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AudioCore: Move samples_per_frame and num_sources into hle/common.h

This commit is contained in:
MerryMage 2016-04-24 13:20:13 +01:00
parent 1ecee38468
commit dda9ffe790
3 changed files with 11 additions and 12 deletions
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2
src/audio_core/audio_core.h
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@ -10,8 +10,6 @@ class VMManager;
namespace AudioCore { 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 constexpr int native_sample_rate = 32728; ///< 32kHz
/// Initialise Audio Core /// Initialise Audio Core

9
src/audio_core/hle/common.h
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@ -7,18 +7,19 @@
#include <algorithm> #include <algorithm>
#include <array> #include <array>
#include "audio_core/audio_core.h"
#include "common/common_types.h" #include "common/common_types.h"
namespace DSP { namespace DSP {
namespace HLE { 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. /// 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. /// 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. * This performs the filter operation defined by FilterT::ProcessSample on the frame in-place.

12
src/audio_core/hle/dsp.h
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@ -7,7 +7,7 @@
#include <cstddef> #include <cstddef>
#include <type_traits> #include <type_traits>
#include "audio_core/audio_core.h" #include "audio_core/hle/common.h"
#include "common/bit_field.h" #include "common/bit_field.h"
#include "common/common_funcs.h" #include "common/common_funcs.h"
@ -305,7 +305,7 @@ struct SourceConfiguration {
u16_le buffer_id; u16_le buffer_id;
}; };
Configuration config[AudioCore::num_sources]; Configuration config[num_sources];
}; };
ASSERT_DSP_STRUCT(SourceConfiguration::Configuration, 192); ASSERT_DSP_STRUCT(SourceConfiguration::Configuration, 192);
ASSERT_DSP_STRUCT(SourceConfiguration::Configuration::Buffer, 20); ASSERT_DSP_STRUCT(SourceConfiguration::Configuration::Buffer, 20);
@ -320,7 +320,7 @@ struct SourceStatus {
INSERT_PADDING_DSPWORDS(1); INSERT_PADDING_DSPWORDS(1);
}; };
Status status[AudioCore::num_sources]; Status status[num_sources];
}; };
ASSERT_DSP_STRUCT(SourceStatus::Status, 12); ASSERT_DSP_STRUCT(SourceStatus::Status, 12);
@ -413,7 +413,7 @@ ASSERT_DSP_STRUCT(DspConfiguration::ReverbEffect, 52);
struct AdpcmCoefficients { struct AdpcmCoefficients {
/// Coefficients are signed fixed point with 11 fractional bits. /// Coefficients are signed fixed point with 11 fractional bits.
/// Each source has 16 coefficients associated with it. /// 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); 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. /// 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. /// When the application writes to this region it has no effect.
struct FinalMixSamples { struct FinalMixSamples {
s16_le pcm16[2 * AudioCore::samples_per_frame]; s16_le pcm16[2 * samples_per_frame];
}; };
ASSERT_DSP_STRUCT(FinalMixSamples, 640); 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. /// Values that exceed s16 range will be clipped by the DSP after further processing.
struct IntermediateMixSamples { struct IntermediateMixSamples {
struct Samples { 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; Samples mix1;