Patroneo/template/calfbox/sampler_gen.c

/*
Calf Box, an open source musical instrument.
Copyright (C) 2010-2013 Krzysztof Foltman

This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.

This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
GNU General Public License for more details.

You should have received a copy of the GNU General Public License
along with this program.  If not, see <http://www.gnu.org/licenses/>.
*/

#include "config-api.h"
#include "dspmath.h"
#include "errors.h"
#include "midi.h"
#include "module.h"
#include "rt.h"
#include "sampler.h"
#include "sfzloader.h"
#include <assert.h>
#include <errno.h>
#include <math.h>
#include <memory.h>
#include <stdio.h>

#define LOW_QUALITY_INTERPOLATION 0

struct resampler_state
{
    float *leftright;
    int offset;
    float lgain, rgain, lgain_delta, rgain_delta;
};

#if USE_NEON

#include <arm_neon.h>

static inline void process_voice_mono_noloop(struct sampler_gen *v, struct resampler_state *rs, const int16_t *srcdata, int endpos)
{
    static const float32x2_t shift1a = {0.f, 1.f}, shift1b = {1.f, 1.f};
    static const float32x2_t shift2a = {-1.f, -1.f}, shift2b = {0.f, 0.f};
    static const float32x2_t shift3a = {-2.f, -2.f}, shift3b = {-2.f, -1.f};
    static const float32x2_t scalinga = {-1 / 6.0, 3 / 6.0}, scalingb = {-3 / 6.0, 1 / 6.0};
    uint64x1_t pos = v->bigpos, delta = v->bigdelta;
    float32x2_t gains = {rs->lgain, rs->rgain};
    const float32x2_t gaindeltas = {rs->lgain_delta, rs->rgain_delta};
    for (uint32_t i = rs->offset; i < endpos; i++)
    {
        float32x2_t posposf = vcvt_n_f32_u32(vreinterpret_u32_u64(pos), 32);
        
        int32x4_t smp = vmovl_s16(vld1_s16(&srcdata[pos >> 32]));
        pos = vadd_u64(pos, delta);
        
        float32x2_t t2 = vdup_n_f32(posposf[0]);
        float32x2_t samplesa = vcvt_f32_s32(vget_low_s32(smp)), samplesb = vcvt_f32_s32(vget_high_s32(smp));
        
        float32x2_t mula = vmul_f32(vmul_f32(vadd_f32(t2, shift1a), vadd_f32(t2, shift2a)), vmul_f32(vadd_f32(t2, shift3a), scalinga));
        float32x2_t mulb = vmul_f32(vmul_f32(vadd_f32(t2, shift1b), vadd_f32(t2, shift2b)), vmul_f32(vadd_f32(t2, shift3b), scalingb));
        float32x2_t v = vmla_f32(vmul_f32(samplesa, mula), samplesb, mulb);
        float32x2_t result = vmul_f32(gains, vadd_f32(v, vrev64_f32(v)));
        gains = vadd_f32(gains, gaindeltas);
        
        rs->leftright[2 * i] = result[0];
        rs->leftright[2 * i + 1] = result[1];
    }
    rs->lgain = gains[0];
    rs->rgain = gains[1];
    v->bigpos = pos;
    rs->offset = endpos;
}

static inline void process_voice_stereo_noloop(struct sampler_gen *v, struct resampler_state *rs, const int16_t *srcdata, int endpos)
{
    static const float32x2_t shift1a = {0.f, 1.f}, shift1b = {1.f, 1.f};
    static const float32x2_t shift2a = {-1.f, -1.f}, shift2b = {0.f, 0.f};
    static const float32x2_t shift3a = {-2.f, -2.f}, shift3b = {-2.f, -1.f};
    static const float32x2_t scalinga = {-1 / 6.0, 3 / 6.0}, scalingb = {-3 / 6.0, 1 / 6.0};
    uint64x1_t pos = v->bigpos, delta = v->bigdelta;
    float32x2_t gains = {rs->lgain, rs->rgain};
    const float32x2_t gaindeltas = {rs->lgain_delta, rs->rgain_delta};
    for (uint32_t i = rs->offset; i < endpos; i++)
    {
        float32x2_t posposf = vcvt_n_f32_u32(vreinterpret_u32_u64(pos), 32);
        
        int16x4x2_t pp = vld2_s16(&srcdata[(pos >> 31) &~ 1]);
        pos = vadd_u64(pos, delta);
        int32x4_t smp_left = vmovl_s16(pp.val[0]), smp_right = vmovl_s16(pp.val[1]);
        
