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Sources: Run clang-format on everything.

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This commit is contained in:
Emmanuel Gil Peyrot 2016-09-18 09:38:01 +09:00
parent fe948af095
commit dc8479928c
386 changed files with 19560 additions and 18080 deletions
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25
src/audio_core/codec.cpp
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@ -15,22 +15,25 @@
namespace Codec {
StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count, const std::array<s16, 16>& adpcm_coeff, ADPCMState& state) {
StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count,
const std::array<s16, 16>& adpcm_coeff, ADPCMState& state) {
// GC-ADPCM with scale factor and variable coefficients.
// Frames are 8 bytes long containing 14 samples each.
// Samples are 4 bits (one nibble) long.
constexpr size_t FRAME_LEN = 8;
constexpr size_t SAMPLES_PER_FRAME = 14;
constexpr std::array<int, 16> SIGNED_NIBBLES {{ 0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1 }};
constexpr std::array<int, 16> SIGNED_NIBBLES{
{0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1}};
const size_t ret_size = sample_count % 2 == 0 ? sample_count : sample_count + 1; // Ensure multiple of two.
const size_t ret_size =
sample_count % 2 == 0 ? sample_count : sample_count + 1; // Ensure multiple of two.
StereoBuffer16 ret(ret_size);
int yn1 = state.yn1,
yn2 = state.yn2;
int yn1 = state.yn1, yn2 = state.yn2;
const size_t NUM_FRAMES = (sample_count + (SAMPLES_PER_FRAME - 1)) / SAMPLES_PER_FRAME; // Round up.
const size_t NUM_FRAMES =
(sample_count + (SAMPLES_PER_FRAME - 1)) / SAMPLES_PER_FRAME; // Round up.
for (size_t framei = 0; framei < NUM_FRAMES; framei++) {
const int frame_header = data[framei * FRAME_LEN];
const int scale = 1 << (frame_header & 0xF);
@ -43,7 +46,8 @@ StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count, cons
// Decodes an audio sample. One nibble produces one sample.
const auto decode_sample = [&](const int nibble) -> s16 {
const int xn = nibble * scale;
// We first transform everything into 11 bit fixed point, perform the second order digital filter, then transform back.
// We first transform everything into 11 bit fixed point, perform the second order
// digital filter, then transform back.
// 0x400 == 0.5 in 11 bit fixed point.
// Filter: y[n] = x[n] + 0.5 + c1 * y[n-1] + c2 * y[n-2]
int val = ((xn << 11) + 0x400 + coef1 * yn1 + coef2 * yn2) >> 11;
@ -82,7 +86,8 @@ static s16 SignExtendS8(u8 x) {
return static_cast<s16>(static_cast<s8>(x));
}
StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data, const size_t sample_count) {
StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data,
const size_t sample_count) {
ASSERT(num_channels == 1 || num_channels == 2);
StereoBuffer16 ret(sample_count);
@ -101,7 +106,8 @@ StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data, con
return ret;
}
StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data, const size_t sample_count) {
StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data,
const size_t sample_count) {
ASSERT(num_channels == 1 || num_channels == 2);
StereoBuffer16 ret(sample_count);
@ -118,5 +124,4 @@ StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data, co
return ret;
}
};