const drwav_uint8* data) +{ + return + ((drwav_uint64)data[0] << 0) | ((drwav_uint64)data[1] << 8) | ((drwav_uint64)data[2] << 16) | ((drwav_uint64)data[3] << 24) | + ((drwav_uint64)data[4] << 32) | ((drwav_uint64)data[5] << 40) | ((drwav_uint64)data[6] << 48) | ((drwav_uint64)data[7] << 56); +} + +static DRWAV_INLINE drwav_int64 drwav__bytes_to_s64(const drwav_uint8* data) +{ + return (drwav_int64)drwav__bytes_to_u64(data); +} + +static DRWAV_INLINE void drwav__bytes_to_guid(const drwav_uint8* data, drwav_uint8* guid) +{ + int i; + for (i = 0; i < 16; ++i) { + guid[i] = data[i]; + } +} + + +static DRWAV_INLINE drwav_uint16 drwav__bswap16(drwav_uint16 n) +{ +#ifdef DRWAV_HAS_BYTESWAP16_INTRINSIC + #if defined(_MSC_VER) + return _byteswap_ushort(n); + #elif defined(__GNUC__) || defined(__clang__) + return __builtin_bswap16(n); + #else + #error "This compiler does not support the byte swap intrinsic." + #endif +#else + return ((n & 0xFF00) >> 8) | + ((n & 0x00FF) << 8); +#endif +} + +static DRWAV_INLINE drwav_uint32 drwav__bswap32(drwav_uint32 n) +{ +#ifdef DRWAV_HAS_BYTESWAP32_INTRINSIC + #if defined(_MSC_VER) + return _byteswap_ulong(n); + #elif defined(__GNUC__) || defined(__clang__) + #if defined(DRWAV_ARM) && (defined(__ARM_ARCH) && __ARM_ARCH >= 6) && !defined(DRWAV_64BIT) /* <-- 64-bit inline assembly has not been tested, so disabling for now. */ + /* Inline assembly optimized implementation for ARM. In my testing, GCC does not generate optimized code with __builtin_bswap32(). */ + drwav_uint32 r; + __asm__ __volatile__ ( + #if defined(DRWAV_64BIT) + "rev %w[out], %w[in]" : [out]"=r"(r) : [in]"r"(n) /* <-- This is untested. If someone in the community could test this, that would be appreciated! */ + #else + "rev %[out], %[in]" : [out]"=r"(r) : [in]"r"(n) + #endif + ); + return r; + #else + return __builtin_bswap32(n); + #endif + #else + #error "This compiler does not support the byte swap intrinsic." + #endif +#else + return ((n & 0xFF000000) >> 24) | + ((n & 0x00FF0000) >> 8) | + ((n & 0x0000FF00) << 8) | + ((n & 0x000000FF) << 24); +#endif +} + +static DRWAV_INLINE drwav_uint64 drwav__bswap64(drwav_uint64 n) +{ +#ifdef DRWAV_HAS_BYTESWAP64_INTRINSIC + #if defined(_MSC_VER) + return _byteswap_uint64(n); + #elif defined(__GNUC__) || defined(__clang__) + return __builtin_bswap64(n); + #else + #error "This compiler does not support the byte swap intrinsic." + #endif +#else + /* Weird "<< 32" bitshift is required for C89 because it doesn't support 64-bit constants. Should be optimized out by a good compiler. */ + return ((n & ((drwav_uint64)0xFF000000 << 32)) >> 56) | + ((n & ((drwav_uint64)0x00FF0000 << 32)) >> 40) | + ((n & ((drwav_uint64)0x0000FF00 << 32)) >> 24) | + ((n & ((drwav_uint64)0x000000FF << 32)) >> 8) | + ((n & ((drwav_uint64)0xFF000000 )) << 8) | + ((n & ((drwav_uint64)0x00FF0000 )) << 24) | + ((n & ((drwav_uint64)0x0000FF00 )) << 40) | + ((n & ((drwav_uint64)0x000000FF )) << 56); +#endif +} + + +static DRWAV_INLINE drwav_int16 drwav__bswap_s16(drwav_int16 n) +{ + return (drwav_int16)drwav__bswap16((drwav_uint16)n); +} + +static DRWAV_INLINE void drwav__bswap_samples_s16(drwav_int16* pSamples, drwav_uint64 sampleCount) +{ + drwav_uint64 iSample; + for (iSample = 0; iSample < sampleCount; iSample += 1) { + pSamples[iSample] = drwav__bswap_s16(pSamples[iSample]); + } +} + + +static DRWAV_INLINE void drwav__bswap_s24(drwav_uint8* p) +{ + drwav_uint8 t; + t = p[0]; + p[0] = p[2]; + p[2] = t; +} + +static DRWAV_INLINE void drwav__bswap_samples_s24(drwav_uint8* pSamples, drwav_uint64 sampleCount) +{ + drwav_uint64 iSample; + for (iSample = 0; iSample < sampleCount; iSample += 1) { + drwav_uint8* pSample = pSamples + (iSample*3); + drwav__bswap_s24(pSample); + } +} + + +static DRWAV_INLINE drwav_int32 drwav__bswap_s32(drwav_int32 n) +{ + return (drwav_int32)drwav__bswap32((drwav_uint32)n); +} + +static DRWAV_INLINE void drwav__bswap_samples_s32(drwav_int32* pSamples, drwav_uint64 sampleCount) +{ + drwav_uint64 iSample; + for (iSample = 0; iSample < sampleCount; iSample += 1) { + pSamples[iSample] = drwav__bswap_s32(pSamples[iSample]); + } +} + + +static DRWAV_INLINE float drwav__bswap_f32(float n) +{ + union { + drwav_uint32 i; + float f; + } x; + x.f = n; + x.i = drwav__bswap32(x.i); + + return x.f; +} + +static DRWAV_INLINE void drwav__bswap_samples_f32(float* pSamples, drwav_uint64 sampleCount) +{ + drwav_uint64 iSample; + for (iSample = 0; iSample < sampleCount; iSample += 1) { + pSamples[iSample] = drwav__bswap_f32(pSamples[iSample]); + } +} + + +static DRWAV_INLINE double drwav__bswap_f64(double n) +{ + union { + drwav_uint64 i; + double f; + } x; + x.f = n; + x.i = drwav__bswap64(x.i); + + return x.f; +} + +static DRWAV_INLINE void drwav__bswap_samples_f64(double* pSamples, drwav_uint64 sampleCount) +{ + drwav_uint64 iSample; + for (iSample = 0; iSample < sampleCount; iSample += 1) { + pSamples[iSample] = drwav__bswap_f64(pSamples[iSample]); + } +} + + +static DRWAV_INLINE void drwav__bswap_samples_pcm(void* pSamples, drwav_uint64 sampleCount, drwav_uint32 bytesPerSample) +{ + /* Assumes integer PCM. Floating point PCM is done in drwav__bswap_samples_ieee(). */ + switch (bytesPerSample) + { + case 2: /* s16, s12 (loosely packed) */ + { + drwav__bswap_samples_s16((drwav_int16*)pSamples, sampleCount); + } break; + case 3: /* s24 */ + { + drwav__bswap_samples_s24((drwav_uint8*)pSamples, sampleCount); + } break; + case 4: /* s32 */ + { + drwav__bswap_samples_s32((drwav_int32*)pSamples, sampleCount); + } break; + default: + { + /* Unsupported format. */ + DRWAV_ASSERT(DRWAV_FALSE); + } break; + } +} + +static DRWAV_INLINE void drwav__bswap_samples_ieee(void* pSamples, drwav_uint64 sampleCount, drwav_uint32 bytesPerSample) +{ + switch (bytesPerSample) + { + #if 0 /* Contributions welcome for f16 support. */ + case 2: /* f16 */ + { + drwav__bswap_samples_f16((drwav_float16*)pSamples, sampleCount); + } break; + #endif + case 4: /* f32 */ + { + drwav__bswap_samples_f32((float*)pSamples, sampleCount); + } break; + case 8: /* f64 */ + { + drwav__bswap_samples_f64((double*)pSamples, sampleCount); + } break; + default: + { + /* Unsupported format. */ + DRWAV_ASSERT(DRWAV_FALSE); + } break; + } +} + +static DRWAV_INLINE void drwav__bswap_samples(void* pSamples, drwav_uint64 sampleCount, drwav_uint32 bytesPerSample, drwav_uint16 format) +{ + switch (format) + { + case DR_WAVE_FORMAT_PCM: + { + drwav__bswap_samples_pcm(pSamples, sampleCount, bytesPerSample); + } break; + + case DR_WAVE_FORMAT_IEEE_FLOAT: + { + drwav__bswap_samples_ieee(pSamples, sampleCount, bytesPerSample); + } break; + + case DR_WAVE_FORMAT_ALAW: + case DR_WAVE_FORMAT_MULAW: + { + drwav__bswap_samples_s16((drwav_int16*)pSamples, sampleCount); + } break; + + case DR_WAVE_FORMAT_ADPCM: + case DR_WAVE_FORMAT_DVI_ADPCM: + default: + { + /* Unsupported format. */ + DRWAV_ASSERT(DRWAV_FALSE); + } break; + } +} + + +static void* drwav__malloc_default(size_t sz, void* pUserData) +{ + (void)pUserData; + return DRWAV_MALLOC(sz); +} + +static void* drwav__realloc_default(void* p, size_t sz, void* pUserData) +{ + (void)pUserData; + return DRWAV_REALLOC(p, sz); +} + +static void drwav__free_default(void* p, void* pUserData) +{ + (void)pUserData; + DRWAV_FREE(p); +} + + +static void* drwav__malloc_from_callbacks(size_t sz, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks == NULL) { + return NULL; + } + + if (pAllocationCallbacks->onMalloc != NULL) { + return pAllocationCallbacks->onMalloc(sz, pAllocationCallbacks->pUserData); + } + + /* Try using realloc(). */ + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(NULL, sz, pAllocationCallbacks->pUserData); + } + + return NULL; +} + +static void* drwav__realloc_from_callbacks(void* p, size_t szNew, size_t szOld, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks == NULL) { + return NULL; + } + + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(p, szNew, pAllocationCallbacks->pUserData); + } + + /* Try emulating realloc() in terms of malloc()/free(). */ + if (pAllocationCallbacks->onMalloc != NULL && pAllocationCallbacks->onFree != NULL) { + void* p2; + + p2 = pAllocationCallbacks->onMalloc(szNew, pAllocationCallbacks->pUserData); + if (p2 == NULL) { + return NULL; + } + + if (p != NULL) { + DRWAV_COPY_MEMORY(p2, p, szOld); + pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData); + } + + return p2; + } + + return NULL; +} + +static void drwav__free_from_callbacks(void* p, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (p == NULL || pAllocationCallbacks == NULL) { + return; + } + + if (pAllocationCallbacks->onFree != NULL) { + pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData); + } +} + + +static drwav_allocation_callbacks drwav_copy_allocation_callbacks_or_defaults(const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks != NULL) { + /* Copy. */ + return *pAllocationCallbacks; + } else { + /* Defaults. */ + drwav_allocation_callbacks allocationCallbacks; + allocationCallbacks.pUserData = NULL; + allocationCallbacks.onMalloc = drwav__malloc_default; + allocationCallbacks.onRealloc = drwav__realloc_default; + allocationCallbacks.onFree = drwav__free_default; + return allocationCallbacks; + } +} + + +static DRWAV_INLINE drwav_bool32 drwav__is_compressed_format_tag(drwav_uint16 formatTag) +{ + return + formatTag == DR_WAVE_FORMAT_ADPCM || + formatTag == DR_WAVE_FORMAT_DVI_ADPCM; +} + +static unsigned int drwav__chunk_padding_size_riff(drwav_uint64 chunkSize) +{ + return (unsigned int)(chunkSize % 2); +} + +static unsigned int drwav__chunk_padding_size_w64(drwav_uint64 chunkSize) +{ + return (unsigned int)(chunkSize % 8); +} + +static drwav_uint64 drwav_read_pcm_frames_s16__msadpcm(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut); +static drwav_uint64 drwav_read_pcm_frames_s16__ima(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut); +static drwav_bool32 drwav_init_write__internal(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount); + +static drwav_result drwav__read_chunk_header(drwav_read_proc onRead, void* pUserData, drwav_container container, drwav_uint64* pRunningBytesReadOut, drwav_chunk_header* pHeaderOut) +{ + if (container == drwav_container_riff || container == drwav_container_rf64) { + drwav_uint8 sizeInBytes[4]; + + if (onRead(pUserData, pHeaderOut->id.fourcc, 4) != 4) { + return DRWAV_AT_END; + } + + if (onRead(pUserData, sizeInBytes, 4) != 4) { + return DRWAV_INVALID_FILE; + } + + pHeaderOut->sizeInBytes = drwav__bytes_to_u32(sizeInBytes); + pHeaderOut->paddingSize = drwav__chunk_padding_size_riff(pHeaderOut->sizeInBytes); + *pRunningBytesReadOut += 8; + } else { + drwav_uint8 sizeInBytes[8]; + + if (onRead(pUserData, pHeaderOut->id.guid, 16) != 16) { + return DRWAV_AT_END; + } + + if (onRead(pUserData, sizeInBytes, 8) != 8) { + return DRWAV_INVALID_FILE; + } + + pHeaderOut->sizeInBytes = drwav__bytes_to_u64(sizeInBytes) - 24; /* <-- Subtract 24 because w64 includes the size of the header. */ + pHeaderOut->paddingSize = drwav__chunk_padding_size_w64(pHeaderOut->sizeInBytes); + *pRunningBytesReadOut += 24; + } + + return DRWAV_SUCCESS; +} + +static drwav_bool32 drwav__seek_forward(drwav_seek_proc onSeek, drwav_uint64 offset, void* pUserData) +{ + drwav_uint64 bytesRemainingToSeek = offset; + while (bytesRemainingToSeek > 0) { + if (bytesRemainingToSeek > 0x7FFFFFFF) { + if (!onSeek(pUserData, 0x7FFFFFFF, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + bytesRemainingToSeek -= 0x7FFFFFFF; + } else { + if (!onSeek(pUserData, (int)bytesRemainingToSeek, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + bytesRemainingToSeek = 0; + } + } + + return DRWAV_TRUE; +} + +static drwav_bool32 drwav__seek_from_start(drwav_seek_proc onSeek, drwav_uint64 offset, void* pUserData) +{ + if (offset <= 0x7FFFFFFF) { + return onSeek(pUserData, (int)offset, drwav_seek_origin_start); + } + + /* Larger than 32-bit seek. */ + if (!onSeek(pUserData, 0x7FFFFFFF, drwav_seek_origin_start)) { + return DRWAV_FALSE; + } + offset -= 0x7FFFFFFF; + + for (;;) { + if (offset <= 0x7FFFFFFF) { + return onSeek(pUserData, (int)offset, drwav_seek_origin_current); + } + + if (!onSeek(pUserData, 0x7FFFFFFF, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + offset -= 0x7FFFFFFF; + } + + /* Should never get here. */ + /*return DRWAV_TRUE; */ +} + + +static drwav_bool32 drwav__read_fmt(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, drwav_container container, drwav_uint64* pRunningBytesReadOut, drwav_fmt* fmtOut) +{ + drwav_chunk_header header; + drwav_uint8 fmt[16]; + + if (drwav__read_chunk_header(onRead, pUserData, container, pRunningBytesReadOut, &header) != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + + /* Skip non-fmt chunks. */ + while (((container == drwav_container_riff || container == drwav_container_rf64) && !drwav__fourcc_equal(header.id.fourcc, "fmt ")) || (container == drwav_container_w64 && !drwav__guid_equal(header.id.guid, drwavGUID_W64_FMT))) { + if (!drwav__seek_forward(onSeek, header.sizeInBytes + header.paddingSize, pUserData)) { + return DRWAV_FALSE; + } + *pRunningBytesReadOut += header.sizeInBytes + header.paddingSize; + + /* Try the next header. */ + if (drwav__read_chunk_header(onRead, pUserData, container, pRunningBytesReadOut, &header) != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + } + + + /* Validation. */ + if (container == drwav_container_riff || container == drwav_container_rf64) { + if (!drwav__fourcc_equal(header.id.fourcc, "fmt ")) { + return DRWAV_FALSE; + } + } else { + if (!drwav__guid_equal(header.id.guid, drwavGUID_W64_FMT)) { + return DRWAV_FALSE; + } + } + + + if (onRead(pUserData, fmt, sizeof(fmt)) != sizeof(fmt)) { + return DRWAV_FALSE; + } + *pRunningBytesReadOut += sizeof(fmt); + + fmtOut->formatTag = drwav__bytes_to_u16(fmt + 0); + fmtOut->channels = drwav__bytes_to_u16(fmt + 2); + fmtOut->sampleRate = drwav__bytes_to_u32(fmt + 4); + fmtOut->avgBytesPerSec = drwav__bytes_to_u32(fmt + 8); + fmtOut->blockAlign = drwav__bytes_to_u16(fmt + 12); + fmtOut->bitsPerSample = drwav__bytes_to_u16(fmt + 14); + + fmtOut->extendedSize = 0; + fmtOut->validBitsPerSample = 0; + fmtOut->channelMask = 0; + memset(fmtOut->subFormat, 0, sizeof(fmtOut->subFormat)); + + if (header.sizeInBytes > 16) { + drwav_uint8 fmt_cbSize[2]; + int bytesReadSoFar = 0; + + if (onRead(pUserData, fmt_cbSize, sizeof(fmt_cbSize)) != sizeof(fmt_cbSize)) { + return DRWAV_FALSE; /* Expecting more data. */ + } + *pRunningBytesReadOut += sizeof(fmt_cbSize); + + bytesReadSoFar = 18; + + fmtOut->extendedSize = drwav__bytes_to_u16(fmt_cbSize); + if (fmtOut->extendedSize > 0) { + /* Simple validation. */ + if (fmtOut->formatTag == DR_WAVE_FORMAT_EXTENSIBLE) { + if (fmtOut->extendedSize != 22) { + return DRWAV_FALSE; + } + } + + if (fmtOut->formatTag == DR_WAVE_FORMAT_EXTENSIBLE) { + drwav_uint8 fmtext[22]; + if (onRead(pUserData, fmtext, fmtOut->extendedSize) != fmtOut->extendedSize) { + return DRWAV_FALSE; /* Expecting more data. */ + } + + fmtOut->validBitsPerSample = drwav__bytes_to_u16(fmtext + 0); + fmtOut->channelMask = drwav__bytes_to_u32(fmtext + 2); + drwav__bytes_to_guid(fmtext + 6, fmtOut->subFormat); + } else { + if (!onSeek(pUserData, fmtOut->extendedSize, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + } + *pRunningBytesReadOut += fmtOut->extendedSize; + + bytesReadSoFar += fmtOut->extendedSize; + } + + /* Seek past any leftover bytes. For w64 the leftover will be defined based on the chunk size. */ + if (!onSeek(pUserData, (int)(header.sizeInBytes - bytesReadSoFar), drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + *pRunningBytesReadOut += (header.sizeInBytes - bytesReadSoFar); + } + + if (header.paddingSize > 0) { + if (!onSeek(pUserData, header.paddingSize, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + *pRunningBytesReadOut += header.paddingSize; + } + + return DRWAV_TRUE; +} + + +static size_t drwav__on_read(drwav_read_proc onRead, void* pUserData, void* pBufferOut, size_t bytesToRead, drwav_uint64* pCursor) +{ + size_t bytesRead; + + DRWAV_ASSERT(onRead != NULL); + DRWAV_ASSERT(pCursor != NULL); + + bytesRead = onRead(pUserData, pBufferOut, bytesToRead); + *pCursor += bytesRead; + return bytesRead; +} + +#if 0 +static drwav_bool32 drwav__on_seek(drwav_seek_proc onSeek, void* pUserData, int offset, drwav_seek_origin origin, drwav_uint64* pCursor) +{ + DRWAV_ASSERT(onSeek != NULL); + DRWAV_ASSERT(pCursor != NULL); + + if (!onSeek(pUserData, offset, origin)) { + return DRWAV_FALSE; + } + + if (origin == drwav_seek_origin_start) { + *pCursor = offset; + } else { + *pCursor += offset; + } + + return DRWAV_TRUE; +} +#endif + + + +static drwav_uint32 drwav_get_bytes_per_pcm_frame(drwav* pWav) +{ + /* + The bytes per frame is a bit ambiguous. It can be either be based on the bits per sample, or the block align. The way I'm doing it here + is that if the bits per sample is a multiple of 8, use floor(bitsPerSample*channels/8), otherwise fall back to the block align. + */ + if ((pWav->bitsPerSample & 0x7) == 0) { + /* Bits per sample is a multiple of 8. */ + return (pWav->bitsPerSample * pWav->fmt.channels) >> 3; + } else { + return pWav->fmt.blockAlign; + } +} + +DRWAV_API drwav_uint16 drwav_fmt_get_format(const drwav_fmt* pFMT) +{ + if (pFMT == NULL) { + return 0; + } + + if (pFMT->formatTag != DR_WAVE_FORMAT_EXTENSIBLE) { + return pFMT->formatTag; + } else { + return drwav__bytes_to_u16(pFMT->subFormat); /* Only the first two bytes are required. */ + } +} + +static drwav_bool32 drwav_preinit(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pReadSeekUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pWav == NULL || onRead == NULL || onSeek == NULL) { + return DRWAV_FALSE; + } + + DRWAV_ZERO_MEMORY(pWav, sizeof(*pWav)); + pWav->onRead = onRead; + pWav->onSeek = onSeek; + pWav->pUserData = pReadSeekUserData; + pWav->allocationCallbacks = drwav_copy_allocation_callbacks_or_defaults(pAllocationCallbacks); + + if (pWav->allocationCallbacks.onFree == NULL || (pWav->allocationCallbacks.onMalloc == NULL && pWav->allocationCallbacks.onRealloc == NULL)) { + return DRWAV_FALSE; /* Invalid allocation callbacks. */ + } + + return DRWAV_TRUE; +} + +static drwav_bool32 drwav_init__internal(drwav* pWav, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags) +{ + /* This function assumes drwav_preinit() has been called beforehand. */ + + drwav_uint64 cursor; /* <-- Keeps track of the byte position so we can seek to specific locations. */ + drwav_bool32 sequential; + drwav_uint8 riff[4]; + drwav_fmt fmt; + unsigned short translatedFormatTag; + drwav_bool32 foundDataChunk; + drwav_uint64 dataChunkSize = 0; /* <-- Important! Don't explicitly set this to 0 anywhere else. Calculation of the size of the data chunk is performed in different paths depending on the container. */ + drwav_uint64 sampleCountFromFactChunk = 0; /* Same as dataChunkSize - make sure this is the only place this is initialized to 0. */ + drwav_uint64 chunkSize; + + cursor = 0; + sequential = (flags & DRWAV_SEQUENTIAL) != 0; + + /* The first 4 bytes should be the RIFF identifier. */ + if (drwav__on_read(pWav->onRead, pWav->pUserData, riff, sizeof(riff), &cursor) != sizeof(riff)) { + return DRWAV_FALSE; + } + + /* + The first 4 bytes can be used to identify the container. For RIFF files it will start with "RIFF" and for + w64 it will start with "riff". + */ + if (drwav__fourcc_equal(riff, "RIFF")) { + pWav->container = drwav_container_riff; + } else if (drwav__fourcc_equal(riff, "riff")) { + int i; + drwav_uint8 riff2[12]; + + pWav->container = drwav_container_w64; + + /* Check the rest of the GUID for validity. */ + if (drwav__on_read(pWav->onRead, pWav->pUserData, riff2, sizeof(riff2), &cursor) != sizeof(riff2)) { + return DRWAV_FALSE; + } + + for (i = 0; i < 12; ++i) { + if (riff2[i] != drwavGUID_W64_RIFF[i+4]) { + return DRWAV_FALSE; + } + } + } else if (drwav__fourcc_equal(riff, "RF64")) { + pWav->container = drwav_container_rf64; + } else { + return DRWAV_FALSE; /* Unknown or unsupported container. */ + } + + + if (pWav->container == drwav_container_riff || pWav->container == drwav_container_rf64) { + drwav_uint8 chunkSizeBytes[4]; + drwav_uint8 wave[4]; + + /* RIFF/WAVE */ + if (drwav__on_read(pWav->onRead, pWav->pUserData, chunkSizeBytes, sizeof(chunkSizeBytes), &cursor) != sizeof(chunkSizeBytes)) { + return DRWAV_FALSE; + } + + if (pWav->container == drwav_container_riff) { + if (drwav__bytes_to_u32(chunkSizeBytes) < 36) { + return DRWAV_FALSE; /* Chunk size should always be at least 36 bytes. */ + } + } else { + if (drwav__bytes_to_u32(chunkSizeBytes) != 0xFFFFFFFF) { + return DRWAV_FALSE; /* Chunk size should always be set to -1/0xFFFFFFFF for RF64. The actual size is retrieved later. */ + } + } + + if (drwav__on_read(pWav->onRead, pWav->pUserData, wave, sizeof(wave), &cursor) != sizeof(wave)) { + return DRWAV_FALSE; + } + + if (!drwav__fourcc_equal(wave, "WAVE")) { + return DRWAV_FALSE; /* Expecting "WAVE". */ + } + } else { + drwav_uint8 chunkSizeBytes[8]; + drwav_uint8 wave[16]; + + /* W64 */ + if (drwav__on_read(pWav->onRead, pWav->pUserData, chunkSizeBytes, sizeof(chunkSizeBytes), &cursor) != sizeof(chunkSizeBytes)) { + return DRWAV_FALSE; + } + + if (drwav__bytes_to_u64(chunkSizeBytes) < 80) { + return DRWAV_FALSE; + } + + if (drwav__on_read(pWav->onRead, pWav->pUserData, wave, sizeof(wave), &cursor) != sizeof(wave)) { + return DRWAV_FALSE; + } + + if (!drwav__guid_equal(wave, drwavGUID_W64_WAVE)) { + return DRWAV_FALSE; + } + } + + + /* For RF64, the "ds64" chunk must come next, before the "fmt " chunk. */ + if (pWav->container == drwav_container_rf64) { + drwav_uint8 sizeBytes[8]; + drwav_uint64 bytesRemainingInChunk; + drwav_chunk_header header; + drwav_result result = drwav__read_chunk_header(pWav->onRead, pWav->pUserData, pWav->container, &cursor, &header); + if (result != