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Add QOA (Quite OK Audio) as a WAV compression mode #91014

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merged 1 commit into from
May 2, 2024
Merged

Add QOA (Quite OK Audio) as a WAV compression mode #91014

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5 changes: 5 additions & 0 deletions COPYRIGHT.txt
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Original file line number Diff line number Diff line change
Expand Up @@ -412,6 +412,11 @@ Comment: PolyPartition / Triangulator
Copyright: 2011-2021, Ivan Fratric and contributors
License: Expat

Files: ./thirdparty/misc/qoa.h
Comment: Quite OK Audio Format
Copyright: 2023, Dominic Szablewski
License: Expat

Files: ./thirdparty/misc/r128.c
./thirdparty/misc/r128.h
Comment: r128 library
Expand Down
5 changes: 4 additions & 1 deletion doc/classes/AudioStreamWAV.xml
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Expand Up @@ -15,7 +15,7 @@
<return type="int" enum="Error" />
<param index="0" name="path" type="String" />
<description>
Saves the AudioStreamWAV as a WAV file to [param path]. Samples with IMA ADPCM format can't be saved.
Saves the AudioStreamWAV as a WAV file to [param path]. Samples with IMA ADPCM or QOA formats can't be saved.
[b]Note:[/b] A [code].wav[/code] extension is automatically appended to [param path] if it is missing.
</description>
</method>
Expand Down Expand Up @@ -56,6 +56,9 @@
<constant name="FORMAT_IMA_ADPCM" value="2" enum="Format">
Audio is compressed using IMA ADPCM.
</constant>
<constant name="FORMAT_QOA" value="3" enum="Format">
Audio is compressed as QOA ([url=https://qoaformat.org/]Quite OK Audio[/url]).
</constant>
<constant name="LOOP_DISABLED" value="0" enum="LoopMode">
Audio does not loop.
</constant>
Expand Down
1 change: 1 addition & 0 deletions doc/classes/ResourceImporterWAV.xml
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Expand Up @@ -14,6 +14,7 @@
The compression mode to use on import.
[b]Disabled:[/b] Imports audio data without any compression. This results in the highest possible quality.
[b]RAM (Ima-ADPCM):[/b] Performs fast lossy compression on import. Low CPU cost, but quality is noticeably decreased compared to Ogg Vorbis or even MP3.
[b]QOA ([url=https://qoaformat.org/]Quite OK Audio[/url]):[/b] Performs lossy compression on import. CPU cost is slightly higher than IMA-ADPCM, but quality is much higher.
</member>
<member name="edit/loop_begin" type="int" setter="" getter="" default="0">
The begin loop point to use when [member edit/loop_mode] is [b]Forward[/b], [b]Ping-Pong[/b] or [b]Backward[/b]. This is set in seconds after the beginning of the audio file.
Expand Down
34 changes: 26 additions & 8 deletions editor/import/resource_importer_wav.cpp
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Expand Up @@ -90,7 +90,7 @@ void ResourceImporterWAV::get_import_options(const String &p_path, List<ImportOp
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));
r_options->push_back(ImportOption(PropertyInfo(Variant::INT, "compress/mode", PROPERTY_HINT_ENUM, "Disabled,RAM (Ima-ADPCM)"), 0));
r_options->push_back(ImportOption(PropertyInfo(Variant::INT, "compress/mode", PROPERTY_HINT_ENUM, "Disabled,RAM (Ima-ADPCM),QOA (Quite OK Audio)"), 0));
}

Error ResourceImporterWAV::import(const String &p_source_file, const String &p_save_path, const HashMap<StringName, Variant> &p_options, List<String> *r_platform_variants, List<String> *r_gen_files, Variant *r_metadata) {
Expand Down Expand Up @@ -456,13 +456,13 @@ Error ResourceImporterWAV::import(const String &p_source_file, const String &p_s
is16 = false;
}

Vector<uint8_t> dst_data;
Vector<uint8_t> pcm_data;
AudioStreamWAV::Format dst_format;

if (compression == 1) {
dst_format = AudioStreamWAV::FORMAT_IMA_ADPCM;
if (format_channels == 1) {
_compress_ima_adpcm(data, dst_data);
_compress_ima_adpcm(data, pcm_data);
} else {
//byte interleave
Vector<float> left;
Expand All @@ -484,9 +484,9 @@ Error ResourceImporterWAV::import(const String &p_source_file, const String &p_s
_compress_ima_adpcm(right, bright);

int dl = bleft.size();
dst_data.resize(dl * 2);
pcm_data.resize(dl * 2);

uint8_t *w = dst_data.ptrw();
uint8_t *w = pcm_data.ptrw();
const uint8_t *rl = bleft.ptr();
const uint8_t *rr = bright.ptr();

