AudioResampler.cpp revision 01d3acba9de861cb2b718338e787cff3566fc5ec 
1/*
2 * Copyright (C) 2007 The Android Open Source Project
3 *
4 * Licensed under the Apache License, Version 2.0 (the "License");
5 * you may not use this file except in compliance with the License.
6 * You may obtain a copy of the License at
7 *
8 *      http://www.apache.org/licenses/LICENSE-2.0
9 *
10 * Unless required by applicable law or agreed to in writing, software
11 * distributed under the License is distributed on an "AS IS" BASIS,
12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
13 * See the License for the specific language governing permissions and
14 * limitations under the License.
15 */
16
17#define LOG_TAG "AudioResampler"
18//#define LOG_NDEBUG 0
19
20#include <stdint.h>
21#include <stdlib.h>
22#include <sys/types.h>
23#include <cutils/log.h>
24#include <cutils/properties.h>
25#include "AudioResampler.h"
26#include "AudioResamplerSinc.h"
27#include "AudioResamplerCubic.h"
28#include "AudioResamplerDyn.h"
29
30#ifdef __arm__
31#include <machine/cpu-features.h>
32#endif
33
34namespace android {
35
36#ifdef __ARM_HAVE_HALFWORD_MULTIPLY // optimized asm option
37    #define ASM_ARM_RESAMP1 // enable asm optimisation for ResamplerOrder1
38#endif // __ARM_HAVE_HALFWORD_MULTIPLY
39// ----------------------------------------------------------------------------
40
41class AudioResamplerOrder1 : public AudioResampler {
42public:
43    AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) :
44        AudioResampler(bitDepth, inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {
45    }
46    virtual void resample(int32_t* out, size_t outFrameCount,
47            AudioBufferProvider* provider);
48private:
49    // number of bits used in interpolation multiply - 15 bits avoids overflow
50    static const int kNumInterpBits = 15;
51
52    // bits to shift the phase fraction down to avoid overflow
53    static const int kPreInterpShift = kNumPhaseBits - kNumInterpBits;
54
55    void init() {}
56    void resampleMono16(int32_t* out, size_t outFrameCount,
57            AudioBufferProvider* provider);
58    void resampleStereo16(int32_t* out, size_t outFrameCount,
59            AudioBufferProvider* provider);
60#ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
61    void AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
62            size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
63            uint32_t &phaseFraction, uint32_t phaseIncrement);
64    void AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
65            size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
66            uint32_t &phaseFraction, uint32_t phaseIncrement);
67#endif  // ASM_ARM_RESAMP1
68
69    static inline int32_t Interp(int32_t x0, int32_t x1, uint32_t f) {
70        return x0 + (((x1 - x0) * (int32_t)(f >> kPreInterpShift)) >> kNumInterpBits);
71    }
72    static inline void Advance(size_t* index, uint32_t* frac, uint32_t inc) {
73        *frac += inc;
74        *index += (size_t)(*frac >> kNumPhaseBits);
75        *frac &= kPhaseMask;
76    }
77    int mX0L;
78    int mX0R;
79};
80
81/*static*/
82const double AudioResampler::kPhaseMultiplier = 1L << AudioResampler::kNumPhaseBits;
83
84bool AudioResampler::qualityIsSupported(src_quality quality)
85{
86    switch (quality) {
87    case DEFAULT_QUALITY:
88    case LOW_QUALITY:
89    case MED_QUALITY:
90    case HIGH_QUALITY:
91    case VERY_HIGH_QUALITY:
92    case DYN_LOW_QUALITY:
93    case DYN_MED_QUALITY:
94    case DYN_HIGH_QUALITY:
95        return true;
96    default:
97        return false;
98    }
99}
100
101// ----------------------------------------------------------------------------
102
103static pthread_once_t once_control = PTHREAD_ONCE_INIT;
104static AudioResampler::src_quality defaultQuality = AudioResampler::DEFAULT_QUALITY;
105
106void AudioResampler::init_routine()
107{
108    char value[PROPERTY_VALUE_MAX];
109    if (property_get("af.resampler.quality", value, NULL) > 0) {
110        char *endptr;
111        unsigned long l = strtoul(value, &endptr, 0);
112        if (*endptr == '\0') {
