Merge remote-tracking branch 'upstream/master' into nx

# Conflicts:
#	src/core/CMakeLists.txt
#	src/core/arm/dynarmic/arm_dynarmic.cpp
#	src/core/arm/dyncom/arm_dyncom.cpp
#	src/core/hle/kernel/process.cpp
#	src/core/hle/kernel/thread.cpp
#	src/core/hle/kernel/thread.h
#	src/core/hle/kernel/vm_manager.cpp
#	src/core/loader/3dsx.cpp
#	src/core/loader/elf.cpp
#	src/core/loader/ncch.cpp
#	src/core/memory.cpp
#	src/core/memory.h
#	src/core/memory_setup.h

6 files changed, 82 insertions, 94 deletions

diff --git a/src/audio_core/codec.cpp b/src/audio_core/codec.cpp
index 7a3bd7eb3..6fba9fdae 100644
--- a/src/audio_core/codec.cpp
+++ b/src/audio_core/codec.cpp
@@ -117,7 +117,9 @@ StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data,
             ret[i].fill(sample);
         }
     } else {
-        std::memcpy(ret.data(), data, sample_count * 2 * sizeof(u16));
+        for (size_t i = 0; i < sample_count; ++i) {
+            std::memcpy(&ret[i], data + i * sizeof(s16) * 2, 2 * sizeof(s16));
+        }
     }
 
     return ret;
diff --git a/src/audio_core/codec.h b/src/audio_core/codec.h
index 2b0c395e6..877b2202d 100644
--- a/src/audio_core/codec.h
+++ b/src/audio_core/codec.h
@@ -5,13 +5,13 @@
 #pragma once
 
 #include <array>
-#include <vector>
+#include <deque>
 #include "common/common_types.h"
 
 namespace Codec {
 
 /// A variable length buffer of signed PCM16 stereo samples.
-using StereoBuffer16 = std::vector<std::array<s16, 2>>;
+using StereoBuffer16 = std::deque<std::array<s16, 2>>;
 
 /// See: Codec::DecodeADPCM
 struct ADPCMState {
diff --git a/src/audio_core/hle/source.cpp b/src/audio_core/hle/source.cpp
index 92484c526..c12287700 100644
--- a/src/audio_core/hle/source.cpp
+++ b/src/audio_core/hle/source.cpp
@@ -244,17 +244,27 @@ void Source::GenerateFrame() {
             break;
         }
 
-        const size_t size_to_copy =
-            std::min(state.current_buffer.size(), current_frame.size() - frame_position);
-
-        std::copy(state.current_buffer.begin(), state.current_buffer.begin() + size_to_copy,
-                  current_frame.begin() + frame_position);
-        state.current_buffer.erase(state.current_buffer.begin(),
-                                   state.current_buffer.begin() + size_to_copy);
-
-        frame_position += size_to_copy;
-        state.next_sample_number += static_cast<u32>(size_to_copy);
+        switch (state.interpolation_mode) {
+        case InterpolationMode::None:
+            AudioInterp::None(state.interp_state, state.current_buffer, state.rate_multiplier,
+                              current_frame, frame_position);
+            break;
+        case InterpolationMode::Linear:
+            AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier,
+                                current_frame, frame_position);
+            break;
+        case InterpolationMode::Polyphase:
+            // TODO(merry): Implement polyphase interpolation
+            LOG_DEBUG(Audio_DSP, "Polyphase interpolation unimplemented; falling back to linear");
+            AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier,
+                                current_frame, frame_position);
+            break;
+        default:
+            UNIMPLEMENTED();
+            break;
+        }
     }
+    state.next_sample_number += static_cast<u32>(frame_position);
 
     state.filters.ProcessFrame(current_frame);
 }
@@ -305,25 +315,6 @@ bool Source::DequeueBuffer() {
         return true;
     }
 
