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Merge pull request #60957 from DeeJayLSP/sample_pcm

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Rémi Verschelde 2022-07-28 19:51:08 +02:00 committed by GitHub
commit 1c820f19b1
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14 changed files with 146 additions and 144 deletions
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6
doc/classes/AudioEffectRecord.xml
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@ -14,7 +14,7 @@
</tutorials>
<methods>
<method name="get_recording" qualifiers="const">
<return type="AudioStreamSample" />
<return type="AudioStreamWAV" />
<description>
Returns the recorded sample.
</description>
@ -34,8 +34,8 @@
</method>
</methods>
<members>
<member name="format" type="int" setter="set_format" getter="get_format" enum="AudioStreamSample.Format" default="1">
Specifies the format in which the sample will be recorded. See [enum AudioStreamSample.Format] for available formats.
<member name="format" type="int" setter="set_format" getter="get_format" enum="AudioStreamWAV.Format" default="1">
Specifies the format in which the sample will be recorded. See [enum AudioStreamWAV.Format] for available formats.
</member>
</members>
</class>

2
doc/classes/AudioStream.xml
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@ -4,7 +4,7 @@
Base class for audio streams.
</brief_description>
<description>
Base class for audio streams. Audio streams are used for sound effects and music playback, and support WAV (via [AudioStreamSample]) and OGG (via [AudioStreamOGGVorbis]) file formats.
Base class for audio streams. Audio streams are used for sound effects and music playback, and support WAV (via [AudioStreamWAV]) and OGG (via [AudioStreamOGGVorbis]) file formats.
</description>
<tutorials>
<link title="Audio streams">$DOCS_URL/tutorials/audio/audio_streams.html</link>

10
doc/classes/AudioStreamSample.xml → doc/classes/AudioStreamWAV.xml
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@ -1,10 +1,10 @@
<?xml version="1.0" encoding="UTF-8" ?>
<class name="AudioStreamSample" inherits="AudioStream" version="4.0" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:noNamespaceSchemaLocation="../class.xsd">
<class name="AudioStreamWAV" inherits="AudioStream" version="4.0" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:noNamespaceSchemaLocation="../class.xsd">
<brief_description>
Stores audio data loaded from WAV files.
</brief_description>
<description>
AudioStreamSample stores sound samples loaded from WAV files. To play the stored sound, use an [AudioStreamPlayer] (for non-positional audio) or [AudioStreamPlayer2D]/[AudioStreamPlayer3D] (for positional audio). The sound can be looped.
AudioStreamWAV stores sound samples loaded from WAV files. To play the stored sound, use an [AudioStreamPlayer] (for non-positional audio) or [AudioStreamPlayer2D]/[AudioStreamPlayer3D] (for positional audio). The sound can be looped.
This class can also be used to store dynamically-generated PCM audio data. See also [AudioStreamGenerator] for procedural audio generation.
</description>
<tutorials>
@ -14,7 +14,7 @@
<return type="int" enum="Error" />
<argument index="0" name="path" type="String" />
<description>
Saves the AudioStreamSample as a WAV file to [code]path[/code]. Samples with IMA ADPCM format can't be saved.
Saves the AudioStreamWAV as a WAV file to [code]path[/code]. Samples with IMA ADPCM format can't be saved.
[b]Note:[/b] A [code].wav[/code] extension is automatically appended to [code]path[/code] if it is missing.
</description>
</method>
@ -24,7 +24,7 @@
Contains the audio data in bytes.
[b]Note:[/b] This property expects signed PCM8 data. To convert unsigned PCM8 to signed PCM8, subtract 128 from each byte.
</member>
<member name="format" type="int" setter="set_format" getter="get_format" enum="AudioStreamSample.Format" default="0">
<member name="format" type="int" setter="set_format" getter="get_format" enum="AudioStreamWAV.Format" default="0">
Audio format. See [enum Format] constants for values.
</member>
<member name="loop_begin" type="int" setter="set_loop_begin" getter="get_loop_begin" default="0">
@ -33,7 +33,7 @@
<member name="loop_end" type="int" setter="set_loop_end" getter="get_loop_end" default="0">
The loop end point (in number of samples, relative to the beginning of the sample). This information will be imported automatically from the WAV file if present.
</member>
<member name="loop_mode" type="int" setter="set_loop_mode" getter="get_loop_mode" enum="AudioStreamSample.LoopMode" default="0">
<member name="loop_mode" type="int" setter="set_loop_mode" getter="get_loop_mode" enum="AudioStreamWAV.LoopMode" default="0">
The loop mode. This information will be imported automatically from the WAV file if present. See [enum LoopMode] constants for values.
</member>
<member name="mix_rate" type="int" setter="set_mix_rate" getter="get_mix_rate" default="44100">

