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4396f8fbd3
Move OggVorbis and MP3 loading code to their AudioStream class, matching how it's done for WAV. The duplicate functions in ResourceImporterOggVorbis are now deprecated. Co-authored-by: MaxIsJoe <34368774+MaxIsJoe@users.noreply.github.com>
309 lines
9.3 KiB
C++
309 lines
9.3 KiB
C++
/**************************************************************************/
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/* audio_stream_wav.h */
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/**************************************************************************/
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/* This file is part of: */
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/* GODOT ENGINE */
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/* https://godotengine.org */
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/**************************************************************************/
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/* Copyright (c) 2014-present Godot Engine contributors (see AUTHORS.md). */
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/* Copyright (c) 2007-2014 Juan Linietsky, Ariel Manzur. */
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/* */
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/* Permission is hereby granted, free of charge, to any person obtaining */
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/* a copy of this software and associated documentation files (the */
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/* "Software"), to deal in the Software without restriction, including */
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/* without limitation the rights to use, copy, modify, merge, publish, */
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/* distribute, sublicense, and/or sell copies of the Software, and to */
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/* permit persons to whom the Software is furnished to do so, subject to */
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/* the following conditions: */
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/* */
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/* The above copyright notice and this permission notice shall be */
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/* included in all copies or substantial portions of the Software. */
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/* */
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/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
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/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
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/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. */
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/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
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/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
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/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
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/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
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/**************************************************************************/
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#ifndef AUDIO_STREAM_WAV_H
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#define AUDIO_STREAM_WAV_H
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#include "servers/audio/audio_stream.h"
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#include "thirdparty/misc/qoa.h"
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class AudioStreamWAV;
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class AudioStreamPlaybackWAV : public AudioStreamPlayback {
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GDCLASS(AudioStreamPlaybackWAV, AudioStreamPlayback);
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enum {
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MIX_FRAC_BITS = 13,
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MIX_FRAC_LEN = (1 << MIX_FRAC_BITS),
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MIX_FRAC_MASK = MIX_FRAC_LEN - 1,
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};
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struct IMA_ADPCM_State {
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int16_t step_index = 0;
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int32_t predictor = 0;
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/* values at loop point */
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int16_t loop_step_index = 0;
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int32_t loop_predictor = 0;
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int32_t last_nibble = 0;
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int32_t loop_pos = 0;
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int32_t window_ofs = 0;
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} ima_adpcm[2];
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struct QOA_State {
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qoa_desc desc = {};
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uint32_t data_ofs = 0;
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uint32_t frame_len = 0;
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LocalVector<int16_t> dec;
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uint32_t dec_len = 0;
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int64_t cache_pos = -1;
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int16_t cache[2] = { 0, 0 };
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int16_t cache_r[2] = { 0, 0 };
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} qoa;
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int64_t offset = 0;
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int sign = 1;
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bool active = false;
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friend class AudioStreamWAV;
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Ref<AudioStreamWAV> base;
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template <typename Depth, bool is_stereo, bool is_ima_adpcm, bool is_qoa>
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void do_resample(const Depth *p_src, AudioFrame *p_dst, int64_t &p_offset, int32_t &p_increment, uint32_t p_amount, IMA_ADPCM_State *p_ima_adpcm, QOA_State *p_qoa);
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bool _is_sample = false;
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Ref<AudioSamplePlayback> sample_playback;
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public:
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virtual void start(double p_from_pos = 0.0) override;
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virtual void stop() override;
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virtual bool is_playing() const override;
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virtual int get_loop_count() const override; //times it looped
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virtual double get_playback_position() const override;
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virtual void seek(double p_time) override;
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virtual int mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) override;
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virtual void tag_used_streams() override;
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virtual void set_is_sample(bool p_is_sample) override;
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virtual bool get_is_sample() const override;
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virtual Ref<AudioSamplePlayback> get_sample_playback() const override;
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virtual void set_sample_playback(const Ref<AudioSamplePlayback> &p_playback) override;
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AudioStreamPlaybackWAV();
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~AudioStreamPlaybackWAV();
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};
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class AudioStreamWAV : public AudioStream {
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GDCLASS(AudioStreamWAV, AudioStream);
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RES_BASE_EXTENSION("sample")
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public:
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enum Format {
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FORMAT_8_BITS,
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FORMAT_16_BITS,
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FORMAT_IMA_ADPCM,
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FORMAT_QOA,
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};
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// Keep the ResourceImporterWAV `edit/loop_mode` enum hint in sync with these options.