113            defaultQuality = (src_quality) l;
114            ALOGD("forcing AudioResampler quality to %d", defaultQuality);
115            if (defaultQuality < DEFAULT_QUALITY || defaultQuality > DYN_HIGH_QUALITY) {
116                defaultQuality = DEFAULT_QUALITY;
117            }
118        }
119    }
120}
121
122uint32_t AudioResampler::qualityMHz(src_quality quality)
123{
124    switch (quality) {
125    default:
126    case DEFAULT_QUALITY:
127    case LOW_QUALITY:
128        return 3;
129    case MED_QUALITY:
130        return 6;
131    case HIGH_QUALITY:
132        return 20;
133    case VERY_HIGH_QUALITY:
134        return 34;
135    case DYN_LOW_QUALITY:
136        return 4;
137    case DYN_MED_QUALITY:
138        return 6;
139    case DYN_HIGH_QUALITY:
140        return 12;
141    }
142}
143
144static const uint32_t maxMHz = 130; // an arbitrary number that permits 3 VHQ, should be tunable
145static pthread_mutex_t mutex = PTHREAD_MUTEX_INITIALIZER;
146static uint32_t currentMHz = 0;
147
148AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
149        int32_t sampleRate, src_quality quality) {
150
151    bool atFinalQuality;
152    if (quality == DEFAULT_QUALITY) {
153        // read the resampler default quality property the first time it is needed
154        int ok = pthread_once(&once_control, init_routine);
155        if (ok != 0) {
156            ALOGE("%s pthread_once failed: %d", __func__, ok);
157        }
158        quality = defaultQuality;
159        atFinalQuality = false;
160    } else {
161        atFinalQuality = true;
162    }
163
164    /* if the caller requests DEFAULT_QUALITY and af.resampler.property
165     * has not been set, the target resampler quality is set to DYN_MED_QUALITY,
166     * and allowed to "throttle" down to DYN_LOW_QUALITY if necessary
167     * due to estimated CPU load of having too many active resamplers
168     * (the code below the if).
169     */
170    if (quality == DEFAULT_QUALITY) {
171        quality = DYN_MED_QUALITY;
172    }
173
174    // naive implementation of CPU load throttling doesn't account for whether resampler is active
175    pthread_mutex_lock(&mutex);
176    for (;;) {
177        uint32_t deltaMHz = qualityMHz(quality);
178        uint32_t newMHz = currentMHz + deltaMHz;
179        if ((qualityIsSupported(quality) && newMHz <= maxMHz) || atFinalQuality) {
180            ALOGV("resampler load %u -> %u MHz due to delta +%u MHz from quality %d",
181                    currentMHz, newMHz, deltaMHz, quality);
182            currentMHz = newMHz;
183            break;
184        }
185        // not enough CPU available for proposed quality level, so try next lowest level
186        switch (quality) {
187        default:
188        case LOW_QUALITY:
189            atFinalQuality = true;
190            break;
191        case MED_QUALITY:
192            quality = LOW_QUALITY;
193            break;
194        case HIGH_QUALITY:
195            quality = MED_QUALITY;
196            break;
197        case VERY_HIGH_QUALITY:
198            quality = HIGH_QUALITY;
199            break;
200        case DYN_LOW_QUALITY:
201            atFinalQuality = true;
202            break;
203        case DYN_MED_QUALITY:
204            quality = DYN_LOW_QUALITY;
205            break;
206        case DYN_HIGH_QUALITY:
207            quality = DYN_MED_QUALITY;
208            break;
209        }
210    }
211    pthread_mutex_unlock(&mutex);
212
213    AudioResampler* resampler;
214
215    switch (quality) {
216    default:
217    case LOW_QUALITY:
218        ALOGV("Create linear Resampler");
219        resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
220        break;
221    case MED_QUALITY:
222        ALOGV("Create cubic Resampler");
223        resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
224        break;
225    case HIGH_QUALITY:
226        ALOGV("Create HIGH_QUALITY sinc Resampler");
227        resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
228        break;
229    case VERY_HIGH_QUALITY:
230        ALOGV("Create VERY_HIGH_QUALITY sinc Resampler = %d", quality);