2
editor/editor_asset_installer.cpp
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@ -100,7 +100,7 @@ void EditorAssetInstaller::open(const String &p_path, int p_depth) {
extension_guess["tga"] = tree->get_theme_icon(SNAME("ImageTexture"), SNAME("EditorIcons"));
extension_guess["webp"] = tree->get_theme_icon(SNAME("ImageTexture"), SNAME("EditorIcons"));
extension_guess["wav"] = tree->get_theme_icon(SNAME("AudioStreamSample"), SNAME("EditorIcons"));
extension_guess["wav"] = tree->get_theme_icon(SNAME("AudioStreamWAV"), SNAME("EditorIcons"));
extension_guess["ogg"] = tree->get_theme_icon(SNAME("AudioStreamOGGVorbis"), SNAME("EditorIcons"));
extension_guess["mp3"] = tree->get_theme_icon(SNAME("AudioStreamMP3"), SNAME("EditorIcons"));

0
editor/icons/AudioStreamSample.svg → editor/icons/AudioStreamWAV.svg
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Side by Side Swipe Overlay

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Width:  |  Height:  |  Size: 1.1 KiB

After

Width:  |  Height:  |  Size: 1.1 KiB

26
editor/import/resource_importer_wav.cpp
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@ -33,7 +33,7 @@
#include "core/io/file_access.h"
#include "core/io/marshalls.h"
#include "core/io/resource_saver.h"
#include "scene/resources/audio_stream_sample.h"
#include "scene/resources/audio_stream_wav.h"
const float TRIM_DB_LIMIT = -50;
const int TRIM_FADE_OUT_FRAMES = 500;
@ -55,7 +55,7 @@ String ResourceImporterWAV::get_save_extension() const {
}
String ResourceImporterWAV::get_resource_type() const {
return "AudioStreamSample";
return "AudioStreamWAV";
}
bool ResourceImporterWAV::get_option_visibility(const String &p_path, const String &p_option, const HashMap<StringName, Variant> &p_options) const {
@ -86,7 +86,7 @@ void ResourceImporterWAV::get_import_options(const String &p_path, List<ImportOp
r_options->push_back(ImportOption(PropertyInfo(Variant::FLOAT, "force/max_rate_hz", PROPERTY_HINT_RANGE, "11025,192000,1,exp"), 44100));
r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL, "edit/trim"), false));
r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL, "edit/normalize"), false));
// Keep the `edit/loop_mode` enum in sync with AudioStreamSample::LoopMode (note: +1 offset due to "Detect From WAV").
// Keep the `edit/loop_mode` enum in sync with AudioStreamWAV::LoopMode (note: +1 offset due to "Detect From WAV").
r_options->push_back(ImportOption(PropertyInfo(Variant::INT, "edit/loop_mode", PROPERTY_HINT_ENUM, "Detect From WAV,Disabled,Forward,Ping-Pong,Backward", PROPERTY_USAGE_DEFAULT | PROPERTY_USAGE_UPDATE_ALL_IF_MODIFIED), 0));
r_options->push_back(ImportOption(PropertyInfo(Variant::INT, "edit/loop_begin"), 0));
r_options->push_back(ImportOption(PropertyInfo(Variant::INT, "edit/loop_end"), -1));
@ -130,7 +130,7 @@ Error ResourceImporterWAV::import(const String &p_source_file, const String &p_s
int format_bits = 0;
int format_channels = 0;
AudioStreamSample::LoopMode loop_mode = AudioStreamSample::LOOP_DISABLED;
AudioStreamWAV::LoopMode loop_mode = AudioStreamWAV::LOOP_DISABLED;
uint16_t compression_code = 1;
bool format_found = false;
bool data_found = false;
@ -282,11 +282,11 @@ Error ResourceImporterWAV::import(const String &p_source_file, const String &p_s
int loop_type = file->get_32();
if (loop_type == 0x00 || loop_type == 0x01 || loop_type == 0x02) {
if (loop_type == 0x00) {
loop_mode = AudioStreamSample::LOOP_FORWARD;
loop_mode = AudioStreamWAV::LOOP_FORWARD;
} else if (loop_type == 0x01) {
loop_mode = AudioStreamSample::LOOP_PINGPONG;
loop_mode = AudioStreamWAV::LOOP_PINGPONG;
} else if (loop_type == 0x02) {
loop_mode = AudioStreamSample::LOOP_BACKWARD;
loop_mode = AudioStreamWAV::LOOP_BACKWARD;
}
loop_begin = file->get_32();
loop_end = file->get_32();
@ -386,7 +386,7 @@ Error ResourceImporterWAV::import(const String &p_source_file, const String &p_s
bool trim = p_options["edit/trim"];
if (trim && (loop_mode != AudioStreamSample::LOOP_DISABLED) && format_channels > 0) {
if (trim && (loop_mode != AudioStreamWAV::LOOP_DISABLED) && format_channels > 0) {
int first = 0;
int last = (frames / format_channels) - 1;
bool found = false;
@ -431,7 +431,7 @@ Error ResourceImporterWAV::import(const String &p_source_file, const String &p_s
}
if (import_loop_mode >= 2) {
loop_mode = (AudioStreamSample::LoopMode)(import_loop_mode - 1);
loop_mode = (AudioStreamWAV::LoopMode)(import_loop_mode - 1);
loop_begin = p_options["edit/loop_begin"];
loop_end = p_options["edit/loop_end"];
// Wrap around to max frames, so `-1` can be used to select the end, etc.
@ -463,10 +463,10 @@ Error ResourceImporterWAV::import(const String &p_source_file, const String &p_s
}
Vector<uint8_t> dst_data;
AudioStreamSample::Format dst_format;
AudioStreamWAV::Format dst_format;
if (compression == 1) {
dst_format = AudioStreamSample::FORMAT_IMA_ADPCM;
dst_format = AudioStreamWAV::FORMAT_IMA_ADPCM;
if (format_channels == 1) {
_compress_ima_adpcm(data, dst_data);
} else {
@ -503,7 +503,7 @@ Error ResourceImporterWAV::import(const String &p_source_file, const String &p_s
}
} else {
dst_format = is16 ? AudioStreamSample::FORMAT_16_BITS : AudioStreamSample::FORMAT_8_BITS;
dst_format = is16 ? AudioStreamWAV::FORMAT_16_BITS : AudioStreamWAV::FORMAT_8_BITS;
dst_data.resize(data.size() * (is16 ? 2 : 1));
{
uint8_t *w = dst_data.ptrw();
@ -521,7 +521,7 @@ Error ResourceImporterWAV::import(const String &p_source_file, const String &p_s
}
}
Ref<AudioStreamSample> sample;
Ref<AudioStreamWAV> sample;
sample.instantiate();
sample->set_data(dst_data);
sample->set_format(dst_format);