Expand All @@ -498,13 +498,14 @@ Error ResourceImporterWAV::import(const String &p_source_file, const String &p_s

} else {
dst_format = is16 ? AudioStreamWAV::FORMAT_16_BITS : AudioStreamWAV::FORMAT_8_BITS;
dst_data.resize(data.size() * (is16 ? 2 : 1));
bool enforce16 = is16 || compression == 2;
pcm_data.resize(data.size() * (enforce16 ? 2 : 1));
{
uint8_t *w = dst_data.ptrw();
uint8_t *w = pcm_data.ptrw();

int ds = data.size();
for (int i = 0; i < ds; i++) {
if (is16) {
if (enforce16) {
int16_t v = CLAMP(data[i] * 32768, -32768, 32767);
encode_uint16(v, &w[i * 2]);
} else {
Expand All @@ -515,6 +516,23 @@ Error ResourceImporterWAV::import(const String &p_source_file, const String &p_s
}
}

Vector<uint8_t> dst_data;
if (compression == 2) {
dst_format = AudioStreamWAV::FORMAT_QOA;
qoa_desc desc = { 0, 0, 0, { { { 0 }, { 0 } } } };
uint32_t qoa_len = 0;

desc.samplerate = rate;
desc.samples = frames;
desc.channels = format_channels;

void *encoded = qoa_encode((short *)pcm_data.ptrw(), &desc, &qoa_len);
dst_data.resize(qoa_len);
memcpy(dst_data.ptrw(), encoded, qoa_len);
} else {
dst_data = pcm_data;
}

Ref<AudioStreamWAV> sample;
sample.instantiate();
sample->set_data(dst_data);
Expand Down
144 changes: 113 additions & 31 deletions scene/resources/audio_stream_wav.cpp
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Expand Up @@ -86,15 +86,15 @@ void AudioStreamPlaybackWAV::seek(double p_time) {
offset = uint64_t(p_time * base->mix_rate) << MIX_FRAC_BITS;
}

template <typename Depth, bool is_stereo, bool is_ima_adpcm>
void AudioStreamPlaybackWAV::do_resample(const Depth *p_src, AudioFrame *p_dst, int64_t &p_offset, int32_t &p_increment, uint32_t p_amount, IMA_ADPCM_State *p_ima_adpcm) {
template <typename Depth, bool is_stereo, bool is_ima_adpcm, bool is_qoa>
void AudioStreamPlaybackWAV::do_resample(const Depth *p_src, AudioFrame *p_dst, int64_t &p_offset, int32_t &p_increment, uint32_t p_amount, IMA_ADPCM_State *p_ima_adpcm, QOA_State *p_qoa) {
// this function will be compiled branchless by any decent compiler

int32_t final, final_r, next, next_r;
int32_t final = 0, final_r = 0, next = 0, next_r = 0;
while (p_amount) {
p_amount--;
int64_t pos = p_offset >> MIX_FRAC_BITS;
if (is_stereo && !is_ima_adpcm) {
if (is_stereo && !is_ima_adpcm && !is_qoa) {
pos <<= 1;
}

Expand Down Expand Up @@ -175,32 +175,77 @@ void AudioStreamPlaybackWAV::do_resample(const Depth *p_src, AudioFrame *p_dst,
}

} else {
final = p_src[pos];
if (is_stereo) {
final_r = p_src[pos + 1];
}
if (is_qoa) {
if (pos != p_qoa->cache_pos) { // Prevents triple decoding on lower mix rates.
for (int i = 0; i < 2; i++) {
// Sign operations prevent triple decoding on backward loops, maxing prevents pop.
uint32_t interp_pos = MIN(pos + (i * sign) + (sign < 0), p_qoa->desc->samples - 1);
uint32_t new_data_ofs = 8 + interp_pos / QOA_FRAME_LEN * p_qoa->frame_len;

