summaryrefslogtreecommitdiffstats
path: root/scene/resources/audio_stream_sample.cpp
diff options
context:
space:
mode:
authorDeeJayLSP <djlsplays@gmail.com>2022-07-23 11:34:36 -0300
committerDeeJayLSP <djlsplays@gmail.com>2022-07-28 13:53:36 -0300
commit4889659227221f137da0bd926ddb6cd867bbd632 (patch)
tree71780267922b1974c2bd28c4f96bfa425f2b3c88 /scene/resources/audio_stream_sample.cpp
parent667cef39b4e9f98ed545d95de442319c447f2164 (diff)
downloadredot-engine-4889659227221f137da0bd926ddb6cd867bbd632.tar.gz
Rename AudioStreamSample to a more discoverable name
Diffstat (limited to 'scene/resources/audio_stream_sample.cpp')
-rw-r--r--scene/resources/audio_stream_sample.cpp667
1 files changed, 0 insertions, 667 deletions
diff --git a/scene/resources/audio_stream_sample.cpp b/scene/resources/audio_stream_sample.cpp
deleted file mode 100644
index dcd36284d4..0000000000
--- a/scene/resources/audio_stream_sample.cpp
+++ /dev/null
@@ -1,667 +0,0 @@
-/*************************************************************************/
-/* audio_stream_sample.cpp */
-/*************************************************************************/
-/* This file is part of: */
-/* GODOT ENGINE */
-/* https://godotengine.org */
-/*************************************************************************/
-/* Copyright (c) 2007-2022 Juan Linietsky, Ariel Manzur. */
-/* Copyright (c) 2014-2022 Godot Engine contributors (cf. AUTHORS.md). */
-/* */
-/* 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.*/
-/* 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. */
-/*************************************************************************/
-
-#include "audio_stream_sample.h"
-
-#include "core/io/file_access.h"
-#include "core/io/marshalls.h"
-
-void AudioStreamPlaybackSample::start(float p_from_pos) {
- if (base->format == AudioStreamSample::FORMAT_IMA_ADPCM) {
- //no seeking in IMA_ADPCM
- for (int i = 0; i < 2; i++) {
- ima_adpcm[i].step_index = 0;
- ima_adpcm[i].predictor = 0;
- ima_adpcm[i].loop_step_index = 0;
- ima_adpcm[i].loop_predictor = 0;
- ima_adpcm[i].last_nibble = -1;
- ima_adpcm[i].loop_pos = 0x7FFFFFFF;
- ima_adpcm[i].window_ofs = 0;
- }
-
- offset = 0;
- } else {
- seek(p_from_pos);
- }
-
- sign = 1;
- active = true;
-}
-
-void AudioStreamPlaybackSample::stop() {
- active = false;
-}
-
-bool AudioStreamPlaybackSample::is_playing() const {
- return active;
-}
-
-int AudioStreamPlaybackSample::get_loop_count() const {
- return 0;
-}
-
-float AudioStreamPlaybackSample::get_playback_position() const {
- return float(offset >> MIX_FRAC_BITS) / base->mix_rate;
-}
-
-void AudioStreamPlaybackSample::seek(float p_time) {
- if (base->format == AudioStreamSample::FORMAT_IMA_ADPCM) {
- return; //no seeking in ima-adpcm
- }
-
- float max = base->get_length();
- if (p_time < 0) {
- p_time = 0;
- } else if (p_time >= max) {
- p_time = max - 0.001;
- }
-
- offset = uint64_t(p_time * base->mix_rate) << MIX_FRAC_BITS;
-}
-
-template <class Depth, bool is_stereo, bool is_ima_adpcm>
-void AudioStreamPlaybackSample::do_resample(const Depth *p_src, AudioFrame *p_dst, int64_t &offset, int32_t &increment, uint32_t amount, IMA_ADPCM_State *ima_adpcm) {