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + if (!drwav__fourcc_equal(header.id.fourcc, "ds64")) { + return DRWAV_FALSE; /* Expecting "ds64". */ + } + + bytesRemainingInChunk = header.sizeInBytes + header.paddingSize; + + /* We don't care about the size of the RIFF chunk - skip it. */ + if (!drwav__seek_forward(pWav->onSeek, 8, pWav->pUserData)) { + return DRWAV_FALSE; + } + bytesRemainingInChunk -= 8; + cursor += 8; + + + /* Next 8 bytes is the size of the "data" chunk. */ + if (drwav__on_read(pWav->onRead, pWav->pUserData, sizeBytes, sizeof(sizeBytes), &cursor) != sizeof(sizeBytes)) { + return DRWAV_FALSE; + } + bytesRemainingInChunk -= 8; + dataChunkSize = drwav__bytes_to_u64(sizeBytes); + + + /* Next 8 bytes is the same count which we would usually derived from the FACT chunk if it was available. */ + if (drwav__on_read(pWav->onRead, pWav->pUserData, sizeBytes, sizeof(sizeBytes), &cursor) != sizeof(sizeBytes)) { + return DRWAV_FALSE; + } + bytesRemainingInChunk -= 8; + sampleCountFromFactChunk = drwav__bytes_to_u64(sizeBytes); + + + /* Skip over everything else. */ + if (!drwav__seek_forward(pWav->onSeek, bytesRemainingInChunk, pWav->pUserData)) { + return DRWAV_FALSE; + } + cursor += bytesRemainingInChunk; + } + + + /* The next bytes should be the "fmt " chunk. */ + if (!drwav__read_fmt(pWav->onRead, pWav->onSeek, pWav->pUserData, pWav->container, &cursor, &fmt)) { + return DRWAV_FALSE; /* Failed to read the "fmt " chunk. */ + } + + /* Basic validation. */ + if ((fmt.sampleRate == 0 || fmt.sampleRate > DRWAV_MAX_SAMPLE_RATE) || + (fmt.channels == 0 || fmt.channels > DRWAV_MAX_CHANNELS) || + (fmt.bitsPerSample == 0 || fmt.bitsPerSample > DRWAV_MAX_BITS_PER_SAMPLE) || + fmt.blockAlign == 0) { + return DRWAV_FALSE; /* Probably an invalid WAV file. */ + } + + + /* Translate the internal format. */ + translatedFormatTag = fmt.formatTag; + if (translatedFormatTag == DR_WAVE_FORMAT_EXTENSIBLE) { + translatedFormatTag = drwav__bytes_to_u16(fmt.subFormat + 0); + } + + + /* + We need to enumerate over each chunk for two reasons: + 1) The "data" chunk may not be the next one + 2) We may want to report each chunk back to the client + + In order to correctly report each chunk back to the client we will need to keep looping until the end of the file. + */ + foundDataChunk = DRWAV_FALSE; + + /* The next chunk we care about is the "data" chunk. This is not necessarily the next chunk so we'll need to loop. */ + for (;;) + { + drwav_chunk_header header; + drwav_result result = drwav__read_chunk_header(pWav->onRead, pWav->pUserData, pWav->container, &cursor, &header); + if (result != DRWAV_SUCCESS) { + if (!foundDataChunk) { + return DRWAV_FALSE; + } else { + break; /* Probably at the end of the file. Get out of the loop. */ + } + } + + /* Tell the client about this chunk. */ + if (!sequential && onChunk != NULL) { + drwav_uint64 callbackBytesRead = onChunk(pChunkUserData, pWav->onRead, pWav->onSeek, pWav->pUserData, &header, pWav->container, &fmt); + + /* + dr_wav may need to read the contents of the chunk, so we now need to seek back to the position before + we called the callback. + */ + if (callbackBytesRead > 0) { + if (!drwav__seek_from_start(pWav->onSeek, cursor, pWav->pUserData)) { + return DRWAV_FALSE; + } + } + } + + + if (!foundDataChunk) { + pWav->dataChunkDataPos = cursor; + } + + chunkSize = header.sizeInBytes; + if (pWav->container == drwav_container_riff || pWav->container == drwav_container_rf64) { + if (drwav__fourcc_equal(header.id.fourcc, "data")) { + foundDataChunk = DRWAV_TRUE; + if (pWav->container != drwav_container_rf64) { /* The data chunk size for RF64 will always be set to 0xFFFFFFFF here. It was set to it's true value earlier. */ + dataChunkSize = chunkSize; + } + } + } else { + if (drwav__guid_equal(header.id.guid, drwavGUID_W64_DATA)) { + foundDataChunk = DRWAV_TRUE; + dataChunkSize = chunkSize; + } + } + + /* + If at this point we have found the data chunk and we're running in sequential mode, we need to break out of this loop. The reason for + this is that we would otherwise require a backwards seek which sequential mode forbids. + */ + if (foundDataChunk && sequential) { + break; + } + + /* Optional. Get the total sample count from the FACT chunk. This is useful for compressed formats. */ + if (pWav->container == drwav_container_riff) { + if (drwav__fourcc_equal(header.id.fourcc, "fact")) { + drwav_uint32 sampleCount; + if (drwav__on_read(pWav->onRead, pWav->pUserData, &sampleCount, 4, &cursor) != 4) { + return DRWAV_FALSE; + } + chunkSize -= 4; + + if (!foundDataChunk) { + pWav->dataChunkDataPos = cursor; + } + + /* + The sample count in the "fact" chunk is either unreliable, or I'm not understanding it properly. For now I am only enabling this + for Microsoft ADPCM formats. + */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + sampleCountFromFactChunk = sampleCount; + } else { + sampleCountFromFactChunk = 0; + } + } + } else if (pWav->container == drwav_container_w64) { + if (drwav__guid_equal(header.id.guid, drwavGUID_W64_FACT)) { + if (drwav__on_read(pWav->onRead, pWav->pUserData, &sampleCountFromFactChunk, 8, &cursor) != 8) { + return DRWAV_FALSE; + } + chunkSize -= 8; + + if (!foundDataChunk) { + pWav->dataChunkDataPos = cursor; + } + } + } else if (pWav->container == drwav_container_rf64) { + /* We retrieved the sample count from the ds64 chunk earlier so no need to do that here. */ + } + + /* "smpl" chunk. */ + if (pWav->container == drwav_container_riff || pWav->container == drwav_container_rf64) { + if (drwav__fourcc_equal(header.id.fourcc, "smpl")) { + drwav_uint8 smplHeaderData[36]; /* 36 = size of the smpl header section, not including the loop data. */ + if (chunkSize >= sizeof(smplHeaderData)) { + drwav_uint64 bytesJustRead = drwav__on_read(pWav->onRead, pWav->pUserData, smplHeaderData, sizeof(smplHeaderData), &cursor); + chunkSize -= bytesJustRead; + + if (bytesJustRead == sizeof(smplHeaderData)) { + drwav_uint32 iLoop; + + pWav->smpl.manufacturer = drwav__bytes_to_u32(smplHeaderData+0); + pWav->smpl.product = drwav__bytes_to_u32(smplHeaderData+4); + pWav->smpl.samplePeriod = drwav__bytes_to_u32(smplHeaderData+8); + pWav->smpl.midiUnityNotes = drwav__bytes_to_u32(smplHeaderData+12); + pWav->smpl.midiPitchFraction = drwav__bytes_to_u32(smplHeaderData+16); + pWav->smpl.smpteFormat = drwav__bytes_to_u32(smplHeaderData+20); + pWav->smpl.smpteOffset = drwav__bytes_to_u32(smplHeaderData+24); + pWav->smpl.numSampleLoops = drwav__bytes_to_u32(smplHeaderData+28); + pWav->smpl.samplerData = drwav__bytes_to_u32(smplHeaderData+32); + + for (iLoop = 0; iLoop < pWav->smpl.numSampleLoops && iLoop < drwav_countof(pWav->smpl.loops); ++iLoop) { + drwav_uint8 smplLoopData[24]; /* 24 = size of a loop section in the smpl chunk. */ + bytesJustRead = drwav__on_read(pWav->onRead, pWav->pUserData, smplLoopData, sizeof(smplLoopData), &cursor); + chunkSize -= bytesJustRead; + + if (bytesJustRead == sizeof(smplLoopData)) { + pWav->smpl.loops[iLoop].cuePointId = drwav__bytes_to_u32(smplLoopData+0); + pWav->smpl.loops[iLoop].type = drwav__bytes_to_u32(smplLoopData+4); + pWav->smpl.loops[iLoop].start = drwav__bytes_to_u32(smplLoopData+8); + pWav->smpl.loops[iLoop].end = drwav__bytes_to_u32(smplLoopData+12); + pWav->smpl.loops[iLoop].fraction = drwav__bytes_to_u32(smplLoopData+16); + pWav->smpl.loops[iLoop].playCount = drwav__bytes_to_u32(smplLoopData+20); + } else { + break; /* Break from the smpl loop for loop. */ + } + } + } + } else { + /* Looks like invalid data. Ignore the chunk. */ + } + } + } else { + if (drwav__guid_equal(header.id.guid, drwavGUID_W64_SMPL)) { + /* + This path will be hit when a W64 WAV file contains a smpl chunk. I don't have a sample file to test this path, so a contribution + is welcome to add support for this. + */ + } + } + + /* Make sure we seek past the padding. */ + chunkSize += header.paddingSize; + if (!drwav__seek_forward(pWav->onSeek, chunkSize, pWav->pUserData)) { + break; + } + cursor += chunkSize; + + if (!foundDataChunk) { + pWav->dataChunkDataPos = cursor; + } + } + + /* If we haven't found a data chunk, return an error. */ + if (!foundDataChunk) { + return DRWAV_FALSE; + } + + /* We may have moved passed the data chunk. If so we need to move back. If running in sequential mode we can assume we are already sitting on the data chunk. */ + if (!sequential) { + if (!drwav__seek_from_start(pWav->onSeek, pWav->dataChunkDataPos, pWav->pUserData)) { + return DRWAV_FALSE; + } + cursor = pWav->dataChunkDataPos; + } + + + /* At this point we should be sitting on the first byte of the raw audio data. */ + + pWav->fmt = fmt; + pWav->sampleRate = fmt.sampleRate; + pWav->channels = fmt.channels; + pWav->bitsPerSample = fmt.bitsPerSample; + pWav->bytesRemaining = dataChunkSize; + pWav->translatedFormatTag = translatedFormatTag; + pWav->dataChunkDataSize = dataChunkSize; + + if (sampleCountFromFactChunk != 0) { + pWav->totalPCMFrameCount = sampleCountFromFactChunk; + } else { + pWav->totalPCMFrameCount = dataChunkSize / drwav_get_bytes_per_pcm_frame(pWav); + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + drwav_uint64 totalBlockHeaderSizeInBytes; + drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign; + + /* Make sure any trailing partial block is accounted for. */ + if ((blockCount * fmt.blockAlign) < dataChunkSize) { + blockCount += 1; + } + + /* We decode two samples per byte. There will be blockCount headers in the data chunk. This is enough to know how to calculate the total PCM frame count. */ + totalBlockHeaderSizeInBytes = blockCount * (6*fmt.channels); + pWav->totalPCMFrameCount = ((dataChunkSize - totalBlockHeaderSizeInBytes) * 2) / fmt.channels; + } + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + drwav_uint64 totalBlockHeaderSizeInBytes; + drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign; + + /* Make sure any trailing partial block is accounted for. */ + if ((blockCount * fmt.blockAlign) < dataChunkSize) { + blockCount += 1; + } + + /* We decode two samples per byte. There will be blockCount headers in the data chunk. This is enough to know how to calculate the total PCM frame count. */ + totalBlockHeaderSizeInBytes = blockCount * (4*fmt.channels); + pWav->totalPCMFrameCount = ((dataChunkSize - totalBlockHeaderSizeInBytes) * 2) / fmt.channels; + + /* The header includes a decoded sample for each channel which acts as the initial predictor sample. */ + pWav->totalPCMFrameCount += blockCount; + } + } + + /* Some formats only support a certain number of channels. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + if (pWav->channels > 2) { + return DRWAV_FALSE; + } + } + +#ifdef DR_WAV_LIBSNDFILE_COMPAT + /* + I use libsndfile as a benchmark for testing, however in the version I'm using (from the Windows installer on the libsndfile website), + it appears the total sample count libsndfile uses for MS-ADPCM is incorrect. It would seem they are computing the total sample count + from the number of blocks, however this results in the inclusion of extra silent samples at the end of the last block. The correct + way to know the total sample count is to inspect the "fact" chunk, which should always be present for compressed formats, and should + always include the sample count. This little block of code below is only used to emulate the libsndfile logic so I can properly run my + correctness tests against libsndfile, and is disabled by default. + */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign; + pWav->totalPCMFrameCount = (((blockCount * (fmt.blockAlign - (6*pWav->channels))) * 2)) / fmt.channels; /* x2 because two samples per byte. */ + } + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign; + pWav->totalPCMFrameCount = (((blockCount * (fmt.blockAlign - (4*pWav->channels))) * 2) + (blockCount * pWav->channels)) / fmt.channels; + } +#endif + + return DRWAV_TRUE; +} + +DRWAV_API drwav_bool32 drwav_init(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_ex(pWav, onRead, onSeek, NULL, pUserData, NULL, 0, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_ex(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_chunk_proc onChunk, void* pReadSeekUserData, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (!drwav_preinit(pWav, onRead, onSeek, pReadSeekUserData, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + return drwav_init__internal(pWav, onChunk, pChunkUserData, flags); +} + + +static drwav_uint32 drwav__riff_chunk_size_riff(drwav_uint64 dataChunkSize) +{ + drwav_uint64 chunkSize = 4 + 24 + dataChunkSize + drwav__chunk_padding_size_riff(dataChunkSize); /* 4 = "WAVE". 24 = "fmt " chunk. */ + if (chunkSize > 0xFFFFFFFFUL) { + chunkSize = 0xFFFFFFFFUL; + } + + return (drwav_uint32)chunkSize; /* Safe cast due to the clamp above. */ +} + +static drwav_uint32 drwav__data_chunk_size_riff(drwav_uint64 dataChunkSize) +{ + if (dataChunkSize <= 0xFFFFFFFFUL) { + return (drwav_uint32)dataChunkSize; + } else { + return 0xFFFFFFFFUL; + } +} + +static drwav_uint64 drwav__riff_chunk_size_w64(drwav_uint64 dataChunkSize) +{ + drwav_uint64 dataSubchunkPaddingSize = drwav__chunk_padding_size_w64(dataChunkSize); + + return 80 + 24 + dataChunkSize + dataSubchunkPaddingSize; /* +24 because W64 includes the size of the GUID and size fields. */ +} + +static drwav_uint64 drwav__data_chunk_size_w64(drwav_uint64 dataChunkSize) +{ + return 24 + dataChunkSize; /* +24 because W64 includes the size of the GUID and size fields. */ +} + +static drwav_uint64 drwav__riff_chunk_size_rf64(drwav_uint64 dataChunkSize) +{ + drwav_uint64 chunkSize = 4 + 36 + 24 + dataChunkSize + drwav__chunk_padding_size_riff(dataChunkSize); /* 4 = "WAVE". 36 = "ds64" chunk. 24 = "fmt " chunk. */ + if (chunkSize > 0xFFFFFFFFUL) { + chunkSize = 0xFFFFFFFFUL; + } + + return chunkSize; +} + +static drwav_uint64 drwav__data_chunk_size_rf64(drwav_uint64 dataChunkSize) +{ + return dataChunkSize; +} + + +static size_t drwav__write(drwav* pWav, const void* pData, size_t dataSize) +{ + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->onWrite != NULL); + + /* Generic write. Assumes no byte reordering required. */ + return pWav->onWrite(pWav->pUserData, pData, dataSize); +} + +static size_t drwav__write_u16ne_to_le(drwav* pWav, drwav_uint16 value) +{ + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->onWrite != NULL); + + if (!drwav__is_little_endian()) { + value = drwav__bswap16(value); + } + + return drwav__write(pWav, &value, 2); +} + +static size_t drwav__write_u32ne_to_le(drwav* pWav, drwav_uint32 value) +{ + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->onWrite != NULL); + + if (!drwav__is_little_endian()) { + value = drwav__bswap32(value); + } + + return drwav__write(pWav, &value, 4); +} + +static size_t drwav__write_u64ne_to_le(drwav* pWav, drwav_uint64 value) +{ + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->onWrite != NULL); + + if (!drwav__is_little_endian()) { + value = drwav__bswap64(value); + } + + return drwav__write(pWav, &value, 8); +} + + +static drwav_bool32 drwav_preinit_write(drwav* pWav, const drwav_data_format* pFormat, drwav_bool32 isSequential, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pWav == NULL || onWrite == NULL) { + return DRWAV_FALSE; + } + + if (!isSequential && onSeek == NULL) { + return DRWAV_FALSE; /* <-- onSeek is required when in non-sequential mode. */ + } + + /* Not currently supporting compressed formats. Will need to add support for the "fact" chunk before we enable this. */ + if (pFormat->format == DR_WAVE_FORMAT_EXTENSIBLE) { + return DRWAV_FALSE; + } + if (pFormat->format == DR_WAVE_FORMAT_ADPCM || pFormat->format == DR_WAVE_FORMAT_DVI_ADPCM) { + return DRWAV_FALSE; + } + + DRWAV_ZERO_MEMORY(pWav, sizeof(*pWav)); + pWav->onWrite = onWrite; + pWav->onSeek = onSeek; + pWav->pUserData = pUserData; + pWav->allocationCallbacks = drwav_copy_allocation_callbacks_or_defaults(pAllocationCallbacks); + + if (pWav->allocationCallbacks.onFree == NULL || (pWav->allocationCallbacks.onMalloc == NULL && pWav->allocationCallbacks.onRealloc == NULL)) { + return DRWAV_FALSE; /* Invalid allocation callbacks. */ + } + + pWav->fmt.formatTag = (drwav_uint16)pFormat->format; + pWav->fmt.channels = (drwav_uint16)pFormat->channels; + pWav->fmt.sampleRate = pFormat->sampleRate; + pWav->fmt.avgBytesPerSec = (drwav_uint32)((pFormat->bitsPerSample * pFormat->sampleRate * pFormat->channels) / 8); + pWav->fmt.blockAlign = (drwav_uint16)((pFormat->channels * pFormat->bitsPerSample) / 8); + pWav->fmt.bitsPerSample = (drwav_uint16)pFormat->bitsPerSample; + pWav->fmt.extendedSize = 0; + pWav->isSequentialWrite = isSequential; + + return DRWAV_TRUE; +} + +static drwav_bool32 drwav_init_write__internal(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount) +{ + /* The function assumes drwav_preinit_write() was called beforehand. */ + + size_t runningPos = 0; + drwav_uint64 initialDataChunkSize = 0; + drwav_uint64 chunkSizeFMT; + + /* + The initial values for the "RIFF" and "data" chunks depends on whether or not we are initializing in sequential mode or not. In + sequential mode we set this to its final values straight away since they can be calculated from the total sample count. In non- + sequential mode we initialize it all to zero and fill it out in drwav_uninit() using a backwards seek. + */ + if (pWav->isSequentialWrite) { + initialDataChunkSize = (totalSampleCount * pWav->fmt.bitsPerSample) / 8; + + /* + The RIFF container has a limit on the number of samples. drwav is not allowing this. There's no practical limits for Wave64 + so for the sake of simplicity I'm not doing any validation for that. + */ + if (pFormat->container == drwav_container_riff) { + if (initialDataChunkSize > (0xFFFFFFFFUL - 36)) { + return DRWAV_FALSE; /* Not enough room to store every sample. */ + } + } + } + + pWav->dataChunkDataSizeTargetWrite = initialDataChunkSize; + + + /* "RIFF" chunk. */ + if (pFormat->container == drwav_container_riff) { + drwav_uint32 chunkSizeRIFF = 28 + (drwav_uint32)initialDataChunkSize; /* +28 = "WAVE" + [sizeof "fmt " chunk] */ + runningPos += drwav__write(pWav, "RIFF", 4); + runningPos += drwav__write_u32ne_to_le(pWav, chunkSizeRIFF); + runningPos += drwav__write(pWav, "WAVE", 4); + } else if (pFormat->container == drwav_container_w64) { + drwav_uint64 chunkSizeRIFF = 80 + 24 + initialDataChunkSize; /* +24 because W64 includes the size of the GUID and size fields. */ + runningPos += drwav__write(pWav, drwavGUID_W64_RIFF, 16); + runningPos += drwav__write_u64ne_to_le(pWav, chunkSizeRIFF); + runningPos += drwav__write(pWav, drwavGUID_W64_WAVE, 16); + } else if (pFormat->container == drwav_container_rf64) { + runningPos += drwav__write(pWav, "RF64", 4); + runningPos += drwav__write_u32ne_to_le(pWav, 0xFFFFFFFF); /* Always 0xFFFFFFFF for RF64. Set to a proper value in the "ds64" chunk. */ + runningPos += drwav__write(pWav, "WAVE", 4); + } + + + /* "ds64" chunk (RF64 only). */ + if (pFormat->container == drwav_container_rf64) { + drwav_uint32 initialds64ChunkSize = 28; /* 28 = [Size of RIFF (8 bytes)] + [Size of DATA (8 bytes)] + [Sample Count (8 bytes)] + [Table Length (4 bytes)]. Table length always set to 0. */ + drwav_uint64 initialRiffChunkSize = 8 + initialds64ChunkSize + initialDataChunkSize; /* +8 for the ds64 header. */ + + runningPos += drwav__write(pWav, "ds64", 4); + runningPos += drwav__write_u32ne_to_le(pWav, initialds64ChunkSize); /* Size of ds64. */ + runningPos += drwav__write_u64ne_to_le(pWav, initialRiffChunkSize); /* Size of RIFF. Set to true value at the end. */ + runningPos += drwav__write_u64ne_to_le(pWav, initialDataChunkSize); /* Size of DATA. Set to true value at the end. */ + runningPos += drwav__write_u64ne_to_le(pWav, totalSampleCount); /* Sample count. */ + runningPos += drwav__write_u32ne_to_le(pWav, 0); /* Table length. Always set to zero in our case since we're not doing any other chunks than "DATA". */ + } + + + /* "fmt " chunk. */ + if (pFormat->container == drwav_container_riff || pFormat->container == drwav_container_rf64) { + chunkSizeFMT = 16; + runningPos += drwav__write(pWav, "fmt ", 4); + runningPos += drwav__write_u32ne_to_le(pWav, (drwav_uint32)chunkSizeFMT); + } else if (pFormat->container == drwav_container_w64) { + chunkSizeFMT = 40; + runningPos += drwav__write(pWav, drwavGUID_W64_FMT, 16); + runningPos += drwav__write_u64ne_to_le(pWav, chunkSizeFMT); + } + + runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.formatTag); + runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.channels); + runningPos += drwav__write_u32ne_to_le(pWav, pWav->fmt.sampleRate); + runningPos += drwav__write_u32ne_to_le(pWav, pWav->fmt.avgBytesPerSec); + runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.blockAlign); + runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.bitsPerSample); + + pWav->dataChunkDataPos = runningPos; + + /* "data" chunk. */ + if (pFormat->container == drwav_container_riff) { + drwav_uint32 chunkSizeDATA = (drwav_uint32)initialDataChunkSize; + runningPos += drwav__write(pWav, "data", 4); + runningPos += drwav__write_u32ne_to_le(pWav, chunkSizeDATA); + } else if (pFormat->container == drwav_container_w64) { + drwav_uint64 chunkSizeDATA = 24 + initialDataChunkSize; /* +24 because W64 includes the size of the GUID and size fields. */ + runningPos += drwav__write(pWav, drwavGUID_W64_DATA, 16); + runningPos += drwav__write_u64ne_to_le(pWav, chunkSizeDATA); + } else if (pFormat->container == drwav_container_rf64) { + runningPos += drwav__write(pWav, "data", 4); + runningPos += drwav__write_u32ne_to_le(pWav, 0xFFFFFFFF); /* Always set to 0xFFFFFFFF for RF64. The true size of the data chunk is specified in the ds64 chunk. */ + } + + /* + The runningPos variable is incremented in the section above but is left unused which is causing some static analysis tools to detect it + as a dead store. I'm leaving this as-is for safety just in case I want to expand this function later to include other tags and want to + keep track of the running position for whatever reason. The line below should silence the static analysis tools. + */ + (void)runningPos; + + /* Set some properties for the client's convenience. */ + pWav->container = pFormat->container; + pWav->channels = (drwav_uint16)pFormat->channels; + pWav->sampleRate = pFormat->sampleRate; + pWav->bitsPerSample = (drwav_uint16)pFormat->bitsPerSample; + pWav->translatedFormatTag = (drwav_uint16)pFormat->format; + + return DRWAV_TRUE; +} + + +DRWAV_API drwav_bool32 drwav_init_write(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (!drwav_preinit_write(pWav, pFormat, DRWAV_FALSE, onWrite, onSeek, pUserData, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + return drwav_init_write__internal(pWav, pFormat, 0); /* DRWAV_FALSE = Not Sequential */ +} + +DRWAV_API drwav_bool32 drwav_init_write_sequential(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (!drwav_preinit_write(pWav, pFormat, DRWAV_TRUE, onWrite, NULL, pUserData, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + return drwav_init_write__internal(pWav, pFormat, totalSampleCount); /* DRWAV_TRUE = Sequential */ +} + +DRWAV_API drwav_bool32 drwav_init_write_sequential_pcm_frames(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pFormat == NULL) { + return DRWAV_FALSE; + } + + return drwav_init_write_sequential(pWav, pFormat, totalPCMFrameCount*pFormat->channels, onWrite, pUserData, pAllocationCallbacks); +} + +DRWAV_API drwav_uint64 drwav_target_write_size_bytes(const drwav_data_format* pFormat, drwav_uint64 totalSampleCount) +{ + /* Casting totalSampleCount to drwav_int64 for VC6 compatibility. No issues in practice because nobody is going to exhaust the whole 63 bits. */ + drwav_uint64 targetDataSizeBytes = (drwav_uint64)((drwav_int64)totalSampleCount * pFormat->channels * pFormat->bitsPerSample/8.0); + drwav_uint64 riffChunkSizeBytes; + drwav_uint64 fileSizeBytes = 0; + + if (pFormat->container == drwav_container_riff) { + riffChunkSizeBytes = drwav__riff_chunk_size_riff(targetDataSizeBytes); + fileSizeBytes = (8 + riffChunkSizeBytes); /* +8 because WAV doesn't include the size of the ChunkID and ChunkSize fields. */ + } else if (pFormat->container == drwav_container_w64) { + riffChunkSizeBytes = drwav__riff_chunk_size_w64(targetDataSizeBytes); + fileSizeBytes = riffChunkSizeBytes; + } else if (pFormat->container == drwav_container_rf64) { + riffChunkSizeBytes = drwav__riff_chunk_size_rf64(targetDataSizeBytes); + fileSizeBytes = (8 + riffChunkSizeBytes); /* +8 because WAV doesn't include the size of the ChunkID and ChunkSize fields. */ + } + + return fileSizeBytes; +} + + +#ifndef DR_WAV_NO_STDIO + +/* drwav_result_from_errno() is only used for fopen() and wfopen() so putting it inside DR_WAV_NO_STDIO for now. If something else needs this later we can move it out. */ +#include +static drwav_result drwav_result_from_errno(int e) +{ + switch (e) + { + case 0: return DRWAV_SUCCESS; + #ifdef EPERM + case EPERM: return DRWAV_INVALID_OPERATION; + #endif + #ifdef ENOENT + case ENOENT: return DRWAV_DOES_NOT_EXIST; + #endif + #ifdef ESRCH + case ESRCH: return DRWAV_DOES_NOT_EXIST; + #endif + #ifdef EINTR + case EINTR: return DRWAV_INTERRUPT; + #endif + #ifdef EIO + case EIO: return DRWAV_IO_ERROR; + #endif + #ifdef ENXIO + case ENXIO: return DRWAV_DOES_NOT_EXIST; + #endif + #ifdef E2BIG + case E2BIG: return DRWAV_INVALID_ARGS; + #endif + #ifdef ENOEXEC + case ENOEXEC: return DRWAV_INVALID_FILE; + #endif + #ifdef EBADF + case EBADF: return DRWAV_INVALID_FILE; + #endif + #ifdef ECHILD + case ECHILD: return DRWAV_ERROR; + #endif + #ifdef EAGAIN + case EAGAIN: return DRWAV_UNAVAILABLE; + #endif + #ifdef ENOMEM + case ENOMEM: return DRWAV_OUT_OF_MEMORY; + #endif + #ifdef EACCES + case EACCES: return DRWAV_ACCESS_DENIED; + #endif + #ifdef EFAULT + case EFAULT: return DRWAV_BAD_ADDRESS; + #endif + #ifdef ENOTBLK + case ENOTBLK: return DRWAV_ERROR; + #endif + #ifdef EBUSY + case EBUSY: return DRWAV_BUSY; + #endif + #ifdef EEXIST + case EEXIST: return DRWAV_ALREADY_EXISTS; + #endif + #ifdef EXDEV + case EXDEV: return DRWAV_ERROR; + #endif + #ifdef ENODEV + case ENODEV: return DRWAV_DOES_NOT_EXIST; + #endif + #ifdef ENOTDIR + case ENOTDIR: return DRWAV_NOT_DIRECTORY; + #endif + #ifdef EISDIR + case EISDIR: return DRWAV_IS_DIRECTORY; + #endif + #ifdef EINVAL + case EINVAL: return DRWAV_INVALID_ARGS; + #endif + #ifdef ENFILE + case ENFILE: return DRWAV_TOO_MANY_OPEN_FILES; + #endif + #ifdef EMFILE + case EMFILE: return DRWAV_TOO_MANY_OPEN_FILES; + #endif + #ifdef ENOTTY + case ENOTTY: return DRWAV_INVALID_OPERATION; + #endif + #ifdef ETXTBSY + case ETXTBSY: return DRWAV_BUSY; + #endif + #ifdef EFBIG + case EFBIG: return DRWAV_TOO_BIG; + #endif + #ifdef ENOSPC + case ENOSPC: return DRWAV_NO_SPACE; + #endif + #ifdef ESPIPE + case ESPIPE: return DRWAV_BAD_SEEK; + #endif + #ifdef EROFS + case EROFS: return DRWAV_ACCESS_DENIED; + #endif + #ifdef EMLINK + case EMLINK: return DRWAV_TOO_MANY_LINKS; + #endif + #ifdef EPIPE + case EPIPE: return DRWAV_BAD_PIPE; + #endif + #ifdef EDOM + case EDOM: return DRWAV_OUT_OF_RANGE; + #endif + #ifdef ERANGE + case ERANGE: return DRWAV_OUT_OF_RANGE; + #endif + #ifdef EDEADLK + case EDEADLK: return DRWAV_DEADLOCK; + #endif + #ifdef ENAMETOOLONG + case ENAMETOOLONG: return DRWAV_PATH_TOO_LONG; + #endif + #ifdef ENOLCK + case ENOLCK: return DRWAV_ERROR; + #endif + #ifdef ENOSYS + case ENOSYS: return DRWAV_NOT_IMPLEMENTED; + #endif + #ifdef ENOTEMPTY + case ENOTEMPTY: return DRWAV_DIRECTORY_NOT_EMPTY; + #endif + #ifdef ELOOP + case ELOOP: return DRWAV_TOO_MANY_LINKS; + #endif + #ifdef ENOMSG + case ENOMSG: return DRWAV_NO_MESSAGE; + #endif + #ifdef EIDRM + case EIDRM: return DRWAV_ERROR; + #endif + #ifdef ECHRNG + case ECHRNG: return DRWAV_ERROR; + #endif + #ifdef EL2NSYNC + case EL2NSYNC: return DRWAV_ERROR; + #endif + #ifdef EL3HLT + case EL3HLT: return DRWAV_ERROR; + #endif + #ifdef EL3RST + case EL3RST: return DRWAV_ERROR; + #endif + #ifdef ELNRNG + case ELNRNG: return DRWAV_OUT_OF_RANGE; + #endif + #ifdef EUNATCH + case EUNATCH: return DRWAV_ERROR; + #endif + #ifdef ENOCSI + case ENOCSI: return DRWAV_ERROR; + #endif + #ifdef EL2HLT + case EL2HLT: return DRWAV_ERROR; + #endif + #ifdef EBADE + case EBADE: return DRWAV_ERROR; + #endif + #ifdef EBADR + case EBADR: return DRWAV_ERROR; + #endif + #ifdef EXFULL + case EXFULL: return DRWAV_ERROR; + #endif + #ifdef ENOANO + case ENOANO: return DRWAV_ERROR; + #endif + #ifdef EBADRQC + case EBADRQC: return DRWAV_ERROR; + #endif + #ifdef EBADSLT + case EBADSLT: return DRWAV_ERROR; + #endif + #ifdef EBFONT + case EBFONT: return DRWAV_INVALID_FILE; + #endif + #ifdef ENOSTR + case ENOSTR: return DRWAV_ERROR; + #endif + #ifdef ENODATA + case ENODATA: return DRWAV_NO_DATA_AVAILABLE; + #endif + #ifdef ETIME + case ETIME: return DRWAV_TIMEOUT; + #endif + #ifdef ENOSR + case ENOSR: return DRWAV_NO_DATA_AVAILABLE; + #endif + #ifdef ENONET + case ENONET: return DRWAV_NO_NETWORK; + #endif + #ifdef ENOPKG + case ENOPKG: return DRWAV_ERROR; + #endif + #ifdef EREMOTE + case EREMOTE: return DRWAV_ERROR; + #endif + #ifdef ENOLINK + case ENOLINK: return DRWAV_ERROR; + #endif + #ifdef EADV + case EADV: return DRWAV_ERROR; + #endif + #ifdef ESRMNT + case ESRMNT: return DRWAV_ERROR; + #endif + #ifdef ECOMM + case ECOMM: return DRWAV_ERROR; + #endif + #ifdef EPROTO + case EPROTO: return DRWAV_ERROR; + #endif + #ifdef EMULTIHOP + case EMULTIHOP: return DRWAV_ERROR; + #endif + #ifdef EDOTDOT + case EDOTDOT: return DRWAV_ERROR; + #endif + #ifdef EBADMSG + case EBADMSG: return DRWAV_BAD_MESSAGE; + #endif + #ifdef EOVERFLOW + case EOVERFLOW: return DRWAV_TOO_BIG; + #endif + #ifdef ENOTUNIQ + case ENOTUNIQ: return DRWAV_NOT_UNIQUE; + #endif + #ifdef EBADFD + case EBADFD: return DRWAV_ERROR; + #endif + #ifdef EREMCHG + case EREMCHG: return DRWAV_ERROR; + #endif + #ifdef ELIBACC + case ELIBACC: return DRWAV_ACCESS_DENIED; + #endif + #ifdef ELIBBAD + case ELIBBAD: return DRWAV_INVALID_FILE; + #endif + #ifdef ELIBSCN + case ELIBSCN: return DRWAV_INVALID_FILE; + #endif + #ifdef ELIBMAX + case ELIBMAX: return DRWAV_ERROR; + #endif + #ifdef ELIBEXEC + case ELIBEXEC: return DRWAV_ERROR; + #endif + #ifdef EILSEQ + case EILSEQ: return DRWAV_INVALID_DATA; + #endif + #ifdef ERESTART + case ERESTART: return DRWAV_ERROR; + #endif + #ifdef ESTRPIPE + case ESTRPIPE: return DRWAV_ERROR; + #endif + #ifdef EUSERS + case EUSERS: return DRWAV_ERROR; + #endif + #ifdef ENOTSOCK + case ENOTSOCK: return DRWAV_NOT_SOCKET; + #endif + #ifdef EDESTADDRREQ + case EDESTADDRREQ: return DRWAV_NO_ADDRESS; + #endif + #ifdef EMSGSIZE + case EMSGSIZE: return DRWAV_TOO_BIG; + #endif + #ifdef EPROTOTYPE + case EPROTOTYPE: return DRWAV_BAD_PROTOCOL; + #endif + #ifdef ENOPROTOOPT + case ENOPROTOOPT: return DRWAV_PROTOCOL_UNAVAILABLE; + #endif + #ifdef EPROTONOSUPPORT + case EPROTONOSUPPORT: return DRWAV_PROTOCOL_NOT_SUPPORTED; + #endif + #ifdef ESOCKTNOSUPPORT + case ESOCKTNOSUPPORT: return DRWAV_SOCKET_NOT_SUPPORTED; + #endif + #ifdef EOPNOTSUPP + case EOPNOTSUPP: return DRWAV_INVALID_OPERATION; + #endif + #ifdef EPFNOSUPPORT + case EPFNOSUPPORT: return DRWAV_PROTOCOL_FAMILY_NOT_SUPPORTED; + #endif + #ifdef EAFNOSUPPORT + case EAFNOSUPPORT: return DRWAV_ADDRESS_FAMILY_NOT_SUPPORTED; + #endif + #ifdef EADDRINUSE + case EADDRINUSE: return DRWAV_ALREADY_IN_USE; + #endif + #ifdef EADDRNOTAVAIL + case EADDRNOTAVAIL: return DRWAV_ERROR; + #endif + #ifdef ENETDOWN + case ENETDOWN: return DRWAV_NO_NETWORK; + #endif + #ifdef ENETUNREACH + case ENETUNREACH: return DRWAV_NO_NETWORK; + #endif + #ifdef ENETRESET + case ENETRESET: return DRWAV_NO_NETWORK; + #endif + #ifdef ECONNABORTED + case ECONNABORTED: return DRWAV_NO_NETWORK; + #endif + #ifdef ECONNRESET + case ECONNRESET: return DRWAV_CONNECTION_RESET; + #endif + #ifdef ENOBUFS + case ENOBUFS: return DRWAV_NO_SPACE; + #endif + #ifdef EISCONN + case EISCONN: return DRWAV_ALREADY_CONNECTED; + #endif + #ifdef ENOTCONN + case ENOTCONN: return DRWAV_NOT_CONNECTED; + #endif + #ifdef ESHUTDOWN + case ESHUTDOWN: return DRWAV_ERROR; + #endif + #ifdef ETOOMANYREFS + case ETOOMANYREFS: return DRWAV_ERROR; + #endif + #ifdef ETIMEDOUT + case ETIMEDOUT: return DRWAV_TIMEOUT; + #endif + #ifdef ECONNREFUSED + case ECONNREFUSED: return DRWAV_CONNECTION_REFUSED; + #endif + #ifdef EHOSTDOWN + case EHOSTDOWN: return DRWAV_NO_HOST; + #endif + #ifdef EHOSTUNREACH + case EHOSTUNREACH: return DRWAV_NO_HOST; + #endif + #ifdef EALREADY + case EALREADY: return DRWAV_IN_PROGRESS; + #endif + #ifdef EINPROGRESS + case EINPROGRESS: return DRWAV_IN_PROGRESS; + #endif + #ifdef ESTALE + case ESTALE: return DRWAV_INVALID_FILE; + #endif + #ifdef EUCLEAN + case EUCLEAN: return DRWAV_ERROR; + #endif + #ifdef ENOTNAM + case ENOTNAM: return DRWAV_ERROR; + #endif + #ifdef ENAVAIL + case ENAVAIL: return DRWAV_ERROR; + #endif + #ifdef EISNAM + case EISNAM: return DRWAV_ERROR; + #endif + #ifdef EREMOTEIO + case EREMOTEIO: return DRWAV_IO_ERROR; + #endif + #ifdef EDQUOT + case EDQUOT: return DRWAV_NO_SPACE; + #endif + #ifdef ENOMEDIUM + case ENOMEDIUM: return DRWAV_DOES_NOT_EXIST; + #endif + #ifdef EMEDIUMTYPE + case EMEDIUMTYPE: return DRWAV_ERROR; + #endif + #ifdef ECANCELED + case ECANCELED: return DRWAV_CANCELLED; + #endif + #ifdef ENOKEY + case ENOKEY: return DRWAV_ERROR; + #endif + #ifdef EKEYEXPIRED + case EKEYEXPIRED: return DRWAV_ERROR; + #endif + #ifdef EKEYREVOKED + case EKEYREVOKED: return DRWAV_ERROR; + #endif + #ifdef EKEYREJECTED + case EKEYREJECTED: return DRWAV_ERROR; + #endif + #ifdef EOWNERDEAD + case EOWNERDEAD: return DRWAV_ERROR; + #endif + #ifdef ENOTRECOVERABLE + case ENOTRECOVERABLE: return DRWAV_ERROR; + #endif + #ifdef ERFKILL + case ERFKILL: return DRWAV_ERROR; + #endif + #ifdef EHWPOISON + case EHWPOISON: return DRWAV_ERROR; + #endif + default: return DRWAV_ERROR; + } +} + +static drwav_result drwav_fopen(FILE** ppFile, const char* pFilePath, const char* pOpenMode) +{ +#if _MSC_VER && _MSC_VER >= 1400 + errno_t err; +#endif + + if (ppFile != NULL) { + *ppFile = NULL; /* Safety. */ + } + + if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) { + return DRWAV_INVALID_ARGS; + } + +#if _MSC_VER && _MSC_VER >= 1400 + err = fopen_s(ppFile, pFilePath, pOpenMode); + if (err != 0) { + return drwav_result_from_errno(err); + } +#else +#if defined(_WIN32) || defined(__APPLE__) + *ppFile = fopen(pFilePath, pOpenMode); +#else + #if defined(_FILE_OFFSET_BITS) && _FILE_OFFSET_BITS == 64 && defined(_LARGEFILE64_SOURCE) + *ppFile = fopen64(pFilePath, pOpenMode); + #else + *ppFile = fopen(pFilePath, pOpenMode); + #endif +#endif + if (*ppFile == NULL) { + drwav_result result = drwav_result_from_errno(errno); + if (result == DRWAV_SUCCESS) { + result = DRWAV_ERROR; /* Just a safety check to make sure we never ever return success when pFile == NULL. */ + } + + return result; + } +#endif + + return DRWAV_SUCCESS; +} + +/* +_wfopen() isn't always available in all compilation environments. + + * Windows only. + * MSVC seems to support it universally as far back as VC6 from what I can tell (haven't checked further back). + * MinGW-64 (both 32- and 64-bit) seems to support it. + * MinGW wraps it in !defined(__STRICT_ANSI__). + * OpenWatcom wraps it in !defined(_NO_EXT_KEYS). + +This can be reviewed as compatibility issues arise. The preference is to use _wfopen_s() and _wfopen() as opposed to the wcsrtombs() +fallback, so if you notice your compiler not detecting this properly I'm happy to look at adding support. +*/ +#if defined(_WIN32) + #if defined(_MSC_VER) || defined(__MINGW64__) || (!defined(__STRICT_ANSI__) && !defined(_NO_EXT_KEYS)) + #define DRWAV_HAS_WFOPEN + #endif +#endif + +static drwav_result drwav_wfopen(FILE** ppFile, const wchar_t* pFilePath, const wchar_t* pOpenMode, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (ppFile != NULL) { + *ppFile = NULL; /* Safety. */ + } + + if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) { + return DRWAV_INVALID_ARGS; + } + +#if defined(DRWAV_HAS_WFOPEN) + { + /* Use _wfopen() on Windows. */ + #if defined(_MSC_VER) && _MSC_VER >= 1400 + errno_t err = _wfopen_s(ppFile, pFilePath, pOpenMode); + if (err != 0) { + return drwav_result_from_errno(err); + } + #else + *ppFile = _wfopen(pFilePath, pOpenMode); + if (*ppFile == NULL) { + return drwav_result_from_errno(errno); + } + #endif + (void)pAllocationCallbacks; + } +#else + /* + Use fopen() on anything other than Windows. Requires a conversion. This is annoying because fopen() is locale specific. The only real way I can + think of to do this is with wcsrtombs(). Note that wcstombs() is apparently not thread-safe because it uses a static global mbstate_t object for + maintaining state. I've checked this with -std=c89 and it works, but if somebody get's a compiler error I'll look into improving compatibility. + */ + { + mbstate_t mbs; + size_t lenMB; + const wchar_t* pFilePathTemp = pFilePath; + char* pFilePathMB = NULL; + char pOpenModeMB[32] = {0}; + + /* Get the length first. */ + DRWAV_ZERO_OBJECT(&mbs); + lenMB = wcsrtombs(NULL, &pFilePathTemp, 0, &mbs); + if (lenMB == (size_t)-1) { + return drwav_result_from_errno(errno); + } + + pFilePathMB = (char*)drwav__malloc_from_callbacks(lenMB + 1, pAllocationCallbacks); + if (pFilePathMB == NULL) { + return DRWAV_OUT_OF_MEMORY; + } + + pFilePathTemp = pFilePath; + DRWAV_ZERO_OBJECT(&mbs); + wcsrtombs(pFilePathMB, &pFilePathTemp, lenMB + 1, &mbs); + + /* The open mode should always consist of ASCII characters so we should be able to do a trivial conversion. */ + { + size_t i = 0; + for (;;) { + if (pOpenMode[i] == 0) { + pOpenModeMB[i] = '\0'; + break; + } + + pOpenModeMB[i] = (char)pOpenMode[i]; + i += 1; + } + } + + *ppFile = fopen(pFilePathMB, pOpenModeMB); + + drwav__free_from_callbacks(pFilePathMB, pAllocationCallbacks); + } + + if (*ppFile == NULL) { + return DRWAV_ERROR; + } +#endif + + return DRWAV_SUCCESS; +} + + +static size_t drwav__on_read_stdio(void* pUserData, void* pBufferOut, size_t bytesToRead) +{ + return fread(pBufferOut, 1, bytesToRead, (FILE*)pUserData); +} + +static size_t drwav__on_write_stdio(void* pUserData, const void* pData, size_t bytesToWrite) +{ + return fwrite(pData, 1, bytesToWrite, (FILE*)pUserData); +} + +static drwav_bool32 drwav__on_seek_stdio(void* pUserData, int offset, drwav_seek_origin origin) +{ + return fseek((FILE*)pUserData, offset, (origin == drwav_seek_origin_current) ? SEEK_CUR : SEEK_SET) == 0; +} + +DRWAV_API drwav_bool32 drwav_init_file(drwav* pWav, const char* filename, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_ex(pWav, filename, NULL, NULL, 0, pAllocationCallbacks); +} + + +static drwav_bool32 drwav_init_file__internal_FILE(drwav* pWav, FILE* pFile, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav_bool32 result; + + result = drwav_preinit(pWav, drwav__on_read_stdio, drwav__on_seek_stdio, (void*)pFile, pAllocationCallbacks); + if (result != DRWAV_TRUE) { + fclose(pFile); + return result; + } + + result = drwav_init__internal(pWav, onChunk, pChunkUserData, flags); + if (result != DRWAV_TRUE) { + fclose(pFile); + return result; + } + + return DRWAV_TRUE; +} + +DRWAV_API drwav_bool32 drwav_init_file_ex(drwav* pWav, const char* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + FILE* pFile; + if (drwav_fopen(&pFile, filename, "rb") != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + /* This takes ownership of the FILE* object. */ + return drwav_init_file__internal_FILE(pWav, pFile, onChunk, pChunkUserData, flags, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_w(drwav* pWav, const wchar_t* filename, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_ex_w(pWav, filename, NULL, NULL, 0, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_ex_w(drwav* pWav, const wchar_t* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + FILE* pFile; + if (drwav_wfopen(&pFile, filename, L"rb", pAllocationCallbacks) != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + /* This takes ownership of the FILE* object. */ + return drwav_init_file__internal_FILE(pWav, pFile, onChunk, pChunkUserData, flags, pAllocationCallbacks); +} + + +static drwav_bool32 drwav_init_file_write__internal_FILE(drwav* pWav, FILE* pFile, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav_bool32 result; + + result = drwav_preinit_write(pWav, pFormat, isSequential, drwav__on_write_stdio, drwav__on_seek_stdio, (void*)pFile, pAllocationCallbacks); + if (result != DRWAV_TRUE) { + fclose(pFile); + return result; + } + + result = drwav_init_write__internal(pWav, pFormat, totalSampleCount); + if (result != DRWAV_TRUE) { + fclose(pFile); + return result; + } + + return DRWAV_TRUE; +} + +static drwav_bool32 drwav_init_file_write__internal(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + FILE* pFile; + if (drwav_fopen(&pFile, filename, "wb") != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + /* This takes ownership of the FILE* object. */ + return drwav_init_file_write__internal_FILE(pWav, pFile, pFormat, totalSampleCount, isSequential, pAllocationCallbacks); +} + +static drwav_bool32 drwav_init_file_write_w__internal(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + FILE* pFile; + if (drwav_wfopen(&pFile, filename, L"wb", pAllocationCallbacks) != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + /* This takes ownership of the FILE* object. */ + return drwav_init_file_write__internal_FILE(pWav, pFile, pFormat, totalSampleCount, isSequential, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_write(drwav* pWav, const char* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_write__internal(pWav, filename, pFormat, 0, DRWAV_FALSE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_write_sequential(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_write__internal(pWav, filename, pFormat, totalSampleCount, DRWAV_TRUE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pFormat == NULL) { + return DRWAV_FALSE; + } + + return drwav_init_file_write_sequential(pWav, filename, pFormat, totalPCMFrameCount*pFormat->channels, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_write_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_write_w__internal(pWav, filename, pFormat, 0, DRWAV_FALSE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_write_w__internal(pWav, filename, pFormat, totalSampleCount, DRWAV_TRUE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pFormat == NULL) { + return DRWAV_FALSE; + } + + return drwav_init_file_write_sequential_w(pWav, filename, pFormat, totalPCMFrameCount*pFormat->channels, pAllocationCallbacks); +} +#endif /* DR_WAV_NO_STDIO */ + + +static size_t drwav__on_read_memory(void* pUserData, void* pBufferOut, size_t bytesToRead) +{ + drwav* pWav = (drwav*)pUserData; + size_t bytesRemaining; + + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->memoryStream.dataSize >= pWav->memoryStream.currentReadPos); + + bytesRemaining = pWav->memoryStream.dataSize - pWav->memoryStream.currentReadPos; + if (bytesToRead > bytesRemaining) { + bytesToRead = bytesRemaining; + } + + if (bytesToRead > 0) { + DRWAV_COPY_MEMORY(pBufferOut, pWav->memoryStream.data + pWav->memoryStream.currentReadPos, bytesToRead); + pWav->memoryStream.currentReadPos += bytesToRead; + } + + return bytesToRead; +} + +static drwav_bool32 drwav__on_seek_memory(void* pUserData, int offset, drwav_seek_origin origin) +{ + drwav* pWav = (drwav*)pUserData; + DRWAV_ASSERT(pWav != NULL); + + if (origin == drwav_seek_origin_current) { + if (offset > 0) { + if (pWav->memoryStream.currentReadPos + offset > pWav->memoryStream.dataSize) { + return DRWAV_FALSE; /* Trying to seek too far forward. */ + } + } else { + if (pWav->memoryStream.currentReadPos < (size_t)-offset) { + return DRWAV_FALSE; /* Trying to seek too far backwards. */ + } + } + + /* This will never underflow thanks to the clamps above. */ + pWav->memoryStream.currentReadPos += offset; + } else { + if ((drwav_uint32)offset <= pWav->memoryStream.dataSize) { + pWav->memoryStream.currentReadPos = offset; + } else { + return DRWAV_FALSE; /* Trying to seek too far forward. */ + } + } + + return DRWAV_TRUE; +} + +static size_t drwav__on_write_memory(void* pUserData, const void* pDataIn, size_t bytesToWrite) +{ + drwav* pWav = (drwav*)pUserData; + size_t bytesRemaining; + + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->memoryStreamWrite.dataCapacity >= pWav->memoryStreamWrite.currentWritePos); + + bytesRemaining = pWav->memoryStreamWrite.dataCapacity - pWav->memoryStreamWrite.currentWritePos; + if (bytesRemaining < bytesToWrite) { + /* Need to reallocate. */ + void* pNewData; + size_t newDataCapacity = (pWav->memoryStreamWrite.dataCapacity == 0) ? 256 : pWav->memoryStreamWrite.dataCapacity * 2; + + /* If doubling wasn't enough, just make it the minimum required size to write the data. */ + if ((newDataCapacity - pWav->memoryStreamWrite.currentWritePos) < bytesToWrite) { + newDataCapacity = pWav->memoryStreamWrite.currentWritePos + bytesToWrite; + } + + pNewData = drwav__realloc_from_callbacks(*pWav->memoryStreamWrite.ppData, newDataCapacity, pWav->memoryStreamWrite.dataCapacity, &pWav->allocationCallbacks); + if (pNewData == NULL) { + return 0; + } + + *pWav->memoryStreamWrite.ppData = pNewData; + pWav->memoryStreamWrite.dataCapacity = newDataCapacity; + } + + DRWAV_COPY_MEMORY(((drwav_uint8*)(*pWav->memoryStreamWrite.ppData)) + pWav->memoryStreamWrite.currentWritePos, pDataIn, bytesToWrite); + + pWav->memoryStreamWrite.currentWritePos += bytesToWrite; + if (pWav->memoryStreamWrite.dataSize < pWav->memoryStreamWrite.currentWritePos) { + pWav->memoryStreamWrite.dataSize = pWav->memoryStreamWrite.currentWritePos; + } + + *pWav->memoryStreamWrite.pDataSize = pWav->memoryStreamWrite.dataSize; + + return bytesToWrite; +} + +static drwav_bool32 drwav__on_seek_memory_write(void* pUserData, int offset, drwav_seek_origin origin) +{ + drwav* pWav = (drwav*)pUserData; + DRWAV_ASSERT(pWav != NULL); + + if (origin == drwav_seek_origin_current) { + if (offset > 0) { + if (pWav->memoryStreamWrite.currentWritePos + offset > pWav->memoryStreamWrite.dataSize) { + offset = (int)(pWav->memoryStreamWrite.dataSize - pWav->memoryStreamWrite.currentWritePos); /* Trying to seek too far forward. */ + } + } else { + if (pWav->memoryStreamWrite.currentWritePos < (size_t)-offset) { + offset = -(int)pWav->memoryStreamWrite.currentWritePos; /* Trying to seek too far backwards. */ + } + } + + /* This will never underflow thanks to the clamps above. */ + pWav->memoryStreamWrite.currentWritePos += offset; + } else { + if ((drwav_uint32)offset <= pWav->memoryStreamWrite.dataSize) { + pWav->memoryStreamWrite.currentWritePos = offset; + } else { + pWav->memoryStreamWrite.currentWritePos = pWav->memoryStreamWrite.dataSize; /* Trying to seek too far forward. */ + } + } + + return DRWAV_TRUE; +} + +DRWAV_API drwav_bool32 drwav_init_memory(drwav* pWav, const void* data, size_t dataSize, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_memory_ex(pWav, data, dataSize, NULL, NULL, 0, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_memory_ex(drwav* pWav, const void* data, size_t dataSize, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (data == NULL || dataSize == 0) { + return DRWAV_FALSE; + } + + if (!drwav_preinit(pWav, drwav__on_read_memory, drwav__on_seek_memory, pWav, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + pWav->memoryStream.data = (const drwav_uint8*)data; + pWav->memoryStream.dataSize = dataSize; + pWav->memoryStream.currentReadPos = 0; + + return drwav_init__internal(pWav, onChunk, pChunkUserData, flags); +} + + +static drwav_bool32 drwav_init_memory_write__internal(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (ppData == NULL || pDataSize == NULL) { + return DRWAV_FALSE; + } + + *ppData = NULL; /* Important because we're using realloc()! */ + *pDataSize = 0; + + if (!drwav_preinit_write(pWav, pFormat, isSequential, drwav__on_write_memory, drwav__on_seek_memory_write, pWav, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + pWav->memoryStreamWrite.ppData = ppData; + pWav->memoryStreamWrite.pDataSize = pDataSize; + pWav->memoryStreamWrite.dataSize = 0; + pWav->memoryStreamWrite.dataCapacity = 0; + pWav->memoryStreamWrite.currentWritePos = 0; + + return drwav_init_write__internal(pWav, pFormat, totalSampleCount); +} + +DRWAV_API drwav_bool32 drwav_init_memory_write(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_memory_write__internal(pWav, ppData, pDataSize, pFormat, 0, DRWAV_FALSE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_memory_write_sequential(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_memory_write__internal(pWav, ppData, pDataSize, pFormat, totalSampleCount, DRWAV_TRUE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_memory_write_sequential_pcm_frames(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pFormat == NULL) { + return DRWAV_FALSE; + } + + return drwav_init_memory_write_sequential(pWav, ppData, pDataSize, pFormat, totalPCMFrameCount*pFormat->channels, pAllocationCallbacks); +} + + + +DRWAV_API drwav_result drwav_uninit(drwav* pWav) +{ + drwav_result result = DRWAV_SUCCESS; + + if (pWav == NULL) { + return DRWAV_INVALID_ARGS; + } + + /* + If the drwav object was opened in write mode we'll need to finalize a few things: + - Make sure the "data" chunk is aligned to 16-bits for RIFF containers, or 64 bits for W64 containers. + - Set the size of the "data" chunk. + */ + if (pWav->onWrite != NULL) { + drwav_uint32 paddingSize = 0; + + /* Padding. Do not adjust pWav->dataChunkDataSize - this should not include the padding. */ + if (pWav->container == drwav_container_riff || pWav->container == drwav_container_rf64) { + paddingSize = drwav__chunk_padding_size_riff(pWav->dataChunkDataSize); + } else { + paddingSize = drwav__chunk_padding_size_w64(pWav->dataChunkDataSize); + } + + if (paddingSize > 0) { + drwav_uint64 paddingData = 0; + drwav__write(pWav, &paddingData, paddingSize); /* Byte order does not matter for this. */ + } + + /* + Chunk sizes. When using sequential mode, these will have been filled in at initialization time. We only need + to do this when using non-sequential mode. + */ + if (pWav->onSeek && !pWav->isSequentialWrite) { + if (pWav->container == drwav_container_riff) { + /* The "RIFF" chunk size. */ + if (pWav->onSeek(pWav->pUserData, 4, drwav_seek_origin_start)) { + drwav_uint32 riffChunkSize = drwav__riff_chunk_size_riff(pWav->dataChunkDataSize); + drwav__write_u32ne_to_le(pWav, riffChunkSize); + } + + /* the "data" chunk size. */ + if (pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos + 4, drwav_seek_origin_start)) { + drwav_uint32 dataChunkSize = drwav__data_chunk_size_riff(pWav->dataChunkDataSize); + drwav__write_u32ne_to_le(pWav, dataChunkSize); + } + } else if (pWav->container == drwav_container_w64) { + /* The "RIFF" chunk size. */ + if (pWav->onSeek(pWav->pUserData, 16, drwav_seek_origin_start)) { + drwav_uint64 riffChunkSize = drwav__riff_chunk_size_w64(pWav->dataChunkDataSize); + drwav__write_u64ne_to_le(pWav, riffChunkSize); + } + + /* The "data" chunk size. */ + if (pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos + 16, drwav_seek_origin_start)) { + drwav_uint64 dataChunkSize = drwav__data_chunk_size_w64(pWav->dataChunkDataSize); + drwav__write_u64ne_to_le(pWav, dataChunkSize); + } + } else if (pWav->container == drwav_container_rf64) { + /* We only need to update the ds64 chunk. The "RIFF" and "data" chunks always have their sizes set to 0xFFFFFFFF for RF64. */ + int ds64BodyPos = 12 + 8; + + /* The "RIFF" chunk size. */ + if (pWav->onSeek(pWav->pUserData, ds64BodyPos + 0, drwav_seek_origin_start)) { + drwav_uint64 riffChunkSize = drwav__riff_chunk_size_rf64(pWav->dataChunkDataSize); + drwav__write_u64ne_to_le(pWav, riffChunkSize); + } + + /* The "data" chunk size. */ + if (pWav->onSeek(pWav->pUserData, ds64BodyPos + 8, drwav_seek_origin_start)) { + drwav_uint64 dataChunkSize = drwav__data_chunk_size_rf64(pWav->dataChunkDataSize); + drwav__write_u64ne_to_le(pWav, dataChunkSize); + } + } + } + + /* Validation for sequential mode. */ + if (pWav->isSequentialWrite) { + if (pWav->dataChunkDataSize != pWav->dataChunkDataSizeTargetWrite) { + result = DRWAV_INVALID_FILE; + } + } + } + +#ifndef DR_WAV_NO_STDIO + /* + If we opened the file with drwav_open_file() we will want to close the file handle. We can know whether or not drwav_open_file() + was used by looking at the onRead and onSeek callbacks. + */ + if (pWav->onRead == drwav__on_read_stdio || pWav->onWrite == drwav__on_write_stdio) { + fclose((FILE*)pWav->pUserData); + } +#endif + + return result; +} + + + +DRWAV_API size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut) +{ + size_t bytesRead; + + if (pWav == NULL || bytesToRead == 0) { + return 0; + } + + if (bytesToRead > pWav->bytesRemaining) { + bytesToRead = (size_t)pWav->bytesRemaining; + } + + if (pBufferOut != NULL) { + bytesRead = pWav->onRead(pWav->pUserData, pBufferOut, bytesToRead); + } else { + /* We need to seek. If we fail, we need to read-and-discard to make sure we get a good byte count. */ + bytesRead = 0; + while (bytesRead < bytesToRead) { + size_t bytesToSeek = (bytesToRead - bytesRead); + if (bytesToSeek > 0x7FFFFFFF) { + bytesToSeek = 0x7FFFFFFF; + } + + if (pWav->onSeek(pWav->pUserData, (int)bytesToSeek, drwav_seek_origin_current) == DRWAV_FALSE) { + break; + } + + bytesRead += bytesToSeek; + } + + /* When we get here we may need to read-and-discard some data. */ + while (bytesRead < bytesToRead) { + drwav_uint8 buffer[4096]; + size_t bytesSeeked; + size_t bytesToSeek = (bytesToRead - bytesRead); + if (bytesToSeek > sizeof(buffer)) { + bytesToSeek = sizeof(buffer); + } + + bytesSeeked = pWav->onRead(pWav->pUserData, buffer, bytesToSeek); + bytesRead += bytesSeeked; + + if (bytesSeeked < bytesToSeek) { + break; /* Reached the end. */ + } + } + } + + pWav->bytesRemaining -= bytesRead; + return bytesRead; +} + + + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_le(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut) +{ + drwav_uint32 bytesPerFrame; + drwav_uint64 bytesToRead; /* Intentionally uint64 instead of size_t so we can do a check that we're not reading too much on 32-bit builds. */ + + if (pWav == NULL || framesToRead == 0) { + return 0; + } + + /* Cannot use this function for compressed formats. */ + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + return 0; + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + bytesToRead = framesToRead * bytesPerFrame; + if (bytesToRead > DRWAV_SIZE_MAX) { + bytesToRead = (DRWAV_SIZE_MAX / bytesPerFrame) * bytesPerFrame; /* Round the number of bytes to read to a clean frame boundary. */ + } + + /* + Doing an explicit check here just to make it clear that we don't want to be attempt to read anything if there's no bytes to read. There + *could* be a time where it evaluates to 0 due to overflowing. + */ + if (bytesToRead == 0) { + return 0; + } + + return drwav_read_raw(pWav, (size_t)bytesToRead, pBufferOut) / bytesPerFrame; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_be(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_le(pWav, framesToRead, pBufferOut); + + if (pBufferOut != NULL) { + drwav__bswap_samples(pBufferOut, framesRead*pWav->channels, drwav_get_bytes_per_pcm_frame(pWav)/pWav->channels, pWav->translatedFormatTag); + } + + return framesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut) +{ + if (drwav__is_little_endian()) { + return drwav_read_pcm_frames_le(pWav, framesToRead, pBufferOut); + } else { + return drwav_read_pcm_frames_be(pWav, framesToRead, pBufferOut); + } +} + + + +DRWAV_API drwav_bool32 drwav_seek_to_first_pcm_frame(drwav* pWav) +{ + if (pWav->onWrite != NULL) { + return DRWAV_FALSE; /* No seeking in write mode. */ + } + + if (!pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos, drwav_seek_origin_start)) { + return DRWAV_FALSE; + } + + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + pWav->compressed.iCurrentPCMFrame = 0; + + /* Cached data needs to be cleared for compressed formats. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + DRWAV_ZERO_OBJECT(&pWav->msadpcm); + } else if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + DRWAV_ZERO_OBJECT(&pWav->ima); + } else { + DRWAV_ASSERT(DRWAV_FALSE); /* If this assertion is triggered it means I've implemented a new compressed format but forgot to add a branch for it here. */ + } + } + + pWav->bytesRemaining = pWav->dataChunkDataSize; + return DRWAV_TRUE; +} + +DRWAV_API drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetFrameIndex) +{ + /* Seeking should be compatible with wave files > 2GB. */ + + if (pWav == NULL || pWav->onSeek == NULL) { + return DRWAV_FALSE; + } + + /* No seeking in write mode. */ + if (pWav->onWrite != NULL) { + return DRWAV_FALSE; + } + + /* If there are no samples, just return DRWAV_TRUE without doing anything. */ + if (pWav->totalPCMFrameCount == 0) { + return DRWAV_TRUE; + } + + /* Make sure the sample is clamped. */ + if (targetFrameIndex >= pWav->totalPCMFrameCount) { + targetFrameIndex = pWav->totalPCMFrameCount - 1; + } + + /* + For compressed formats we just use a slow generic seek. If we are seeking forward we just seek forward. If we are going backwards we need + to seek back to the start. + */ + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + /* TODO: This can be optimized. */ + + /* + If we're seeking forward it's simple - just keep reading samples until we hit the sample we're requesting. If we're seeking backwards, + we first need to seek back to the start and then just do the same thing as a forward seek. + */ + if (targetFrameIndex < pWav->compressed.iCurrentPCMFrame) { + if (!drwav_seek_to_first_pcm_frame(pWav)) { + return DRWAV_FALSE; + } + } + + if (targetFrameIndex > pWav->compressed.iCurrentPCMFrame) { + drwav_uint64 offsetInFrames = targetFrameIndex - pWav->compressed.iCurrentPCMFrame; + + drwav_int16 devnull[2048]; + while (offsetInFrames > 0) { + drwav_uint64 framesRead = 0; + drwav_uint64 framesToRead = offsetInFrames; + if (framesToRead > drwav_countof(devnull)/pWav->channels) { + framesToRead = drwav_countof(devnull)/pWav->channels; + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + framesRead = drwav_read_pcm_frames_s16__msadpcm(pWav, framesToRead, devnull); + } else if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + framesRead = drwav_read_pcm_frames_s16__ima(pWav, framesToRead, devnull); + } else { + DRWAV_ASSERT(DRWAV_FALSE); /* If this assertion is triggered it means I've implemented a new compressed format but forgot to add a branch for it here. */ + } + + if (framesRead != framesToRead) { + return DRWAV_FALSE; + } + + offsetInFrames -= framesRead; + } + } + } else { + drwav_uint64 totalSizeInBytes; + drwav_uint64 currentBytePos; + drwav_uint64 targetBytePos; + drwav_uint64 offset; + + totalSizeInBytes = pWav->totalPCMFrameCount * drwav_get_bytes_per_pcm_frame(pWav); + DRWAV_ASSERT(totalSizeInBytes >= pWav->bytesRemaining); + + currentBytePos = totalSizeInBytes - pWav->bytesRemaining; + targetBytePos = targetFrameIndex * drwav_get_bytes_per_pcm_frame(pWav); + + if (currentBytePos < targetBytePos) { + /* Offset forwards. */ + offset = (targetBytePos - currentBytePos); + } else { + /* Offset backwards. */ + if (!drwav_seek_to_first_pcm_frame(pWav)) { + return DRWAV_FALSE; + } + offset = targetBytePos; + } + + while (offset > 0) { + int offset32 = ((offset > INT_MAX) ? INT_MAX : (int)offset); + if (!pWav->onSeek(pWav->pUserData, offset32, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + + pWav->bytesRemaining -= offset32; + offset -= offset32; + } + } + + return DRWAV_TRUE; +} + + +DRWAV_API size_t drwav_write_raw(drwav* pWav, size_t bytesToWrite, const void* pData) +{ + size_t bytesWritten; + + if (pWav == NULL || bytesToWrite == 0 || pData == NULL) { + return 0; + } + + bytesWritten = pWav->onWrite(pWav->pUserData, pData, bytesToWrite); + pWav->dataChunkDataSize += bytesWritten; + + return bytesWritten; +} + + +DRWAV_API drwav_uint64 drwav_write_pcm_frames_le(drwav* pWav, drwav_uint64 framesToWrite, const void* pData) +{ + drwav_uint64 bytesToWrite; + drwav_uint64 bytesWritten; + const drwav_uint8* pRunningData; + + if (pWav == NULL || framesToWrite == 0 || pData == NULL) { + return 0; + } + + bytesToWrite = ((framesToWrite * pWav->channels * pWav->bitsPerSample) / 8); + if (bytesToWrite > DRWAV_SIZE_MAX) { + return 0; + } + + bytesWritten = 0; + pRunningData = (const drwav_uint8*)pData; + + while (bytesToWrite > 0) { + size_t bytesJustWritten; + drwav_uint64 bytesToWriteThisIteration; + + bytesToWriteThisIteration = bytesToWrite; + DRWAV_ASSERT(bytesToWriteThisIteration <= DRWAV_SIZE_MAX); /* <-- This is checked above. */ + + bytesJustWritten = drwav_write_raw(pWav, (size_t)bytesToWriteThisIteration, pRunningData); + if (bytesJustWritten == 0) { + break; + } + + bytesToWrite -= bytesJustWritten; + bytesWritten += bytesJustWritten; + pRunningData += bytesJustWritten; + } + + return (bytesWritten * 8) / pWav->bitsPerSample / pWav->channels; +} + +DRWAV_API drwav_uint64 drwav_write_pcm_frames_be(drwav* pWav, drwav_uint64 framesToWrite, const void* pData) +{ + drwav_uint64 bytesToWrite; + drwav_uint64 bytesWritten; + drwav_uint32 bytesPerSample; + const drwav_uint8* pRunningData; + + if (pWav == NULL || framesToWrite == 0 || pData == NULL) { + return 0; + } + + bytesToWrite = ((framesToWrite * pWav->channels * pWav->bitsPerSample) / 8); + if (bytesToWrite > DRWAV_SIZE_MAX) { + return 0; + } + + bytesWritten = 0; + pRunningData = (const drwav_uint8*)pData; + + bytesPerSample = drwav_get_bytes_per_pcm_frame(pWav) / pWav->channels; + + while (bytesToWrite > 0) { + drwav_uint8 temp[4096]; + drwav_uint32 sampleCount; + size_t bytesJustWritten; + drwav_uint64 bytesToWriteThisIteration; + + bytesToWriteThisIteration = bytesToWrite; + DRWAV_ASSERT(bytesToWriteThisIteration <= DRWAV_SIZE_MAX); /* <-- This is checked above. */ + + /* + WAV files are always little-endian. We need to byte swap on big-endian architectures. Since our input buffer is read-only we need + to use an intermediary buffer for the conversion. + */ + sampleCount = sizeof(temp)/bytesPerSample; + + if (bytesToWriteThisIteration > ((drwav_uint64)sampleCount)*bytesPerSample) { + bytesToWriteThisIteration = ((drwav_uint64)sampleCount)*bytesPerSample; + } + + DRWAV_COPY_MEMORY(temp, pRunningData, (size_t)bytesToWriteThisIteration); + drwav__bswap_samples(temp, sampleCount, bytesPerSample, pWav->translatedFormatTag); + + bytesJustWritten = drwav_write_raw(pWav, (size_t)bytesToWriteThisIteration, temp); + if (bytesJustWritten == 0) { + break; + } + + bytesToWrite -= bytesJustWritten; + bytesWritten += bytesJustWritten; + pRunningData += bytesJustWritten; + } + + return (bytesWritten * 8) / pWav->bitsPerSample / pWav->channels; +} + +DRWAV_API drwav_uint64 drwav_write_pcm_frames(drwav* pWav, drwav_uint64 framesToWrite, const void* pData) +{ + if (drwav__is_little_endian()) { + return drwav_write_pcm_frames_le(pWav, framesToWrite, pData); + } else { + return drwav_write_pcm_frames_be(pWav, framesToWrite, pData); + } +} + + +static drwav_uint64 drwav_read_pcm_frames_s16__msadpcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead = 0; + + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(framesToRead > 0); + + /* TODO: Lots of room for optimization here. */ + + while (framesToRead > 0 && pWav->compressed.iCurrentPCMFrame < pWav->totalPCMFrameCount) { + /* If there are no cached frames we need to load a new block. */ + if (pWav->msadpcm.cachedFrameCount == 0 && pWav->msadpcm.bytesRemainingInBlock == 0) { + if (pWav->channels == 1) { + /* Mono. */ + drwav_uint8 header[7]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalFramesRead; + } + pWav->msadpcm.