3
editor/project_converter_3_to_4.cpp
View File

@ -120,7 +120,7 @@ static const char *enum_renames[][2] = {
{ "JOINT_PIN", "JOINT_TYPE_PIN" }, // PhysicsServer2D
{ "JOINT_SLIDER", "JOINT_TYPE_SLIDER" }, // PhysicsServer3D
{ "KEY_CONTROL", "KEY_CTRL" }, // Globals
{ "LOOP_PING_PONG", "LOOP_PINGPONG" }, //AudioStreamSample
{ "LOOP_PING_PONG", "LOOP_PINGPONG" }, // AudioStreamWAV
{ "MATH_RAND", "MATH_RANDF_RANGE" }, // VisualScriptBuiltinFunc
{ "MATH_RANDOM", "MATH_RANDI_RANGE" }, // VisualScriptBuiltinFunc
{ "MATH_STEPIFY", "MATH_STEP_DECIMALS" }, // VisualScriptBuiltinFunc
@ -1251,6 +1251,7 @@ static const char *class_renames[][2] = {
{ "AnimationTreePlayer", "AnimationTree" },
{ "Area", "Area3D" }, // Be careful, this will be used everywhere
{ "AudioStreamRandomPitch", "AudioStreamRandomizer" },
{ "AudioStreamSample", "AudioStreamWAV" },
{ "BakedLightmap", "LightmapGI" },
{ "BakedLightmapData", "LightmapGIData" },
{ "BitmapFont", "FontFile" },

5
scene/register_scene_types.cpp
View File

@ -141,7 +141,7 @@
#include "scene/multiplayer/scene_replication_interface.h"
#include "scene/multiplayer/scene_rpc_interface.h"
#include "scene/resources/animation_library.h"
#include "scene/resources/audio_stream_sample.h"
#include "scene/resources/audio_stream_wav.h"
#include "scene/resources/bit_map.h"
#include "scene/resources/bone_map.h"
#include "scene/resources/box_shape_3d.h"
@ -904,7 +904,7 @@ void register_scene_types() {
GDREGISTER_CLASS(AudioStreamPlayer3D);
#endif
GDREGISTER_ABSTRACT_CLASS(VideoStream);
GDREGISTER_CLASS(AudioStreamSample);
GDREGISTER_CLASS(AudioStreamWAV);
OS::get_singleton()->yield(); // may take time to init
@ -1091,6 +1091,7 @@ void register_scene_types() {
ClassDB::add_compatibility_class("World", "World3D");
// Renamed during 4.0 alpha, added to ease transition between alphas.
ClassDB::add_compatibility_class("AudioStreamSample", "AudioStreamWAV");
ClassDB::add_compatibility_class("StreamCubemap", "CompressedCubemap");
ClassDB::add_compatibility_class("StreamCubemapArray", "CompressedCubemapArray");
ClassDB::add_compatibility_class("StreamTexture2D", "CompressedTexture2D");