- // this function will be compiled branchless by any decent compiler
-
- int32_t final, final_r, next, next_r;
- while (amount) {
- amount--;
- int64_t pos = offset >> MIX_FRAC_BITS;
- if (is_stereo && !is_ima_adpcm) {
- pos <<= 1;
- }
-
- if (is_ima_adpcm) {
- int64_t sample_pos = pos + ima_adpcm[0].window_ofs;
-
- while (sample_pos > ima_adpcm[0].last_nibble) {
- 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
- };
-
- for (int i = 0; i < (is_stereo ? 2 : 1); i++) {
- int16_t nibble, diff, step;
-
- ima_adpcm[i].last_nibble++;
- const uint8_t *src_ptr = (const uint8_t *)base->data;
- src_ptr += AudioStreamSample::DATA_PAD;
-
- uint8_t nbb = src_ptr[(ima_adpcm[i].last_nibble >> 1) * (is_stereo ? 2 : 1) + i];
- nibble = (ima_adpcm[i].last_nibble & 1) ? (nbb >> 4) : (nbb & 0xF);
- step = _ima_adpcm_step_table[ima_adpcm[i].step_index];
-
- ima_adpcm[i].step_index += _ima_adpcm_index_table[nibble];
- if (ima_adpcm[i].step_index < 0) {
- ima_adpcm[i].step_index = 0;
- }
- if (ima_adpcm[i].step_index > 88) {
- ima_adpcm[i].step_index = 88;
- }
-
- diff = step >> 3;
- if (nibble & 1) {
- diff += step >> 2;
- }
- if (nibble & 2) {
- diff += step >> 1;
- }
- if (nibble & 4) {
- diff += step;
- }
- if (nibble & 8) {
- diff = -diff;
- }
-
- ima_adpcm[i].predictor += diff;
- if (ima_adpcm[i].predictor < -0x8000) {
- ima_adpcm[i].predictor = -0x8000;
- } else if (ima_adpcm[i].predictor > 0x7FFF) {
- ima_adpcm[i].predictor = 0x7FFF;
- }
-
- /* store loop if there */
- if (ima_adpcm[i].last_nibble == ima_adpcm[i].loop_pos) {
- ima_adpcm[i].loop_step_index = ima_adpcm[i].step_index;
- ima_adpcm[i].loop_predictor = ima_adpcm[i].predictor;
- }
-
- //printf("%i - %i - pred %i\n",int(ima_adpcm[i].last_nibble),int(nibble),int(ima_adpcm[i].predictor));
- }
- }
-
- final = ima_adpcm[0].predictor;
- if (is_stereo) {
- final_r = ima_adpcm[1].predictor;
- }
-
- } else {
- final = p_src[pos];
- if (is_stereo) {
- final_r = p_src[pos + 1];
- }
-
- if (sizeof(Depth) == 1) { /* conditions will not exist anymore when compiled! */
- final <<= 8;
- if (is_stereo) {
- final_r <<= 8;
- }
- }
-
- if (is_stereo) {
- next = p_src[pos + 2];
- next_r = p_src[pos + 3];
- } else {
- next = p_src[pos + 1];
- }
-
- if (sizeof(Depth) == 1) {
- next <<= 8;
- if (is_stereo) {
- next_r <<= 8;
- }
- }
-
- int32_t frac = int64_t(offset & MIX_FRAC_MASK);
-
- final = final + ((next - final) * frac >> MIX_FRAC_BITS);
- if (is_stereo) {
- final_r = final_r + ((next_r - final_r) * frac >> MIX_FRAC_BITS);
- }
- }
-
- if (!is_stereo) {
- final_r = final; //copy to right channel if stereo
- }
-
- p_dst->l = final / 32767.0;
- p_dst->r = final_r / 32767.0;
- p_dst++;
-
- offset += increment;
- }
-}
-
-int AudioStreamPlaybackSample::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
- if (!base->data || !active) {
- for (int i = 0; i < p_frames; i++) {
- p_buffer[i] = AudioFrame(0, 0);
- }
- return 0;
- }
-
- int len = base->data_bytes;
- switch (base->format) {
- case AudioStreamSample::FORMAT_8_BITS:
- len /= 1;
- break;