if (p_qoa->data_ofs != new_data_ofs) {
p_qoa->data_ofs = new_data_ofs;
const uint8_t *src_ptr = (const uint8_t *)base->data;
src_ptr += p_qoa->data_ofs + AudioStreamWAV::DATA_PAD;
qoa_decode_frame(src_ptr, p_qoa->frame_len, p_qoa->desc, p_qoa->dec, &p_qoa->dec_len);
}

if constexpr (sizeof(Depth) == 1) { /* conditions will not exist anymore when compiled! */
final <<= 8;
uint32_t dec_idx = (interp_pos % QOA_FRAME_LEN) * p_qoa->desc->channels;

if ((sign > 0 && i == 0) || (sign < 0 && i == 1)) {
final = p_qoa->dec[dec_idx];
p_qoa->cache[0] = final;
if (is_stereo) {
final_r = p_qoa->dec[dec_idx + 1];
p_qoa->cache_r[0] = final_r;
}
} else {
next = p_qoa->dec[dec_idx];
p_qoa->cache[1] = next;
if (is_stereo) {
next_r = p_qoa->dec[dec_idx + 1];
p_qoa->cache_r[1] = next_r;
}
}
}
p_qoa->cache_pos = pos;
} else {
final = p_qoa->cache[0];
if (is_stereo) {
final_r = p_qoa->cache_r[0];
}

next = p_qoa->cache[1];
if (is_stereo) {
next_r = p_qoa->cache_r[1];
}
}
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} else {
final = p_src[pos];
if (is_stereo) {
final_r <<= 8;
final_r = p_src[pos + 1];
}
}

if (is_stereo) {
next = p_src[pos + 2];
next_r = p_src[pos + 3];
} else {
next = p_src[pos + 1];
}
if constexpr (sizeof(Depth) == 1) { /* conditions will not exist anymore when compiled! */
final <<= 8;
if (is_stereo) {
final_r <<= 8;
}
}

if constexpr (sizeof(Depth) == 1) {
next <<= 8;
if (is_stereo) {
next_r <<= 8;
next = p_src[pos + 2];
next_r = p_src[pos + 3];
} else {
next = p_src[pos + 1];
}
}

if constexpr (sizeof(Depth) == 1) {
next <<= 8;
if (is_stereo) {
next_r <<= 8;
}
}
}
int32_t frac = int64_t(p_offset & MIX_FRAC_MASK);

final = final + ((next - final) * frac >> MIX_FRAC_BITS);
Expand Down Expand Up @@ -240,6 +285,9 @@ int AudioStreamPlaybackWAV::mix(AudioFrame *p_buffer, float p_rate_scale, int p_
case AudioStreamWAV::FORMAT_IMA_ADPCM:
len *= 2;
break;
case AudioStreamWAV::FORMAT_QOA:
len = qoa.desc->samples * qoa.desc->channels;
break;
}

if (base->stereo) {
Expand Down Expand Up @@ -368,27 +416,34 @@ int AudioStreamPlaybackWAV::mix(AudioFrame *p_buffer, float p_rate_scale, int p_
switch (base->format) {
case AudioStreamWAV::FORMAT_8_BITS: {
if (is_stereo) {
do_resample<int8_t, true, false>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm);
do_resample<int8_t, true, false, false>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm, &qoa);
} else {
do_resample<int8_t, false, false>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm);
do_resample<int8_t, false, false, false>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm, &qoa);
}
} break;
case AudioStreamWAV::FORMAT_16_BITS: {
if (is_stereo) {
do_resample<int16_t, true, false>((int16_t *)data, dst_buff, offset, increment, target, ima_adpcm);
do_resample<int16_t, true, false, false>((int16_t *)data, dst_buff, offset, increment, target, ima_adpcm, &qoa);
} else {
do_resample<int16_t, false, false>((int16_t *)data, dst_buff, offset, increment, target, ima_adpcm);
do_resample<int16_t, false, false, false>((int16_t *)data, dst_buff, offset, increment, target, ima_adpcm, &qoa);
}

} break;
case AudioStreamWAV::FORMAT_IMA_ADPCM: {
if (is_stereo) {
do_resample<int8_t, true, true>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm);
do_resample<int8_t, true, true, false>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm, &qoa);
} else {
do_resample<int8_t, false, true>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm);
do_resample<int8_t, false, true, false>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm, &qoa);
}