168
scene/resources/audio_stream_sample.cpp → scene/resources/audio_stream_wav.cpp
View File

@ -1,5 +1,5 @@
/*************************************************************************/
/* audio_stream_sample.cpp */
/* audio_stream_wav.cpp */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
@ -28,13 +28,13 @@
/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
/*************************************************************************/
#include "audio_stream_sample.h"
#include "audio_stream_wav.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) {
void AudioStreamPlaybackWAV::start(float p_from_pos) {
if (base->format == AudioStreamWAV::FORMAT_IMA_ADPCM) {
//no seeking in IMA_ADPCM
for (int i = 0; i < 2; i++) {
ima_adpcm[i].step_index = 0;
@ -55,24 +55,24 @@ void AudioStreamPlaybackSample::start(float p_from_pos) {
active = true;
}
void AudioStreamPlaybackSample::stop() {
void AudioStreamPlaybackWAV::stop() {
active = false;
}
bool AudioStreamPlaybackSample::is_playing() const {
bool AudioStreamPlaybackWAV::is_playing() const {
return active;
}
int AudioStreamPlaybackSample::get_loop_count() const {
int AudioStreamPlaybackWAV::get_loop_count() const {
return 0;
}
float AudioStreamPlaybackSample::get_playback_position() const {
float AudioStreamPlaybackWAV::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) {
void AudioStreamPlaybackWAV::seek(float p_time) {
if (base->format == AudioStreamWAV::FORMAT_IMA_ADPCM) {
return; //no seeking in ima-adpcm
}
@ -87,7 +87,7 @@ void AudioStreamPlaybackSample::seek(float p_time) {
}
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) {
void AudioStreamPlaybackWAV::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;
@ -124,7 +124,7 @@ void AudioStreamPlaybackSample::do_resample(const Depth *p_src, AudioFrame *p_ds
ima_adpcm[i].last_nibble++;
const uint8_t *src_ptr = (const uint8_t *)base->data;
src_ptr += AudioStreamSample::DATA_PAD;
src_ptr += AudioStreamWAV::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);
@ -221,7 +221,7 @@ void AudioStreamPlaybackSample::do_resample(const Depth *p_src, AudioFrame *p_ds
}
}
int AudioStreamPlaybackSample::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
int AudioStreamPlaybackWAV::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);
@ -231,13 +231,13 @@ int AudioStreamPlaybackSample::mix(AudioFrame *p_buffer, float p_rate_scale, int
int len = base->data_bytes;
switch (base->format) {
case AudioStreamSample::FORMAT_8_BITS:
case AudioStreamWAV::FORMAT_8_BITS:
len /= 1;
break;
case AudioStreamSample::FORMAT_16_BITS:
case AudioStreamWAV::FORMAT_16_BITS:
len /= 2;
break;
case AudioStreamSample::FORMAT_IMA_ADPCM:
case AudioStreamWAV::FORMAT_IMA_ADPCM:
len *= 2;
break;
}
@ -251,13 +251,13 @@ int AudioStreamPlaybackSample::mix(AudioFrame *p_buffer, float p_rate_scale, int
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;
int64_t begin_limit = (base->loop_mode != AudioStreamWAV::LOOP_DISABLED) ? loop_begin_fp : 0;
int64_t end_limit = (base->loop_mode != AudioStreamWAV::LOOP_DISABLED) ? loop_end_fp : length_fp;
bool is_stereo = base->stereo;
int32_t todo = p_frames;
if (base->loop_mode == AudioStreamSample::LOOP_BACKWARD) {
if (base->loop_mode == AudioStreamWAV::LOOP_BACKWARD) {
sign = -1;
}
@ -271,20 +271,20 @@ int AudioStreamPlaybackSample::mix(AudioFrame *p_buffer, float p_rate_scale, int
//looping
AudioStreamSample::LoopMode loop_format = base->loop_mode;
AudioStreamSample::Format format = base->format;
AudioStreamWAV::LoopMode loop_format = base->loop_mode;
AudioStreamWAV::Format format = base->format;
/* audio data */
uint8_t *dataptr = (uint8_t *)base->data;
const void *data = dataptr + AudioStreamSample::DATA_PAD;
const void *data = dataptr + AudioStreamWAV::DATA_PAD;
AudioFrame *dst_buff = p_buffer;
if (format == AudioStreamSample::FORMAT_IMA_ADPCM) {
if (loop_format != AudioStreamSample::LOOP_DISABLED) {
if (format == AudioStreamWAV::FORMAT_IMA_ADPCM) {
if (loop_format != AudioStreamWAV::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;
loop_format = AudioStreamWAV::LOOP_FORWARD;
}
}
@ -297,9 +297,9 @@ int AudioStreamPlaybackSample::mix(AudioFrame *p_buffer, float p_rate_scale, int
if (increment < 0) {
/* going backwards */