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enum LoopMode {
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LOOP_DISABLED,
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LOOP_FORWARD,
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LOOP_PINGPONG,
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LOOP_BACKWARD
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};
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private:
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friend class AudioStreamPlaybackWAV;
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enum {
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DATA_PAD = 16 //padding for interpolation
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};
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Format format = FORMAT_8_BITS;
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LoopMode loop_mode = LOOP_DISABLED;
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bool stereo = false;
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int loop_begin = 0;
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int loop_end = 0;
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int mix_rate = 44100;
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LocalVector<uint8_t> data;
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uint32_t data_bytes = 0;
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protected:
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static void _bind_methods();
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public:
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static Ref<AudioStreamWAV> load_from_buffer(const Vector<uint8_t> &p_stream_data, const Dictionary &p_options);
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static Ref<AudioStreamWAV> load_from_file(const String &p_path, const Dictionary &p_options);
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void set_format(Format p_format);
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Format get_format() const;
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void set_loop_mode(LoopMode p_loop_mode);
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LoopMode get_loop_mode() const;
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void set_loop_begin(int p_frame);
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int get_loop_begin() const;
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void set_loop_end(int p_frame);
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int get_loop_end() const;
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void set_mix_rate(int p_hz);
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int get_mix_rate() const;
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void set_stereo(bool p_enable);
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bool is_stereo() const;
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virtual double get_length() const override; //if supported, otherwise return 0
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virtual bool is_monophonic() const override;
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void set_data(const Vector<uint8_t> &p_data);
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Vector<uint8_t> get_data() const;
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Error save_to_wav(const String &p_path);
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virtual Ref<AudioStreamPlayback> instantiate_playback() override;
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virtual String get_stream_name() const override;
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virtual bool can_be_sampled() const override {
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return true;
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}
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virtual Ref<AudioSample> generate_sample() const override;
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static void _compress_ima_adpcm(const Vector<float> &p_data, Vector<uint8_t> &r_dst_data) {
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static const int16_t _ima_adpcm_step_table[89] = {
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7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
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19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
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50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
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130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
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337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
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876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
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2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
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5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
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15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
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};
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static const int8_t _ima_adpcm_index_table[16] = {
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-1, -1, -1, -1, 2, 4, 6, 8,
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-1, -1, -1, -1, 2, 4, 6, 8
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};
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int datalen = p_data.size();
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int datamax = datalen;
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if (datalen & 1) {
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datalen++;
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}
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r_dst_data.resize(datalen / 2 + 4);
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uint8_t *w = r_dst_data.ptrw();
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int i, step_idx = 0, prev = 0;
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uint8_t *out = w;
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const float *in = p_data.ptr();
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// Initial value is zero.
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*(out++) = 0;
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*(out++) = 0;
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// Table index initial value.
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*(out++) = 0;
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// Unused.
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*(out++) = 0;
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for (i = 0; i < datalen; i++) {
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int step, diff, vpdiff, mask;
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uint8_t nibble;
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int16_t xm_sample;
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if (i >= datamax) {
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xm_sample = 0;
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} else {
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xm_sample = CLAMP(in[i] * 32767.0, -32768, 32767);
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}
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diff = (int)xm_sample - prev;
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nibble = 0;
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step = _ima_adpcm_step_table[step_idx];
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vpdiff = step >> 3;
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if (diff < 0) {
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nibble = 8;
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diff = -diff;
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}
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mask = 4;
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while (mask) {
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if (diff >= step) {
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nibble |= mask;
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diff -= step;
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vpdiff += step;
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}
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step >>= 1;
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mask >>= 1;
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}
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if (nibble & 8) {
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prev -= vpdiff;
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} else {
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prev += vpdiff;
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}
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prev = CLAMP(prev, -32768, 32767);
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step_idx += _ima_adpcm_index_table[nibble];
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step_idx = CLAMP(step_idx, 0, 88);
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if (i & 1) {
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*out |= nibble << 4;
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out++;
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} else {
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*out = nibble;
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}
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}
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}
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static void _compress_qoa(const Vector<float> &p_data, Vector<uint8_t> &dst_data, qoa_desc *p_desc) {
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uint32_t frames_len = (p_desc->samples + QOA_FRAME_LEN - 1) / QOA_FRAME_LEN * (QOA_LMS_LEN * 4 * p_desc->channels + 8);
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uint32_t slices_len = (p_desc->samples + QOA_SLICE_LEN - 1) / QOA_SLICE_LEN * 8 * p_desc->channels;
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dst_data.resize(8 + frames_len + slices_len);
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for (uint32_t c = 0; c < p_desc->channels; c++) {
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memset(p_desc->lms[c].history, 0, sizeof(p_desc->lms[c].history));
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memset(p_desc->lms[c].weights, 0, sizeof(p_desc->lms[c].weights));
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p_desc->lms[c].weights[2] = -(1 << 13);
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p_desc->lms[c].weights[3] = (1 << 14);
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}
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LocalVector<int16_t> data16;
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data16.resize(QOA_FRAME_LEN * p_desc->channels);
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uint8_t *dst_ptr = dst_data.ptrw();
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dst_ptr += qoa_encode_header(p_desc, dst_data.ptrw());
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uint32_t frame_len = QOA_FRAME_LEN;
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for (uint32_t s = 0; s < p_desc->samples; s += frame_len) {
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frame_len = MIN(frame_len, p_desc->samples - s);
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for (uint32_t i = 0; i < frame_len * p_desc->channels; i++) {
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data16[i] = CLAMP(p_data[s * p_desc->channels + i] * 32767.0, -32768, 32767);
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}
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dst_ptr += qoa_encode_frame(data16.ptr(), p_desc, frame_len, dst_ptr);
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}
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}
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AudioStreamWAV();
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~AudioStreamWAV();
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};
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VARIANT_ENUM_CAST(AudioStreamWAV::Format)
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VARIANT_ENUM_CAST(AudioStreamWAV::LoopMode)
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#endif // AUDIO_STREAM_WAV_H
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