-    switch (state.interpolation_mode) {
-    case InterpolationMode::None:
-        state.current_buffer =
-            AudioInterp::None(state.interp_state, state.current_buffer, state.rate_multiplier);
-        break;
-    case InterpolationMode::Linear:
-        state.current_buffer =
-            AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier);
-        break;
-    case InterpolationMode::Polyphase:
-        // TODO(merry): Implement polyphase interpolation
-        state.current_buffer =
-            AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier);
-        break;
-    default:
-        UNIMPLEMENTED();
-        break;
-    }
-
     // the first playthrough starts at play_position, loops start at the beginning of the buffer
     state.current_sample_number = (!buf.has_played) ? buf.play_position : 0;
     state.next_sample_number = state.current_sample_number;
diff --git a/src/audio_core/hle/source.h b/src/audio_core/hle/source.h
index ccb7f064f..c4d2debc2 100644
--- a/src/audio_core/hle/source.h
+++ b/src/audio_core/hle/source.h
@@ -108,7 +108,7 @@ private:
 
         u32 current_sample_number = 0;
         u32 next_sample_number = 0;
-        std::vector<std::array<s16, 2>> current_buffer;
+        AudioInterp::StereoBuffer16 current_buffer;
 
         // buffer_id state
 
diff --git a/src/audio_core/interpolate.cpp b/src/audio_core/interpolate.cpp
index 8a5d4181a..83573d772 100644
--- a/src/audio_core/interpolate.cpp
+++ b/src/audio_core/interpolate.cpp
@@ -13,74 +13,64 @@ namespace AudioInterp {
 constexpr u64 scale_factor = 1 << 24;
 constexpr u64 scale_mask = scale_factor - 1;
 
-/// Here we step over the input in steps of rate_multiplier, until we consume all of the input.
+/// Here we step over the input in steps of rate, until we consume all of the input.
 /// Three adjacent samples are passed to fn each step.
 template <typename Function>
-static StereoBuffer16 StepOverSamples(State& state, const StereoBuffer16& input,
-                                      float rate_multiplier, Function fn) {
-    ASSERT(rate_multiplier > 0);
+static void StepOverSamples(State& state, StereoBuffer16& input, float rate,
+                            DSP::HLE::StereoFrame16& output, size_t& outputi, Function fn) {
+    ASSERT(rate > 0);
 
-    if (input.size() < 2)
-        return {};
+    if (input.empty())
+        return;
 
-    StereoBuffer16 output;
-    output.reserve(static_cast<size_t>(input.size() / rate_multiplier));
+    input.insert(input.begin(), {state.xn2, state.xn1});
 
-    u64 step_size = static_cast<u64>(rate_multiplier * scale_factor);
+    const u64 step_size = static_cast<u64>(rate * scale_factor);
+    u64 fposition = state.fposition;
+    size_t inputi = 0;
 
-    u64 fposition = 0;
-    const u64 max_fposition = input.size() * scale_factor;
+    while (outputi < output.size()) {
+        inputi = static_cast<size_t>(fposition / scale_factor);
 
-    while (fposition < 1 * scale_factor) {
-        u64 fraction = fposition & scale_mask;
-
-        output.push_back(fn(fraction, state.xn2, state.xn1, input[0]));
-
-        fposition += step_size;
-    }
-
-    while (fposition < 2 * scale_factor) {
-        u64 fraction = fposition & scale_mask;
-
-        output.push_back(fn(fraction, state.xn1, input[0], input[1]));
-
-        fposition += step_size;
-    }
+        if (inputi + 2 >= input.size()) {
+            inputi = input.size() - 2;
+            break;
+        }
 
-    while (fposition < max_fposition) {
         u64 fraction = fposition & scale_mask;
-
-        size_t index = static_cast<size_t>(fposition / scale_factor);
-        output.push_back(fn(fraction, input[index - 2], input[index - 1], input[index]));
+        output[outputi++] = fn(fraction, input[inputi], input[inputi + 1], input[inputi + 2]);
 
         fposition += step_size;
     }
 
-    state.xn2 = input[input.size() - 2];
-    state.xn1 = input[input.size() - 1];
+    state.xn2 = input[inputi];
+    state.xn1 = input[inputi + 1];
+    state.fposition = fposition - inputi * scale_factor;
 
-    return output;
+    input.erase(input.begin(), std::next(input.begin(), inputi + 2));
 }
 
-StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier) {
-    return StepOverSamples(
-        state, input, rate_multiplier,
+void None(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
+          size_t& outputi) {
+    StepOverSamples(
+        state, input, rate, output, outputi,
         [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) { return x0; });
 }
 
-StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier) {
+void Linear(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
+            size_t& outputi) {
     // Note on accuracy: Some values that this produces are +/- 1 from the actual firmware.
-    return StepOverSamples(state, input, rate_multiplier,
-                           [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) {
-                               // This is a saturated subtraction. (Verified by black-box fuzzing.)
-                               s64 delta0 = MathUtil::Clamp<s64>(x1[0] - x0[0], -32768, 32767);
-                               s64 delta1 = MathUtil::Clamp<s64>(x1[1] - x0[1], -32768, 32767);
-
-                               return std::array<s16, 2>{
-                                   static_cast<s16>(x0[0] + fraction * delta0 / scale_factor),
-                                   static_cast<s16>(x0[1] + fraction * delta1 / scale_factor),
-                               };
-                           });
+    StepOverSamples(state, input, rate, output, outputi,
+                    [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) {
+                        // This is a saturated subtraction. (Verified by black-box fuzzing.)
+                        s64 delta0 = MathUtil::Clamp<s64>(x1[0] - x0[0], -32768, 32767);
+                        s64 delta1 = MathUtil::Clamp<s64>(x1[1] - x0[1], -32768, 32767);
+
+                        return std::array<s16, 2>{
+                            static_cast<s16>(x0[0] + fraction * delta0 / scale_factor),
+                            static_cast<s16>(x0[1] + fraction * delta1 / scale_factor),
+                        };
+                    });
 }
 
 } // namespace AudioInterp
diff --git a/src/audio_core/interpolate.h b/src/audio_core/interpolate.h
index 19a7b66cb..8dff6111a 100644
--- a/src/audio_core/interpolate.h
+++ b/src/audio_core/interpolate.h
@@ -5,40 +5,45 @@
 #pragma once
 
 #include <array>
-#include <vector>
+#include <deque>
+#include "audio_core/hle/common.h"
 #include "common/common_types.h"
 
 namespace AudioInterp {
 
 /// A variable length buffer of signed PCM16 stereo samples.
-using StereoBuffer16 = std::vector<std::array<s16, 2>>;
+using StereoBuffer16 = std::deque<std::array<s16, 2>>;
 
 struct State {
-    // Two historical samples.
+    /// Two historical samples.
     std::array<s16, 2> xn1 = {}; ///< x[n-1]
     std::array<s16, 2> xn2 = {}; ///< x[n-2]
+    /// Current fractional position.
+    u64 fposition = 0;
 };
 
 /**
  * No interpolation. This is equivalent to a zero-order hold. There is a two-sample predelay.
  * @param state Interpolation state.
  * @param input Input buffer.
- * @param rate_multiplier Stretch factor. Must be a positive non-zero value.
- *                        rate_multiplier > 1.0 performs decimation and rate_multipler < 1.0
- *                        performs upsampling.
- * @return The resampled audio buffer.
+ * @param rate Stretch factor. Must be a positive non-zero value.
+ *             rate > 1.0 performs decimation and rate < 1.0 performs upsampling.
+ * @param output The resampled audio buffer.
+ * @param outputi The index of output to start writing to.
  */
-StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier);
+void None(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
+          size_t& outputi);
 
 /**
  * Linear interpolation. This is equivalent to a first-order hold. There is a two-sample predelay.
  * @param state Interpolation state.
  * @param input Input buffer.
- * @param rate_multiplier Stretch factor. Must be a positive non-zero value.
- *                        rate_multiplier > 1.0 performs decimation and rate_multipler < 1.0
- *                        performs upsampling.
- * @return The resampled audio buffer.
+ * @param rate Stretch factor. Must be a positive non-zero value.
+ *             rate > 1.0 performs decimation and rate < 1.0 performs upsampling.
+ * @param output The resampled audio buffer.
+ * @param outputi The index of output to start writing to.
  */
-StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier);
+void Linear(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
+            size_t& outputi);
 
 } // namespace AudioInterp