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-rw-r--r--src/audio_core/hle/source.cpp49
-rw-r--r--src/audio_core/interpolate.cpp86
-rw-r--r--src/audio_core/interpolate.h27
3 files changed, 74 insertions, 88 deletions
diff --git a/src/audio_core/hle/source.cpp b/src/audio_core/hle/source.cpp
index 92484c526..de4e88cae 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 += 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/interpolate.cpp b/src/audio_core/interpolate.cpp
index 8a5d4181a..16e68bc5c 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(), 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..59f59bc14 100644
--- a/src/audio_core/interpolate.h
+++ b/src/audio_core/interpolate.h
@@ -6,6 +6,7 @@
#include <array>
#include <vector>
+#include "audio_core/hle/common.h"
#include "common/common_types.h"
namespace AudioInterp {
@@ -14,31 +15,35 @@ namespace AudioInterp {
using StereoBuffer16 = std::vector<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
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-17 06:00:47 +0100