        float32x2_t t2 = vdup_n_f32(posposf[0]);
        float32x2_t samplesLa = vcvt_f32_s32(vget_low_s32(smp_left)), samplesLb = vcvt_f32_s32(vget_high_s32(smp_left));
        float32x2_t samplesRa = vcvt_f32_s32(vget_low_s32(smp_right)), samplesRb = vcvt_f32_s32(vget_high_s32(smp_right));
        
        float32x2_t mula = vmul_f32(vmul_f32(vadd_f32(t2, shift1a), vadd_f32(t2, shift2a)), vmul_f32(vadd_f32(t2, shift3a), scalinga));
        float32x2_t mulb = vmul_f32(vmul_f32(vadd_f32(t2, shift1b), vadd_f32(t2, shift2b)), vmul_f32(vadd_f32(t2, shift3b), scalingb));
        float32x2_t vL = vmla_f32(vmul_f32(samplesLa, mula), samplesLb, mulb);
        float32x2_t vR = vmla_f32(vmul_f32(samplesRa, mula), samplesRb, mulb);
        float32x2x2_t transposed = vtrn_f32(vL, vR);
        float32x2_t result = vmul_f32(gains, vadd_f32(transposed.val[0], transposed.val[1]));
        gains = vadd_f32(gains, gaindeltas);
        
        rs->leftright[2 * i] = result[0];
        rs->leftright[2 * i + 1] = result[1];
    }
    rs->lgain = gains[0];
    rs->rgain = gains[1];
    v->bigpos = pos;
    rs->offset = endpos;
}

#elif USE_SSE

#include <xmmintrin.h>

typedef __m128 V4SF;

static const V4SF shift1 = {0, 1, 1, 1};
static const V4SF shift2 = {-1, -1, 0, 0};
static const V4SF shift3 = {-2, -2, -2, -1};
static const V4SF scaling = {-1, 3, -3, 1};
static const V4SF zero = {0, 0, 0, 0};


static inline void process_voice_mono_noloop(struct sampler_gen *v, struct resampler_state *rs, const int16_t *srcdata, int endpos)
{
    uint64_t pos = v->bigpos;
    const float ffrac = 1.0f / 6.0f;
    const float _scaler = 1.f / (128.f * 16777216.f);
    for (int i = rs->offset; i < endpos; i++)
    {
        //float t = ((pos >> 8) & 0x00FFFFFF) * scaler;
        const int16_t *p = &srcdata[pos >> 32];
        
        V4SF t2 = __builtin_ia32_cvtsi2ss(zero, (pos & 0xFFFFFFFF) >> 1) * _scaler;
        pos += v->bigdelta;
        
        V4SF t4 = __builtin_ia32_shufps(t2, t2, 0);
        V4SF v4mul = (t4 + shift1) * (t4 + shift2) * (t4 + shift3) * scaling;
        V4SF samples = {p[0], p[1], p[2], p[3]};
        v4mul = __builtin_ia32_mulps(samples, v4mul);
        
        float c = (v4mul[0] + v4mul[1] + v4mul[2] + v4mul[3]) * ffrac;

        rs->leftright[2 * i] = rs->lgain * c;
        rs->leftright[2 * i + 1] = rs->rgain * c;
        rs->lgain += rs->lgain_delta;
        rs->rgain += rs->rgain_delta;
    }
    v->bigpos = pos;
    rs->offset = endpos;
}

static inline void process_voice_stereo_noloop(struct sampler_gen *v, struct resampler_state *rs, const int16_t *srcdata, int endpos)
{
    uint64_t pos = v->bigpos;
    const float ffrac = 1.0f / 6.0f;
    const float _scaler = 1.f / (128.f * 16777216.f);
    for (int i = rs->offset; i < endpos; i++)
    {
        //float t = ((pos >> 8) & 0x00FFFFFF) * scaler;
        const int16_t *p = &srcdata[(pos >> 31) & ~1];
        
        V4SF t2 = __builtin_ia32_cvtsi2ss(zero, (pos & 0xFFFFFFFF) >> 1) * _scaler;
        pos += v->bigdelta;
        
        V4SF t4 = __builtin_ia32_shufps(t2, t2, 0);
        V4SF v4mul = (t4 + shift1) * (t4 + shift2) * (t4 + shift3) * scaling;
        V4SF samples_left = {p[0], p[2], p[4], p[6]};
        samples_left = __builtin_ia32_mulps(samples_left, v4mul);
        V4SF samples_right = {p[1], p[3], p[5], p[7]};
        samples_right = __builtin_ia32_mulps(samples_right, v4mul);
        
        float cl = (samples_left[0] + samples_left[1] + samples_left[2] + samples_left[3]) * ffrac;
        float cr = (samples_right[0] + samples_right[1] + samples_right[2] + samples_right[3]) * ffrac;

        rs->leftright[2 * i] = rs->lgain * cl;
        rs->leftright[2 * i + 1] = rs->rgain * cr;
        rs->lgain += rs->lgain_delta;
        rs->rgain += rs->rgain_delta;
    }
    v->bigpos = pos;
    rs->offset = endpos;
}