if (loop_format != AudioStreamSample::LOOP_DISABLED && offset < loop_begin_fp) {
if (loop_format != AudioStreamWAV::LOOP_DISABLED && offset < loop_begin_fp) {
/* loopstart reached */
if (loop_format == AudioStreamSample::LOOP_PINGPONG) {
if (loop_format == AudioStreamWAV::LOOP_PINGPONG) {
/* bounce ping pong */
offset = loop_begin_fp + (loop_begin_fp - offset);
increment = -increment;
@ -317,10 +317,10 @@ int AudioStreamPlaybackSample::mix(AudioFrame *p_buffer, float p_rate_scale, int
}
} else {
/* going forward */
if (loop_format != AudioStreamSample::LOOP_DISABLED && offset >= loop_end_fp) {
if (loop_format != AudioStreamWAV::LOOP_DISABLED && offset >= loop_end_fp) {
/* loopend reached */
if (loop_format == AudioStreamSample::LOOP_PINGPONG) {
if (loop_format == AudioStreamWAV::LOOP_PINGPONG) {
/* bounce ping pong */
offset = loop_end_fp - (offset - loop_end_fp);
increment = -increment;
@ -328,7 +328,7 @@ int AudioStreamPlaybackSample::mix(AudioFrame *p_buffer, float p_rate_scale, int
} else {
/* go to loop-begin */
if (format == AudioStreamSample::FORMAT_IMA_ADPCM) {
if (format == AudioStreamWAV::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;
@ -366,14 +366,14 @@ int AudioStreamPlaybackSample::mix(AudioFrame *p_buffer, float p_rate_scale, int
todo -= target;
switch (base->format) {
case AudioStreamSample::FORMAT_8_BITS: {
case AudioStreamWAV::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: {
case AudioStreamWAV::FORMAT_16_BITS: {
if (is_stereo) {
do_resample<int16_t, true, false>((int16_t *)data, dst_buff, offset, increment, target, ima_adpcm);
} else {
@ -381,7 +381,7 @@ int AudioStreamPlaybackSample::mix(AudioFrame *p_buffer, float p_rate_scale, int
}
} break;
case AudioStreamSample::FORMAT_IMA_ADPCM: {
case AudioStreamWAV::FORMAT_IMA_ADPCM: {
if (is_stereo) {
do_resample<int8_t, true, true>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm);
} else {
@ -406,73 +406,73 @@ int AudioStreamPlaybackSample::mix(AudioFrame *p_buffer, float p_rate_scale, int
return p_frames;
}
void AudioStreamPlaybackSample::tag_used_streams() {
void AudioStreamPlaybackWAV::tag_used_streams() {
base->tag_used(get_playback_position());
}
AudioStreamPlaybackSample::AudioStreamPlaybackSample() {}
AudioStreamPlaybackWAV::AudioStreamPlaybackWAV() {}
/////////////////////
void AudioStreamSample::set_format(Format p_format) {
void AudioStreamWAV::set_format(Format p_format) {
format = p_format;
}
AudioStreamSample::Format AudioStreamSample::get_format() const {
AudioStreamWAV::Format AudioStreamWAV::get_format() const {
return format;
}
void AudioStreamSample::set_loop_mode(LoopMode p_loop_mode) {
void AudioStreamWAV::set_loop_mode(LoopMode p_loop_mode) {
loop_mode = p_loop_mode;
}
AudioStreamSample::LoopMode AudioStreamSample::get_loop_mode() const {
AudioStreamWAV::LoopMode AudioStreamWAV::get_loop_mode() const {
return loop_mode;
}
void AudioStreamSample::set_loop_begin(int p_frame) {
void AudioStreamWAV::set_loop_begin(int p_frame) {
loop_begin = p_frame;
}
int AudioStreamSample::get_loop_begin() const {
int AudioStreamWAV::get_loop_begin() const {
return loop_begin;
}
void AudioStreamSample::set_loop_end(int p_frame) {
void AudioStreamWAV::set_loop_end(int p_frame) {
loop_end = p_frame;
}
int AudioStreamSample::get_loop_end() const {
int AudioStreamWAV::get_loop_end() const {
return loop_end;
}
void AudioStreamSample::set_mix_rate(int p_hz) {
void AudioStreamWAV::set_mix_rate(int p_hz) {
ERR_FAIL_COND(p_hz == 0);
mix_rate = p_hz;
}
int AudioStreamSample::get_mix_rate() const {
int AudioStreamWAV::get_mix_rate() const {
return mix_rate;
}
void AudioStreamSample::set_stereo(bool p_enable) {
void AudioStreamWAV::set_stereo(bool p_enable) {
stereo = p_enable;
}
bool AudioStreamSample::is_stereo() const {
bool AudioStreamWAV::is_stereo() const {
return stereo;
}
float AudioStreamSample::get_length() const {
float AudioStreamWAV::get_length() const {
int len = data_bytes;
switch (format) {
case AudioStreamSample::FORMAT_8_BITS:
case AudioStreamWAV::FORMAT_8_BITS:
len /= 1;
break;
case AudioStreamSample::FORMAT_16_BITS:
case AudioStreamWAV::FORMAT_16_BITS:
len /= 2;
break;
case AudioStreamSample::FORMAT_IMA_ADPCM:
case AudioStreamWAV::FORMAT_IMA_ADPCM:
len *= 2;
break;
}