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + pWav->msadpcm.predictor[0] = header[0]; + pWav->msadpcm.delta[0] = drwav__bytes_to_s16(header + 1); + pWav->msadpcm.prevFrames[0][1] = (drwav_int32)drwav__bytes_to_s16(header + 3); + pWav->msadpcm.prevFrames[0][0] = (drwav_int32)drwav__bytes_to_s16(header + 5); + pWav->msadpcm.cachedFrames[2] = pWav->msadpcm.prevFrames[0][0]; + pWav->msadpcm.cachedFrames[3] = pWav->msadpcm.prevFrames[0][1]; + pWav->msadpcm.cachedFrameCount = 2; + } else { + /* Stereo. */ + drwav_uint8 header[14]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalFramesRead; + } + pWav->msadpcm.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + pWav->msadpcm.predictor[0] = header[0]; + pWav->msadpcm.predictor[1] = header[1]; + pWav->msadpcm.delta[0] = drwav__bytes_to_s16(header + 2); + pWav->msadpcm.delta[1] = drwav__bytes_to_s16(header + 4); + pWav->msadpcm.prevFrames[0][1] = (drwav_int32)drwav__bytes_to_s16(header + 6); + pWav->msadpcm.prevFrames[1][1] = (drwav_int32)drwav__bytes_to_s16(header + 8); + pWav->msadpcm.prevFrames[0][0] = (drwav_int32)drwav__bytes_to_s16(header + 10); + pWav->msadpcm.prevFrames[1][0] = (drwav_int32)drwav__bytes_to_s16(header + 12); + + pWav->msadpcm.cachedFrames[0] = pWav->msadpcm.prevFrames[0][0]; + pWav->msadpcm.cachedFrames[1] = pWav->msadpcm.prevFrames[1][0]; + pWav->msadpcm.cachedFrames[2] = pWav->msadpcm.prevFrames[0][1]; + pWav->msadpcm.cachedFrames[3] = pWav->msadpcm.prevFrames[1][1]; + pWav->msadpcm.cachedFrameCount = 2; + } + } + + /* Output anything that's cached. */ + while (framesToRead > 0 && pWav->msadpcm.cachedFrameCount > 0 && pWav->compressed.iCurrentPCMFrame < pWav->totalPCMFrameCount) { + if (pBufferOut != NULL) { + drwav_uint32 iSample = 0; + for (iSample = 0; iSample < pWav->channels; iSample += 1) { + pBufferOut[iSample] = (drwav_int16)pWav->msadpcm.cachedFrames[(drwav_countof(pWav->msadpcm.cachedFrames) - (pWav->msadpcm.cachedFrameCount*pWav->channels)) + iSample]; + } + + pBufferOut += pWav->channels; + } + + framesToRead -= 1; + totalFramesRead += 1; + pWav->compressed.iCurrentPCMFrame += 1; + pWav->msadpcm.cachedFrameCount -= 1; + } + + if (framesToRead == 0) { + return totalFramesRead; + } + + + /* + If there's nothing left in the cache, just go ahead and load more. If there's nothing left to load in the current block we just continue to the next + loop iteration which will trigger the loading of a new block. + */ + if (pWav->msadpcm.cachedFrameCount == 0) { + if (pWav->msadpcm.bytesRemainingInBlock == 0) { + continue; + } else { + static drwav_int32 adaptationTable[] = { + 230, 230, 230, 230, 307, 409, 512, 614, + 768, 614, 512, 409, 307, 230, 230, 230 + }; + static drwav_int32 coeff1Table[] = { 256, 512, 0, 192, 240, 460, 392 }; + static drwav_int32 coeff2Table[] = { 0, -256, 0, 64, 0, -208, -232 }; + + drwav_uint8 nibbles; + drwav_int32 nibble0; + drwav_int32 nibble1; + + if (pWav->onRead(pWav->pUserData, &nibbles, 1) != 1) { + return totalFramesRead; + } + pWav->msadpcm.bytesRemainingInBlock -= 1; + + /* TODO: Optimize away these if statements. */ + nibble0 = ((nibbles & 0xF0) >> 4); if ((nibbles & 0x80)) { nibble0 |= 0xFFFFFFF0UL; } + nibble1 = ((nibbles & 0x0F) >> 0); if ((nibbles & 0x08)) { nibble1 |= 0xFFFFFFF0UL; } + + if (pWav->channels == 1) { + /* Mono. */ + drwav_int32 newSample0; + drwav_int32 newSample1; + + newSample0 = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8; + newSample0 += nibble0 * pWav->msadpcm.delta[0]; + newSample0 = drwav_clamp(newSample0, -32768, 32767); + + pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8; + if (pWav->msadpcm.delta[0] < 16) { + pWav->msadpcm.delta[0] = 16; + } + + pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1]; + pWav->msadpcm.prevFrames[0][1] = newSample0; + + + newSample1 = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8; + newSample1 += nibble1 * pWav->msadpcm.delta[0]; + newSample1 = drwav_clamp(newSample1, -32768, 32767); + + pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[0]) >> 8; + if (pWav->msadpcm.delta[0] < 16) { + pWav->msadpcm.delta[0] = 16; + } + + pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1]; + pWav->msadpcm.prevFrames[0][1] = newSample1; + + + pWav->msadpcm.cachedFrames[2] = newSample0; + pWav->msadpcm.cachedFrames[3] = newSample1; + pWav->msadpcm.cachedFrameCount = 2; + } else { + /* Stereo. */ + drwav_int32 newSample0; + drwav_int32 newSample1; + + /* Left. */ + newSample0 = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8; + newSample0 += nibble0 * pWav->msadpcm.delta[0]; + newSample0 = drwav_clamp(newSample0, -32768, 32767); + + pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8; + if (pWav->msadpcm.delta[0] < 16) { + pWav->msadpcm.delta[0] = 16; + } + + pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1]; + pWav->msadpcm.prevFrames[0][1] = newSample0; + + + /* Right. */ + newSample1 = ((pWav->msadpcm.prevFrames[1][1] * coeff1Table[pWav->msadpcm.predictor[1]]) + (pWav->msadpcm.prevFrames[1][0] * coeff2Table[pWav->msadpcm.predictor[1]])) >> 8; + newSample1 += nibble1 * pWav->msadpcm.delta[1]; + newSample1 = drwav_clamp(newSample1, -32768, 32767); + + pWav->msadpcm.delta[1] = (adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[1]) >> 8; + if (pWav->msadpcm.delta[1] < 16) { + pWav->msadpcm.delta[1] = 16; + } + + pWav->msadpcm.prevFrames[1][0] = pWav->msadpcm.prevFrames[1][1]; + pWav->msadpcm.prevFrames[1][1] = newSample1; + + pWav->msadpcm.cachedFrames[2] = newSample0; + pWav->msadpcm.cachedFrames[3] = newSample1; + pWav->msadpcm.cachedFrameCount = 1; + } + } + } + } + + return totalFramesRead; +} + + +static drwav_uint64 drwav_read_pcm_frames_s16__ima(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead = 0; + drwav_uint32 iChannel; + + static drwav_int32 indexTable[16] = { + -1, -1, -1, -1, 2, 4, 6, 8, + -1, -1, -1, -1, 2, 4, 6, 8 + }; + + static drwav_int32 stepTable[89] = { + 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, + 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, + 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, + 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, + 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, + 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, + 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, + 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, + 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 + }; + + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(framesToRead > 0); + + /* TODO: Lots of room for optimization here. */ + + while (framesToRead > 0 && pWav->compressed.iCurrentPCMFrame < pWav->totalPCMFrameCount) { + /* If there are no cached samples we need to load a new block. */ + if (pWav->ima.cachedFrameCount == 0 && pWav->ima.bytesRemainingInBlock == 0) { + if (pWav->channels == 1) { + /* Mono. */ + drwav_uint8 header[4]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalFramesRead; + } + pWav->ima.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + if (header[2] >= drwav_countof(stepTable)) { + pWav->onSeek(pWav->pUserData, pWav->ima.bytesRemainingInBlock, drwav_seek_origin_current); + pWav->ima.bytesRemainingInBlock = 0; + return totalFramesRead; /* Invalid data. */ + } + + pWav->ima.predictor[0] = drwav__bytes_to_s16(header + 0); + pWav->ima.stepIndex[0] = header[2]; + pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 1] = pWav->ima.predictor[0]; + pWav->ima.cachedFrameCount = 1; + } else { + /* Stereo. */ + drwav_uint8 header[8]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalFramesRead; + } + pWav->ima.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + if (header[2] >= drwav_countof(stepTable) || header[6] >= drwav_countof(stepTable)) { + pWav->onSeek(pWav->pUserData, pWav->ima.bytesRemainingInBlock, drwav_seek_origin_current); + pWav->ima.bytesRemainingInBlock = 0; + return totalFramesRead; /* Invalid data. */ + } + + pWav->ima.predictor[0] = drwav__bytes_to_s16(header + 0); + pWav->ima.stepIndex[0] = header[2]; + pWav->ima.predictor[1] = drwav__bytes_to_s16(header + 4); + pWav->ima.stepIndex[1] = header[6]; + + pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 2] = pWav->ima.predictor[0]; + pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 1] = pWav->ima.predictor[1]; + pWav->ima.cachedFrameCount = 1; + } + } + + /* Output anything that's cached. */ + while (framesToRead > 0 && pWav->ima.cachedFrameCount > 0 && pWav->compressed.iCurrentPCMFrame < pWav->totalPCMFrameCount) { + if (pBufferOut != NULL) { + drwav_uint32 iSample; + for (iSample = 0; iSample < pWav->channels; iSample += 1) { + pBufferOut[iSample] = (drwav_int16)pWav->ima.cachedFrames[(drwav_countof(pWav->ima.cachedFrames) - (pWav->ima.cachedFrameCount*pWav->channels)) + iSample]; + } + pBufferOut += pWav->channels; + } + + framesToRead -= 1; + totalFramesRead += 1; + pWav->compressed.iCurrentPCMFrame += 1; + pWav->ima.cachedFrameCount -= 1; + } + + if (framesToRead == 0) { + return totalFramesRead; + } + + /* + If there's nothing left in the cache, just go ahead and load more. If there's nothing left to load in the current block we just continue to the next + loop iteration which will trigger the loading of a new block. + */ + if (pWav->ima.cachedFrameCount == 0) { + if (pWav->ima.bytesRemainingInBlock == 0) { + continue; + } else { + /* + From what I can tell with stereo streams, it looks like every 4 bytes (8 samples) is for one channel. So it goes 4 bytes for the + left channel, 4 bytes for the right channel. + */ + pWav->ima.cachedFrameCount = 8; + for (iChannel = 0; iChannel < pWav->channels; ++iChannel) { + drwav_uint32 iByte; + drwav_uint8 nibbles[4]; + if (pWav->onRead(pWav->pUserData, &nibbles, 4) != 4) { + pWav->ima.cachedFrameCount = 0; + return totalFramesRead; + } + pWav->ima.bytesRemainingInBlock -= 4; + + for (iByte = 0; iByte < 4; ++iByte) { + drwav_uint8 nibble0 = ((nibbles[iByte] & 0x0F) >> 0); + drwav_uint8 nibble1 = ((nibbles[iByte] & 0xF0) >> 4); + + drwav_int32 step = stepTable[pWav->ima.stepIndex[iChannel]]; + drwav_int32 predictor = pWav->ima.predictor[iChannel]; + + drwav_int32 diff = step >> 3; + if (nibble0 & 1) diff += step >> 2; + if (nibble0 & 2) diff += step >> 1; + if (nibble0 & 4) diff += step; + if (nibble0 & 8) diff = -diff; + + predictor = drwav_clamp(predictor + diff, -32768, 32767); + pWav->ima.predictor[iChannel] = predictor; + pWav->ima.stepIndex[iChannel] = drwav_clamp(pWav->ima.stepIndex[iChannel] + indexTable[nibble0], 0, (drwav_int32)drwav_countof(stepTable)-1); + pWav->ima.cachedFrames[(drwav_countof(pWav->ima.cachedFrames) - (pWav->ima.cachedFrameCount*pWav->channels)) + (iByte*2+0)*pWav->channels + iChannel] = predictor; + + + step = stepTable[pWav->ima.stepIndex[iChannel]]; + predictor = pWav->ima.predictor[iChannel]; + + diff = step >> 3; + if (nibble1 & 1) diff += step >> 2; + if (nibble1 & 2) diff += step >> 1; + if (nibble1 & 4) diff += step; + if (nibble1 & 8) diff = -diff; + + predictor = drwav_clamp(predictor + diff, -32768, 32767); + pWav->ima.predictor[iChannel] = predictor; + pWav->ima.stepIndex[iChannel] = drwav_clamp(pWav->ima.stepIndex[iChannel] + indexTable[nibble1], 0, (drwav_int32)drwav_countof(stepTable)-1); + pWav->ima.cachedFrames[(drwav_countof(pWav->ima.cachedFrames) - (pWav->ima.cachedFrameCount*pWav->channels)) + (iByte*2+1)*pWav->channels + iChannel] = predictor; + } + } + } + } + } + + return totalFramesRead; +} + + +#ifndef DR_WAV_NO_CONVERSION_API +static unsigned short g_drwavAlawTable[256] = { + 0xEA80, 0xEB80, 0xE880, 0xE980, 0xEE80, 0xEF80, 0xEC80, 0xED80, 0xE280, 0xE380, 0xE080, 0xE180, 0xE680, 0xE780, 0xE480, 0xE580, + 0xF540, 0xF5C0, 0xF440, 0xF4C0, 0xF740, 0xF7C0, 0xF640, 0xF6C0, 0xF140, 0xF1C0, 0xF040, 0xF0C0, 0xF340, 0xF3C0, 0xF240, 0xF2C0, + 0xAA00, 0xAE00, 0xA200, 0xA600, 0xBA00, 0xBE00, 0xB200, 0xB600, 0x8A00, 0x8E00, 0x8200, 0x8600, 0x9A00, 0x9E00, 0x9200, 0x9600, + 0xD500, 0xD700, 0xD100, 0xD300, 0xDD00, 0xDF00, 0xD900, 0xDB00, 0xC500, 0xC700, 0xC100, 0xC300, 0xCD00, 0xCF00, 0xC900, 0xCB00, + 0xFEA8, 0xFEB8, 0xFE88, 0xFE98, 0xFEE8, 0xFEF8, 0xFEC8, 0xFED8, 0xFE28, 0xFE38, 0xFE08, 0xFE18, 0xFE68, 0xFE78, 0xFE48, 0xFE58, + 0xFFA8, 0xFFB8, 0xFF88, 0xFF98, 0xFFE8, 0xFFF8, 0xFFC8, 0xFFD8, 0xFF28, 0xFF38, 0xFF08, 0xFF18, 0xFF68, 0xFF78, 0xFF48, 0xFF58, + 0xFAA0, 0xFAE0, 0xFA20, 0xFA60, 0xFBA0, 0xFBE0, 0xFB20, 0xFB60, 0xF8A0, 0xF8E0, 0xF820, 0xF860, 0xF9A0, 0xF9E0, 0xF920, 0xF960, + 0xFD50, 0xFD70, 0xFD10, 0xFD30, 0xFDD0, 0xFDF0, 0xFD90, 0xFDB0, 0xFC50, 0xFC70, 0xFC10, 0xFC30, 0xFCD0, 0xFCF0, 0xFC90, 0xFCB0, + 0x1580, 0x1480, 0x1780, 0x1680, 0x1180, 0x1080, 0x1380, 0x1280, 0x1D80, 0x1C80, 0x1F80, 0x1E80, 0x1980, 0x1880, 0x1B80, 0x1A80, + 0x0AC0, 0x0A40, 0x0BC0, 0x0B40, 0x08C0, 0x0840, 0x09C0, 0x0940, 0x0EC0, 0x0E40, 0x0FC0, 0x0F40, 0x0CC0, 0x0C40, 0x0DC0, 0x0D40, + 0x5600, 0x5200, 0x5E00, 0x5A00, 0x4600, 0x4200, 0x4E00, 0x4A00, 0x7600, 0x7200, 0x7E00, 0x7A00, 0x6600, 0x6200, 0x6E00, 0x6A00, + 0x2B00, 0x2900, 0x2F00, 0x2D00, 0x2300, 0x2100, 0x2700, 0x2500, 0x3B00, 0x3900, 0x3F00, 0x3D00, 0x3300, 0x3100, 0x3700, 0x3500, + 0x0158, 0x0148, 0x0178, 0x0168, 0x0118, 0x0108, 0x0138, 0x0128, 0x01D8, 0x01C8, 0x01F8, 0x01E8, 0x0198, 0x0188, 0x01B8, 0x01A8, + 0x0058, 0x0048, 0x0078, 0x0068, 0x0018, 0x0008, 0x0038, 0x0028, 0x00D8, 0x00C8, 0x00F8, 0x00E8, 0x0098, 0x0088, 0x00B8, 0x00A8, + 0x0560, 0x0520, 0x05E0, 0x05A0, 0x0460, 0x0420, 0x04E0, 0x04A0, 0x0760, 0x0720, 0x07E0, 0x07A0, 0x0660, 0x0620, 0x06E0, 0x06A0, + 0x02B0, 0x0290, 0x02F0, 0x02D0, 0x0230, 0x0210, 0x0270, 0x0250, 0x03B0, 0x0390, 0x03F0, 0x03D0, 0x0330, 0x0310, 0x0370, 0x0350 +}; + +static unsigned short g_drwavMulawTable[256] = { + 0x8284, 0x8684, 0x8A84, 0x8E84, 0x9284, 0x9684, 0x9A84, 0x9E84, 0xA284, 0xA684, 0xAA84, 0xAE84, 0xB284, 0xB684, 0xBA84, 0xBE84, + 0xC184, 0xC384, 0xC584, 0xC784, 0xC984, 0xCB84, 0xCD84, 0xCF84, 0xD184, 0xD384, 0xD584, 0xD784, 0xD984, 0xDB84, 0xDD84, 0xDF84, + 0xE104, 0xE204, 0xE304, 0xE404, 0xE504, 0xE604, 0xE704, 0xE804, 0xE904, 0xEA04, 0xEB04, 0xEC04, 0xED04, 0xEE04, 0xEF04, 0xF004, + 0xF0C4, 0xF144, 0xF1C4, 0xF244, 0xF2C4, 0xF344, 0xF3C4, 0xF444, 0xF4C4, 0xF544, 0xF5C4, 0xF644, 0xF6C4, 0xF744, 0xF7C4, 0xF844, + 0xF8A4, 0xF8E4, 0xF924, 0xF964, 0xF9A4, 0xF9E4, 0xFA24, 0xFA64, 0xFAA4, 0xFAE4, 0xFB24, 0xFB64, 0xFBA4, 0xFBE4, 0xFC24, 0xFC64, + 0xFC94, 0xFCB4, 0xFCD4, 0xFCF4, 0xFD14, 0xFD34, 0xFD54, 0xFD74, 0xFD94, 0xFDB4, 0xFDD4, 0xFDF4, 0xFE14, 0xFE34, 0xFE54, 0xFE74, + 0xFE8C, 0xFE9C, 0xFEAC, 0xFEBC, 0xFECC, 0xFEDC, 0xFEEC, 0xFEFC, 0xFF0C, 0xFF1C, 0xFF2C, 0xFF3C, 0xFF4C, 0xFF5C, 0xFF6C, 0xFF7C, + 0xFF88, 0xFF90, 0xFF98, 0xFFA0, 0xFFA8, 0xFFB0, 0xFFB8, 0xFFC0, 0xFFC8, 0xFFD0, 0xFFD8, 0xFFE0, 0xFFE8, 0xFFF0, 0xFFF8, 0x0000, + 0x7D7C, 0x797C, 0x757C, 0x717C, 0x6D7C, 0x697C, 0x657C, 0x617C, 0x5D7C, 0x597C, 0x557C, 0x517C, 0x4D7C, 0x497C, 0x457C, 0x417C, + 0x3E7C, 0x3C7C, 0x3A7C, 0x387C, 0x367C, 0x347C, 0x327C, 0x307C, 0x2E7C, 0x2C7C, 0x2A7C, 0x287C, 0x267C, 0x247C, 0x227C, 0x207C, + 0x1EFC, 0x1DFC, 0x1CFC, 0x1BFC, 0x1AFC, 0x19FC, 0x18FC, 0x17FC, 0x16FC, 0x15FC, 0x14FC, 0x13FC, 0x12FC, 0x11FC, 0x10FC, 0x0FFC, + 0x0F3C, 0x0EBC, 0x0E3C, 0x0DBC, 0x0D3C, 0x0CBC, 0x0C3C, 0x0BBC, 0x0B3C, 0x0ABC, 0x0A3C, 0x09BC, 0x093C, 0x08BC, 0x083C, 0x07BC, + 0x075C, 0x071C, 0x06DC, 0x069C, 0x065C, 0x061C, 0x05DC, 0x059C, 0x055C, 0x051C, 0x04DC, 0x049C, 0x045C, 0x041C, 0x03DC, 0x039C, + 0x036C, 0x034C, 0x032C, 0x030C, 0x02EC, 0x02CC, 0x02AC, 0x028C, 0x026C, 0x024C, 0x022C, 0x020C, 0x01EC, 0x01CC, 0x01AC, 0x018C, + 0x0174, 0x0164, 0x0154, 0x0144, 0x0134, 0x0124, 0x0114, 0x0104, 0x00F4, 0x00E4, 0x00D4, 0x00C4, 0x00B4, 0x00A4, 0x0094, 0x0084, + 0x0078, 0x0070, 0x0068, 0x0060, 0x0058, 0x0050, 0x0048, 0x0040, 0x0038, 0x0030, 0x0028, 0x0020, 0x0018, 0x0010, 0x0008, 0x0000 +}; + +static DRWAV_INLINE drwav_int16 drwav__alaw_to_s16(drwav_uint8 sampleIn) +{ + return (short)g_drwavAlawTable[sampleIn]; +} + +static DRWAV_INLINE drwav_int16 drwav__mulaw_to_s16(drwav_uint8 sampleIn) +{ + return (short)g_drwavMulawTable[sampleIn]; +} + + + +static void drwav__pcm_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample) +{ + unsigned int i; + + /* Special case for 8-bit sample data because it's treated as unsigned. */ + if (bytesPerSample == 1) { + drwav_u8_to_s16(pOut, pIn, totalSampleCount); + return; + } + + + /* Slightly more optimal implementation for common formats. */ + if (bytesPerSample == 2) { + for (i = 0; i < totalSampleCount; ++i) { + *pOut++ = ((const drwav_int16*)pIn)[i]; + } + return; + } + if (bytesPerSample == 3) { + drwav_s24_to_s16(pOut, pIn, totalSampleCount); + return; + } + if (bytesPerSample == 4) { + drwav_s32_to_s16(pOut, (const drwav_int32*)pIn, totalSampleCount); + return; + } + + + /* Anything more than 64 bits per sample is not supported. */ + if (bytesPerSample > 8) { + DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut)); + return; + } + + + /* Generic, slow converter. */ + for (i = 0; i < totalSampleCount; ++i) { + drwav_uint64 sample = 0; + unsigned int shift = (8 - bytesPerSample) * 8; + + unsigned int j; + for (j = 0; j < bytesPerSample; j += 1) { + DRWAV_ASSERT(j < 8); + sample |= (drwav_uint64)(pIn[j]) << shift; + shift += 8; + } + + pIn += j; + *pOut++ = (drwav_int16)((drwav_int64)sample >> 48); + } +} + +static void drwav__ieee_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample) +{ + if (bytesPerSample == 4) { + drwav_f32_to_s16(pOut, (const float*)pIn, totalSampleCount); + return; + } else if (bytesPerSample == 8) { + drwav_f64_to_s16(pOut, (const double*)pIn, totalSampleCount); + return; + } else { + /* Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. */ + DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut)); + return; + } +} + +static drwav_uint64 drwav_read_pcm_frames_s16__pcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint32 bytesPerFrame; + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096]; + + /* Fast path. */ + if ((pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM && pWav->bitsPerSample == 16) || pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData); + if (framesRead == 0) { + break; + } + + drwav__pcm_to_s16(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels), bytesPerFrame/pWav->channels); + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +static drwav_uint64 drwav_read_pcm_frames_s16__ieee(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096]; + drwav_uint32 bytesPerFrame; + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData); + if (framesRead == 0) { + break; + } + + drwav__ieee_to_s16(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels), bytesPerFrame/pWav->channels); + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +static drwav_uint64 drwav_read_pcm_frames_s16__alaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096]; + drwav_uint32 bytesPerFrame; + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData); + if (framesRead == 0) { + break; + } + + drwav_alaw_to_s16(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels)); + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +static drwav_uint64 drwav_read_pcm_frames_s16__mulaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096]; + drwav_uint32 bytesPerFrame; + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData); + if (framesRead == 0) { + break; + } + + drwav_mulaw_to_s16(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels)); + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + if (pWav == NULL || framesToRead == 0) { + return 0; + } + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + if (framesToRead * pWav->channels * sizeof(drwav_int16) > DRWAV_SIZE_MAX) { + framesToRead = DRWAV_SIZE_MAX / sizeof(drwav_int16) / pWav->channels; + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) { + return drwav_read_pcm_frames_s16__pcm(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) { + return drwav_read_pcm_frames_s16__ieee(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) { + return drwav_read_pcm_frames_s16__alaw(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + return drwav_read_pcm_frames_s16__mulaw(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + return drwav_read_pcm_frames_s16__msadpcm(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + return drwav_read_pcm_frames_s16__ima(pWav, framesToRead, pBufferOut); + } + + return 0; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16le(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_FALSE) { + drwav__bswap_samples_s16(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16be(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_TRUE) { + drwav__bswap_samples_s16(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + + +DRWAV_API void drwav_u8_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + int x = pIn[i]; + r = x << 8; + r = r - 32768; + pOut[i] = (short)r; + } +} + +DRWAV_API void drwav_s24_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + int x = ((int)(((unsigned int)(((const drwav_uint8*)pIn)[i*3+0]) << 8) | ((unsigned int)(((const drwav_uint8*)pIn)[i*3+1]) << 16) | ((unsigned int)(((const drwav_uint8*)pIn)[i*3+2])) << 24)) >> 8; + r = x >> 8; + pOut[i] = (short)r; + } +} + +DRWAV_API void drwav_s32_to_s16(drwav_int16* pOut, const drwav_int32* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + int x = pIn[i]; + r = x >> 16; + pOut[i] = (short)r; + } +} + +DRWAV_API void drwav_f32_to_s16(drwav_int16* pOut, const float* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + float x = pIn[i]; + float c; + c = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); + c = c + 1; + r = (int)(c * 32767.5f); + r = r - 32768; + pOut[i] = (short)r; + } +} + +DRWAV_API void