- case AudioStreamSample::FORMAT_16_BITS:
- len /= 2;
- break;
- case AudioStreamSample::FORMAT_IMA_ADPCM:
- len *= 2;
- break;
- }
-
- if (base->stereo) {
- len /= 2;
- }
-
- /* some 64-bit fixed point precaches */
-
- int64_t loop_begin_fp = ((int64_t)base->loop_begin << MIX_FRAC_BITS);
- int64_t loop_end_fp = ((int64_t)base->loop_end << MIX_FRAC_BITS);
- int64_t length_fp = ((int64_t)len << MIX_FRAC_BITS);
- int64_t begin_limit = (base->loop_mode != AudioStreamSample::LOOP_DISABLED) ? loop_begin_fp : 0;
- int64_t end_limit = (base->loop_mode != AudioStreamSample::LOOP_DISABLED) ? loop_end_fp : length_fp;
- bool is_stereo = base->stereo;
-
- int32_t todo = p_frames;
-
- if (base->loop_mode == AudioStreamSample::LOOP_BACKWARD) {
- sign = -1;
- }
-
- float base_rate = AudioServer::get_singleton()->get_mix_rate();
- float srate = base->mix_rate;
- srate *= p_rate_scale;
- float playback_speed_scale = AudioServer::get_singleton()->get_playback_speed_scale();
- float fincrement = (srate * playback_speed_scale) / base_rate;
- int32_t increment = int32_t(MAX(fincrement * MIX_FRAC_LEN, 1));
- increment *= sign;
-
- //looping
-
- AudioStreamSample::LoopMode loop_format = base->loop_mode;
- AudioStreamSample::Format format = base->format;
-
- /* audio data */
-
- uint8_t *dataptr = (uint8_t *)base->data;
- const void *data = dataptr + AudioStreamSample::DATA_PAD;
- AudioFrame *dst_buff = p_buffer;
-
- if (format == AudioStreamSample::FORMAT_IMA_ADPCM) {
- if (loop_format != AudioStreamSample::LOOP_DISABLED) {
- ima_adpcm[0].loop_pos = loop_begin_fp >> MIX_FRAC_BITS;
- ima_adpcm[1].loop_pos = loop_begin_fp >> MIX_FRAC_BITS;
- loop_format = AudioStreamSample::LOOP_FORWARD;
- }
- }
-
- while (todo > 0) {
- int64_t limit = 0;
- int32_t target = 0, aux = 0;
-
- /** LOOP CHECKING **/
-
- if (increment < 0) {
- /* going backwards */
-
- if (loop_format != AudioStreamSample::LOOP_DISABLED && offset < loop_begin_fp) {
- /* loopstart reached */
- if (loop_format == AudioStreamSample::LOOP_PINGPONG) {
- /* bounce ping pong */
- offset = loop_begin_fp + (loop_begin_fp - offset);
- increment = -increment;
- sign *= -1;
- } else {
- /* go to loop-end */
- offset = loop_end_fp - (loop_begin_fp - offset);
- }
- } else {
- /* check for sample not reaching beginning */
- if (offset < 0) {
- active = false;
- break;
- }
- }
- } else {
- /* going forward */
- if (loop_format != AudioStreamSample::LOOP_DISABLED && offset >= loop_end_fp) {
- /* loopend reached */
-
- if (loop_format == AudioStreamSample::LOOP_PINGPONG) {
- /* bounce ping pong */
- offset = loop_end_fp - (offset - loop_end_fp);
- increment = -increment;
- sign *= -1;
- } else {
- /* go to loop-begin */
-
- if (format == AudioStreamSample::FORMAT_IMA_ADPCM) {
- for (int i = 0; i < 2; i++) {
- ima_adpcm[i].step_index = ima_adpcm[i].loop_step_index;
- ima_adpcm[i].predictor = ima_adpcm[i].loop_predictor;
- ima_adpcm[i].last_nibble = loop_begin_fp >> MIX_FRAC_BITS;
- }
- offset = loop_begin_fp;
- } else {
- offset = loop_begin_fp + (offset - loop_end_fp);
- }
- }
- } else {
- /* no loop, check for end of sample */
- if (offset >= length_fp) {