@ -484,11 +484,11 @@ float AudioStreamSample::get_length() const {
return float(len) / mix_rate;
}
bool AudioStreamSample::is_monophonic() const {
bool AudioStreamWAV::is_monophonic() const {
return false;
}
void AudioStreamSample::set_data(const Vector<uint8_t> &p_data) {
void AudioStreamWAV::set_data(const Vector<uint8_t> &p_data) {
AudioServer::get_singleton()->lock();
if (data) {
memfree(data);
@ -510,7 +510,7 @@ void AudioStreamSample::set_data(const Vector<uint8_t> &p_data) {
AudioServer::get_singleton()->unlock();
}
Vector<uint8_t> AudioStreamSample::get_data() const {
Vector<uint8_t> AudioStreamWAV::get_data() const {
Vector<uint8_t> pv;
if (data) {
@ -525,8 +525,8 @@ Vector<uint8_t> AudioStreamSample::get_data() const {
return pv;
}
Error AudioStreamSample::save_to_wav(const String &p_path) {
if (format == AudioStreamSample::FORMAT_IMA_ADPCM) {
Error AudioStreamWAV::save_to_wav(const String &p_path) {
if (format == AudioStreamWAV::FORMAT_IMA_ADPCM) {
WARN_PRINT("Saving IMA_ADPC samples are not supported yet");
return ERR_UNAVAILABLE;
}
@ -544,13 +544,13 @@ Error AudioStreamSample::save_to_wav(const String &p_path) {
int byte_pr_sample = 0;
switch (format) {
case AudioStreamSample::FORMAT_8_BITS:
case AudioStreamWAV::FORMAT_8_BITS:
byte_pr_sample = 1;
break;
case AudioStreamSample::FORMAT_16_BITS:
case AudioStreamWAV::FORMAT_16_BITS:
byte_pr_sample = 2;
break;
case AudioStreamSample::FORMAT_IMA_ADPCM:
case AudioStreamWAV::FORMAT_IMA_ADPCM:
byte_pr_sample = 4;
break;
}
@ -583,19 +583,19 @@ Error AudioStreamSample::save_to_wav(const String &p_path) {
Vector<uint8_t> data = get_data();
const uint8_t *read_data = data.ptr();
switch (format) {
case AudioStreamSample::FORMAT_8_BITS:
case AudioStreamWAV::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:
case AudioStreamWAV::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:
case AudioStreamWAV::FORMAT_IMA_ADPCM:
//Unimplemented
break;
}
@ -603,40 +603,40 @@ Error AudioStreamSample::save_to_wav(const String &p_path) {
return OK;
}
Ref<AudioStreamPlayback> AudioStreamSample::instantiate_playback() {
Ref<AudioStreamPlaybackSample> sample;
Ref<AudioStreamPlayback> AudioStreamWAV::instantiate_playback() {
Ref<AudioStreamPlaybackWAV> sample;
sample.instantiate();
sample->base = Ref<AudioStreamSample>(this);
sample->base = Ref<AudioStreamWAV>(this);
return sample;
}
String AudioStreamSample::get_stream_name() const {
String AudioStreamWAV::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);
void AudioStreamWAV::_bind_methods() {
ClassDB::bind_method(D_METHOD("set_data", "data"), &AudioStreamWAV::set_data);
ClassDB::bind_method(D_METHOD("get_data"), &AudioStreamWAV::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_format", "format"), &AudioStreamWAV::set_format);
ClassDB::bind_method(D_METHOD("get_format"), &AudioStreamWAV::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_mode", "loop_mode"), &AudioStreamWAV::set_loop_mode);
ClassDB::bind_method(D_METHOD("get_loop_mode"), &AudioStreamWAV::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_begin", "loop_begin"), &AudioStreamWAV::set_loop_begin);
ClassDB::bind_method(D_METHOD("get_loop_begin"), &AudioStreamWAV::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_loop_end", "loop_end"), &AudioStreamWAV::set_loop_end);
ClassDB::bind_method(D_METHOD("get_loop_end"), &AudioStreamWAV::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_mix_rate", "mix_rate"), &AudioStreamWAV::set_mix_rate);
ClassDB::bind_method(D_METHOD("get_mix_rate"), &AudioStreamWAV::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("set_stereo", "stereo"), &AudioStreamWAV::set_stereo);
ClassDB::bind_method(D_METHOD("is_stereo"), &AudioStreamWAV::is_stereo);
ClassDB::bind_method(D_METHOD("save_to_wav", "path"), &AudioStreamSample::save_to_wav);
ClassDB::bind_method(D_METHOD("save_to_wav", "path"), &AudioStreamWAV::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");
@ -656,9 +656,9 @@ void AudioStreamSample::_bind_methods() {
BIND_ENUM_CONSTANT(LOOP_BACKWARD);
}
AudioStreamSample::AudioStreamSample() {}
AudioStreamWAV::AudioStreamWAV() {}
AudioStreamSample::~AudioStreamSample() {
AudioStreamWAV::~AudioStreamWAV() {
if (data) {
memfree(data);
data = nullptr;

