diff options
Diffstat (limited to 'thirdparty/misc')
| -rw-r--r-- | thirdparty/misc/patches/qoa-min-fix.patch | 155 | ||||
| -rw-r--r-- | thirdparty/misc/qoa.h | 728 |
2 files changed, 883 insertions, 0 deletions
diff --git a/thirdparty/misc/patches/qoa-min-fix.patch b/thirdparty/misc/patches/qoa-min-fix.patch new file mode 100644 index 0000000000..1043d8bbe7 --- /dev/null +++ b/thirdparty/misc/patches/qoa-min-fix.patch @@ -0,0 +1,155 @@ +diff --git a/qoa.h b/qoa.h +index aa8fb59434..2dde8df098 100644 +--- a/qoa.h ++++ b/qoa.h +@@ -140,14 +140,14 @@ typedef struct { + #endif + } qoa_desc; + +-unsigned int qoa_encode_header(qoa_desc *qoa, unsigned char *bytes); +-unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned int frame_len, unsigned char *bytes); +-void *qoa_encode(const short *sample_data, qoa_desc *qoa, unsigned int *out_len); ++inline unsigned int qoa_encode_header(qoa_desc *qoa, unsigned char *bytes); ++inline unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned int frame_len, unsigned char *bytes); ++inline void *qoa_encode(const short *sample_data, qoa_desc *qoa, unsigned int *out_len); + +-unsigned int qoa_max_frame_size(qoa_desc *qoa); +-unsigned int qoa_decode_header(const unsigned char *bytes, int size, qoa_desc *qoa); +-unsigned int qoa_decode_frame(const unsigned char *bytes, unsigned int size, qoa_desc *qoa, short *sample_data, unsigned int *frame_len); +-short *qoa_decode(const unsigned char *bytes, int size, qoa_desc *file); ++inline unsigned int qoa_max_frame_size(qoa_desc *qoa); ++inline unsigned int qoa_decode_header(const unsigned char *bytes, int size, qoa_desc *qoa); ++inline unsigned int qoa_decode_frame(const unsigned char *bytes, unsigned int size, qoa_desc *qoa, short *sample_data, unsigned int *frame_len); ++inline short *qoa_decode(const unsigned char *bytes, int size, qoa_desc *file); + + #ifndef QOA_NO_STDIO + +@@ -366,7 +366,7 @@ unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned + ), bytes, &p); + + +- for (int c = 0; c < channels; c++) { ++ for (unsigned int c = 0; c < channels; c++) { + /* Write the current LMS state */ + qoa_uint64_t weights = 0; + qoa_uint64_t history = 0; +@@ -380,9 +380,9 @@ unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned + + /* We encode all samples with the channels interleaved on a slice level. + E.g. for stereo: (ch-0, slice 0), (ch 1, slice 0), (ch 0, slice 1), ...*/ +- for (int sample_index = 0; sample_index < frame_len; sample_index += QOA_SLICE_LEN) { ++ for (unsigned int sample_index = 0; sample_index < frame_len; sample_index += QOA_SLICE_LEN) { + +- for (int c = 0; c < channels; c++) { ++ for (unsigned int c = 0; c < channels; c++) { + int slice_len = qoa_clamp(QOA_SLICE_LEN, 0, frame_len - sample_index); + int slice_start = sample_index * channels + c; + int slice_end = (sample_index + slice_len) * channels + c; +@@ -391,10 +391,9 @@ unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned + 16 scalefactors, encode all samples for the current slice and + meassure the total squared error. */ + qoa_uint64_t best_rank = -1; +- qoa_uint64_t best_error = -1; +- qoa_uint64_t best_slice; +- qoa_lms_t best_lms; +- int best_scalefactor; ++ qoa_uint64_t best_slice = -1; ++ qoa_lms_t best_lms = {{-1, -1, -1, -1}, {-1, -1, -1, -1}}; ++ int best_scalefactor = -1; + + for (int sfi = 0; sfi < 16; sfi++) { + /* There is a strong correlation between the scalefactors of +@@ -408,7 +407,6 @@ unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned + qoa_lms_t lms = qoa->lms[c]; + qoa_uint64_t slice = scalefactor; + qoa_uint64_t current_rank = 0; +- qoa_uint64_t current_error = 0; + + for (int si = slice_start; si < slice_end; si += channels) { + int sample = sample_data[si]; +@@ -438,7 +436,6 @@ unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned + qoa_uint64_t error_sq = error * error; + + current_rank += error_sq + weights_penalty * weights_penalty; +- current_error += error_sq; + if (current_rank > best_rank) { + break; + } +@@ -449,7 +446,6 @@ unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned + + if (current_rank < best_rank) { + best_rank = current_rank; +- best_error = current_error; + best_slice = slice; + best_lms = lms; + best_scalefactor = scalefactor; +@@ -492,9 +488,9 @@ void *qoa_encode(const short *sample_data, qoa_desc *qoa, unsigned int *out_len) + num_frames * QOA_LMS_LEN * 4 * qoa->channels + /* 4 * 4 bytes lms state per channel */ + num_slices * 8 * qoa->channels; /* 8 byte slices */ + +- unsigned char *bytes = QOA_MALLOC(encoded_size); ++ unsigned char *bytes = (unsigned char *)QOA_MALLOC(encoded_size); + +- for (int c = 0; c < qoa->channels; c++) { ++ for (unsigned int c = 0; c < qoa->channels; c++) { + /* Set the initial LMS weights to {0, 0, -1, 2}. This helps with the + prediction of the first few ms of a file. */ + qoa->lms[c].weights[0] = 0; +@@ -517,7 +513,7 @@ void *qoa_encode(const short *sample_data, qoa_desc *qoa, unsigned int *out_len) + #endif + + int frame_len = QOA_FRAME_LEN; +- for (int sample_index = 0; sample_index < qoa->samples; sample_index += frame_len) { ++ for (unsigned int sample_index = 0; sample_index < qoa->samples; sample_index += frame_len) { + frame_len = qoa_clamp(QOA_FRAME_LEN, 0, qoa->samples - sample_index); + const short *frame_samples = sample_data + sample_index * qoa->channels; + unsigned int frame_size = qoa_encode_frame(frame_samples, qoa, frame_len, bytes + p); +@@ -580,14 +576,14 @@ unsigned int qoa_decode_frame(const unsigned char *bytes, unsigned int size, qoa + + /* Read and verify the frame header */ + qoa_uint64_t frame_header = qoa_read_u64(bytes, &p); +- int channels = (frame_header >> 56) & 0x0000ff; +- int samplerate = (frame_header >> 32) & 0xffffff; +- int samples = (frame_header >> 16) & 0x00ffff; +- int frame_size = (frame_header ) & 0x00ffff; ++ unsigned int channels = (frame_header >> 56) & 0x0000ff; ++ unsigned int samplerate = (frame_header >> 32) & 0xffffff; ++ unsigned int samples = (frame_header >> 16) & 0x00ffff; ++ unsigned int frame_size = (frame_header ) & 0x00ffff; + + int data_size = frame_size - 8 - QOA_LMS_LEN * 4 * channels; + int num_slices = data_size / 8; +- int max_total_samples = num_slices * QOA_SLICE_LEN; ++ unsigned int max_total_samples = num_slices * QOA_SLICE_LEN; + + if ( + channels != qoa->channels || +@@ -600,7 +596,7 @@ unsigned int qoa_decode_frame(const unsigned char *bytes, unsigned int size, qoa + + + /* Read the LMS state: 4 x 2 bytes history, 4 x 2 bytes weights per channel */ +- for (int c = 0; c < channels; c++) { ++ for (unsigned int c = 0; c < channels; c++) { + qoa_uint64_t history = qoa_read_u64(bytes, &p); + qoa_uint64_t weights = qoa_read_u64(bytes, &p); + for (int i = 0; i < QOA_LMS_LEN; i++) { +@@ -613,8 +609,8 @@ unsigned int qoa_decode_frame(const unsigned char *bytes, unsigned int size, qoa + + + /* Decode all slices for all channels in this frame */ +- for (int sample_index = 0; sample_index < samples; sample_index += QOA_SLICE_LEN) { +- for (int c = 0; c < channels; c++) { ++ for (unsigned int sample_index = 0; sample_index < samples; sample_index += QOA_SLICE_LEN) { ++ for (unsigned int c = 0; c < channels; c++) { + qoa_uint64_t slice = qoa_read_u64(bytes, &p); + + int scalefactor = (slice >> 60) & 0xf; +@@ -647,7 +643,7 @@ short *qoa_decode(const unsigned char *bytes, int size, qoa_desc *qoa) { + + /* Calculate the required size of the sample buffer and allocate */ + int total_samples = qoa->samples * qoa->channels; +- short *sample_data = QOA_MALLOC(total_samples * sizeof(short)); ++ short *sample_data = (short *)QOA_MALLOC(total_samples * sizeof(short)); + + unsigned int sample_index = 0; + unsigned int frame_len; diff --git a/thirdparty/misc/qoa.h b/thirdparty/misc/qoa.h new file mode 100644 index 0000000000..2dde8df098 --- /dev/null +++ b/thirdparty/misc/qoa.h @@ -0,0 +1,728 @@ +/* + +Copyright (c) 2023, Dominic Szablewski - https://phoboslab.org +SPDX-License-Identifier: MIT + +QOA - The "Quite OK Audio" format for fast, lossy audio compression + + +-- Data Format + +QOA encodes pulse-code modulated (PCM) audio data with up to 255 channels, +sample rates from 1 up to 16777215 hertz and a bit depth of 16 bits. + +The compression method employed in QOA is lossy; it discards some information +from the uncompressed PCM data. For many types of audio signals this compression +is "transparent", i.e. the difference from the original file is often not +audible. + +QOA encodes 20 samples of 16 bit PCM data into slices of 64 bits. A single +sample therefore requires 3.2 bits of storage space, resulting in a 5x +compression (16 / 3.2). + +A QOA file consists of an 8 byte file header, followed by a number of frames. +Each frame contains an 8 byte frame header, the current 16 byte en-/decoder +state per channel and 256 slices per channel. Each slice is 8 bytes wide and +encodes 20 samples of audio data. + +All values, including the slices, are big endian. The file layout is as follows: + +struct { + struct { + char magic[4]; // magic bytes "qoaf" + uint32_t samples; // samples per channel in this file + } file_header; + + struct { + struct { + uint8_t num_channels; // no. of channels + uint24_t samplerate; // samplerate in hz + uint16_t fsamples; // samples per channel in this frame + uint16_t fsize; // frame size (includes this header) + } frame_header; + + struct { + int16_t history[4]; // most recent last + int16_t weights[4]; // most recent last + } lms_state[num_channels]; + + qoa_slice_t slices[256][num_channels]; + + } frames[ceil(samples / (256 * 20))]; +} qoa_file_t; + +Each `qoa_slice_t` contains a quantized scalefactor `sf_quant` and 20 quantized +residuals `qrNN`: + +.- QOA_SLICE -- 64 bits, 20 samples --------------------------/ /------------. +| Byte[0] | Byte[1] | Byte[2] \ \ Byte[7] | +| 7 6 5 4 3 2 1 0 | 7 6 5 4 3 2 1 0 | 7 6 5 / / 2 1 0 | +|------------+--------+--------+--------+---------+---------+-\ \--+---------| +| sf_quant | qr00 | qr01 | qr02 | qr03 | qr04 | / / | qr19 | +`-------------------------------------------------------------\ \------------` + +Each frame except the last must contain exactly 256 slices per channel. The last +frame may contain between 1 .. 256 (inclusive) slices per channel. The last +slice (for each channel) in the last frame may contain less than 20 samples; the +slice still must be 8 bytes wide, with the unused samples zeroed out. + +Channels are interleaved per slice. E.g. for 2 channel stereo: +slice[0] = L, slice[1] = R, slice[2] = L, slice[3] = R ... + +A valid QOA file or stream must have at least one frame. Each frame must contain +at least one channel and one sample with a samplerate between 1 .. 16777215 +(inclusive). + +If the total number of samples is not known by the encoder, the samples in the +file header may be set to 0x00000000 to indicate that the encoder is +"streaming". In a streaming context, the samplerate and number of channels may +differ from frame to frame. For static files (those with samples set to a +non-zero value), each frame must have the same number of channels and same +samplerate. + +Note that this implementation of QOA only handles files with a known total +number of samples. + +A decoder should support at least 8 channels. The channel layout for channel +counts 1 .. 