- active = false;
- break;
- }
- }
- }
-
- /** MIXCOUNT COMPUTING **/
-
- /* next possible limit (looppoints or sample begin/end */
- limit = (increment < 0) ? begin_limit : end_limit;
-
- /* compute what is shorter, the todo or the limit? */
- aux = (limit - offset) / increment + 1;
- target = (aux < todo) ? aux : todo; /* mix target is the shorter buffer */
-
- /* check just in case */
- if (target <= 0) {
- active = false;
- break;
- }
-
- todo -= target;
-
- switch (base->format) {
- case AudioStreamSample::FORMAT_8_BITS: {
- if (is_stereo) {
- do_resample<int8_t, true, false>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm);
- } else {
- do_resample<int8_t, false, false>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm);
- }
- } break;
- case AudioStreamSample::FORMAT_16_BITS: {
- if (is_stereo) {
- do_resample<int16_t, true, false>((int16_t *)data, dst_buff, offset, increment, target, ima_adpcm);
- } else {
- do_resample<int16_t, false, false>((int16_t *)data, dst_buff, offset, increment, target, ima_adpcm);
- }
-
- } break;
- case AudioStreamSample::FORMAT_IMA_ADPCM: {
- if (is_stereo) {
- do_resample<int8_t, true, true>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm);
- } else {
- do_resample<int8_t, false, true>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm);
- }
-
- } break;
- }
-
- dst_buff += target;
- }
-
- if (todo) {
- int mixed_frames = p_frames - todo;
- //bit was missing from mix
- int todo_ofs = p_frames - todo;
- for (int i = todo_ofs; i < p_frames; i++) {
- p_buffer[i] = AudioFrame(0, 0);
- }
- return mixed_frames;
- }
- return p_frames;
-}
-
-void AudioStreamPlaybackSample::tag_used_streams() {
- base->tag_used(get_playback_position());
-}
-
-AudioStreamPlaybackSample::AudioStreamPlaybackSample() {}
-
-/////////////////////
-
-void AudioStreamSample::set_format(Format p_format) {
- format = p_format;
-}
-
-AudioStreamSample::Format AudioStreamSample::get_format() const {
- return format;
-}
-
-void AudioStreamSample::set_loop_mode(LoopMode p_loop_mode) {
- loop_mode = p_loop_mode;
-}
-
-AudioStreamSample::LoopMode AudioStreamSample::get_loop_mode() const {
- return loop_mode;
-}
-
-void AudioStreamSample::set_loop_begin(int p_frame) {
- loop_begin = p_frame;
-}
-
-int AudioStreamSample::get_loop_begin() const {
- return loop_begin;
-}
-
-void AudioStreamSample::set_loop_end(int p_frame) {
- loop_end = p_frame;
-}
-
-int AudioStreamSample::get_loop_end() const {
- return loop_end;
-}
-
-void AudioStreamSample::set_mix_rate(int p_hz) {
- ERR_FAIL_COND(p_hz == 0);
- mix_rate = p_hz;
-}
-
-int AudioStreamSample::get_mix_rate() const {
- return mix_rate;
-}
-
-void AudioStreamSample::set_stereo(bool p_enable) {
- stereo = p_enable;
-}
-
-bool AudioStreamSample::is_stereo() const {
- return stereo;
-}
-
-float AudioStreamSample::get_length() const {
- int len = data_bytes;
- switch (format) {
- case AudioStreamSample::FORMAT_8_BITS:
- len /= 1;
- break;
- case AudioStreamSample::FORMAT_16_BITS:
- len /= 2;
- break;
- case AudioStreamSample::FORMAT_IMA_ADPCM:
- len *= 2;
- break;
- }
-
- if (stereo) {
- len /= 2;
- }
