From 933508e2a2f7923cebc15d679b78933df8fb9ee5 Mon Sep 17 00:00:00 2001 From: MerryMage Date: Thu, 3 Aug 2017 12:22:51 +0100 Subject: interpolate: Interpolate on a frame-by-frame basis --- src/audio_core/interpolate.h | 27 ++++++++++++++++----------- 1 file changed, 16 insertions(+), 11 deletions(-) (limited to 'src/audio_core/interpolate.h') 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 #include +#include "audio_core/hle/common.h" #include "common/common_types.h" namespace AudioInterp { @@ -14,31 +15,35 @@ namespace AudioInterp { using StereoBuffer16 = std::vector>; struct State { - // Two historical samples. + /// Two historical samples. std::array xn1 = {}; ///< x[n-1] std::array 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 -- cgit v1.2.3