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authorbunnei <bunneidev@gmail.com>2018-08-03 21:30:01 +0200
committerbunnei <bunneidev@gmail.com>2018-08-05 00:22:58 +0200
commitf1cb3903ac358183dcdc562ba19dc469b056e73f (patch)
tree0179ee55d3c573cdd9189cb708e2b69e02a1b166 /src/audio_core/codec.cpp
parentcubeb_sink: Support variable sample_rate and num_channels. (diff)
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-rw-r--r--src/audio_core/codec.cpp77
1 files changed, 77 insertions, 0 deletions
diff --git a/src/audio_core/codec.cpp b/src/audio_core/codec.cpp
new file mode 100644
index 000000000..c3021403f
--- /dev/null
+++ b/src/audio_core/codec.cpp
@@ -0,0 +1,77 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <algorithm>
+
+#include "audio_core/codec.h"
+
+namespace AudioCore::Codec {
+
+std::vector<s16> DecodeADPCM(const u8* const data, size_t size, const ADPCM_Coeff& coeff,
+ ADPCMState& state) {
+ // GC-ADPCM with scale factor and variable coefficients.
+ // Frames are 8 bytes long containing 14 samples each.
+ // Samples are 4 bits (one nibble) long.
+
+ constexpr size_t FRAME_LEN = 8;
+ constexpr size_t SAMPLES_PER_FRAME = 14;
+ constexpr std::array<int, 16> SIGNED_NIBBLES = {
+ {0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1}};
+
+ const size_t sample_count = (size / FRAME_LEN) * SAMPLES_PER_FRAME;
+ const size_t ret_size =
+ sample_count % 2 == 0 ? sample_count : sample_count + 1; // Ensure multiple of two.
+ std::vector<s16> ret(ret_size);
+
+ int yn1 = state.yn1, yn2 = state.yn2;
+
+ const size_t NUM_FRAMES =
+ (sample_count + (SAMPLES_PER_FRAME - 1)) / SAMPLES_PER_FRAME; // Round up.
+ for (size_t framei = 0; framei < NUM_FRAMES; framei++) {
+ const int frame_header = data[framei * FRAME_LEN];
+ const int scale = 1 << (frame_header & 0xF);
+ const int idx = (frame_header >> 4) & 0x7;
+
+ // Coefficients are fixed point with 11 bits fractional part.
+ const int coef1 = coeff[idx * 2 + 0];
+ const int coef2 = coeff[idx * 2 + 1];
+
+ // Decodes an audio sample. One nibble produces one sample.
+ const auto decode_sample = [&](const int nibble) -> s16 {
+ const int xn = nibble * scale;
+ // We first transform everything into 11 bit fixed point, perform the second order
+ // digital filter, then transform back.
+ // 0x400 == 0.5 in 11 bit fixed point.
+ // Filter: y[n] = x[n] + 0.5 + c1 * y[n-1] + c2 * y[n-2]
+ int val = ((xn << 11) + 0x400 + coef1 * yn1 + coef2 * yn2) >> 11;
+ // Clamp to output range.
+ val = std::clamp<s32>(val, -32768, 32767);
+ // Advance output feedback.
+ yn2 = yn1;
+ yn1 = val;
+ return static_cast<s16>(val);
+ };
+
+ size_t outputi = framei * SAMPLES_PER_FRAME;
+ size_t datai = framei * FRAME_LEN + 1;
+ for (size_t i = 0; i < SAMPLES_PER_FRAME && outputi < sample_count; i += 2) {
+ const s16 sample1 = decode_sample(SIGNED_NIBBLES[data[datai] >> 4]);
+ ret[outputi] = sample1;
+ outputi++;
+
+ const s16 sample2 = decode_sample(SIGNED_NIBBLES[data[datai] & 0xF]);
+ ret[outputi] = sample2;
+ outputi++;
+
+ datai++;
+ }
+ }
+
+ state.yn1 = yn1;
+ state.yn2 = yn2;
+
+ return ret;
+}
+
+} // namespace AudioCore::Codec