#else

static inline void process_voice_mono_noloop(struct sampler_gen *v, struct resampler_state *rs, const int16_t *srcdata, int endpos)
{
    const float ffrac = 1.0f / 6.0f;
    const float scaler = 1.f / 16777216.f;

    for (int i = rs->offset; i < endpos; i++)
    {
        float t = ((v->bigpos >> 8) & 0x00FFFFFF) * scaler;
        const int16_t *p = &srcdata[v->bigpos >> 32];
#if LOW_QUALITY_INTERPOLATION
        float c = (1.f - t) * p[1] + t * p[2];
#else
        float b0 = -t*(t-1.f)*(t-2.f);
        float b1 = 3.f*(t+1.f)*(t-1.f)*(t-2.f);
        float c = (b0 * p[0] + b1 * p[1] - 3.f*(t+1.f)*t*(t-2.f) * p[2] + (t+1.f)*t*(t-1.f) * p[3]) * ffrac;
#endif        
        rs->leftright[2 * i] = rs->lgain * c;
        rs->leftright[2 * i + 1] = rs->rgain * c;
        rs->lgain += rs->lgain_delta;
        rs->rgain += rs->rgain_delta;
        v->bigpos += v->bigdelta;
    }
    rs->offset = endpos;
}

static inline void process_voice_stereo_noloop(struct sampler_gen *v, struct resampler_state *rs, const int16_t *srcdata, int endpos)
{
    const float ffrac = 1.0f / 6.0f;
    const float scaler = 1.f / 16777216.f;

    for (int i = rs->offset; i < endpos; i++)
    {
        float t = ((v->bigpos >> 8) & 0x00FFFFFF) * scaler;
        const int16_t *p = &srcdata[(v->bigpos >> 31) & ~1];
#if LOW_QUALITY_INTERPOLATION
        float c0 = (1.f - t) * p[2] + t * p[4];
        float c1 = (1.f - t) * p[3] + t * p[5];
#else
        float b0 = -t*(t-1.f)*(t-2.f);
        float b1 = 3.f*(t+1.f)*(t-1.f)*(t-2.f);
        float c0 = (b0 * p[0] + b1 * p[2] - 3.f*(t+1.f)*t*(t-2.f) * p[4] + (t+1.f)*t*(t-1.f) * p[6]) * ffrac;
        float c1 = (b0 * p[1] + b1 * p[3] - 3.f*(t+1.f)*t*(t-2.f) * p[5] + (t+1.f)*t*(t-1.f) * p[7]) * ffrac;
#endif
        rs->leftright[2 * i] = rs->lgain * c0;
        rs->leftright[2 * i + 1] = rs->rgain * c1;
        rs->lgain += rs->lgain_delta;
        rs->rgain += rs->rgain_delta;
        v->bigpos += v->bigdelta;
    }
    rs->offset = endpos;
}

#endif

static inline uint32_t process_voice_noloop(struct sampler_gen *v, struct resampler_state *rs, const int16_t *srcdata, uint32_t pos_offset, uint32_t usable_sample_end)
{
    uint32_t out_frames = CBOX_BLOCK_SIZE - rs->offset;

    uint64_t sample_end64 = ((uint64_t)usable_sample_end) << 32;
    // Check how many frames can be written to output buffer without going
    // past usable_sample_end.
    if (__builtin_expect(v->bigpos + (out_frames - 1) * v->bigdelta >= sample_end64, 0))
        out_frames = (sample_end64 - v->bigpos) / v->bigdelta + 1;
    
    assert(out_frames > 0 && out_frames <= (uint32_t)(CBOX_BLOCK_SIZE - rs->offset));
    uint32_t oldpos = v->bigpos >> 32;
    if (v->mode == spt_stereo16)
        process_voice_stereo_noloop(v, rs, srcdata - (pos_offset << 1), rs->offset + out_frames);
    else
        process_voice_mono_noloop(v, rs, srcdata - pos_offset, rs->offset + out_frames);
    return (v->bigpos >> 32) - oldpos;
}

static void process_voice_withloop(struct sampler_gen *v, struct resampler_state *rs)
{
    // This is the first frame where interpolation will cross the loop boundary
    uint32_t loop_end = v->loop_end;
    uint32_t loop_edge = loop_end - MAX_INTERPOLATION_ORDER;
    