34
scene/resources/audio_stream_sample.h → scene/resources/audio_stream_wav.h
View File

@ -1,5 +1,5 @@
/*************************************************************************/
/* audio_stream_sample.h */
/* audio_stream_wav.h */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
@ -28,15 +28,15 @@
/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
/*************************************************************************/
#ifndef AUDIO_STREAM_SAMPLE_H
#define AUDIO_STREAM_SAMPLE_H
#ifndef AUDIO_STREAM_WAV_H
#define AUDIO_STREAM_WAV_H
#include "servers/audio/audio_stream.h"
class AudioStreamSample;
class AudioStreamWAV;
class AudioStreamPlaybackSample : public AudioStreamPlayback {
GDCLASS(AudioStreamPlaybackSample, AudioStreamPlayback);
class AudioStreamPlaybackWAV : public AudioStreamPlayback {
GDCLASS(AudioStreamPlaybackWAV, AudioStreamPlayback);
enum {
MIX_FRAC_BITS = 13,
MIX_FRAC_LEN = (1 << MIX_FRAC_BITS),
@ -57,8 +57,8 @@ class AudioStreamPlaybackSample : public AudioStreamPlayback {
int64_t offset = 0;
int sign = 1;
bool active = false;
friend class AudioStreamSample;
Ref<AudioStreamSample> base;
friend class AudioStreamWAV;
Ref<AudioStreamWAV> base;
template <class Depth, bool is_stereo, bool is_ima_adpcm>
void do_resample(const Depth *p_src, AudioFrame *p_dst, int64_t &offset, int32_t &increment, uint32_t amount, IMA_ADPCM_State *ima_adpcm);
@ -77,11 +77,11 @@ public:
virtual void tag_used_streams() override;
AudioStreamPlaybackSample();
AudioStreamPlaybackWAV();
};
class AudioStreamSample : public AudioStream {
GDCLASS(AudioStreamSample, AudioStream);
class AudioStreamWAV : public AudioStream {
GDCLASS(AudioStreamWAV, AudioStream);
RES_BASE_EXTENSION("sample")
public:
@ -100,7 +100,7 @@ public:
};
private:
friend class AudioStreamPlaybackSample;
friend class AudioStreamPlaybackWAV;
enum {
DATA_PAD = 16 //padding for interpolation
@ -149,11 +149,11 @@ public:
virtual Ref<AudioStreamPlayback> instantiate_playback() override;
virtual String get_stream_name() const override;
AudioStreamSample();
~AudioStreamSample();
AudioStreamWAV();
~AudioStreamWAV();
};
VARIANT_ENUM_CAST(AudioStreamSample::Format)
VARIANT_ENUM_CAST(AudioStreamSample::LoopMode)
VARIANT_ENUM_CAST(AudioStreamWAV::Format)
VARIANT_ENUM_CAST(AudioStreamWAV::LoopMode)
#endif // AUDIO_STREAM_SAMPLE_H
#endif // AUDIO_STREAM_WAV_H