-
- return float(len) / mix_rate;
-}
-
-bool AudioStreamSample::is_monophonic() const {
- return false;
-}
-
-void AudioStreamSample::set_data(const Vector<uint8_t> &p_data) {
- AudioServer::get_singleton()->lock();
- if (data) {
- memfree(data);
- data = nullptr;
- data_bytes = 0;
- }
-
- int datalen = p_data.size();
- if (datalen) {
- const uint8_t *r = p_data.ptr();
- int alloc_len = datalen + DATA_PAD * 2;
- data = memalloc(alloc_len); //alloc with some padding for interpolation
- memset(data, 0, alloc_len);
- uint8_t *dataptr = (uint8_t *)data;
- memcpy(dataptr + DATA_PAD, r, datalen);
- data_bytes = datalen;
- }
-
- AudioServer::get_singleton()->unlock();
-}
-
-Vector<uint8_t> AudioStreamSample::get_data() const {
- Vector<uint8_t> pv;
-
- if (data) {
- pv.resize(data_bytes);
- {
- uint8_t *w = pv.ptrw();
- uint8_t *dataptr = (uint8_t *)data;
- memcpy(w, dataptr + DATA_PAD, data_bytes);
- }
- }
-
- return pv;
-}
-
-Error AudioStreamSample::save_to_wav(const String &p_path) {
- if (format == AudioStreamSample::FORMAT_IMA_ADPCM) {
- WARN_PRINT("Saving IMA_ADPC samples are not supported yet");
- return ERR_UNAVAILABLE;
- }
-
- int sub_chunk_2_size = data_bytes; //Subchunk2Size = Size of data in bytes
-
- // Format code
- // 1:PCM format (for 8 or 16 bit)
- // 3:IEEE float format
- int format_code = (format == FORMAT_IMA_ADPCM) ? 3 : 1;
-
- int n_channels = stereo ? 2 : 1;
-
- long sample_rate = mix_rate;
-
- int byte_pr_sample = 0;
- switch (format) {
- case AudioStreamSample::FORMAT_8_BITS:
- byte_pr_sample = 1;
- break;
- case AudioStreamSample::FORMAT_16_BITS:
- byte_pr_sample = 2;
- break;
- case AudioStreamSample::FORMAT_IMA_ADPCM:
- byte_pr_sample = 4;
- break;
- }
-
- String file_path = p_path;
- if (!(file_path.substr(file_path.length() - 4, 4) == ".wav")) {
- file_path += ".wav";
- }
-
- Ref<FileAccess> file = FileAccess::open(file_path, FileAccess::WRITE); //Overrides existing file if present
-
- ERR_FAIL_COND_V(file.is_null(), ERR_FILE_CANT_WRITE);
-
- // Create WAV Header
- file->store_string("RIFF"); //ChunkID
- file->store_32(sub_chunk_2_size + 36); //ChunkSize = 36 + SubChunk2Size (size of entire file minus the 8 bits for this and previous header)
- file->store_string("WAVE"); //Format
- file->store_string("fmt "); //Subchunk1ID
- file->store_32(16); //Subchunk1Size = 16
- file->store_16(format_code); //AudioFormat
- file->store_16(n_channels); //Number of Channels
- file->store_32(sample_rate); //SampleRate
- file->store_32(sample_rate * n_channels * byte_pr_sample); //ByteRate
- file->store_16(n_channels * byte_pr_sample); //BlockAlign = NumChannels * BytePrSample
- file->store_16(byte_pr_sample * 8); //BitsPerSample
- file->store_string("data"); //Subchunk2ID
- file->store_32(sub_chunk_2_size); //Subchunk2Size
-
- // Add data
- Vector<uint8_t> data = get_data();
- const uint8_t *read_data = data.ptr();
- switch (format) {
- case AudioStreamSample::FORMAT_8_BITS:
- for (unsigned int i = 0; i < data_bytes; i++) {
- uint8_t data_point = (read_data[i] + 128);
- file->store_8(data_point);
- }
- break;
- case AudioStreamSample::FORMAT_16_BITS:
- for (unsigned int i = 0; i < data_bytes / 2; i++) {
- uint16_t data_point = decode_uint16(&read_data[i * 2]);
- file->store_16(data_point);
- }
- break;
- case AudioStreamSample::FORMAT_IMA_ADPCM:
- //Unimplemented
- break;
- }
-
- return OK;
-}
-
-Ref<AudioStreamPlayback> AudioStreamSample::instantiate_playback() {
- Ref<AudioStreamPlaybackSample> sample;
- sample.instantiate();
- sample->base = Ref<AudioStreamSample>(this);
- return sample;
-}
-
-String AudioStreamSample::get_stream_name() const {
- return "";
-}
-
-void AudioStreamSample::_bind_methods() {
- ClassDB::bind_method(D_METHOD("set_data", "data"), &AudioStreamSample::set_data);
- ClassDB::bind_method(D_METHOD("get_data"), &AudioStreamSample::get_data);
-
- ClassDB::bind_method(D_METHOD("set_format", "format"), &AudioStreamSample::set_format);
- ClassDB::bind_method(D_METHOD("get_format"), &AudioStreamSample::get_format);
-
- ClassDB::bind_method(D_METHOD("set_loop_mode", "loop_mode"), &AudioStreamSample::set_loop_mode);
- ClassDB::bind_method(D_METHOD("get_loop_mode"), &AudioStreamSample::get_loop_mode);
-
- ClassDB::bind_method(D_METHOD("set_loop_begin", "loop_begin"), &AudioStreamSample::set_loop_begin);
- ClassDB::bind_method(D_METHOD("get_loop_begin"), &AudioStreamSample::get_loop_begin);
-
- ClassDB::bind_method(D_METHOD("set_loop_end", "loop_end"), &AudioStreamSample::set_loop_end);
- ClassDB::bind_method(D_METHOD("get_loop_end"), &AudioStreamSample::get_loop_end);
-
- ClassDB::bind_method(D_METHOD("set_mix_rate", "mix_rate"), &AudioStreamSample::set_mix_rate);
- ClassDB::bind_method(D_METHOD("get_mix_rate"), &AudioStreamSample::get_mix_rate);
-
- ClassDB::bind_method(D_METHOD("set_stereo", "stereo"), &AudioStreamSample::set_stereo);
- ClassDB::bind_method(D_METHOD("is_stereo"), &AudioStreamSample::is_stereo);
-
- ClassDB::bind_method(D_METHOD("save_to_wav", "path"), &AudioStreamSample::save_to_wav);
-
- ADD_PROPERTY(PropertyInfo(Variant::PACKED_BYTE_ARRAY, "data", PROPERTY_HINT_NONE, "", PROPERTY_USAGE_NO_EDITOR), "set_data", "get_data");
- ADD_PROPERTY(PropertyInfo(Variant::INT, "format", PROPERTY_HINT_ENUM, "8-Bit,16-Bit,IMA-ADPCM"), "set_format", "get_format");
- ADD_PROPERTY(PropertyInfo(Variant::INT, "loop_mode", PROPERTY_HINT_ENUM, "Disabled,Forward,Ping-Pong,Backward"), "set_loop_mode", "get_loop_mode");
- ADD_PROPERTY(PropertyInfo(Variant::INT, "loop_begin"), "set_loop_begin", "get_loop_begin");
- ADD_PROPERTY(PropertyInfo(Variant::INT, "loop_end"), "set_loop_end", "get_loop_end");
- ADD_PROPERTY(PropertyInfo(Variant::INT, "mix_rate"), "set_mix_rate", "get_mix_rate");
- ADD_PROPERTY(PropertyInfo(Variant::BOOL, "stereo"), "set_stereo", "is_stereo");
-
- BIND_ENUM_CONSTANT(FORMAT_8_BITS);
- BIND_ENUM_CONSTANT(FORMAT_16_BITS);
- BIND_ENUM_CONSTANT(FORMAT_IMA_ADPCM);
-
- BIND_ENUM_CONSTANT(LOOP_DISABLED);
- BIND_ENUM_CONSTANT(LOOP_FORWARD);
- BIND_ENUM_CONSTANT(LOOP_PINGPONG);
- BIND_ENUM_CONSTANT(LOOP_BACKWARD);
-}
-
-AudioStreamSample::AudioStreamSample() {}
-
-AudioStreamSample::~AudioStreamSample() {
- if (data) {
- memfree(data);
- data = nullptr;
- data_bytes = 0;
- }
-}