The-Homebrew-Cloud/Lime3DS
2
0
Fork 0
mirror of https://github.com/Lime3DS/Lime3DS.git synced 2024-11-27 20:14:17 +01:00
Code Issues Packages Projects Releases Wiki Activity

Merge pull request #1727 from MerryMage/minor-commit

Browse Source 
AudioCore: Move samples_per_frame and num_sources into hle/common.h
This commit is contained in:
bunnei 2016-04-28 09:47:08 -04:00
commit fda578e19d
3 changed files with 11 additions and 12 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
src/audio_core/audio_core.h
View File

@ -10,8 +10,6 @@ class VMManager;
namespace AudioCore {
constexpr int num_sources = 24;
constexpr int samples_per_frame = 160; ///< Samples per audio frame at native sample rate
constexpr int native_sample_rate = 32728; ///< 32kHz
/// Initialise Audio Core

9
src/audio_core/hle/common.h
View File

@ -7,18 +7,19 @@
#include <algorithm>
#include <array>
#include "audio_core/audio_core.h"
#include "common/common_types.h"
namespace DSP {
namespace HLE {
constexpr int num_sources = 24;
constexpr int samples_per_frame = 160; ///< Samples per audio frame at native sample rate
/// The final output to the speakers is stereo. Preprocessing output in Source is also stereo.
using StereoFrame16 = std::array<std::array<s16, 2>, AudioCore::samples_per_frame>;
using StereoFrame16 = std::array<std::array<s16, 2>, samples_per_frame>;
/// The DSP is quadraphonic internally.
using QuadFrame32 = std::array<std::array<s32, 4>, AudioCore::samples_per_frame>;
using QuadFrame32 = std::array<std::array<s32, 4>, samples_per_frame>;
/**
* This performs the filter operation defined by FilterT::ProcessSample on the frame in-place.

12
src/audio_core/hle/dsp.h
View File

@ -7,7 +7,7 @@
#include <cstddef>
#include <type_traits>
#include "audio_core/audio_core.h"
#include "audio_core/hle/common.h"
#include "common/bit_field.h"
#include "common/common_funcs.h"
@ -305,7 +305,7 @@ struct SourceConfiguration {
u16_le buffer_id;
};
Configuration config[AudioCore::num_sources];
Configuration config[num_sources];
};
ASSERT_DSP_STRUCT(SourceConfiguration::Configuration, 192);
ASSERT_DSP_STRUCT(SourceConfiguration::Configuration::Buffer, 20);
@ -320,7 +320,7 @@ struct SourceStatus {
INSERT_PADDING_DSPWORDS(1);
};
Status status[AudioCore::num_sources];
Status status[num_sources];
};
ASSERT_DSP_STRUCT(SourceStatus::Status, 12);
@ -413,7 +413,7 @@ ASSERT_DSP_STRUCT(DspConfiguration::ReverbEffect, 52);
struct AdpcmCoefficients {
/// Coefficients are signed fixed point with 11 fractional bits.
/// Each source has 16 coefficients associated with it.
s16_le coeff[AudioCore::num_sources][16];
s16_le coeff[num_sources][16];
};
ASSERT_DSP_STRUCT(AdpcmCoefficients, 768);
@ -427,7 +427,7 @@ ASSERT_DSP_STRUCT(DspStatus, 32);
/// Final mixed output in PCM16 stereo format, what you hear out of the speakers.
/// When the application writes to this region it has no effect.
struct FinalMixSamples {
s16_le pcm16[2 * AudioCore::samples_per_frame];
s16_le pcm16[2 * samples_per_frame];
};
ASSERT_DSP_STRUCT(FinalMixSamples, 640);
@ -437,7 +437,7 @@ ASSERT_DSP_STRUCT(FinalMixSamples, 640);
/// Values that exceed s16 range will be clipped by the DSP after further processing.
struct IntermediateMixSamples {
struct Samples {
s32_le pcm32[4][AudioCore::samples_per_frame]; ///< Little-endian as opposed to DSP middle-endian.
s32_le pcm32[4][samples_per_frame]; ///< Little-endian as opposed to DSP middle-endian.
};
Samples mix1;