8 is: + + 1. Mono + 2. L, R + 3. L, R, C + 4. FL, FR, B/SL, B/SR + 5. FL, FR, C, B/SL, B/SR + 6. FL, FR, C, LFE, B/SL, B/SR + 7. FL, FR, C, LFE, B, SL, SR + 8. FL, FR, C, LFE, BL, BR, SL, SR + +QOA predicts each audio sample based on the previously decoded ones using a +"Sign-Sign Least Mean Squares Filter" (LMS). This prediction plus the +dequantized residual forms the final output sample. + +*/ + + + +/* ----------------------------------------------------------------------------- + Header - Public functions */ + +#ifndef QOA_H +#define QOA_H + +#ifdef __cplusplus +extern "C" { +#endif + +#define QOA_MIN_FILESIZE 16 +#define QOA_MAX_CHANNELS 8 + +#define QOA_SLICE_LEN 20 +#define QOA_SLICES_PER_FRAME 256 +#define QOA_FRAME_LEN (QOA_SLICES_PER_FRAME * QOA_SLICE_LEN) +#define QOA_LMS_LEN 4 +#define QOA_MAGIC 0x716f6166 /* 'qoaf' */ + +#define QOA_FRAME_SIZE(channels, slices) \ + (8 + QOA_LMS_LEN * 4 * channels + 8 * slices * channels) + +typedef struct { + int history[QOA_LMS_LEN]; + int weights[QOA_LMS_LEN]; +} qoa_lms_t; + +typedef struct { + unsigned int channels; + unsigned int samplerate; + unsigned int samples; + qoa_lms_t lms[QOA_MAX_CHANNELS]; + #ifdef QOA_RECORD_TOTAL_ERROR + double error; + #endif +} qoa_desc; + +inline unsigned int qoa_encode_header(qoa_desc *qoa, unsigned char *bytes); +inline unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned int frame_len, unsigned char *bytes); +inline void *qoa_encode(const short *sample_data, qoa_desc *qoa, unsigned int *out_len); + +inline unsigned int qoa_max_frame_size(qoa_desc *qoa); +inline unsigned int qoa_decode_header(const unsigned char *bytes, int size, qoa_desc *qoa); +inline unsigned int qoa_decode_frame(const unsigned char *bytes, unsigned int size, qoa_desc *qoa, short *sample_data, unsigned int *frame_len); +inline short *qoa_decode(const unsigned char *bytes, int size, qoa_desc *file); + +#ifndef QOA_NO_STDIO + +int qoa_write(const char *filename, const short *sample_data, qoa_desc *qoa); +void *qoa_read(const char *filename, qoa_desc *qoa); + +#endif /* QOA_NO_STDIO */ + + +#ifdef __cplusplus +} +#endif +#endif /* QOA_H */ + + +/* ----------------------------------------------------------------------------- + Implementation */ + +#ifdef QOA_IMPLEMENTATION +#include <stdlib.h> + +#ifndef QOA_MALLOC + #define QOA_MALLOC(sz) malloc(sz) + #define QOA_FREE(p) free(p) +#endif + +typedef unsigned long long qoa_uint64_t; + + +/* The quant_tab provides an index into the dequant_tab for residuals in the +range of -8 .. 8. It maps this range to just 3bits and becomes less accurate at +the higher end. Note that the residual zero is identical to the lowest positive +value. This is mostly fine, since the qoa_div() function always rounds away +from zero. */ + +static const int qoa_quant_tab[17] = { + 7, 7, 7, 5, 5, 3, 3, 1, /* -8..-1 */ + 0, /* 0 */ + 0, 2, 2, 4, 4, 6, 6, 6 /* 1.. 8 */ +}; + + +/* We have 16 different scalefactors. Like the quantized residuals these become +less accurate at the higher end. In theory, the highest scalefactor that we +would need to encode the highest 16bit residual is (2**16)/8 = 8192. However we +rely on the LMS filter to predict samples accurately enough that a maximum +residual of one quarter of the 16 bit range is sufficient. I.e. with the +scalefactor 2048 times the quant range of 8 we can encode residuals up to 2**14. + +The scalefactor values are computed as: +scalefactor_tab[s] <- round(pow(s + 1, 2.75)) */ + +static const int qoa_scalefactor_tab[16] = { + 1, 7, 21, 45, 84, 138, 211, 304, 421, 562, 731, 928, 1157, 1419, 1715, 2048 +}; + + +/* The reciprocal_tab maps each of the 16 scalefactors to their rounded +reciprocals 1/scalefactor. This allows us to calculate the scaled residuals in +the encoder with just one multiplication instead of an expensive division. We +do this in .16 fixed point with integers, instead of floats. + +The reciprocal_tab is computed as: +reciprocal_tab[s] <- ((1<<16) + scalefactor_tab[s] - 1) / scalefactor_tab[s] */ + +static const int qoa_reciprocal_tab[16] = { + 65536, 9363, 3121, 1457, 781, 475, 311, 216, 156, 117, 90, 71, 57, 47, 39, 32 +}; + + +/* The dequant_tab maps each of the scalefactors and quantized residuals to +their unscaled & dequantized version. + +Since qoa_div rounds away from the zero, the smallest entries are mapped to 3/4 +instead of 1. The dequant_tab assumes the following dequantized