20
servers/audio/effects/audio_effect_record.cpp
View File

@ -199,16 +199,16 @@ bool AudioEffectRecord::is_recording_active() const {
return recording_active;
}
void AudioEffectRecord::set_format(AudioStreamSample::Format p_format) {
void AudioEffectRecord::set_format(AudioStreamWAV::Format p_format) {
format = p_format;
}
AudioStreamSample::Format AudioEffectRecord::get_format() const {
AudioStreamWAV::Format AudioEffectRecord::get_format() const {
return format;
}
Ref<AudioStreamSample> AudioEffectRecord::get_recording() const {
AudioStreamSample::Format dst_format = format;
Ref<AudioStreamWAV> AudioEffectRecord::get_recording() const {
AudioStreamWAV::Format dst_format = format;
bool stereo = true; //forcing mono is not implemented
Vector<uint8_t> dst_data;
@ -216,7 +216,7 @@ Ref<AudioStreamSample> AudioEffectRecord::get_recording() const {
ERR_FAIL_COND_V(current_instance.is_null(), nullptr);
ERR_FAIL_COND_V(current_instance->recording_data.size() == 0, nullptr);
if (dst_format == AudioStreamSample::FORMAT_8_BITS) {
if (dst_format == AudioStreamWAV::FORMAT_8_BITS) {
int data_size = current_instance->recording_data.size();
dst_data.resize(data_size);
uint8_t *w = dst_data.ptrw();
@ -225,7 +225,7 @@ Ref<AudioStreamSample> AudioEffectRecord::get_recording() const {
int8_t v = CLAMP(current_instance->recording_data[i] * 128, -128, 127);
w[i] = v;
}
} else if (dst_format == AudioStreamSample::FORMAT_16_BITS) {
} else if (dst_format == AudioStreamWAV::FORMAT_16_BITS) {
int data_size = current_instance->recording_data.size();
dst_data.resize(data_size * 2);
uint8_t *w = dst_data.ptrw();
@ -234,7 +234,7 @@ Ref<AudioStreamSample> AudioEffectRecord::get_recording() const {
int16_t v = CLAMP(current_instance->recording_data[i] * 32768, -32768, 32767);
encode_uint16(v, &w[i * 2]);
}
} else if (dst_format == AudioStreamSample::FORMAT_IMA_ADPCM) {
} else if (dst_format == AudioStreamWAV::FORMAT_IMA_ADPCM) {
//byte interleave
Vector<float> left;
Vector<float> right;
@ -273,12 +273,12 @@ Ref<AudioStreamSample> AudioEffectRecord::get_recording() const {
ERR_PRINT("Format not implemented.");
}
Ref<AudioStreamSample> sample;
Ref<AudioStreamWAV> sample;
sample.instantiate();
sample->set_data(dst_data);
sample->set_format(dst_format);
sample->set_mix_rate(AudioServer::get_singleton()->get_mix_rate());
sample->set_loop_mode(AudioStreamSample::LOOP_DISABLED);
sample->set_loop_mode(AudioStreamWAV::LOOP_DISABLED);
sample->set_loop_begin(0);
sample->set_loop_end(0);
sample->set_stereo(stereo);
@ -297,6 +297,6 @@ void AudioEffectRecord::_bind_methods() {
}
AudioEffectRecord::AudioEffectRecord() {
format = AudioStreamSample::FORMAT_16_BITS;
format = AudioStreamWAV::FORMAT_16_BITS;
recording_active = false;
}

10
servers/audio/effects/audio_effect_record.h
View File

@ -35,7 +35,7 @@
#include "core/io/marshalls.h"
#include "core/os/os.h"
#include "core/os/thread.h"
#include "scene/resources/audio_stream_sample.h"
#include "scene/resources/audio_stream_wav.h"
#include "servers/audio/audio_effect.h"
#include "servers/audio_server.h"
@ -85,7 +85,7 @@ class AudioEffectRecord : public AudioEffect {
bool recording_active;
Ref<AudioEffectRecordInstance> current_instance;
AudioStreamSample::Format format;
AudioStreamWAV::Format format;
void ensure_thread_stopped();
@ -96,9 +96,9 @@ public:
Ref<AudioEffectInstance> instantiate() override;
void set_recording_active(bool p_record);
bool is_recording_active() const;
void set_format(AudioStreamSample::Format p_format);
AudioStreamSample::Format get_format() const;
Ref<AudioStreamSample> get_recording() const;
void set_format(AudioStreamWAV::Format p_format);
AudioStreamWAV::Format get_format() const;
Ref<AudioStreamWAV> get_recording() const;
AudioEffectRecord();
};

2
servers/audio_server.cpp
View File

@ -39,7 +39,7 @@
#include "core/os/os.h"
#include "core/string/string_name.h"
#include "core/templates/pair.h"
#include "scene/resources/audio_stream_sample.h"
#include "scene/resources/audio_stream_wav.h"
#include "servers/audio/audio_driver_dummy.h"
#include "servers/audio/effects/audio_effect_compressor.h"

2
servers/audio_server.h
View File

@ -43,7 +43,7 @@
class AudioDriverDummy;
class AudioStream;
class AudioStreamSample;
class AudioStreamWAV;
class AudioStreamPlayback;
class AudioDriver {