DSP: Implement audio codecs (PCM8, PCM16, ADPCM)

3 files changed, 174 insertions, 0 deletions

diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt
index b0d1c7eb6..c4bad8cb0 100644
--- a/src/audio_core/CMakeLists.txt
+++ b/src/audio_core/CMakeLists.txt
@@ -1,11 +1,13 @@
 set(SRCS
             audio_core.cpp
+            codec.cpp
             hle/dsp.cpp
             hle/pipe.cpp
             )
 
 set(HEADERS
             audio_core.h
+            codec.h
             hle/dsp.h
             hle/pipe.h
             sink.h
diff --git a/src/audio_core/codec.cpp b/src/audio_core/codec.cpp
new file mode 100644
index 000000000..ab65514b7
--- /dev/null
+++ b/src/audio_core/codec.cpp
@@ -0,0 +1,122 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <array>
+#include <cstddef>
+#include <cstring>
+#include <vector>
+
+#include "audio_core/codec.h"
+
+#include "common/assert.h"
+#include "common/common_types.h"
+#include "common/math_util.h"
+
+namespace Codec {
+
+StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count, const std::array<s16, 16>& adpcm_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 ret_size = sample_count % 2 == 0 ? sample_count : sample_count + 1; // Ensure multiple of two.
+    StereoBuffer16 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 = adpcm_coeff[idx * 2 + 0];
+        const int coef2 = adpcm_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 = MathUtil::Clamp(val, -32768, 32767);
+            // Advance output feedback.
+            yn2 = yn1;
+            yn1 = val;
+            return (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] & 0xF]);
+            ret[outputi].fill(sample1);
+            outputi++;
+
+            const s16 sample2 = decode_sample(SIGNED_NIBBLES[data[datai] >> 4]);
+            ret[outputi].fill(sample2);
+            outputi++;
+
+            datai++;
+        }
+    }
+
+    state.yn1 = yn1;
+    state.yn2 = yn2;
+
+    return ret;
+}
+
+static s16 SignExtendS8(u8 x) {
+    // The data is actually signed PCM8.
+    // We sign extend this to signed PCM16.
+    return static_cast<s16>(static_cast<s8>(x));
+}
+
+StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data, const size_t sample_count) {
+    ASSERT(num_channels == 1 || num_channels == 2);
+
+    StereoBuffer16 ret(sample_count);
+
+    if (num_channels == 1) {
+        for (size_t i = 0; i < sample_count; i++) {
+            ret[i].fill(SignExtendS8(data[i]));
+        }
+    } else {
+        for (size_t i = 0; i < sample_count; i++) {
+            ret[i][0] = SignExtendS8(data[i * 2 + 0]);
+            ret[i][1] = SignExtendS8(data[i * 2 + 1]);
+        }
+    }
+
+    return ret;
+}
+
+StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data, const size_t sample_count) {
+    ASSERT(num_channels == 1 || num_channels == 2);
+
+    StereoBuffer16 ret(sample_count);
+
+    if (num_channels == 1) {
+        for (size_t i = 0; i < sample_count; i++) {
+            s16 sample;
+            std::memcpy(&sample, data + i * sizeof(s16), sizeof(s16));
+            ret[i].fill(sample);
+        }
+    } else {
+        std::memcpy(ret.data(), data, sample_count * 2 * sizeof(u16));
+    }
+
+    return ret;
+}
+
+};
diff --git a/src/audio_core/codec.h b/src/audio_core/codec.h
new file mode 100644
index 000000000..e695f2edc
--- /dev/null
+++ b/src/audio_core/codec.h
@@ -0,0 +1,50 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <array>
+#include <vector>
+
+#include "common/common_types.h"
+
+namespace Codec {
+
+/// A variable length buffer of signed PCM16 stereo samples.
+using StereoBuffer16 = std::vector<std::array<s16, 2>>;
+
+/// See: Codec::DecodeADPCM
+struct ADPCMState {
+    // Two historical samples from previous processed buffer,
+    // required for ADPCM decoding
+    s16 yn1; ///< y[n-1]
+    s16 yn2; ///< y[n-2]
+};
+
+/**
+ * @param data Pointer to buffer that contains ADPCM data to decode
+ * @param sample_count Length of buffer in terms of number of samples
+ * @param adpcm_coeff ADPCM coefficients
+ * @param state ADPCM state, this is updated with new state
+ * @return Decoded stereo signed PCM16 data, sample_count in length
+ */
+StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count, const std::array<s16, 16>& adpcm_coeff, ADPCMState& state);
+
+/**
+ * @param num_channels Number of channels
+ * @param data Pointer to buffer that contains PCM8 data to decode
+ * @param sample_count Length of buffer in terms of number of samples
+ * @return Decoded stereo signed PCM16 data, sample_count in length
+ */
+StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data, const size_t sample_count);
+
+/**
+ * @param num_channels Number of channels
+ * @param data Pointer to buffer that contains PCM16 data to decode
+ * @param sample_count Length of buffer in terms of number of samples
+ * @return Decoded stereo signed PCM16 data, sample_count in length
+ */
+StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data, const size_t sample_count);
+
+};