values for each +of the quant_tab indices and is computed as: +float dqt[8] = {0.75, -0.75, 2.5, -2.5, 4.5, -4.5, 7, -7}; +dequant_tab[s][q] <- round_ties_away_from_zero(scalefactor_tab[s] * dqt[q]) + +The rounding employed here is "to nearest, ties away from zero", i.e. positive +and negative values are treated symmetrically. +*/ + +static const int qoa_dequant_tab[16][8] = { + { 1, -1, 3, -3, 5, -5, 7, -7}, + { 5, -5, 18, -18, 32, -32, 49, -49}, + { 16, -16, 53, -53, 95, -95, 147, -147}, + { 34, -34, 113, -113, 203, -203, 315, -315}, + { 63, -63, 210, -210, 378, -378, 588, -588}, + { 104, -104, 345, -345, 621, -621, 966, -966}, + { 158, -158, 528, -528, 950, -950, 1477, -1477}, + { 228, -228, 760, -760, 1368, -1368, 2128, -2128}, + { 316, -316, 1053, -1053, 1895, -1895, 2947, -2947}, + { 422, -422, 1405, -1405, 2529, -2529, 3934, -3934}, + { 548, -548, 1828, -1828, 3290, -3290, 5117, -5117}, + { 696, -696, 2320, -2320, 4176, -4176, 6496, -6496}, + { 868, -868, 2893, -2893, 5207, -5207, 8099, -8099}, + {1064, -1064, 3548, -3548, 6386, -6386, 9933, -9933}, + {1286, -1286, 4288, -4288, 7718, -7718, 12005, -12005}, + {1536, -1536, 5120, -5120, 9216, -9216, 14336, -14336}, +}; + + +/* The Least Mean Squares Filter is the heart of QOA. It predicts the next +sample based on the previous 4 reconstructed samples. It does so by continuously +adjusting 4 weights based on the residual of the previous prediction. + +The next sample is predicted as the sum of (weight[i] * history[i]). + +The adjustment of the weights is done with a "Sign-Sign-LMS" that adds or +subtracts the residual to each weight, based on the corresponding sample from +the history. This, surprisingly, is sufficient to get worthwhile predictions. + +This is all done with fixed point integers. Hence the right-shifts when updating +the weights and calculating the prediction. */ + +static int qoa_lms_predict(qoa_lms_t *lms) { + int prediction = 0; + for (int i = 0; i < QOA_LMS_LEN; i++) { + prediction += lms->weights[i] * lms->history[i]; + } + return prediction >> 13; +} + +static void qoa_lms_update(qoa_lms_t *lms, int sample, int residual) { + int delta = residual >> 4; + for (int i = 0; i < QOA_LMS_LEN; i++) { + lms->weights[i] += lms->history[i] < 0 ? -delta : delta; + } + + for (int i = 0; i < QOA_LMS_LEN-1; i++) { + lms->history[i] = lms->history[i+1]; + } + lms->history[QOA_LMS_LEN-1] = sample; +} + + +/* qoa_div() implements a rounding division, but avoids rounding to zero for +small numbers. E.g. 0.1 will be rounded to 1. Note that 0 itself still +returns as 0, which is handled in the qoa_quant_tab[]. +qoa_div() takes an index into the .16 fixed point qoa_reciprocal_tab as an +argument, so it can do the division with a cheaper integer multiplication. */ + +static inline int qoa_div(int v, int scalefactor) { + int reciprocal = qoa_reciprocal_tab[scalefactor]; + int n = (v * reciprocal + (1 << 15)) >> 16; + n = n + ((v > 0) - (v < 0)) - ((n > 0) - (n < 0)); /* round away from 0 */ + return n; +} + +static inline int qoa_clamp(int v, int min, int max) { + if (v < min) { return min; } + if (v > max) { return max; } + return v; +} + +/* This specialized clamp function for the signed 16 bit range improves decode +performance quite a bit. The extra if() statement works nicely with the CPUs +branch prediction as this branch is rarely taken. */ + +static inline int qoa_clamp_s16(int v) { + if ((unsigned int)(v + 32768) > 65535) { + if (v < -32768) { return -32768; } + if (v > 32767) { return 32767; } + } + return v; +} + +static inline qoa_uint64_t qoa_read_u64(const unsigned char *bytes, unsigned int *p) { + bytes += *p; + *p += 8; + return + ((qoa_uint64_t)(bytes[0]) << 56) | ((qoa_uint64_t)(bytes[1]) << 48) | + ((qoa_uint64_t)(bytes[2]) << 40) | ((qoa_uint64_t)(bytes[3]) << 32) | + ((qoa_uint64_t)(bytes[4]) << 24) | ((qoa_uint64_t)(bytes[5]) << 16) | + ((qoa_uint64_t)(bytes[6]) << 