/**************************************************************************/
/* audio_stream_wav.h */
/**************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
/**************************************************************************/
/* Copyright (c) 2014-present Godot Engine contributors (see AUTHORS.md). */
/* Copyright (c) 2007-2014 Juan Linietsky, Ariel Manzur. */
/* */
/* 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, subject to */
/* the following conditions: */
/* */
/* The above copyright notice and this permission notice shall be */
/* included in all copies or substantial portions of the Software. */
/* */
/* 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. */
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/**************************************************************************/
#pragma once
#include "servers/audio/audio_stream.h"
#include <thirdparty/misc/qoa.h>
class AudioStreamWAV;
class AudioStreamPlaybackWAV : public AudioStreamPlaybackResampled {
GDCLASS(AudioStreamPlaybackWAV, AudioStreamPlaybackResampled);
struct IMA_ADPCM_State {
int16_t step_index = 0;
int32_t predictor = 0;
/* values at loop point */
int16_t loop_step_index = 0;
int32_t loop_predictor = 0;
int32_t last_nibble = 0;
int32_t loop_pos = 0;
int32_t window_ofs = 0;
} ima_adpcm[2];
struct QOA_State {
qoa_desc desc = {};
uint64_t data_ofs = 0;
uint32_t frame_len = 0;
TightLocalVector<int16_t> dec;
uint32_t dec_len = 0;
} qoa;
int64_t offset = 0;
int8_t sign = 1;
bool active = false;
friend class AudioStreamWAV;
Ref<AudioStreamWAV> base;
template <typename Depth, bool is_stereo, bool is_ima_adpcm, bool is_qoa>
void decode_samples(const Depth *p_src, AudioFrame *p_dst, int64_t &p_offset, int8_t &p_increment, uint32_t p_amount, IMA_ADPCM_State *p_ima_adpcm, QOA_State *p_qoa);
bool _is_sample = false;
Ref<AudioSamplePlayback> sample_playback;
protected:
virtual int _mix_internal(AudioFrame *p_buffer, int p_frames) override;
virtual float get_stream_sampling_rate() override;
public:
virtual void start(double p_from_pos = 0.0) override;
virtual void stop() override;
virtual bool is_playing() const override;
virtual int get_loop_count() const override; //times it looped
virtual double get_playback_position() const override;
virtual void seek(double p_time) override;
virtual void tag_used_streams() override;
virtual void set_is_sample(bool p_is_sample) override;
virtual bool get_is_sample() const override;
virtual Ref<AudioSamplePlayback> get_sample_playback() const override;
virtual void set_sample_playback(const Ref<AudioSamplePlayback> &p_playback) override;
};
class AudioStreamWAV : public AudioStream {
GDCLASS(AudioStreamWAV, AudioStream);
RES_BASE_EXTENSION("sample")
public:
enum Format {
FORMAT_8_BITS,
FORMAT_16_BITS,
FORMAT_IMA_ADPCM,
FORMAT_QOA,
};
// Keep the ResourceImporterWAV `edit/loop_mode` enum hint in sync with these options.