231        resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality);
232        break;
233    case DYN_LOW_QUALITY:
234    case DYN_MED_QUALITY:
235    case DYN_HIGH_QUALITY:
236        ALOGV("Create dynamic Resampler = %d", quality);
237        resampler = new AudioResamplerDyn(bitDepth, inChannelCount, sampleRate, quality);
238        break;
239    }
240
241    // initialize resampler
242    resampler->init();
243    return resampler;
244}
245
246AudioResampler::AudioResampler(int bitDepth, int inChannelCount,
247        int32_t sampleRate, src_quality quality) :
248    mBitDepth(bitDepth), mChannelCount(inChannelCount),
249            mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
250            mPhaseFraction(0), mLocalTimeFreq(0),
251            mPTS(AudioBufferProvider::kInvalidPTS), mQuality(quality) {
252    // sanity check on format
253    if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) {
254        ALOGE("Unsupported sample format, %d bits, %d channels", bitDepth,
255                inChannelCount);
256        // ALOG_ASSERT(0);
257    }
258    if (sampleRate <= 0) {
259        ALOGE("Unsupported sample rate %d Hz", sampleRate);
260    }
261
262    // initialize common members
263    mVolume[0] = mVolume[1] = 0;
264    mBuffer.frameCount = 0;
265
266}
267
268AudioResampler::~AudioResampler() {
269    pthread_mutex_lock(&mutex);
270    src_quality quality = getQuality();
271    uint32_t deltaMHz = qualityMHz(quality);
272    int32_t newMHz = currentMHz - deltaMHz;
273    ALOGV("resampler load %u -> %d MHz due to delta -%u MHz from quality %d",
274            currentMHz, newMHz, deltaMHz, quality);
275    LOG_ALWAYS_FATAL_IF(newMHz < 0, "negative resampler load %d MHz", newMHz);
276    currentMHz = newMHz;
277    pthread_mutex_unlock(&mutex);
278}
279
280void AudioResampler::setSampleRate(int32_t inSampleRate) {
281    mInSampleRate = inSampleRate;
282    mPhaseIncrement = (uint32_t)((kPhaseMultiplier * inSampleRate) / mSampleRate);
283}
284
285void AudioResampler::setVolume(int16_t left, int16_t right) {
286    // TODO: Implement anti-zipper filter
287    mVolume[0] = left;
288    mVolume[1] = right;
289}
290
291void AudioResampler::setLocalTimeFreq(uint64_t freq) {
292    mLocalTimeFreq = freq;
293}
294
295void AudioResampler::setPTS(int64_t pts) {
296    mPTS = pts;
297}
298
299int64_t AudioResampler::calculateOutputPTS(int outputFrameIndex) {
300
301    if (mPTS == AudioBufferProvider::kInvalidPTS) {
302        return AudioBufferProvider::kInvalidPTS;
303    } else {
304        return mPTS + ((outputFrameIndex * mLocalTimeFreq) / mSampleRate);
305    }
306}
307
308void AudioResampler::reset() {
309    mInputIndex = 0;
310    mPhaseFraction = 0;
311    mBuffer.frameCount = 0;
312}
313
314// ----------------------------------------------------------------------------
315
316void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
317        AudioBufferProvider* provider) {
318
319    // should never happen, but we overflow if it does
320    // ALOG_ASSERT(outFrameCount < 32767);
321
322    // select the appropriate resampler
323    switch (mChannelCount) {
324    case 1:
325        resampleMono16(out, outFrameCount, provider);
326        break;
327    case 2:
328        resampleStereo16(out, outFrameCount, provider);
329        break;
330    }
331}
332
333void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
334        AudioBufferProvider* provider) {
335
336    int32_t vl = mVolume[0];
337    int32_t vr = mVolume[1];
338
339    size_t inputIndex = mInputIndex;
340    uint32_t phaseFraction = mPhaseFraction;
341    uint32_t phaseIncrement = mPhaseIncrement;
342    size_t outputIndex = 0;
343    size_t outputSampleCount = outFrameCount * 2;
344    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
345
346    // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
347    //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
348
349    while (outputIndex < outputSampleCount) {
350
351        // buffer is empty, fetch a new one
352        while (mBuffer.frameCount == 0) {
353            mBuffer.frameCount = inFrameCount;
354            provider->getNextBuffer(&mBuffer,