drwav_f64_to_s16(drwav_int16* pOut, const double* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + double x = pIn[i]; + double c; + c = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); + c = c + 1; + r = (int)(c * 32767.5); + r = r - 32768; + pOut[i] = (short)r; + } +} + +DRWAV_API void drwav_alaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + for (i = 0; i < sampleCount; ++i) { + pOut[i] = drwav__alaw_to_s16(pIn[i]); + } +} + +DRWAV_API void drwav_mulaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + for (i = 0; i < sampleCount; ++i) { + pOut[i] = drwav__mulaw_to_s16(pIn[i]); + } +} + + + +static void drwav__pcm_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount, unsigned int bytesPerSample) +{ + unsigned int i; + + /* Special case for 8-bit sample data because it's treated as unsigned. */ + if (bytesPerSample == 1) { + drwav_u8_to_f32(pOut, pIn, sampleCount); + return; + } + + /* Slightly more optimal implementation for common formats. */ + if (bytesPerSample == 2) { + drwav_s16_to_f32(pOut, (const drwav_int16*)pIn, sampleCount); + return; + } + if (bytesPerSample == 3) { + drwav_s24_to_f32(pOut, pIn, sampleCount); + return; + } + if (bytesPerSample == 4) { + drwav_s32_to_f32(pOut, (const drwav_int32*)pIn, sampleCount); + return; + } + + + /* Anything more than 64 bits per sample is not supported. */ + if (bytesPerSample > 8) { + DRWAV_ZERO_MEMORY(pOut, sampleCount * sizeof(*pOut)); + return; + } + + + /* Generic, slow converter. */ + for (i = 0; i < sampleCount; ++i) { + drwav_uint64 sample = 0; + unsigned int shift = (8 - bytesPerSample) * 8; + + unsigned int j; + for (j = 0; j < bytesPerSample; j += 1) { + DRWAV_ASSERT(j < 8); + sample |= (drwav_uint64)(pIn[j]) << shift; + shift += 8; + } + + pIn += j; + *pOut++ = (float)((drwav_int64)sample / 9223372036854775807.0); + } +} + +static void drwav__ieee_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount, unsigned int bytesPerSample) +{ + if (bytesPerSample == 4) { + unsigned int i; + for (i = 0; i < sampleCount; ++i) { + *pOut++ = ((const float*)pIn)[i]; + } + return; + } else if (bytesPerSample == 8) { + drwav_f64_to_f32(pOut, (const double*)pIn, sampleCount); + return; + } else { + /* Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. */ + DRWAV_ZERO_MEMORY(pOut, sampleCount * sizeof(*pOut)); + return; + } +} + + +static drwav_uint64 drwav_read_pcm_frames_f32__pcm(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096]; + + drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData); + if (framesRead == 0) { + break; + } + + drwav__pcm_to_f32(pBufferOut, sampleData, (size_t)framesRead*pWav->channels, bytesPerFrame/pWav->channels); + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +static drwav_uint64 drwav_read_pcm_frames_f32__msadpcm(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + /* + We're just going to borrow the implementation from the drwav_read_s16() since ADPCM is a little bit more complicated than other formats and I don't + want to duplicate that code. + */ + drwav_uint64 totalFramesRead = 0; + drwav_int16 samples16[2048]; + while (framesToRead > 0) { + drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels), samples16); + if (framesRead == 0) { + break; + } + + drwav_s16_to_f32(pBufferOut, samples16, (size_t)(framesRead*pWav->channels)); /* <-- Safe cast because we're clamping to 2048. */ + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +static drwav_uint64 drwav_read_pcm_frames_f32__ima(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + /* + We're just going to borrow the implementation from the drwav_read_s16() since IMA-ADPCM is a little bit more complicated than other formats and I don't + want to duplicate that code. + */ + drwav_uint64 totalFramesRead = 0; + drwav_int16 samples16[2048]; + while (framesToRead > 0) { + drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels), samples16); + if (framesRead == 0) { + break; + } + + drwav_s16_to_f32(pBufferOut, samples16, (size_t)(framesRead*pWav->channels)); /* <-- Safe cast because we're clamping to 2048. */ + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +static drwav_uint64 drwav_read_pcm_frames_f32__ieee(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096]; + drwav_uint32 bytesPerFrame; + + /* Fast path. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT && pWav->bitsPerSample == 32) { + return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData); + if (framesRead == 0) { + break; + } + + drwav__ieee_to_f32(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels), bytesPerFrame/pWav->channels); + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +static drwav_uint64 drwav_read_pcm_frames_f32__alaw(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096]; + drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData); + if (framesRead == 0) { + break; + } + + drwav_alaw_to_f32(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels)); + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +static drwav_uint64 drwav_read_pcm_frames_f32__mulaw(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096]; + + drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData); + if (framesRead == 0) { + break; + } + + drwav_mulaw_to_f32(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels)); + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + if (pWav == NULL || framesToRead == 0) { + return 0; + } + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + if (framesToRead * pWav->channels * sizeof(float) > DRWAV_SIZE_MAX) { + framesToRead = DRWAV_SIZE_MAX / sizeof(float) / pWav->channels; + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) { + return drwav_read_pcm_frames_f32__pcm(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + return drwav_read_pcm_frames_f32__msadpcm(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) { + return drwav_read_pcm_frames_f32__ieee(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) { + return drwav_read_pcm_frames_f32__alaw(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + return drwav_read_pcm_frames_f32__mulaw(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + return drwav_read_pcm_frames_f32__ima(pWav, framesToRead, pBufferOut); + } + + return 0; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32le(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_f32(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_FALSE) { + drwav__bswap_samples_f32(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32be(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_f32(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_TRUE) { + drwav__bswap_samples_f32(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + + +DRWAV_API void drwav_u8_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + +#ifdef DR_WAV_LIBSNDFILE_COMPAT + /* + It appears libsndfile uses slightly different logic for the u8 -> f32 conversion to dr_wav, which in my opinion is incorrect. It appears + libsndfile performs the conversion something like "f32 = (u8 / 256) * 2 - 1", however I think it should be "f32 = (u8 / 255) * 2 - 1" (note + the divisor of 256 vs 255). I use libsndfile as a benchmark for testing, so I'm therefore leaving this block here just for my automated + correctness testing. This is disabled by default. + */ + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (pIn[i] / 256.0f) * 2 - 1; + } +#else + for (i = 0; i < sampleCount; ++i) { + float x = pIn[i]; + x = x * 0.00784313725490196078f; /* 0..255 to 0..2 */ + x = x - 1; /* 0..2 to -1..1 */ + + *pOut++ = x; + } +#endif +} + +DRWAV_API void drwav_s16_to_f32(float* pOut, const drwav_int16* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = pIn[i] * 0.000030517578125f; + } +} + +DRWAV_API void drwav_s24_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + double x; + drwav_uint32 a = ((drwav_uint32)(pIn[i*3+0]) << 8); + drwav_uint32 b = ((drwav_uint32)(pIn[i*3+1]) << 16); + drwav_uint32 c = ((drwav_uint32)(pIn[i*3+2]) << 24); + + x = (double)((drwav_int32)(a | b | c) >> 8); + *pOut++ = (float)(x * 0.00000011920928955078125); + } +} + +DRWAV_API void drwav_s32_to_f32(float* pOut, const drwav_int32* pIn, size_t sampleCount) +{ + size_t i; + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (float)(pIn[i] / 2147483648.0); + } +} + +DRWAV_API void drwav_f64_to_f32(float* pOut, const double* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (float)pIn[i]; + } +} + +DRWAV_API void drwav_alaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = drwav__alaw_to_s16(pIn[i]) / 32768.0f; + } +} + +DRWAV_API void drwav_mulaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = drwav__mulaw_to_s16(pIn[i]) / 32768.0f; + } +} + + + +static void drwav__pcm_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample) +{ + unsigned int i; + + /* Special case for 8-bit sample data because it's treated as unsigned. */ + if (bytesPerSample == 1) { + drwav_u8_to_s32(pOut, pIn, totalSampleCount); + return; + } + + /* Slightly more optimal implementation for common formats. */ + if (bytesPerSample == 2) { + drwav_s16_to_s32(pOut, (const drwav_int16*)pIn, totalSampleCount); + return; + } + if (bytesPerSample == 3) { + drwav_s24_to_s32(pOut, pIn, totalSampleCount); + return; + } + if (bytesPerSample == 4) { + for (i = 0; i < totalSampleCount; ++i) { + *pOut++ = ((const drwav_int32*)pIn)[i]; + } + return; + } + + + /* Anything more than 64 bits per sample is not supported. */ + if (bytesPerSample > 8) { + DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut)); + return; + } + + + /* Generic, slow converter. */ + for (i = 0; i < totalSampleCount; ++i) { + drwav_uint64 sample = 0; + unsigned int shift = (8 - bytesPerSample) * 8; + + unsigned int j; + for (j = 0; j < bytesPerSample; j += 1) { + DRWAV_ASSERT(j < 8); + sample |= (drwav_uint64)(pIn[j]) << shift; + shift += 8; + } + + pIn += j; + *pOut++ = (drwav_int32)((drwav_int64)sample >> 32); + } +} + +static void drwav__ieee_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample) +{ + if (bytesPerSample == 4) { + drwav_f32_to_s32(pOut, (const float*)pIn, totalSampleCount); + return; + } else if (bytesPerSample == 8) { + drwav_f64_to_s32(pOut, (const double*)pIn, totalSampleCount); + return; + } else { + /* Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. */ + DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut)); + return; + } +} + + +static drwav_uint64 drwav_read_pcm_frames_s32__pcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096]; + drwav_uint32 bytesPerFrame; + + /* Fast path. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM && pWav->bitsPerSample == 32) { + return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData); + if (framesRead == 0) { + break; + } + + drwav__pcm_to_s32(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels), bytesPerFrame/pWav->channels); + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +static drwav_uint64 drwav_read_pcm_frames_s32__msadpcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + /* + We're just going to borrow the implementation from the drwav_read_s16() since ADPCM is a little bit more complicated than other formats and I don't + want to duplicate that code. + */ + drwav_uint64 totalFramesRead = 0; + drwav_int16 samples16[2048]; + while (framesToRead > 0) { + drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels), samples16); + if (framesRead == 0) { + break; + } + + drwav_s16_to_s32(pBufferOut, samples16, (size_t)(framesRead*pWav->channels)); /* <-- Safe cast because we're clamping to 2048. */ + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +static drwav_uint64 drwav_read_pcm_frames_s32__ima(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + /* + We're just going to borrow the implementation from the drwav_read_s16() since IMA-ADPCM is a little bit more complicated than other formats and I don't + want to duplicate that code. + */ + drwav_uint64 totalFramesRead = 0; + drwav_int16 samples16[2048]; + while (framesToRead > 0) { + drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels), samples16); + if (framesRead == 0) { + break; + } + + drwav_s16_to_s32(pBufferOut, samples16, (size_t)(framesRead*pWav->channels)); /* <-- Safe cast because we're clamping to 2048. */ + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +static drwav_uint64 drwav_read_pcm_frames_s32__ieee(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096]; + + drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData); + if (framesRead == 0) { + break; + } + + drwav__ieee_to_s32(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels), bytesPerFrame/pWav->channels); + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +static drwav_uint64 drwav_read_pcm_frames_s32__alaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096]; + + drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData); + if (framesRead == 0) { + break; + } + + drwav_alaw_to_s32(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels)); + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +static drwav_uint64 drwav_read_pcm_frames_s32__mulaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096]; + + drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData); + if (framesRead == 0) { + break; + } + + drwav_mulaw_to_s32(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels)); + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + if (pWav == NULL || framesToRead == 0) { + return 0; + } + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + if (framesToRead * pWav->channels * sizeof(drwav_int32) > DRWAV_SIZE_MAX) { + framesToRead = DRWAV_SIZE_MAX / sizeof(drwav_int32) / pWav->channels; + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) { + return drwav_read_pcm_frames_s32__pcm(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + return drwav_read_pcm_frames_s32__msadpcm(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) { + return drwav_read_pcm_frames_s32__ieee(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) { + return drwav_read_pcm_frames_s32__alaw(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + return drwav_read_pcm_frames_s32__mulaw(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + return drwav_read_pcm_frames_s32__ima(pWav, framesToRead, pBufferOut); + } + + return 0; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32le(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_s32(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_FALSE) { + drwav__bswap_samples_s32(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32be(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_s32(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_TRUE) { + drwav__bswap_samples_s32(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + + +DRWAV_API void drwav_u8_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = ((int)pIn[i] - 128) << 24; + } +} + +DRWAV_API void drwav_s16_to_s32(drwav_int32* pOut, const drwav_int16* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = pIn[i] << 16; + } +} + +DRWAV_API void drwav_s24_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + unsigned int s0 = pIn[i*3 + 0]; + unsigned int s1 = pIn[i*3 + 1]; + unsigned int s2 = pIn[i*3 + 2]; + + drwav_int32 sample32 = (drwav_int32)((s0 << 8) | (s1 << 16) | (s2 << 24)); + *pOut++ = sample32; + } +} + +DRWAV_API void drwav_f32_to_s32(drwav_int32* pOut, const float* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (drwav_int32)(2147483648.0 * pIn[i]); + } +} + +DRWAV_API void drwav_f64_to_s32(drwav_int32* pOut, const double* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (drwav_int32)(2147483648.0 * pIn[i]); + } +} + +DRWAV_API void drwav_alaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = ((drwav_int32)drwav__alaw_to_s16(pIn[i])) << 16; + } +} + +DRWAV_API void drwav_mulaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i= 0; i < sampleCount; ++i) { + *pOut++ = ((drwav_int32)drwav__mulaw_to_s16(pIn[i])) << 16; + } +} + + + +static drwav_int16* drwav__read_pcm_frames_and_close_s16(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount) +{ + drwav_uint64 sampleDataSize; + drwav_int16* pSampleData; + drwav_uint64 framesRead; + + DRWAV_ASSERT(pWav != NULL); + + sampleDataSize = pWav->totalPCMFrameCount * pWav->channels * sizeof(drwav_int16); + if (sampleDataSize > DRWAV_SIZE_MAX) { + drwav_uninit(pWav); + return NULL; /* File's too big. */ + } + + pSampleData = (drwav_int16*)drwav__malloc_from_callbacks((size_t)sampleDataSize, &pWav->allocationCallbacks); /* <-- Safe cast due to the check above. */ + if (pSampleData == NULL) { + drwav_uninit(pWav); + return NULL; /* Failed to allocate memory. */ + } + + framesRead = drwav_read_pcm_frames_s16(pWav, (size_t)pWav->totalPCMFrameCount, pSampleData); + if (framesRead != pWav->totalPCMFrameCount) { + drwav__free_from_callbacks(pSampleData, &pWav->allocationCallbacks); + drwav_uninit(pWav); + return NULL; /* There was an error reading the samples. */ + } + + drwav_uninit(pWav); + + if (sampleRate) { + *sampleRate = pWav->sampleRate; + } + if (channels) { + *channels = pWav->channels; + } + if (totalFrameCount) { + *totalFrameCount = pWav->totalPCMFrameCount; + } + + return pSampleData; +} + +static float* drwav__read_pcm_frames_and_close_f32(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount) +{ + drwav_uint64 sampleDataSize; + float* pSampleData; + drwav_uint64 framesRead; + + DRWAV_ASSERT(pWav != NULL); + + sampleDataSize = pWav->totalPCMFrameCount * pWav->channels * sizeof(float); + if (sampleDataSize > DRWAV_SIZE_MAX) { + drwav_uninit(pWav); + return NULL; /* File's too big. */ + } + + pSampleData = (float*)drwav__malloc_from_callbacks((size_t)sampleDataSize, &pWav->allocationCallbacks); /* <-- Safe cast due to the check above. */ + if (pSampleData == NULL) { + drwav_uninit(pWav); + return NULL; /* Failed to allocate memory. */ + } + + framesRead = drwav_read_pcm_frames_f32(pWav, (size_t)pWav->totalPCMFrameCount, pSampleData); + if (framesRead != pWav->totalPCMFrameCount) { + drwav__free_from_callbacks(pSampleData, &pWav->allocationCallbacks); + drwav_uninit(pWav); + return NULL; /* There was an error reading the samples. */ + } + + drwav_uninit(pWav); + + if (sampleRate) { + *sampleRate = pWav->sampleRate; + } + if (channels) { + *channels = pWav->channels; + } + if (totalFrameCount) { + *totalFrameCount = pWav->totalPCMFrameCount; + } + + return pSampleData; +} + +static drwav_int32* drwav__read_pcm_frames_and_close_s32(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount) +{ + drwav_uint64 sampleDataSize; + drwav_int32* pSampleData; + drwav_uint64 framesRead; + + DRWAV_ASSERT(pWav != NULL); + + sampleDataSize = pWav->totalPCMFrameCount * pWav->channels * sizeof(drwav_int32); + if (sampleDataSize > DRWAV_SIZE_MAX) { + drwav_uninit(pWav); + return NULL; /* File's too big. */ + } + + pSampleData = (drwav_int32*)drwav__malloc_from_callbacks((size_t)sampleDataSize, &pWav->allocationCallbacks); /* <-- Safe cast due to the check above. */ + if (pSampleData == NULL) { + drwav_uninit(pWav); + return NULL; /* Failed to allocate memory. */ + } + + framesRead = drwav_read_pcm_frames_s32(pWav, (size_t)pWav->totalPCMFrameCount, pSampleData); + if (framesRead != pWav->totalPCMFrameCount) { + drwav__free_from_callbacks(pSampleData, &pWav->allocationCallbacks); + drwav_uninit(pWav); + return NULL; /* There was an error reading the samples. */ + } + + drwav_uninit(pWav); + + if (sampleRate) { + *sampleRate = pWav->sampleRate; + } + if (channels) { + *channels = pWav->channels; + } + if (totalFrameCount) { + *totalFrameCount = pWav->totalPCMFrameCount; + } + + return pSampleData; +} + + + +DRWAV_API drwav_int16* drwav_open_and_read_pcm_frames_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init(&wav, onRead, onSeek, pUserData, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API float* drwav_open_and_read_pcm_frames_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init(&wav, onRead, onSeek, pUserData, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API drwav_int32* drwav_open_and_read_pcm_frames_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init(&wav, onRead, onSeek, pUserData, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +#ifndef DR_WAV_NO_STDIO +DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + + +DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (channelsOut) { + *channelsOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file_w(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (channelsOut) { + *channelsOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file_w(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (channelsOut) { + *channelsOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file_w(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} +#endif + +DRWAV_API drwav_int16* drwav_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_memory(&wav, data, dataSize, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API float* drwav_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_memory(&wav, data, dataSize, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API drwav_int32* drwav_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_memory(&wav, data, dataSize, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} +#endif /* DR_WAV_NO_CONVERSION_API */ + + +DRWAV_API void drwav_free(void* p, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks != NULL) { + drwav__free_from_callbacks(p, pAllocationCallbacks); + } else { + drwav__free_default(p, NULL); + } +} + +DRWAV_API drwav_uint16 drwav_bytes_to_u16(const drwav_uint8* data) +{ + return drwav__bytes_to_u16(data); +} + +DRWAV_API drwav_int16 drwav_bytes_to_s16(const drwav_uint8* data) +{ + return drwav__bytes_to_s16(data); +} + +DRWAV_API drwav_uint32 drwav_bytes_to_u32(const drwav_uint8* data) +{ + return drwav__bytes_to_u32(data); +} + +DRWAV_API drwav_int32 drwav_bytes_to_s32(const drwav_uint8* data) +{ + return drwav__bytes_to_s32(data); +} + +DRWAV_API drwav_uint64 drwav_bytes_to_u64(const drwav_uint8* data) +{ + return drwav__bytes_to_u64(data); +} + +DRWAV_API drwav_int64 drwav_bytes_to_s64(const drwav_uint8* data) +{ + return drwav__bytes_to_s64(data); +} + + +DRWAV_API drwav_bool32 drwav_guid_equal(const drwav_uint8 a[16], const drwav_uint8 b[16]) +{ + return drwav__guid_equal(a, b); +} + +DRWAV_API drwav_bool32 drwav_fourcc_equal(const drwav_uint8* a, const char* b) +{ + return drwav__fourcc_equal(a, b); +} + +#endif /* dr_wav_c */ +#endif /* DR_WAV_IMPLEMENTATION */ + +/* +RELEASE NOTES - v0.11.0 +======================= +Version 0.11.0 has breaking API changes. + +Improved Client-Defined Memory Allocation +----------------------------------------- +The main change with this release is the addition of a more flexible way of implementing custom memory allocation routines. The +existing system of DRWAV_MALLOC, DRWAV_REALLOC and DRWAV_FREE are still in place and will be used by default when no custom +allocation callbacks are specified. + +To use the new system, you pass in a pointer to a drwav_allocation_callbacks object to drwav_init() and family, like this: + + void* my_malloc(size_t sz, void* pUserData) + { + return malloc(sz); + } + void* my_realloc(void* p, size_t sz, void* pUserData) + { + return realloc(p, sz); + } + void my_free(void* p, void* pUserData) + { + free(p); + } + + ... + + drwav_allocation_callbacks allocationCallbacks; + allocationCallbacks.pUserData = &myData; + allocationCallbacks.onMalloc = my_malloc; + allocationCallbacks.onRealloc = my_realloc; + allocationCallbacks.onFree = my_free; + drwav_init_file(&wav, "my_file.wav", &allocationCallbacks); + +The advantage of this new system is that it allows you to specify user data which will be passed in to the allocation routines. + +Passing in null for the allocation callbacks object will cause dr_wav to use defaults which is the same as DRWAV_MALLOC, +DRWAV_REALLOC and DRWAV_FREE and the equivalent of how it worked in previous versions. + +Every API that opens a drwav object now takes this extra parameter. These include the following: + + drwav_init() + drwav_init_ex() + drwav_init_file() + drwav_init_file_ex() + drwav_init_file_w() + drwav_init_file_w_ex() + drwav_init_memory() + drwav_init_memory_ex() + drwav_init_write() + drwav_init_write_sequential() + drwav_init_write_sequential_pcm_frames() + drwav_init_file_write() + drwav_init_file_write_sequential() + drwav_init_file_write_sequential_pcm_frames() + drwav_init_file_write_w() + drwav_init_file_write_sequential_w() + drwav_init_file_write_sequential_pcm_frames_w() + drwav_init_memory_write() + drwav_init_memory_write_sequential() + drwav_init_memory_write_sequential_pcm_frames() + drwav_open_and_read_pcm_frames_s16() + drwav_open_and_read_pcm_frames_f32() + drwav_open_and_read_pcm_frames_s32() + drwav_open_file_and_read_pcm_frames_s16() + drwav_open_file_and_read_pcm_frames_f32() + drwav_open_file_and_read_pcm_frames_s32() + drwav_open_file_and_read_pcm_frames_s16_w() + drwav_open_file_and_read_pcm_frames_f32_w() + drwav_open_file_and_read_pcm_frames_s32_w() + drwav_open_memory_and_read_pcm_frames_s16() + drwav_open_memory_and_read_pcm_frames_f32() + drwav_open_memory_and_read_pcm_frames_s32() + +Endian Improvements +------------------- +Previously, the following APIs returned little-endian audio data. These now return native-endian data. This improves compatibility +on big-endian architectures. + + drwav_read_pcm_frames() + drwav_read_pcm_frames_s16() + drwav_read_pcm_frames_s32() + drwav_read_pcm_frames_f32() + drwav_open_and_read_pcm_frames_s16() + drwav_open_and_read_pcm_frames_s32() + drwav_open_and_read_pcm_frames_f32() + drwav_open_file_and_read_pcm_frames_s16() + drwav_open_file_and_read_pcm_frames_s32() + drwav_open_file_and_read_pcm_frames_f32() + drwav_open_file_and_read_pcm_frames_s16_w() + drwav_open_file_and_read_pcm_frames_s32_w() + drwav_open_file_and_read_pcm_frames_f32_w() + drwav_open_memory_and_read_pcm_frames_s16() + drwav_open_memory_and_read_pcm_frames_s32() + drwav_open_memory_and_read_pcm_frames_f32() + +APIs have been added to give you explicit control over whether or not audio data is read or written in big- or little-endian byte +order: + + drwav_read_pcm_frames_le() + drwav_read_pcm_frames_be() + drwav_read_pcm_frames_s16le() + drwav_read_pcm_frames_s16be() + drwav_read_pcm_frames_f32le() + drwav_read_pcm_frames_f32be() + drwav_read_pcm_frames_s32le() + drwav_read_pcm_frames_s32be() + drwav_write_pcm_frames_le() + drwav_write_pcm_frames_be() + +Removed APIs +------------ +The following APIs were deprecated in version 0.10.0 and have now been removed: + + drwav_open() + drwav_open_ex() + drwav_open_write() + drwav_open_write_sequential() + drwav_open_file() + drwav_open_file_ex() + drwav_open_file_write() + drwav_open_file_write_sequential() + drwav_open_memory() + drwav_open_memory_ex() + drwav_open_memory_write() + drwav_open_memory_write_sequential() + drwav_close() + + + +RELEASE NOTES - v0.10.0 +======================= +Version 0.10.0 has breaking API changes. There are no significant bug fixes in this release, so if you are affected you do +not need to upgrade. + +Removed APIs +------------ +The following APIs were deprecated in version 0.9.0 and have been completely removed in version 0.10.0: + + drwav_read() + drwav_read_s16() + drwav_read_f32() + drwav_read_s32() + drwav_seek_to_sample() + drwav_write() + drwav_open_and_read_s16() + drwav_open_and_read_f32() + drwav_open_and_read_s32() + drwav_open_file_and_read_s16() + drwav_open_file_and_read_f32() + drwav_open_file_and_read_s32() + drwav_open_memory_and_read_s16() + drwav_open_memory_and_read_f32() + drwav_open_memory_and_read_s32() + drwav::totalSampleCount + +See release notes for version 0.9.0 at the bottom of this file for replacement APIs. + +Deprecated APIs +--------------- +The following APIs have been deprecated. There is a confusing and completely arbitrary difference between drwav_init*() and +drwav_open*(), where drwav_init*() initializes a pre-allocated drwav object, whereas drwav_open*() will first allocated a +drwav object on the heap and then initialize it. drwav_open*() has been deprecated which means you must now use a pre- +allocated drwav object with drwav_init*(). If you need the previous functionality, you can just do a malloc() followed by +a called to one of the drwav_init*() APIs. + + drwav_open() + drwav_open_ex() + drwav_open_write() + drwav_open_write_sequential() + drwav_open_file() + drwav_open_file_ex() + drwav_open_file_write() + drwav_open_file_write_sequential() + drwav_open_memory() + drwav_open_memory_ex() + drwav_open_memory_write() + drwav_open_memory_write_sequential() + drwav_close() + +These APIs will be removed completely in a future version. The rationale for this change is to remove confusion between the +two different ways to initialize a drwav object. +*/ + +/* +REVISION HISTORY +================ +v0.12.16 - 2020-12-02 + - Fix a bug when trying to read more bytes than can fit in a size_t. + +v0.12.15 - 2020-11-21 + - Fix compilation with OpenWatcom. + +v0.12.14 - 2020-11-13 + - Minor code clean up. + +v0.12.13 - 2020-11-01 + - Improve compiler support for older versions of GCC. + +v0.12.12 - 2020-09-28 + - Add support for RF64. + - Fix a bug in writing mode where the size of the RIFF chunk incorrectly includes the header section. + +v0.12.11 - 2020-09-08 + - Fix a compilation error on older compilers. + +v0.12.10 - 2020-08-24 + - Fix a bug when seeking with ADPCM formats. + +v0.12.9 - 2020-08-02 + - Simplify sized types. + +v0.12.8 - 2020-07-25 + - Fix a compilation warning. + +v0.12.7 - 2020-07-15 + - Fix some bugs on big-endian architectures. + - Fix an error in s24 to f32 conversion. + +v0.12.6 - 2020-06-23 + - Change drwav_read_*() to allow NULL to be passed in as the output buffer which is equivalent to a forward seek. + - Fix a buffer overflow when trying to decode invalid IMA-ADPCM files. + - Add include guard for the implementation section. + +v0.12.5 - 2020-05-27 + - Minor documentation fix. + +v0.12.4 - 2020-05-16 + - Replace assert() with DRWAV_ASSERT(). + - Add compile-time and run-time version querying. + - DRWAV_VERSION_MINOR + - DRWAV_VERSION_MAJOR + - DRWAV_VERSION_REVISION + - DRWAV_VERSION_STRING + - drwav_version() + - drwav_version_string() + +v0.12.3 - 2020-04-30 + - Fix compilation errors with VC6. + +v0.12.2 - 2020-04-21 + - Fix a bug where drwav_init_file() does not close the file handle after attempting to load an erroneous file. + +v0.12.1 - 2020-04-13 + - Fix some pedantic warnings. + +v0.12.0 - 2020-04-04 + - API CHANGE: Add container and format parameters to the chunk callback. + - Minor documentation updates. + +v0.11.5 - 2020-03-07 + - Fix compilation error with Visual Studio .NET 2003. + +v0.11.4 - 2020-01-29 + - Fix some static analysis warnings. + - Fix a bug when reading f32 samples from an A-law encoded stream. + +v0.11.3 - 2020-01-12 + - Minor changes to some f32 format conversion routines. + - Minor bug fix for ADPCM conversion when end of file is reached. + +v0.11.2 - 2019-12-02 + - Fix a possible crash when using custom memory allocators without a custom realloc() implementation. + - Fix an integer overflow bug. + - Fix a null pointer dereference bug. + - Add limits to sample rate, channels and bits per sample to tighten up some validation. + +v0.11.1 - 2019-10-07 + - Internal code clean up. + +v0.11.0 - 2019-10-06 + - API CHANGE: Add support for user defined memory allocation routines. This system allows the program to specify their own memory allocation + routines with a user data pointer for client-specific contextual data. This adds an extra parameter to the end of the following APIs: + - drwav_init() + - drwav_init_ex() + - drwav_init_file() + - drwav_init_file_ex() + - drwav_init_file_w() + - drwav_init_file_w_ex() + - drwav_init_memory() + - drwav_init_memory_ex() + - drwav_init_write() + - drwav_init_write_sequential() + - drwav_init_write_sequential_pcm_frames() + - drwav_init_file_write() + - drwav_init_file_write_sequential() + - drwav_init_file_write_sequential_pcm_frames() + - drwav_init_file_write_w() + - drwav_init_file_write_sequential_w() + - drwav_init_file_write_sequential_pcm_frames_w() + - drwav_init_memory_write() + - drwav_init_memory_write_sequential() + - drwav_init_memory_write_sequential_pcm_frames() + - drwav_open_and_read_pcm_frames_s16() + - drwav_open_and_read_pcm_frames_f32() + - drwav_open_and_read_pcm_frames_s32() + - drwav_open_file_and_read_pcm_frames_s16() + - drwav_open_file_and_read_pcm_frames_f32() + - drwav_open_file_and_read_pcm_frames_s32() + - drwav_open_file_and_read_pcm_frames_s16_w() + - drwav_open_file_and_read_pcm_frames_f32_w() + - drwav_open_file_and_read_pcm_frames_s32_w() + - drwav_open_memory_and_read_pcm_frames_s16() + - drwav_open_memory_and_read_pcm_frames_f32() + - drwav_open_memory_and_read_pcm_frames_s32() + Set this extra parameter to NULL to use defaults which is the same as the previous behaviour. Setting this NULL will use + DRWAV_MALLOC, DRWAV_REALLOC and DRWAV_FREE. + - Add support for reading and writing PCM frames in an explicit endianness. New APIs: + - drwav_read_pcm_frames_le() + - drwav_read_pcm_frames_be() + - drwav_read_pcm_frames_s16le() + - drwav_read_pcm_frames_s16be() + - drwav_read_pcm_frames_f32le() + - drwav_read_pcm_frames_f32be() + - drwav_read_pcm_frames_s32le() + - drwav_read_pcm_frames_s32be() + - drwav_write_pcm_frames_le() + - drwav_write_pcm_frames_be() + - Remove deprecated APIs. + - API CHANGE: The following APIs now return native-endian data. Previously they returned little-endian data. + - drwav_read_pcm_frames() + - drwav_read_pcm_frames_s16() + - drwav_read_pcm_frames_s32() + - drwav_read_pcm_frames_f32() + - drwav_open_and_read_pcm_frames_s16() + - drwav_open_and_read_pcm_frames_s32() + - drwav_open_and_read_pcm_frames_f32() + - drwav_open_file_and_read_pcm_frames_s16() + - drwav_open_file_and_read_pcm_frames_s32() + - drwav_open_file_and_read_pcm_frames_f32() + - drwav_open_file_and_read_pcm_frames_s16_w() + - drwav_open_file_and_read_pcm_frames_s32_w() + - drwav_open_file_and_read_pcm_frames_f32_w() + - drwav_open_memory_and_read_pcm_frames_s16() + - drwav_open_memory_and_read_pcm_frames_s32() + - drwav_open_memory_and_read_pcm_frames_f32() + +v0.10.1 - 2019-08-31 + - Correctly handle partial trailing ADPCM blocks. + +v0.10.0 - 2019-08-04 + - Remove deprecated APIs. + - Add wchar_t variants for file loading APIs: + drwav_init_file_w() + drwav_init_file_ex_w() + drwav_init_file_write_w() + drwav_init_file_write_sequential_w() + - Add drwav_target_write_size_bytes() which calculates the total size in bytes of a WAV file given a format and sample count. + - Add APIs for specifying the PCM frame count instead of the sample count when opening in sequential write mode: + drwav_init_write_sequential_pcm_frames() + drwav_init_file_write_sequential_pcm_frames() + drwav_init_file_write_sequential_pcm_frames_w() + drwav_init_memory_write_sequential_pcm_frames() + - Deprecate drwav_open*() and drwav_close(): + drwav_open() + drwav_open_ex() + drwav_open_write() + drwav_open_write_sequential() + drwav_open_file() + drwav_open_file_ex() + drwav_open_file_write() + drwav_open_file_write_sequential() + drwav_open_memory() + drwav_open_memory_ex() + drwav_open_memory_write() + drwav_open_memory_write_sequential() + drwav_close() + - Minor documentation updates. + +v0.9.2 - 2019-05-21 + - Fix warnings. + +v0.9.1 - 2019-05-05 + - Add support for C89. + - Change license to choice of public domain or MIT-0. + +v0.9.0 - 2018-12-16 + - API CHANGE: Add new reading APIs for reading by PCM frames instead of samples. Old APIs have been deprecated and + will be removed in v0.10.0. Deprecated APIs and their replacements: + drwav_read() -> drwav_read_pcm_frames() + drwav_read_s16() -> drwav_read_pcm_frames_s16() + drwav_read_f32() -> drwav_read_pcm_frames_f32() + drwav_read_s32() -> drwav_read_pcm_frames_s32() + drwav_seek_to_sample() -> drwav_seek_to_pcm_frame() + drwav_write() -> drwav_write_pcm_frames() + drwav_open_and_read_s16() -> drwav_open_and_read_pcm_frames_s16() + drwav_open_and_read_f32() -> drwav_open_and_read_pcm_frames_f32() + drwav_open_and_read_s32() -> drwav_open_and_read_pcm_frames_s32() + drwav_open_file_and_read_s16() -> drwav_open_file_and_read_pcm_frames_s16() + drwav_open_file_and_read_f32() -> drwav_open_file_and_read_pcm_frames_f32() + drwav_open_file_and_read_s32() -> drwav_open_file_and_read_pcm_frames_s32() + drwav_open_memory_and_read_s16() -> drwav_open_memory_and_read_pcm_frames_s16() + drwav_open_memory_and_read_f32() -> drwav_open_memory_and_read_pcm_frames_f32() + drwav_open_memory_and_read_s32() -> drwav_open_memory_and_read_pcm_frames_s32() + drwav::totalSampleCount -> drwav::totalPCMFrameCount + - API CHANGE: Rename drwav_open_and_read_file_*() to drwav_open_file_and_read_*(). + - API CHANGE: Rename drwav_open_and_read_memory_*() to drwav_open_memory_and_read_*(). + - Add built-in support for smpl chunks. + - Add support for firing a callback for each chunk in the file at initialization time. + - This is enabled through the drwav_init_ex(), etc. family of APIs. + - Handle invalid FMT chunks more robustly. + +v0.8.5 - 2018-09-11 + - Const correctness. + - Fix a potential stack overflow. + +v0.8.4 - 2018-08-07 + - Improve 64-bit detection. + +v0.8.3 - 2018-08-05 + - Fix C++ build on older versions of GCC. + +v0.8.2 - 2018-08-02 + - Fix some big-endian bugs. + +v0.8.1 - 2018-06-29 + - Add support for sequential writing APIs. + - Disable seeking in write mode. + - Fix bugs with Wave64. + - Fix typos. + +v0.8 - 2018-04-27 + - Bug fix. + - Start using major.minor.revision versioning. + +v0.7f - 2018-02-05 + - Restrict ADPCM formats to a maximum of 2 channels. + +v0.7e - 2018-02-02 + - Fix a crash. + +v0.7d - 2018-02-01 + - Fix a crash. + +v0.7c - 2018-02-01 + - Set drwav.bytesPerSample to 0 for all compressed formats. + - Fix a crash when reading 16-bit floating point WAV files. In this case dr_wav will output silence for + all format conversion reading APIs (*_s16, *_s32, *_f32 APIs). + - Fix some divide-by-zero errors. + +v0.7b - 2018-01-22 + - Fix errors with seeking of compressed formats. + - Fix compilation error when DR_WAV_NO_CONVERSION_API + +v0.7a - 2017-11-17 + - Fix some GCC warnings. + +v0.7 - 2017-11-04 + - Add writing APIs. + +v0.6 - 2017-08-16 + - API CHANGE: Rename dr_* types to drwav_*. + - Add support for custom implementations of malloc(), realloc(), etc. + - Add support for Microsoft ADPCM. + - Add support for IMA ADPCM (DVI, format code 0x11). + - Optimizations to drwav_read_s16(). + - Bug fixes. + +v0.5g - 2017-07-16 + - Change underlying type for booleans to unsigned. + +v0.5f - 2017-04-04 + - Fix a minor bug with drwav_open_and_read_s16() and family. + +v0.5e - 2016-12-29 + - Added support for reading samples as signed 16-bit integers. Use the _s16() family of APIs for this. + - Minor fixes to documentation. + +v0.5d - 2016-12-28 + - Use drwav_int* and drwav_uint* sized types to improve compiler support. + +v0.5c - 2016-11-11 + - Properly handle JUNK chunks that come before the FMT chunk. + +v0.5b - 2016-10-23 + - A minor change to drwav_bool8 and drwav_bool32 types. + +v0.5a - 2016-10-11 + - Fixed a bug with drwav_open_and_read() and family due to incorrect argument ordering. + - Improve A-law and mu-law efficiency. + +v0.5 - 2016-09-29 + - API CHANGE. Swap the order of "channels" and "sampleRate" parameters in drwav_open_and_read*(). Rationale for this is to + keep it consistent with dr_audio and dr_flac. + +v0.4b - 2016-09-18 + - Fixed a typo in documentation. + +v0.4a - 2016-09-18 + - Fixed a typo. + - Change date format to ISO 8601 (YYYY-MM-DD) + +v0.4 - 2016-07-13 + - API CHANGE. Make onSeek consistent with dr_flac. + - API CHANGE. Rename drwav_seek() to drwav_seek_to_sample() for clarity and consistency with dr_flac. + - Added support for Sony Wave64. + +v0.3a - 2016-05-28 + - API CHANGE. Return drwav_bool32 instead of int in onSeek callback. + - Fixed a memory leak. + +v0.3 - 2016-05-22 + - Lots of API changes for consistency. + +v0.2a - 2016-05-16 + - Fixed Linux/GCC build. + +v0.2 - 2016-05-11 + - Added support for reading data as signed 32-bit PCM for consistency with dr_flac. + +v0.1a - 2016-05-07 + - Fixed a bug in drwav_open_file() where the file handle would not be closed if the loader failed to initialize. + +v0.1 - 2016-05-04 + - Initial versioned release. +*/ + +/* +This software is available as a choice of the following licenses. Choose +whichever you prefer. + +=============================================================================== +ALTERNATIVE 1 - Public Domain (www.unlicense.org) +=============================================================================== +This is free and unencumbered software released into the public domain. + +Anyone is free to copy, modify, publish, use, compile, sell, or distribute this +software, either in source code form or as a compiled binary, for any purpose, +commercial or non-commercial, and by any means. + +In jurisdictions that recognize copyright laws, the author or authors of this +software dedicate any and all copyright interest in the software to the public +domain. We make this dedication for the benefit of the public at large and to +the detriment of our heirs and successors. We intend this dedication to be an +overt act of relinquishment in perpetuity of all present and future rights to +this software under copyright law. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN +ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION +WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + +For more information, please refer to + +=============================================================================== +ALTERNATIVE 2 - MIT No Attribution +=============================================================================== +Copyright 2020 David Reid + +Permission is hereby granted, free of charge, to any person obtaining a copy of +this software and associated documentation files (the "Software"), to deal in +the Software without restriction, including without limitation the rights to +use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies +of the Software, and to permit persons to whom the Software is furnished to do +so. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. +*/ \ No newline at end of file diff --git a/vall_e.cpp/include/encodec.h b/vall_e.cpp/include/encodec.h new file mode 100644 index 0000000..0a8d8b9 --- /dev/null +++ b/vall_e.cpp/include/encodec.h @@ -0,0 +1,184 @@ +/* +╞══════════════════════════════════════════════════════════════════════════════╡ +│ Copyright 2024 Pierre-Antoine Bannier │ +│ │ +│ Permission to use, copy, modify, and/or distribute this software for │ +│ any purpose with or without fee is hereby granted, provided that the │ +│ above copyright notice and this permission notice appear in all copies. │ +│ │ +│ THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL │ +│ WARRANTIES WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED │ +│ WARRANTIES OF MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE │ +│ AUTHOR BE LIABLE FOR ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL │ +│ DAMAGES OR ANY DAMAGES WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR │ +│ PROFITS, WHETHER IN AN ACTION OF CONTRACT, NEGLIGENCE OR OTHER │ +│ TORTIOUS ACTION, ARISING OUT OF OR IN CONNECTION WITH THE USE OR │ +│ PERFORMANCE OF THIS SOFTWARE. │ +╚─────────────────────────────────────────────────────────────────────────────*/ +/* + * This file contains the declarations of the structs and functions used in the encodec library. + * The library provides functionality for audio compression and decompression using a custom model. + * The model consists of an encoder, a quantizer and a decoder, each with their own set of parameters. + * The library also provides functions for loading and freeing the model, as well as compressing and decompressing audio data. + * + */ +#pragma once + +#include "ggml-alloc.h" +#include "ggml-backend.h" +#include "ggml.h" + +#ifdef __cplusplus +extern "C" { +#endif + struct encodec_context; + + struct encodec_statistics { + // The time taken to load the model. + int64_t t_load_us; + // The time taken to compute the model. + int64_t t_compute_us; + }; + + /** + * Loads an encodec model from the specified file path. + * + * @param model_path The file path to the encodec model. + * @param offset The offset (in bytes) to the start of the model in the file. + * @param n_gpu_layers The number of GPU layers to use. + * @return A pointer to the encodec context struct. + */ + struct encodec_context *encodec_load_model( + const char *model_path, + const int offset, + int n_gpu_layers); + + /** + * Sets the target bandwidth for the given encodec context. + * + * @param ectx The encodec context to set the target bandwidth for. + * @param bandwidth The target bandwidth to set, in bits per second. + */ + void encodec_set_target_bandwidth( + struct encodec_context *ectx, + int bandwidth); + + /** + * Sets the sample rate for the given encodec context. + * + * @param ectx The encodec context to set the target bandwidth for. + * @param sample_rate The sample rate to set. + */ + void encodec_set_sample_rate( + struct encodec_context *ectx, + int sample_rate); + + /** + * Reconstructs audio from raw audio data using the specified encodec context. + * + * @param ectx The encodec context to use for reconstruction. + * @param raw_audio The raw audio data to reconstruct. + * @param n_samples The number of samples in the raw audio buffer. + * @param n_threads The number of threads to use for reconstruction. + * @return True if the reconstruction was successful, false otherwise. + */ + bool encodec_reconstruct_audio( + struct encodec_context *ectx, + const float *raw_audio, + const int n_samples, + int n_threads); + + /** + * Compresses audio data using the specified encodec context. + * + * @param ectx The encodec context to use for compression. + * @param raw_audio The raw audio data to compress. + * @param n_samples The number of samples in the raw audio buffer. + * @param n_threads The number of threads to use for compression. + * @return True if the compression was successful, false otherwise. + */ + bool encodec_compress_audio( + struct encodec_context *ectx, + const float *raw_audio, + const int n_samples, + int n_threads); + + /** + * Decompresses audio data using the specified encodec context. + * + * @param ectx The encodec context to use for decompression. + * @param codes The compressed audio data to decompress. + * @param n_codes The number of codes in the codes buffer. + * @param n_threads The number of threads to use for decompression. + * @return True if the audio data was successfully decompressed, false otherwise. + */ + bool encodec_decompress_audio( + struct encodec_context *ectx, + const int32_t *codes, + const int n_codes, + int n_threads); + + /** + * Gets the audio data from the given encodec context. + * + * @param ectx The encodec context to get the audio data from. + * @return A pointer to the audio data. + */ + float * encodec_get_audio( + struct encodec_context *ectx); + + /** + * Gets the size of the audio data from the given encodec context. + * + * @param ectx The encodec context to get the audio size from. + * @return The size of the audio data. + */ + int encodec_get_audio_size( + struct encodec_context *ectx); + + /** + * Gets the code data from the given encodec context. + * + * @param ectx The encodec context to get the code data from. + * @return A pointer to the code data. + */ + int32_t * encodec_get_codes( + struct encodec_context *ectx); + + /** + * Gets the size of the code data from the given encodec context. + * + * @param ectx The encodec context to get the code size from. + * @return The size of the code data. + */ + int encodec_get_codes_size( + struct encodec_context *ectx); + + /** + * Gets the statistics for the given encodec context. + * + * @param ectx The encodec context to get the statistics for. + * @return A pointer to the statistics struct. + */ + const struct encodec_statistics* encodec_get_statistics( + struct encodec_context *ectx); + + /** + * Reset the statistics for the given encodec context. + * + * @param ectx The encodec context to reset the statistics for. + */ + void encodec_reset_statistics( + struct encodec_context *ectx); + + /** + * @brief Frees the memory allocated for an encodec context. + * + * @param ectx The encodec context to free. + */ + void encodec_free( + struct encodec_context *ectx); + +#ifdef __cplusplus +} +#endif \ No newline at end of file diff --git a/vall_e.cpp/include/encoder.h b/vall_e.cpp/include/encoder.h new file mode 100644 index 0000000..15b4e3f --- /dev/null +++ b/vall_e.cpp/include/encoder.h @@ -0,0 +1,109 @@ +#pragma once + +#include + +#include "ggml.h" +#include "lstm.h" + +// res + downsample block at some ratio +struct encodec_encoder_block { + // conv1 + struct ggml_tensor *conv_1_w; + struct ggml_tensor *conv_1_b; + + // conv2 + struct ggml_tensor *conv_2_w; + struct ggml_tensor *conv_2_b; + + // shortcut + struct ggml_tensor *conv_sc_w; + struct ggml_tensor *conv_sc_b; + + // downsampling layers + struct ggml_tensor *ds_conv_w; + struct ggml_tensor *ds_conv_b; +}; + +struct encodec_encoder { + struct ggml_tensor *init_conv_w; + struct ggml_tensor *init_conv_b; + + encodec_lstm lstm; + + struct ggml_tensor *final_conv_w; + struct ggml_tensor *final_conv_b; + + std::vector blocks; +}; + +struct ggml_tensor *encodec_forward_encoder( + const struct encodec_encoder *encoder, struct ggml_context *ctx0, + struct ggml_tensor *inp, const int * ratios, const int kernel_size, const int res_kernel_size, + const int stride) { + + if (!inp) { + fprintf(stderr, "%s: null input tensor\n", __func__); + return NULL; + } + + struct ggml_tensor *inpL = strided_conv_1d( + ctx0, inp, encoder->init_conv_w, encoder->init_conv_b, stride); + + for (int layer_ix = 0; layer_ix < 4; layer_ix++) { + encodec_encoder_block block = encoder->blocks[layer_ix]; + + struct ggml_tensor *current = inpL; + + // shortcut + struct ggml_tensor *shortcut = strided_conv_1d( + ctx0, inpL, block.conv_sc_w, block.conv_sc_b, stride); + + // conv1 + current = ggml_elu(ctx0, current); + + current = strided_conv_1d( + ctx0, current, block.conv_1_w, block.conv_1_b, stride); + + // conv2 + current = ggml_elu(ctx0, current); + + current = strided_conv_1d( + ctx0, current, block.conv_2_w, block.conv_2_b, stride); + + // residual connection + inpL = ggml_add(ctx0, current, shortcut); + + // downsampling layers + inpL = ggml_elu(ctx0, inpL); + + inpL = strided_conv_1d( + ctx0, inpL, block.ds_conv_w, block.ds_conv_b, ratios[3 - layer_ix]); + } + + // lstm + { + struct ggml_tensor *cur = inpL; + + const encodec_lstm lstm = encoder->lstm; + + // first lstm layer + char l0_prefix[7] = "enc_l0"; + struct ggml_tensor *hs1 = forward_pass_lstm_unilayer( + ctx0, cur, lstm.l0_ih_w, lstm.l0_hh_w, lstm.l0_ih_b, lstm.l0_hh_b, l0_prefix); + + // second lstm layer + char l1_prefix[7] = "enc_l1"; + struct ggml_tensor *out = forward_pass_lstm_unilayer( + ctx0, hs1, lstm.l1_ih_w, lstm.l1_hh_w, lstm.l1_ih_b, lstm.l1_hh_b, l1_prefix); + + inpL = ggml_add(ctx0, inpL, out); + } + + // final conv + inpL = ggml_elu(ctx0, inpL); + + struct ggml_tensor *encoded_inp = strided_conv_1d( + ctx0, inpL, encoder->final_conv_w, encoder->final_conv_b, stride); + + return encoded_inp; +} diff --git a/vall_e.cpp/include/espeak-ng/encoding.h b/vall_e.cpp/include/espeak-ng/encoding.h new file mode 100644 index 0000000..9cc8a5d --- /dev/null +++ b/vall_e.cpp/include/espeak-ng/encoding.h @@ -0,0 +1,103 @@ +/* + * Copyright (C) 2017 Reece H. Dunn + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU 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 General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see: . + */ +#ifndef ESPEAK_NG_ENCODING_H +#define ESPEAK_NG_ENCODING_H + +#include + +#ifdef __cplusplus +extern "C" +{ +#endif + +typedef enum +{ + ESPEAKNG_ENCODING_UNKNOWN, + ESPEAKNG_ENCODING_US_ASCII, + ESPEAKNG_ENCODING_ISO_8859_1, + ESPEAKNG_ENCODING_ISO_8859_2, + ESPEAKNG_ENCODING_ISO_8859_3, + ESPEAKNG_ENCODING_ISO_8859_4, + ESPEAKNG_ENCODING_ISO_8859_5, + ESPEAKNG_ENCODING_ISO_8859_6, + ESPEAKNG_ENCODING_ISO_8859_7, + ESPEAKNG_ENCODING_ISO_8859_8, + ESPEAKNG_ENCODING_ISO_8859_9, + ESPEAKNG_ENCODING_ISO_8859_10, + ESPEAKNG_ENCODING_ISO_8859_11, + // ISO-8859-12 is not a valid encoding. + ESPEAKNG_ENCODING_ISO_8859_13, + ESPEAKNG_ENCODING_ISO_8859_14, + ESPEAKNG_ENCODING_ISO_8859_15, + ESPEAKNG_ENCODING_ISO_8859_16, + ESPEAKNG_ENCODING_KOI8_R, + ESPEAKNG_ENCODING_ISCII, + ESPEAKNG_ENCODING_UTF_8, + ESPEAKNG_ENCODING_ISO_10646_UCS_2, +} espeak_ng_ENCODING; + +ESPEAK_NG_API espeak_ng_ENCODING +espeak_ng_EncodingFromName(const char *encoding); + +typedef struct espeak_ng_TEXT_DECODER_ espeak_ng_TEXT_DECODER; + +ESPEAK_NG_API espeak_ng_TEXT_DECODER * +create_text_decoder(void); + +ESPEAK_NG_API void +destroy_text_decoder(espeak_ng_TEXT_DECODER *decoder); + +ESPEAK_NG_API espeak_ng_STATUS +text_decoder_decode_string(espeak_ng_TEXT_DECODER *decoder, + const char *string, + int length, + espeak_ng_ENCODING encoding); + +ESPEAK_NG_API espeak_ng_STATUS +text_decoder_decode_string_auto(espeak_ng_TEXT_DECODER *decoder, + const char *string, + int length, + espeak_ng_ENCODING encoding); + +ESPEAK_NG_API espeak_ng_STATUS +text_decoder_decode_wstring(espeak_ng_TEXT_DECODER *decoder, + const wchar_t *string, + int length); + +ESPEAK_NG_API espeak_ng_STATUS +text_decoder_decode_string_multibyte(espeak_ng_TEXT_DECODER *decoder, + const void *input, + espeak_ng_ENCODING encoding, + int flags); + +ESPEAK_NG_API int +text_decoder_eof(espeak_ng_TEXT_DECODER *decoder); + +ESPEAK_NG_API uint32_t +text_decoder_getc(espeak_ng_TEXT_DECODER *decoder); + +ESPEAK_NG_API uint32_t +text_decoder_peekc(espeak_ng_TEXT_DECODER *decoder); + +ESPEAK_NG_API const void * +text_decoder_get_buffer(espeak_ng_TEXT_DECODER *decoder); + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/vall_e.cpp/include/espeak-ng/espeak_ng.h b/vall_e.cpp/include/espeak-ng/espeak_ng.h new file mode 100644 index 0000000..81988e4 --- /dev/null +++ b/vall_e.cpp/include/espeak-ng/espeak_ng.h @@ -0,0 +1,223 @@ +/* eSpeak NG API. + * + * Copyright (C) 2015-2017 Reece H. Dunn + * + * This program is free software: you can redistribute it and/or modify + * it under the terms of the GNU 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 General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + */ + +#ifndef ESPEAK_NG_H +#define ESPEAK_NG_H + +#include + +#ifdef __cplusplus +extern "C" +{ +#endif + +#if defined(_WIN32) || defined(_WIN64) +#ifdef LIBESPEAK_NG_EXPORT +#define ESPEAK_NG_API __declspec(dllexport) +#else +#define ESPEAK_NG_API __declspec(dllimport) +#endif +#else +#define ESPEAK_NG_API +#endif + +#define ESPEAKNG_DEFAULT_VOICE "en" + +typedef enum { + ENS_GROUP_MASK = 0x70000000, + ENS_GROUP_ERRNO = 0x00000000, /* Values 0-255 map to errno error codes. */ + ENS_GROUP_ESPEAK_NG = 0x10000000, /* eSpeak NG error codes. */ + + /* eSpeak NG 1.49.0 */ + ENS_OK = 0, + ENS_COMPILE_ERROR = 0x100001FF, + ENS_VERSION_MISMATCH = 0x100002FF, + ENS_FIFO_BUFFER_FULL = 0x100003FF, + ENS_NOT_INITIALIZED = 0x100004FF, + ENS_AUDIO_ERROR = 0x100005FF, + ENS_VOICE_NOT_FOUND = 0x100006FF, + ENS_MBROLA_NOT_FOUND = 0x100007FF, + ENS_MBROLA_VOICE_NOT_FOUND = 0x100008FF, + ENS_EVENT_BUFFER_FULL = 0x100009FF, + ENS_NOT_SUPPORTED = 0x10000AFF, + ENS_UNSUPPORTED_PHON_FORMAT = 0x10000BFF, + ENS_NO_SPECT_FRAMES = 0x10000CFF, + ENS_EMPTY_PHONEME_MANIFEST = 0x10000DFF, + ENS_SPEECH_STOPPED = 0x10000EFF, + + /* eSpeak NG 1.49.2 */ + ENS_UNKNOWN_PHONEME_FEATURE = 0x10000FFF, + ENS_UNKNOWN_TEXT_ENCODING = 0x100010FF, +} espeak_ng_STATUS; + +typedef enum { + ENOUTPUT_MODE_SYNCHRONOUS = 0x0001, + ENOUTPUT_MODE_SPEAK_AUDIO = 0x0002, +} espeak_ng_OUTPUT_MODE; + +typedef enum { + ENGENDER_UNKNOWN = 0, + ENGENDER_MALE = 1, + ENGENDER_FEMALE = 2, + ENGENDER_NEUTRAL = 3, +} espeak_ng_VOICE_GENDER; + +typedef struct +{ + void (*outputPhoSymbol)(char* pho_code,int pho_type); + void (*outputSilence)(short echo_tail); + void (*outputVoiced)(short sample); + void (*outputUnvoiced)(short sample); +} espeak_ng_OUTPUT_HOOKS; + +/* eSpeak NG 1.49.0 */ + +typedef struct espeak_ng_ERROR_CONTEXT_ *espeak_ng_ERROR_CONTEXT; + +ESPEAK_NG_API void +espeak_ng_ClearErrorContext(espeak_ng_ERROR_CONTEXT *context); + +ESPEAK_NG_API void +espeak_ng_GetStatusCodeMessage(espeak_ng_STATUS status, + char *buffer, + size_t length); + +ESPEAK_NG_API void +espeak_ng_PrintStatusCodeMessage(espeak_ng_STATUS status, + FILE *out, + espeak_ng_ERROR_CONTEXT context); + +ESPEAK_NG_API void +espeak_ng_InitializePath(const char *path); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_Initialize(espeak_ng_ERROR_CONTEXT *context); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_InitializeOutput(espeak_ng_OUTPUT_MODE output_mode, + int buffer_length, + const char *device); + +ESPEAK_NG_API int +espeak_ng_GetSampleRate(void); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_SetParameter(espeak_PARAMETER parameter, + int value, + int relative); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_SetPhonemeEvents(int enable, int ipa); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_SetPunctuationList(const wchar_t *punctlist); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_SetVoiceByName(const char *name); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_SetVoiceByFile(const char *filename); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_SetVoiceByProperties(espeak_VOICE *voice_selector); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_Synthesize(const void *text, + size_t size, + unsigned int position, + espeak_POSITION_TYPE position_type, + unsigned int end_position, + unsigned int flags, + unsigned int *unique_identifier, + void *user_data); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_SynthesizeMark(const void *text, + size_t size, + const char *index_mark, + unsigned int end_position, + unsigned int flags, + unsigned int *unique_identifier, + void *user_data); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_SpeakKeyName(const char *key_name); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_SpeakCharacter(wchar_t character); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_Cancel(void); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_Synchronize(void); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_Terminate(void); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_CompileDictionary(const char *dsource, + const char *dict_name, + FILE *log, + int flags, + espeak_ng_ERROR_CONTEXT *context); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_CompileMbrolaVoice(const char *path, + FILE *log, + espeak_ng_ERROR_CONTEXT *context); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_CompilePhonemeData(long rate, + FILE *log, + espeak_ng_ERROR_CONTEXT *context); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_CompileIntonation(FILE *log, + espeak_ng_ERROR_CONTEXT *context); + + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_CompileIntonationPath(const char *source_path, + const char *destination_path, + FILE *log, + espeak_ng_ERROR_CONTEXT *context); + +/* eSpeak NG 1.49.1 */ + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_CompilePhonemeDataPath(long rate, + const char *source_path, + const char *destination_path, + FILE *log, + espeak_ng_ERROR_CONTEXT *context); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_SetOutputHooks(espeak_ng_OUTPUT_HOOKS* hooks); +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_SetConstF0(int f0); + +ESPEAK_NG_API espeak_ng_STATUS +espeak_ng_SetRandSeed(long seed); + + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/vall_e.cpp/include/espeak-ng/speak_lib.h b/vall_e.cpp/include/espeak-ng/speak_lib.h new file mode 100644 index 0000000..9c0e273 --- /dev/null +++ b/vall_e.cpp/include/espeak-ng/speak_lib.h @@ -0,0 +1,709 @@ +#ifndef SPEAK_LIB_H +#define SPEAK_LIB_H +/*************************************************************************** + * Copyright (C) 2005 to 2012 by Jonathan Duddington * + * email: jonsd@users.sourceforge.net * + * * + * This program is free software; you can redistribute it and/or modify * + * it under the terms of the GNU 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 General Public License for more details. * + * * + * You should have received a copy of the GNU General Public License * + * along with this program; if not, see: * + * . * + ***************************************************************************/ + + +/*************************************************************/ +/* This is the header file for the library version of espeak */ +/* */ +/*************************************************************/ + +#include +#include + +#if defined(_WIN32) || defined(_WIN64) +#ifdef LIBESPEAK_NG_EXPORT +#define ESPEAK_API __declspec(dllexport) +#else +#define ESPEAK_API __declspec(dllimport) +#endif +#else +#define ESPEAK_API +#endif + +#define ESPEAK_API_REVISION 12 +/* +Revision 2 + Added parameter "options" to eSpeakInitialize() + +Revision 3 + Added espeakWORDGAP to espeak_PARAMETER + +Revision 4 + Added flags parameter to espeak_CompileDictionary() + +Revision 5 + Added espeakCHARS_16BIT + +Revision 6 + Added macros: espeakRATE_MINIMUM, espeakRATE_MAXIMUM, espeakRATE_NORMAL + +Revision 7 24.Dec.2011 + Changed espeak_EVENT structure to add id.string[] for phoneme mnemonics. + Added espeakINITIALIZE_PHONEME_IPA option for espeak_Initialize() to report phonemes as IPA names. + +Revision 8 26.Apr.2013 + Added function espeak_TextToPhonemes(). + +Revision 9 30.May.2013 + Changed function espeak_TextToPhonemes(). + +Revision 10 29.Aug.2014 + Changed phonememode parameter to espeak_TextToPhonemes() and espeak_SetPhonemeTrace + +Revision 11 (espeak-ng) + Made ESPEAK_API import/export symbols correctly on Windows. + +Revision 12 (espeak-ng) + Exposed espeak_SetPhonemeCallback. This is available in eSpeak, but was not exposed in this header. + +*/ + /********************/ + /* Initialization */ + /********************/ + +// values for 'value' in espeak_SetParameter(espeakRATE, value, 0), nominally in words-per-minute +#define espeakRATE_MINIMUM 80 +#define espeakRATE_MAXIMUM 450 +#define espeakRATE_NORMAL 175 + + +typedef enum { + espeakEVENT_LIST_TERMINATED = 0, // Retrieval mode: terminates the event list. + espeakEVENT_WORD = 1, // Start of word + espeakEVENT_SENTENCE = 2, // Start of sentence + espeakEVENT_MARK = 3, // Mark + espeakEVENT_PLAY = 4, // Audio element + espeakEVENT_END = 5, // End of sentence or clause + espeakEVENT_MSG_TERMINATED = 6, // End of message + espeakEVENT_PHONEME = 7, // Phoneme, if enabled in espeak_Initialize() + espeakEVENT_SAMPLERATE = 8 // Set sample rate +} espeak_EVENT_TYPE; + + + +typedef struct { + espeak_EVENT_TYPE type; + unsigned int unique_identifier; // message identifier (or 0 for key or character) + int text_position; // the number of characters from the start of the text + int length; // word length, in characters (for espeakEVENT_WORD) + int audio_position; // the time in mS within the generated speech output data + int sample; // sample id (internal use) + void* user_data; // pointer supplied by the calling program + union { + int number; // used for WORD and SENTENCE events. + const char *name; // used for MARK and PLAY events. UTF8 string + char string[8]; // used for phoneme names (UTF8). Terminated by a zero byte unless the name needs the full 8 bytes. + } id; +} espeak_EVENT; +/* + When a message is supplied to espeak_synth, the request is buffered and espeak_synth returns. When the message is really processed, the callback function will be repetedly called. + + + In RETRIEVAL mode, the callback function supplies to the calling program the audio data and an event list terminated by 0 (LIST_TERMINATED). + + In PLAYBACK mode, the callback function is called as soon as an event happens. + + For example suppose that the following message is supplied to espeak_Synth: + "hello, hello." + + + * Once processed in RETRIEVAL mode, it could lead to 3 calls of the callback function : + + ** Block 1: +