From efd1c3f8c3f649b0fa3fec0b236e9a748fc34e98 Mon Sep 17 00:00:00 2001
From: MerryMage <MerryMage@users.noreply.github.com>
Date: Thu, 24 Mar 2016 00:12:54 +0000
Subject: [PATCH] DSP: Implement audio codecs (PCM8, PCM16, ADPCM)
---
src/audio_core/CMakeLists.txt | 2 +
src/audio_core/codec.cpp | 122 ++++++++++++++++++++++++++++++++++
src/audio_core/codec.h | 50 ++++++++++++++
3 files changed, 174 insertions(+)
create mode 100644 src/audio_core/codec.cpp
create mode 100644 src/audio_core/codec.h
diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt
index b0d1c7eb66..c4bad8cb06 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 0000000000..ab65514b73
--- /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 0000000000..e695f2edcd
--- /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);
+
+};
--
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