355                                    calculateOutputPTS(outputIndex / 2));
356            if (mBuffer.raw == NULL) {
357                goto resampleStereo16_exit;
358            }
359
360            // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
361            if (mBuffer.frameCount > inputIndex) break;
362
363            inputIndex -= mBuffer.frameCount;
364            mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
365            mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
366            provider->releaseBuffer(&mBuffer);
367            // mBuffer.frameCount == 0 now so we reload a new buffer
368        }
369
370        int16_t *in = mBuffer.i16;
371
372        // handle boundary case
373        while (inputIndex == 0) {
374            // ALOGE("boundary case");
375            out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction);
376            out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction);
377            Advance(&inputIndex, &phaseFraction, phaseIncrement);
378            if (outputIndex == outputSampleCount) {
379                break;
380            }
381        }
382
383        // process input samples
384        // ALOGE("general case");
385
386#ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
387        if (inputIndex + 2 < mBuffer.frameCount) {
388            int32_t* maxOutPt;
389            int32_t maxInIdx;
390
391            maxOutPt = out + (outputSampleCount - 2);   // 2 because 2 frames per loop
392            maxInIdx = mBuffer.frameCount - 2;
393            AsmStereo16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
394                    phaseFraction, phaseIncrement);
395        }
396#endif  // ASM_ARM_RESAMP1
397
398        while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) {
399            out[outputIndex++] += vl * Interp(in[inputIndex*2-2],
400                    in[inputIndex*2], phaseFraction);
401            out[outputIndex++] += vr * Interp(in[inputIndex*2-1],
402                    in[inputIndex*2+1], phaseFraction);
403            Advance(&inputIndex, &phaseFraction, phaseIncrement);
404        }
405
406        // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
407
408        // if done with buffer, save samples
409        if (inputIndex >= mBuffer.frameCount) {
410            inputIndex -= mBuffer.frameCount;
411
412            // ALOGE("buffer done, new input index %d", inputIndex);
413
414            mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
415            mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
416            provider->releaseBuffer(&mBuffer);
417
418            // verify that the releaseBuffer resets the buffer frameCount
419            // ALOG_ASSERT(mBuffer.frameCount == 0);
420        }
421    }
422
423    // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
424
425resampleStereo16_exit:
426    // save state
427    mInputIndex = inputIndex;
428    mPhaseFraction = phaseFraction;
429}
430
431void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
432        AudioBufferProvider* provider) {
433
434    int32_t vl = mVolume[0];
435    int32_t vr = mVolume[1];
436
437    size_t inputIndex = mInputIndex;
438    uint32_t phaseFraction = mPhaseFraction;
439    uint32_t phaseIncrement = mPhaseIncrement;
440    size_t outputIndex = 0;
441    size_t outputSampleCount = outFrameCount * 2;
442    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
443
444    // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
445    //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
446    while (outputIndex < outputSampleCount) {
447        // buffer is empty, fetch a new one
448        while (mBuffer.frameCount == 0) {
449            mBuffer.frameCount = inFrameCount;
450            provider->getNextBuffer(&mBuffer,
451                                    calculateOutputPTS(outputIndex / 2));
452            if (mBuffer.raw == NULL) {
453                mInputIndex = inputIndex;
454                mPhaseFraction = phaseFraction;
455                goto resampleMono16_exit;
456            }
457            // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
