diff options
author | bunnei <bunneidev@gmail.com> | 2016-04-28 15:47:08 +0200 |
---|---|---|
committer | bunnei <bunneidev@gmail.com> | 2016-04-28 15:47:08 +0200 |
commit | fda578e19d236d2c2f138c126f926e638ca3a818 (patch) | |
tree | 3c502f28221b915766c0503e534eb99f62752cf2 /src | |
parent | Merge pull request #1722 from MerryMage/soundtouch (diff) | |
parent | AudioCore: Move samples_per_frame and num_sources into hle/common.h (diff) | |
download | yuzu-fda578e19d236d2c2f138c126f926e638ca3a818.tar yuzu-fda578e19d236d2c2f138c126f926e638ca3a818.tar.gz yuzu-fda578e19d236d2c2f138c126f926e638ca3a818.tar.bz2 yuzu-fda578e19d236d2c2f138c126f926e638ca3a818.tar.lz yuzu-fda578e19d236d2c2f138c126f926e638ca3a818.tar.xz yuzu-fda578e19d236d2c2f138c126f926e638ca3a818.tar.zst yuzu-fda578e19d236d2c2f138c126f926e638ca3a818.zip |
Diffstat (limited to 'src')
-rw-r--r-- | src/audio_core/audio_core.h | 2 | ||||
-rw-r--r-- | src/audio_core/hle/common.h | 9 | ||||
-rw-r--r-- | src/audio_core/hle/dsp.h | 12 |
3 files changed, 11 insertions, 12 deletions
diff --git a/src/audio_core/audio_core.h b/src/audio_core/audio_core.h index 64c330914..b349895ea 100644 --- a/src/audio_core/audio_core.h +++ b/src/audio_core/audio_core.h @@ -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 diff --git a/src/audio_core/hle/common.h b/src/audio_core/hle/common.h index 37d441eb2..7910f42ae 100644 --- a/src/audio_core/hle/common.h +++ b/src/audio_core/hle/common.h @@ -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. diff --git a/src/audio_core/hle/dsp.h b/src/audio_core/hle/dsp.h index c15ef0b7a..c76350bdd 100644 --- a/src/audio_core/hle/dsp.h +++ b/src/audio_core/hle/dsp.h @@ -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; |