enum LoopMode {
LOOP_DISABLED,
LOOP_FORWARD,
LOOP_PINGPONG,
LOOP_BACKWARD
};
private:
friend class AudioStreamPlaybackWAV;
Format format = FORMAT_8_BITS;
LoopMode loop_mode = LOOP_DISABLED;
bool stereo = false;
int64_t loop_begin = 0;
int64_t loop_end = 0;
uint32_t mix_rate = 44100;
TightLocalVector<uint8_t, uint64_t> data;
uint32_t data_bytes = 0;
Dictionary tags;
protected:
static void _bind_methods();
public:
static Ref<AudioStreamWAV> load_from_buffer(const Vector<uint8_t> &p_stream_data, const Dictionary &p_options);
static Ref<AudioStreamWAV> load_from_file(const String &p_path, const Dictionary &p_options);
void set_format(Format p_format);
Format get_format() const;
void set_loop_mode(LoopMode p_loop_mode);
LoopMode get_loop_mode() const;
void set_loop_begin(int p_frame);
int get_loop_begin() const;
void set_loop_end(int p_frame);
int get_loop_end() const;
void set_mix_rate(int p_hz);
int get_mix_rate() const;
void set_stereo(bool p_enable);
bool is_stereo() const;
void set_tags(const Dictionary &p_tags);
virtual Dictionary get_tags() const override;
virtual double get_length() const override; //if supported, otherwise return 0
virtual bool is_monophonic() const override;
void set_data(const Vector<uint8_t> &p_data);
Vector<uint8_t> get_data() const;
Error save_to_wav(const String &p_path);
virtual Ref<AudioStreamPlayback> instantiate_playback() override;
virtual String get_stream_name() const override;
virtual bool can_be_sampled() const override {
return true;
}
virtual Ref<AudioSample> generate_sample() const override;
static void _compress_ima_adpcm(const Vector<float> &p_data, Vector<uint8_t> &r_dst_data) {
static const int16_t _ima_adpcm_step_table[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
};
static const int8_t _ima_adpcm_index_table[16] = {
-1, -1, -1, -1, 2, 4, 6, 8,
-1, -1, -1, -1, 2, 4, 6, 8
};
int datalen = p_data.size();
int datamax = datalen;
if (datalen & 1) {
datalen++;
}
r_dst_data.resize(datalen / 2 + 4);
uint8_t *w = r_dst_data.ptrw();
int i, step_idx = 0, prev = 0;
uint8_t *out = w;
const float *in = p_data.ptr();
// Initial value is zero.
*(out++) = 0;
*(out++) = 0;
// Table index initial value.
*(out++) = 0;
// Unused.
*(out++) = 0;
for (i = 0; i < datalen; i++) {
int step, diff, vpdiff, mask;
uint8_t nibble;
int16_t xm_sample;
if (i >= datamax) {
xm_sample = 0;
} else {
xm_sample = CLAMP(in[i] * 32767.0, -32768, 32767);
}
diff = (int)xm_sample - prev;
nibble = 0;
step = _ima_adpcm_step_table[step_idx];
vpdiff = step >> 3;
if (diff < 0) {
nibble = 8;
diff = -diff;
}
mask = 4;
while (mask) {
if (diff >= step) {
nibble |= mask;
diff -= step;
vpdiff += step;
}
step >>= 1;
mask >>= 1;
}
if (nibble & 8) {
prev -= vpdiff;
} else {
prev += vpdiff;
}
prev = CLAMP(prev, -32768, 32767);
step_idx += _ima_adpcm_index_table[nibble];
step_idx = CLAMP(step_idx, 0, 88);
if (i & 1) {
*out |= nibble << 4;
out++;
} else {
*out = nibble;
}
}
}
static void _compress_qoa(const Vector<float> &p_data, Vector<uint8_t> &dst_data, qoa_desc *p_desc) {
uint32_t frames_len = (p_desc->samples + QOA_FRAME_LEN - 1) / QOA_FRAME_LEN * (QOA_LMS_LEN * 4 * p_desc->channels + 8);
uint32_t slices_len = (p_desc->samples + QOA_SLICE_LEN - 1) / QOA_SLICE_LEN * 8 * p_desc->channels;
dst_data.resize(8 + frames_len + slices_len);
for (uint32_t c = 0; c < p_desc->channels; c++) {
memset(p_desc->lms[c].history, 0, sizeof(p_desc->lms[c].history));
memset(p_desc->lms[c].weights, 0, sizeof(p_desc->lms[c].weights));
p_desc->lms[c].weights[2] = -(1 << 13);
p_desc->lms[c].weights[3] = (1 << 14);
}
TightLocalVector<int16_t> data16;
data16.resize(QOA_FRAME_LEN * p_desc->channels);
uint8_t *dst_ptr = dst_data.ptrw();
dst_ptr += qoa_encode_header(p_desc, dst_data.ptrw());
uint32_t frame_len = QOA_FRAME_LEN;
for (uint32_t s = 0; s < p_desc->samples; s += frame_len) {
frame_len = MIN(frame_len, p_desc->samples - s);
for (uint32_t i = 0; i < frame_len * p_desc->channels; i++) {
data16[i] = CLAMP(p_data[s * p_desc->channels + i] * 32767.0, -32768, 32767);
}
dst_ptr += qoa_encode_frame(data16.ptr(), p_desc, frame_len, dst_ptr);
}
}
};
VARIANT_ENUM_CAST(AudioStreamWAV::Format)
VARIANT_ENUM_CAST(AudioStreamWAV::LoopMode)