458            if (mBuffer.frameCount >  inputIndex) break;
459
460            inputIndex -= mBuffer.frameCount;
461            mX0L = mBuffer.i16[mBuffer.frameCount-1];
462            provider->releaseBuffer(&mBuffer);
463            // mBuffer.frameCount == 0 now so we reload a new buffer
464        }
465        int16_t *in = mBuffer.i16;
466
467        // handle boundary case
468        while (inputIndex == 0) {
469            // ALOGE("boundary case");
470            int32_t sample = Interp(mX0L, in[0], phaseFraction);
471            out[outputIndex++] += vl * sample;
472            out[outputIndex++] += vr * sample;
473            Advance(&inputIndex, &phaseFraction, phaseIncrement);
474            if (outputIndex == outputSampleCount) {
475                break;
476            }
477        }
478
479        // process input samples
480        // ALOGE("general case");
481
482#ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
483        if (inputIndex + 2 < mBuffer.frameCount) {
484            int32_t* maxOutPt;
485            int32_t maxInIdx;
486
487            maxOutPt = out + (outputSampleCount - 2);
488            maxInIdx = (int32_t)mBuffer.frameCount - 2;
489                AsmMono16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
490                        phaseFraction, phaseIncrement);
491        }
492#endif  // ASM_ARM_RESAMP1
493
494        while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) {
495            int32_t sample = Interp(in[inputIndex-1], in[inputIndex],
496                    phaseFraction);
497            out[outputIndex++] += vl * sample;
498            out[outputIndex++] += vr * sample;
499            Advance(&inputIndex, &phaseFraction, phaseIncrement);
500        }
501
502
503        // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
504
505        // if done with buffer, save samples
506        if (inputIndex >= mBuffer.frameCount) {
507            inputIndex -= mBuffer.frameCount;
508
509            // ALOGE("buffer done, new input index %d", inputIndex);
510
511            mX0L = mBuffer.i16[mBuffer.frameCount-1];
512            provider->releaseBuffer(&mBuffer);
513
514            // verify that the releaseBuffer resets the buffer frameCount
515            // ALOG_ASSERT(mBuffer.frameCount == 0);
516        }
517    }
518
519    // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
520
521resampleMono16_exit:
522    // save state
523    mInputIndex = inputIndex;
524    mPhaseFraction = phaseFraction;
525}
526
527#ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
528
529/*******************************************************************
530*
531*   AsmMono16Loop
532*   asm optimized monotonic loop version; one loop is 2 frames
533*   Input:
534*       in : pointer on input samples
535*       maxOutPt : pointer on first not filled
536*       maxInIdx : index on first not used
537*       outputIndex : pointer on current output index
538*       out : pointer on output buffer
539*       inputIndex : pointer on current input index
540*       vl, vr : left and right gain
541*       phaseFraction : pointer on current phase fraction
542*       phaseIncrement
543*   Ouput:
544*       outputIndex :
545*       out : updated buffer
546*       inputIndex : index of next to use
547*       phaseFraction : phase fraction for next interpolation
548*
549*******************************************************************/
550__attribute__((noinline))
551void AudioResamplerOrder1::AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
552            size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
553            uint32_t &phaseFraction, uint32_t phaseIncrement)
554{
555    (void)maxOutPt; // remove unused parameter warnings
556    (void)maxInIdx;
557    (void)outputIndex;
558    (void)out;
559    (void)inputIndex;
560    (void)vl;
561    (void)vr;
562    (void)phaseFraction;
563    (void)phaseIncrement;
564    (void)in;
565#define MO_PARAM5   "36"        // offset of parameter 5 (outputIndex)
566
567    asm(
568        "stmfd  sp!, {r4, r5, r6, r7, r8, r9, r10, r11, lr}\n"