} break;
case AudioStreamWAV::FORMAT_QOA: {
if (is_stereo) {
do_resample<uint8_t, true, false, true>((uint8_t *)data, dst_buff, offset, increment, target, ima_adpcm, &qoa);
} else {
do_resample<uint8_t, false, false, true>((uint8_t *)data, dst_buff, offset, increment, target, ima_adpcm, &qoa);
}
} break;
}

dst_buff += target;
Expand All @@ -412,6 +467,16 @@ void AudioStreamPlaybackWAV::tag_used_streams() {

AudioStreamPlaybackWAV::AudioStreamPlaybackWAV() {}

AudioStreamPlaybackWAV::~AudioStreamPlaybackWAV() {
if (qoa.desc) {
memfree(qoa.desc);
}

if (qoa.dec) {
memfree(qoa.dec);
}
}

/////////////////////

void AudioStreamWAV::set_format(Format p_format) {
Expand Down Expand Up @@ -475,6 +540,10 @@ double AudioStreamWAV::get_length() const {
case AudioStreamWAV::FORMAT_IMA_ADPCM:
len *= 2;
break;
case AudioStreamWAV::FORMAT_QOA:
qoa_desc desc = { 0, 0, 0, { { { 0 }, { 0 } } } };
qoa_decode_header((uint8_t *)data + DATA_PAD, QOA_MIN_FILESIZE, &desc);
len = desc.samples * desc.channels;
}

if (stereo) {
Expand Down Expand Up @@ -526,8 +595,8 @@ Vector<uint8_t> AudioStreamWAV::get_data() const {
}

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");
if (format == AudioStreamWAV::FORMAT_IMA_ADPCM || format == AudioStreamWAV::FORMAT_QOA) {
WARN_PRINT("Saving IMA_ADPCM and QOA samples is not supported yet");
return ERR_UNAVAILABLE;
}

Expand All @@ -548,6 +617,7 @@ Error AudioStreamWAV::save_to_wav(const String &p_path) {
byte_pr_sample = 1;
break;
case AudioStreamWAV::FORMAT_16_BITS:
case AudioStreamWAV::FORMAT_QOA:
byte_pr_sample = 2;
break;
case AudioStreamWAV::FORMAT_IMA_ADPCM:
Expand Down Expand Up @@ -590,6 +660,7 @@ Error AudioStreamWAV::save_to_wav(const String &p_path) {
}
break;
case AudioStreamWAV::FORMAT_16_BITS:
case AudioStreamWAV::FORMAT_QOA:
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);
Expand All @@ -607,6 +678,16 @@ Ref<AudioStreamPlayback> AudioStreamWAV::instantiate_playback() {
Ref<AudioStreamPlaybackWAV> sample;
sample.instantiate();
sample->base = Ref<AudioStreamWAV>(this);

if (format == AudioStreamWAV::FORMAT_QOA) {
sample->qoa.desc = (qoa_desc *)memalloc(sizeof(qoa_desc));
qoa_decode_header((uint8_t *)data + DATA_PAD, QOA_MIN_FILESIZE, sample->qoa.desc);
sample->qoa.frame_len = qoa_max_frame_size(sample->qoa.desc);
int samples_len = (sample->qoa.desc->samples > QOA_FRAME_LEN ? QOA_FRAME_LEN : sample->qoa.desc->samples);
int alloc_len = sample->qoa.desc->channels * samples_len * sizeof(int16_t);
sample->qoa.dec = (int16_t *)memalloc(alloc_len);
}

return sample;
}

Expand Down Expand Up @@ -639,7 +720,7 @@ void AudioStreamWAV::_bind_methods() {
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");
ADD_PROPERTY(PropertyInfo(Variant::INT, "format", PROPERTY_HINT_ENUM, "8-Bit,16-Bit,IMA-ADPCM,QOA"), "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");
Expand All @@ -649,6 +730,7 @@ void AudioStreamWAV::_bind_methods() {
BIND_ENUM_CONSTANT(FORMAT_8_BITS);
BIND_ENUM_CONSTANT(FORMAT_16_BITS);
BIND_ENUM_CONSTANT(FORMAT_IMA_ADPCM);
BIND_ENUM_CONSTANT(FORMAT_QOA);

BIND_ENUM_CONSTANT(LOOP_DISABLED);
BIND_ENUM_CONSTANT(LOOP_FORWARD);
Expand Down
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