8) | ((qoa_uint64_t)(bytes[7]) << 0); +} + +static inline void qoa_write_u64(qoa_uint64_t v, unsigned char *bytes, unsigned int *p) { + bytes += *p; + *p += 8; + bytes[0] = (v >> 56) & 0xff; + bytes[1] = (v >> 48) & 0xff; + bytes[2] = (v >> 40) & 0xff; + bytes[3] = (v >> 32) & 0xff; + bytes[4] = (v >> 24) & 0xff; + bytes[5] = (v >> 16) & 0xff; + bytes[6] = (v >> 8) & 0xff; + bytes[7] = (v >> 0) & 0xff; +} + + +/* ----------------------------------------------------------------------------- + Encoder */ + +unsigned int qoa_encode_header(qoa_desc *qoa, unsigned char *bytes) { + unsigned int p = 0; + qoa_write_u64(((qoa_uint64_t)QOA_MAGIC << 32) | qoa->samples, bytes, &p); + return p; +} + +unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned int frame_len, unsigned char *bytes) { + unsigned int channels = qoa->channels; + + unsigned int p = 0; + unsigned int slices = (frame_len + QOA_SLICE_LEN - 1) / QOA_SLICE_LEN; + unsigned int frame_size = QOA_FRAME_SIZE(channels, slices); + int prev_scalefactor[QOA_MAX_CHANNELS] = {0}; + + /* Write the frame header */ + qoa_write_u64(( + (qoa_uint64_t)qoa->channels << 56 | + (qoa_uint64_t)qoa->samplerate << 32 | + (qoa_uint64_t)frame_len << 16 | + (qoa_uint64_t)frame_size + ), bytes, &p); + + + for (unsigned int c = 0; c < channels; c++) { + /* Write the current LMS state */ + qoa_uint64_t weights = 0; + qoa_uint64_t history = 0; + for (int i = 0; i < QOA_LMS_LEN; i++) { + history = (history << 16) | (qoa->lms[c].history[i] & 0xffff); + weights = (weights << 16) | (qoa->lms[c].weights[i] & 0xffff); + } + qoa_write_u64(history, bytes, &p); + qoa_write_u64(weights, bytes, &p); + } + + /* We encode all samples with the channels interleaved on a slice level. + E.g. for stereo: (ch-0, slice 0), (ch 1, slice 0), (ch 0, slice 1), ...*/ + for (unsigned int sample_index = 0; sample_index < frame_len; sample_index += QOA_SLICE_LEN) { + + for (unsigned int c = 0; c < channels; c++) { + int slice_len = qoa_clamp(QOA_SLICE_LEN, 0, frame_len - sample_index); + int slice_start = sample_index * channels + c; + int slice_end = (sample_index + slice_len) * channels + c; + + /* Brute for search for the best scalefactor. Just go through all + 16 scalefactors, encode all samples for the current slice and + meassure the total squared error. */ + qoa_uint64_t best_rank = -1; + qoa_uint64_t best_slice = -1; + qoa_lms_t best_lms = {{-1, -1, -1, -1}, {-1, -1, -1, -1}}; + int best_scalefactor = -1; + + for (int sfi = 0; sfi < 16; sfi++) { + /* There is a strong correlation between the scalefactors of + neighboring slices. As an optimization, start testing + the best scalefactor of the previous slice first. */ + int scalefactor = (sfi + prev_scalefactor[c]) % 16; + + /* We have to reset the LMS state to the last known good one + before trying each scalefactor, as each pass updates the LMS + state when encoding. */ + qoa_lms_t lms = qoa->lms[c]; + qoa_uint64_t slice = scalefactor; + qoa_uint64_t current_rank = 0; + + for (int si = slice_start; si < slice_end; si += channels) { + int sample = sample_data[si]; + int predicted = qoa_lms_predict(&lms); + + int residual = sample - predicted; + int scaled = qoa_div(residual, scalefactor); + int clamped = qoa_clamp(scaled, -8, 8); + int quantized = qoa_quant_tab[clamped + 8]; + int dequantized = qoa_dequant_tab[scalefactor][quantized]; + int reconstructed = qoa_clamp_s16(predicted + dequantized); + + + /* If the weights have grown too large, we introduce a penalty + here. This prevents pops/clicks in certain problem cases */ + int weights_penalty = (( + lms.weights[0] * lms.weights[0] + + lms.weights[1] * lms.weights[1] + + lms.weights[2] * lms.weights[2] + + lms.weights[3] * lms.weights[3] + ) >> 18) - 0x8ff; + if (weights_penalty < 0) { + weights_penalty = 0; + } + + long long error = (sample - reconstructed); + qoa_uint64_t error_sq = error * error; + + current_rank += error_sq + weights_penalty * weights_penalty; + if (current_rank > best_rank) { + break; + } + + qoa_lms_update(&lms, reconstructed, dequantized); + slice = (slice << 3) | quantized; + } + + if (current_rank < best_rank) { + best_rank = current_rank; + best_slice = slice; + best_lms = lms; + best_scalefactor = scalefactor; + } + } + + prev_scalefactor[c] = best_scalefactor; + + qoa->lms[c] = best_lms; + #ifdef QOA_RECORD_TOTAL_ERROR + qoa->error += best_error; + #endif + + /* If this slice was shorter than QOA_SLICE_LEN, we have to left- + shift all encoded data, to ensure the rightmost bits are the empty + ones. This should only happen in the last frame of a file as all + slices are completely filled otherwise. */ + best_slice <<= (QOA_SLICE_LEN - slice_len) * 3; + qoa_write_u64(best_slice, bytes, &p); + } + } + + return p; +} + +void *qoa_encode(const short *sample_data, qoa_desc *qoa, unsigned int *out_len) { + if ( + qoa->samples == 0 || + qoa->samplerate == 0 || qoa->samplerate > 0xffffff || + qoa->channels == 0 || qoa->channels > QOA_MAX_CHANNELS + ) { + return NULL; + } + + /* Calculate the encoded size and allocate */ + unsigned int num_frames = (qoa->samples + QOA_FRAME_LEN-1) / QOA_FRAME_LEN; + unsigned int num_slices = (qoa->samples + QOA_SLICE_LEN-1) / QOA_SLICE_LEN; + unsigned int encoded_size = 8 + /* 8 byte file header */ + num_frames * 8 + /* 8 byte frame headers */ + num_frames * QOA_LMS_LEN * 4 * qoa->channels + /* 4 * 4 bytes lms state per channel */ + num_slices * 8 * qoa->channels; /* 8 byte slices */ + + unsigned char *bytes = (unsigned char *)QOA_MALLOC(encoded_size); + + for (unsigned int c = 0; c < qoa->channels; c++) { + /* Set the initial LMS weights to {0, 0, -1, 2}. This helps with the + prediction of the first few ms of a file. */ + qoa->lms[c].weights[0] = 0; + qoa->lms[c].weights[1] = 0; + qoa->lms[c].weights[2] = -(1<<13); + qoa->lms[c].weights[3] = (1<<14); + + /* Explicitly set the history samples to 0, as we might have some + garbage in there. */ + for (int i = 0; i < QOA_LMS_LEN; i++) { + qoa->lms[c].history[i] = 0; + } + } + + + /* Encode the header and go through all frames */ + unsigned int p = qoa_encode_header(qoa, bytes); + #ifdef QOA_RECORD_TOTAL_ERROR + qoa->error = 0; + #endif + + int frame_len = QOA_FRAME_LEN; + for (unsigned int sample_index = 0; sample_index < qoa->samples; sample_index += frame_len) { + frame_len = qoa_clamp(QOA_FRAME_LEN, 0, qoa->samples - sample_index); + const short *frame_samples = sample_data + sample_index * qoa->channels; + unsigned int frame_size = qoa_encode_frame(frame_samples, qoa, frame_len, bytes + p); + p += frame_size; + } + + *out_len = p; + return bytes; +} + + + +/* ----------------------------------------------------------------------------- + Decoder */ + +unsigned int qoa_max_frame_size(qoa_desc *qoa) { + return QOA_FRAME_SIZE(qoa->channels, QOA_SLICES_PER_FRAME); +} + +unsigned int qoa_decode_header(const unsigned char *bytes, int size, qoa_desc *qoa) { + unsigned int p = 0; + if (size < QOA_MIN_FILESIZE) { + return 0; + } + + + /* Read the file header, verify the magic number ('qoaf') and read the + total number of samples. */ + qoa_uint64_t file_header = qoa_read_u64(bytes, &p); + + if ((file_header >> 32) != QOA_MAGIC) { + return 0; + } + + qoa->samples = file_header & 0xffffffff; + if (!qoa->samples) { + return 0; + } + + /* Peek into the first frame header to get the number of channels and + the samplerate. */ + qoa_uint64_t frame_header = qoa_read_u64(bytes, &p); + qoa->channels = (frame_header >> 56) & 0x0000ff; + qoa->samplerate = (frame_header >> 32) & 0xffffff; + + if (qoa->channels == 0 || qoa->samples == 0 || qoa->samplerate == 0) { + return 0; + } + + return 8; +} + +unsigned int qoa_decode_frame(const unsigned char *bytes, unsigned int size, qoa_desc *qoa, short *sample_data, unsigned int *frame_len) { + unsigned int p = 0; + *frame_len = 0; + + if (size < 8 + QOA_LMS_LEN * 4 * qoa->channels) { + return 