569        // get parameters
570        "   ldr r6, [sp, #" MO_PARAM5 " + 20]\n"    // &phaseFraction
571        "   ldr r6, [r6]\n"                         // phaseFraction
572        "   ldr r7, [sp, #" MO_PARAM5 " + 8]\n"     // &inputIndex
573        "   ldr r7, [r7]\n"                         // inputIndex
574        "   ldr r8, [sp, #" MO_PARAM5 " + 4]\n"     // out
575        "   ldr r0, [sp, #" MO_PARAM5 " + 0]\n"     // &outputIndex
576        "   ldr r0, [r0]\n"                         // outputIndex
577        "   add r8, r8, r0, asl #2\n"               // curOut
578        "   ldr r9, [sp, #" MO_PARAM5 " + 24]\n"    // phaseIncrement
579        "   ldr r10, [sp, #" MO_PARAM5 " + 12]\n"   // vl
580        "   ldr r11, [sp, #" MO_PARAM5 " + 16]\n"   // vr
581
582        // r0 pin, x0, Samp
583
584        // r1 in
585        // r2 maxOutPt
586        // r3 maxInIdx
587
588        // r4 x1, i1, i3, Out1
589        // r5 out0
590
591        // r6 frac
592        // r7 inputIndex
593        // r8 curOut
594
595        // r9 inc
596        // r10 vl
597        // r11 vr
598
599        // r12
600        // r13 sp
601        // r14
602
603        // the following loop works on 2 frames
604
605        "1:\n"
606        "   cmp r8, r2\n"                   // curOut - maxCurOut
607        "   bcs 2f\n"
608
609#define MO_ONE_FRAME \
610    "   add r0, r1, r7, asl #1\n"       /* in + inputIndex */\
611    "   ldrsh r4, [r0]\n"               /* in[inputIndex] */\
612    "   ldr r5, [r8]\n"                 /* out[outputIndex] */\
613    "   ldrsh r0, [r0, #-2]\n"          /* in[inputIndex-1] */\
614    "   bic r6, r6, #0xC0000000\n"      /* phaseFraction & ... */\
615    "   sub r4, r4, r0\n"               /* in[inputIndex] - in[inputIndex-1] */\
616    "   mov r4, r4, lsl #2\n"           /* <<2 */\
617    "   smulwt r4, r4, r6\n"            /* (x1-x0)*.. */\
618    "   add r6, r6, r9\n"               /* phaseFraction + phaseIncrement */\
619    "   add r0, r0, r4\n"               /* x0 - (..) */\
620    "   mla r5, r0, r10, r5\n"          /* vl*interp + out[] */\
621    "   ldr r4, [r8, #4]\n"             /* out[outputIndex+1] */\
622    "   str r5, [r8], #4\n"             /* out[outputIndex++] = ... */\
623    "   mla r4, r0, r11, r4\n"          /* vr*interp + out[] */\
624    "   add r7, r7, r6, lsr #30\n"      /* inputIndex + phaseFraction>>30 */\
625    "   str r4, [r8], #4\n"             /* out[outputIndex++] = ... */
626
627        MO_ONE_FRAME    // frame 1
628        MO_ONE_FRAME    // frame 2
629
630        "   cmp r7, r3\n"                   // inputIndex - maxInIdx
631        "   bcc 1b\n"
632        "2:\n"
633
634        "   bic r6, r6, #0xC0000000\n"             // phaseFraction & ...
635        // save modified values
636        "   ldr r0, [sp, #" MO_PARAM5 " + 20]\n"    // &phaseFraction
637        "   str r6, [r0]\n"                         // phaseFraction
638        "   ldr r0, [sp, #" MO_PARAM5 " + 8]\n"     // &inputIndex
639        "   str r7, [r0]\n"                         // inputIndex
640        "   ldr r0, [sp, #" MO_PARAM5 " + 4]\n"     // out
641        "   sub r8, r0\n"                           // curOut - out
642        "   asr r8, #2\n"                           // new outputIndex
643        "   ldr r0, [sp, #" MO_PARAM5 " + 0]\n"     // &outputIndex
644        "   str r8, [r0]\n"                         // save outputIndex
645
646        "   ldmfd   sp!, {r4, r5, r6, r7, r8, r9, r10, r11, pc}\n"
647    );
648}
649
650/*******************************************************************
651*
652*   AsmStereo16Loop
653*   asm optimized stereo loop version; one loop is 2 frames
654*   Input:
655*       in : pointer on input samples
656*       maxOutPt : pointer on first not filled
657*       maxInIdx : index on first not used
658*       outputIndex : pointer on current output index
659*       out : pointer on output buffer
660*       inputIndex : pointer on current input index
661*       vl, vr : left and right gain
662*       phaseFraction : pointer on current phase fraction
663*       phaseIncrement
664*   Ouput:
665*       outputIndex :