    while ( rs->offset < CBOX_BLOCK_SIZE ) {
        uint64_t startframe = v->bigpos >> 32;
        
        int16_t *source_data = v->sample_data;
        uint32_t source_offset = 0;
        uint32_t usable_sample_end = loop_edge;
        // if the first frame to play is already within 3 frames of loop end
        // (we need consecutive 4 frames for cubic interpolation) then
        // "straighten out" the area around the loop, and play that
        if (__builtin_expect(startframe >= loop_edge, 0))
        {
            // if fully past the loop end, then it's normal wraparound
            // (or end of the sample if not looping)
            if (startframe >= loop_end)
            {
                if (v->loop_start == (uint32_t)-1)
                {
                    v->mode = spt_inactive;
                    return;
                }
                v->play_count++;
                if (v->loop_count && v->play_count >= v->loop_count)
                {
                    v->mode = spt_inactive;
                    return;
                }
                v->bigpos -= (uint64_t)(loop_end - v->loop_start) << 32;
                continue;
            }

            usable_sample_end = loop_end;
            source_data = v->scratch;
            source_offset = loop_edge;
        }

        process_voice_noloop(v, rs, source_data, source_offset, usable_sample_end);
    }
}

static void process_voice_streaming(struct sampler_gen *v, struct resampler_state *rs, uint32_t limit)
{
    if (v->consumed_credit > 0)
    {
        if (v->consumed_credit >= limit)
        {
            v->consumed_credit -= limit;
            return;
        }
        limit -= v->consumed_credit;
        v->consumed_credit = 0;
    }
    // This is the first frame where interpolation will cross the loop boundary
    int16_t scratch[2 * MAX_INTERPOLATION_ORDER * 2];
    
    while ( limit && rs->offset < CBOX_BLOCK_SIZE ) {
        uint64_t startframe = v->bigpos >> 32;
        
        int16_t *source_data = v->in_streaming_buffer ? v->streaming_buffer : v->sample_data;
        uint32_t loop_start = v->in_streaming_buffer ? 0 : v->loop_start;
        uint32_t loop_end = v->in_streaming_buffer ? v->streaming_buffer_frames : v->loop_end;
        uint32_t loop_edge = loop_end - MAX_INTERPOLATION_ORDER;
        uint32_t source_offset = 0;
        uint32_t usable_sample_end = loop_edge;
        // if the first frame to play is already within 3 frames of loop end
        // (we need consecutive 4 frames for cubic interpolation) then
        // "straighten out" the area around the loop, and play that
        if (startframe >= loop_edge)
        {
            // if fully past the loop end, then it's normal wraparound
            // (or end of the sample if not looping)
            if (startframe >= loop_end)
            {
                if (v->loop_start == (uint32_t)-1)
                {
                    v->mode = spt_inactive;
                    return;
                }
                v->bigpos -= (uint64_t)(loop_end - loop_start) << 32;
                if (v->prefetch_only_loop)
                    v->consumed -= (loop_end - loop_start);
                else
                    v->in_streaming_buffer = TRUE;
                continue;
            }

            int shift = (v->mode == spt_stereo16) ? 1 : 0;
            
            // 'linearize' the virtual circular buffer - write 3 (or N) frames before end of the loop
            // and 3 (N) frames at the start of the loop, and play it; in rare cases this will need to be
            // repeated twice if output write pointer is close to CBOX_BLOCK_SIZE or playback rate is very low,
            // but that's OK.
            uint32_t halfscratch = MAX_INTERPOLATION_ORDER << shift;
            memcpy(&scratch[0], &source_data[loop_edge << shift], halfscratch * sizeof(int16_t) );
            if (v->loop_start == (uint32_t)-1)
                memset(scratch + halfscratch, 0, halfscratch * sizeof(int16_t));
            else
                memcpy(scratch + halfscratch, &v->streaming_buffer[v->loop_start << shift], halfscratch * sizeof(int16_t));

            usable_sample_end = loop_end;
            source_data = scratch;
            source_offset = loop_edge;
        }
        if (limit != (uint32_t)-1 && usable_sample_end - startframe > limit)
            usable_sample_end = startframe + limit;

        uint32_t consumed = process_voice_noloop(v, rs, source_data, source_offset, usable_sample_end);
        if (consumed > limit)
        {
            // The number of frames 'consumed' may be greater than the amount
            // available because of sample-skipping (at least that's the only
            // *legitimate* reason). This should be accounted for in the,
            // consumed sample counter (hence temporary storage of the 
            // 'buffer overconsumption' in the consumed_credit field), but is not
            // actually causing any use of missing data, as the missing samples
            // have been skipped.
            assert(v->consumed_credit == 0);
            v->consumed_credit = consumed - limit;
            assert (v->consumed_credit <= 1 + (v->bigdelta >> 32));
            consumed = limit;
        }
        v->consumed += consumed;
        if (consumed < limit)
            limit -= consumed;
        else
            break;
    }
}

