Merge pull request #60957 from DeeJayLSP/sample_pcm

14 files changed, 146 insertions, 144 deletions

diff --git a/doc/classes/AudioEffectRecord.xml b/doc/classes/AudioEffectRecord.xml
index 9728011bb2..32a6aea340 100644
--- a/doc/classes/AudioEffectRecord.xml
+++ b/doc/classes/AudioEffectRecord.xml
@@ -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>
diff --git a/doc/classes/AudioStream.xml b/doc/classes/AudioStream.xml
index 68f64505d0..0793f2efef 100644
--- a/doc/classes/AudioStream.xml
+++ b/doc/classes/AudioStream.xml
@@ -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>
diff --git a/doc/classes/AudioStreamSample.xml b/doc/classes/AudioStreamWAV.xml
index 62f27ce876..17595aec2f 100644
--- a/doc/classes/AudioStreamSample.xml
+++ b/doc/classes/AudioStreamWAV.xml
@@ -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">
diff --git a/editor/editor_asset_installer.cpp b/editor/editor_asset_installer.cpp
index 8fa486408e..aea962f344 100644
--- a/editor/editor_asset_installer.cpp
+++ b/editor/editor_asset_installer.cpp
@@ -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"));
 
diff --git a/editor/icons/AudioStreamSample.svg b/editor/icons/AudioStreamWAV.svg
index 2e54de9faa..2e54de9faa 100644
--- a/editor/icons/AudioStreamSample.svg
+++ b/editor/icons/AudioStreamWAV.svg
diff --git a/editor/import/resource_importer_wav.cpp b/editor/import/resource_importer_wav.cpp
index f0ba1eb7a1..3a47bfb29f 100644
--- a/editor/import/resource_importer_wav.cpp
+++ b/editor/import/resource_importer_wav.cpp
@@ -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);
diff --git a/editor/project_converter_3_to_4.cpp b/editor/project_converter_3_to_4.cpp
index 6437e19404..5be6e9d059 100644
--- a/editor/project_converter_3_to_4.cpp
+++ b/editor/project_converter_3_to_4.cpp
@@ -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" },
diff --git a/scene/register_scene_types.cpp b/scene/register_scene_types.cpp
index ef5ac36114..3475422edd 100644
--- a/scene/register_scene_types.cpp
+++ b/scene/register_scene_types.cpp
@@ -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");
diff --git a/scene/resources/audio_stream_sample.cpp b/scene/resources/audio_stream_wav.cpp
index dcd36284d4..a87c8272ea 100644
--- a/scene/resources/audio_stream_sample.cpp
+++ b/scene/resources/audio_stream_wav.cpp
@@ -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;
diff --git a/scene/resources/audio_stream_sample.h b/scene/resources/audio_stream_wav.h
index 2e694cffe2..d800388d96 100644
--- a/scene/resources/audio_stream_sample.h
+++ b/scene/resources/audio_stream_wav.h
@@ -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
diff --git a/servers/audio/effects/audio_effect_record.cpp b/servers/audio/effects/audio_effect_record.cpp
index a6553e1431..fff6dbc32a 100644
--- a/servers/audio/effects/audio_effect_record.cpp
+++ b/servers/audio/effects/audio_effect_record.cpp
@@ -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;
 }
diff --git a/servers/audio/effects/audio_effect_record.h b/servers/audio/effects/audio_effect_record.h
index b23b63dbd8..e89d8adbde 100644
--- a/servers/audio/effects/audio_effect_record.h
+++ b/servers/audio/effects/audio_effect_record.h
@@ -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();
 };
diff --git a/servers/audio_server.cpp b/servers/audio_server.cpp
index 1054073377..9052f8e05e 100644
--- a/servers/audio_server.cpp
+++ b/servers/audio_server.cpp
@@ -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"
 
diff --git a/servers/audio_server.h b/servers/audio_server.h
index 287a18ecde..5613267909 100644
--- a/servers/audio_server.h
+++ b/servers/audio_server.h
@@ -43,7 +43,7 @@
 
 class AudioDriverDummy;
 class AudioStream;
-class AudioStreamSample;
+class AudioStreamWAV;
 class AudioStreamPlayback;
 
 class AudioDriver {