0; + } + + /* Read and verify the frame header */ + qoa_uint64_t frame_header = qoa_read_u64(bytes, &p); + unsigned int channels = (frame_header >> 56) & 0x0000ff; + unsigned int samplerate = (frame_header >> 32) & 0xffffff; + unsigned int samples = (frame_header >> 16) & 0x00ffff; + unsigned int frame_size = (frame_header ) & 0x00ffff; + + int data_size = frame_size - 8 - QOA_LMS_LEN * 4 * channels; + int num_slices = data_size / 8; + unsigned int max_total_samples = num_slices * QOA_SLICE_LEN; + + if ( + channels != qoa->channels || + samplerate != qoa->samplerate || + frame_size > size || + samples * channels > max_total_samples + ) { + return 0; + } + + + /* Read the LMS state: 4 x 2 bytes history, 4 x 2 bytes weights per channel */ + for (unsigned int c = 0; c < channels; c++) { + qoa_uint64_t history = qoa_read_u64(bytes, &p); + qoa_uint64_t weights = qoa_read_u64(bytes, &p); + for (int i = 0; i < QOA_LMS_LEN; i++) { + qoa->lms[c].history[i] = ((signed short)(history >> 48)); + history <<= 16; + qoa->lms[c].weights[i] = ((signed short)(weights >> 48)); + weights <<= 16; + } + } + + + /* Decode all slices for all channels in this frame */ + for (unsigned int sample_index = 0; sample_index < samples; sample_index += QOA_SLICE_LEN) { + for (unsigned int c = 0; c < channels; c++) { + qoa_uint64_t slice = qoa_read_u64(bytes, &p); + + int scalefactor = (slice >> 60) & 0xf; + int slice_start = sample_index * channels + c; + int slice_end = qoa_clamp(sample_index + QOA_SLICE_LEN, 0, samples) * channels + c; + + for (int si = slice_start; si < slice_end; si += channels) { + int predicted = qoa_lms_predict(&qoa->lms[c]); + int quantized = (slice >> 57) & 0x7; + int dequantized = qoa_dequant_tab[scalefactor][quantized]; + int reconstructed = qoa_clamp_s16(predicted + dequantized); + + sample_data[si] = reconstructed; + slice <<= 3; + + qoa_lms_update(&qoa->lms[c], reconstructed, dequantized); + } + } + } + + *frame_len = samples; + return p; +} + +short *qoa_decode(const unsigned char *bytes, int size, qoa_desc *qoa) { + unsigned int p = qoa_decode_header(bytes, size, qoa); + if (!p) { + return NULL; + } + + /* Calculate the required size of the sample buffer and allocate */ + int total_samples = qoa->samples * qoa->channels; + short *sample_data = (short *)QOA_MALLOC(total_samples * sizeof(short)); + + unsigned int sample_index = 0; + unsigned int frame_len; + unsigned int frame_size; + + /* Decode all frames */ + do { + short *sample_ptr = sample_data + sample_index * qoa->channels; + frame_size = qoa_decode_frame(bytes + p, size - p, qoa, sample_ptr, &frame_len); + + p += frame_size; + sample_index += frame_len; + } while (frame_size && sample_index < qoa->samples); + + qoa->samples = sample_index; + return sample_data; +} + + + +/* ----------------------------------------------------------------------------- + File read/write convenience functions */ + +#ifndef QOA_NO_STDIO +#include <stdio.h> + +int qoa_write(const char *filename, const short *sample_data, qoa_desc *qoa) { + FILE *f = fopen(filename, "wb"); + unsigned int size; + void *encoded; + + if (!f) { + return 0; + } + + encoded = qoa_encode(sample_data, qoa, &size); + if (!encoded) { + fclose(f); + return 0; + } + + fwrite(encoded, 1, size, f); + fclose(f); + + QOA_FREE(encoded); + return size; +} + +void *qoa_read(const char *filename, qoa_desc *qoa) { + FILE *f = fopen(filename, "rb"); + int size, bytes_read; + void *data; + short *sample_data; + + if (!f) { + return NULL; + } + + fseek(f, 0, SEEK_END); + size = ftell(f); + if (size <= 0) { + fclose(f); + return NULL; + } + fseek(f, 0, SEEK_SET); + + data = QOA_MALLOC(size); + if (!data) { + fclose(f); + return NULL; + } + + bytes_read = fread(data, 1, size, f); + fclose(f); + + sample_data = qoa_decode(data, bytes_read, qoa); + QOA_FREE(data); + return sample_data; +} + +#endif /* QOA_NO_STDIO */ +#endif /* QOA_IMPLEMENTATION */ |