void sampler_gen_reset(struct sampler_gen *v)
{
    v->mode = spt_inactive;
    v->bigpos = 0;
    v->last_lgain = 0.f;
    v->last_rgain = 0.f;
    v->play_count = 0;
    v->consumed = 0;
    v->consumed_credit = 0;
    v->streaming_buffer = NULL;
    v->in_streaming_buffer = FALSE;
    v->prefetch_only_loop = FALSE;
    v->fadein_counter = -1.f;
}

uint32_t sampler_gen_sample_playback(struct sampler_gen *v, float *leftright, uint32_t limit)
{
    struct resampler_state rs;
    rs.leftright = leftright;
    rs.offset = 0;
    rs.lgain = v->last_lgain;
    rs.rgain = v->last_rgain;
    rs.lgain_delta = (v->lgain - v->last_lgain) * (1.f / CBOX_BLOCK_SIZE);
    rs.rgain_delta = (v->rgain - v->last_rgain) * (1.f / CBOX_BLOCK_SIZE);
    if (v->streaming_buffer)
        process_voice_streaming(v, &rs, limit);
    else
    {
        process_voice_withloop(v, &rs);
    }
    uint32_t written = rs.offset;
    
    if (!v->streaming_buffer)
    {
        v->virtpos += written * v->virtdelta;
        if (v->virtpos != v->bigpos)
        {
            while ((v->virtpos >> 32) >= v->loop_end && v->loop_start != SAMPLER_NO_LOOP)
                v->virtpos -= ((uint64_t)(v->loop_end - v->loop_start)) << 32;
        }
        // XXXKF looping
        if (v->fadein_counter == -1 && fabs((v->bigpos - v->virtpos) / (65536.0 * 65536.0)) > v->stretching_jump)
        {
            int64_t jump = (int64_t)(v->stretching_jump * 65536.0 * 65536.0);
            int64_t newpos = v->bigpos > v->virtpos ? v->bigpos - jump : v->bigpos + jump;
            if (newpos < 0)
                newpos = 0;
            // XXXKF beware of extremely short loops
            while ((newpos >> 32) >= v->loop_end && v->loop_start != SAMPLER_NO_LOOP)
                newpos -= ((uint64_t)(v->loop_end - v->loop_start)) << 32;
            if ((newpos >> 32) >= v->cur_sample_end - 4)
                newpos = ((uint64_t)v->cur_sample_end - 4)<< 32;
            v->fadein_pos = newpos;
            v->fadein_counter = 0;
        }
        else if (v->fadein_counter != -1)
        {
            float leftright_fadein[2 * CBOX_BLOCK_SIZE];

            rs.offset = 0;
            rs.leftright = leftright_fadein;
            rs.lgain = v->last_lgain;
            rs.rgain = v->last_rgain;

            uint64_t oldpos = v->bigpos;
            v->bigpos = v->fadein_pos;            
            process_voice_withloop(v, &rs);
            v->fadein_pos = v->bigpos;
            v->bigpos = oldpos;
            
            uint32_t written2 = rs.offset;
            
            // XXXKF not the best set of special cases
            uint32_t i;
            if (written2 > written)
            {
                for (i = 2 * written; i < 2 * written2; i += 2)
                    leftright[i] = leftright[i + 1] = 0.f;
                written = written2;
            }
            if (written2 < written)
            {
                for (i = 2 * written2; i < 2 * written; i += 2)
                    leftright_fadein[i] = leftright_fadein[i + 1] = 0.f;
                written2 = written;
            }
            float cnt = v->fadein_counter;
            float scl = v->bigdelta / (v->stretching_crossfade * v->virtdelta);
            for (i = 0; i < 2 * written2; i += 2)
            {
                leftright[i] += (leftright_fadein[i] - leftright[i]) * cnt;
                leftright[i + 1] += (leftright_fadein[i + 1] - leftright[i + 1]) * cnt;
                cnt += scl;
                if (cnt > 1.f)
                    cnt = 1.f;
            }
            if (cnt >= 1.f)
            {
                cnt = -1.f;
                v->bigpos = v->fadein_pos;
            }
            v->fadein_counter = cnt;
        }
    }
    v->last_lgain = v->lgain;
    v->last_rgain = v->rgain;
    return written;
}