666*       out : updated buffer
667*       inputIndex : index of next to use
668*       phaseFraction : phase fraction for next interpolation
669*
670*******************************************************************/
671__attribute__((noinline))
672void AudioResamplerOrder1::AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
673            size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
674            uint32_t &phaseFraction, uint32_t phaseIncrement)
675{
676    (void)maxOutPt; // remove unused parameter warnings
677    (void)maxInIdx;
678    (void)outputIndex;
679    (void)out;
680    (void)inputIndex;
681    (void)vl;
682    (void)vr;
683    (void)phaseFraction;
684    (void)phaseIncrement;
685    (void)in;
686#define ST_PARAM5    "40"     // offset of parameter 5 (outputIndex)
687    asm(
688        "stmfd  sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, lr}\n"
689        // get parameters
690        "   ldr r6, [sp, #" ST_PARAM5 " + 20]\n"    // &phaseFraction
691        "   ldr r6, [r6]\n"                         // phaseFraction
692        "   ldr r7, [sp, #" ST_PARAM5 " + 8]\n"     // &inputIndex
693        "   ldr r7, [r7]\n"                         // inputIndex
694        "   ldr r8, [sp, #" ST_PARAM5 " + 4]\n"     // out
695        "   ldr r0, [sp, #" ST_PARAM5 " + 0]\n"     // &outputIndex
696        "   ldr r0, [r0]\n"                         // outputIndex
697        "   add r8, r8, r0, asl #2\n"               // curOut
698        "   ldr r9, [sp, #" ST_PARAM5 " + 24]\n"    // phaseIncrement
699        "   ldr r10, [sp, #" ST_PARAM5 " + 12]\n"   // vl
700        "   ldr r11, [sp, #" ST_PARAM5 " + 16]\n"   // vr
701
702        // r0 pin, x0, Samp
703
704        // r1 in
705        // r2 maxOutPt
706        // r3 maxInIdx
707
708        // r4 x1, i1, i3, out1
709        // r5 out0
710
711        // r6 frac
712        // r7 inputIndex
713        // r8 curOut
714
715        // r9 inc
716        // r10 vl
717        // r11 vr
718
719        // r12 temporary
720        // r13 sp
721        // r14
722
723        "3:\n"
724        "   cmp r8, r2\n"                   // curOut - maxCurOut
725        "   bcs 4f\n"
726
727#define ST_ONE_FRAME \
728    "   bic r6, r6, #0xC0000000\n"      /* phaseFraction & ... */\
729\
730    "   add r0, r1, r7, asl #2\n"       /* in + 2*inputIndex */\
731\
732    "   ldrsh r4, [r0]\n"               /* in[2*inputIndex] */\
733    "   ldr r5, [r8]\n"                 /* out[outputIndex] */\
734    "   ldrsh r12, [r0, #-4]\n"         /* in[2*inputIndex-2] */\
735    "   sub r4, r4, r12\n"              /* in[2*InputIndex] - in[2*InputIndex-2] */\
736    "   mov r4, r4, lsl #2\n"           /* <<2 */\
737    "   smulwt r4, r4, r6\n"            /* (x1-x0)*.. */\
738    "   add r12, r12, r4\n"             /* x0 - (..) */\
739    "   mla r5, r12, r10, r5\n"         /* vl*interp + out[] */\
740    "   ldr r4, [r8, #4]\n"             /* out[outputIndex+1] */\
741    "   str r5, [r8], #4\n"             /* out[outputIndex++] = ... */\
742\
743    "   ldrsh r12, [r0, #+2]\n"         /* in[2*inputIndex+1] */\
744    "   ldrsh r0, [r0, #-2]\n"          /* in[2*inputIndex-1] */\
745    "   sub r12, r12, r0\n"             /* in[2*InputIndex] - in[2*InputIndex-2] */\
746    "   mov r12, r12, lsl #2\n"         /* <<2 */\
747    "   smulwt r12, r12, r6\n"          /* (x1-x0)*.. */\
748    "   add r12, r0, r12\n"             /* x0 - (..) */\
749    "   mla r4, r12, r11, r4\n"         /* vr*interp + out[] */\
750    "   str r4, [r8], #4\n"             /* out[outputIndex++] = ... */\
751\
752    "   add r6, r6, r9\n"               /* phaseFraction + phaseIncrement */\
753    "   add r7, r7, r6, lsr #30\n"      /* inputIndex + phaseFraction>>30 */
754
755    ST_ONE_FRAME    // frame 1
756    ST_ONE_FRAME    // frame 1
757
758        "   cmp r7, r3\n"                       // inputIndex - maxInIdx
759        "   bcc 3b\n"
760        "4:\n"
761
762        "   bic r6, r6, #0xC0000000\n"              // phaseFraction & ...
763        // save modified values
764        "   ldr r0, [sp, #" ST_PARAM5 " + 20]\n"    // &phaseFraction
765        "   str r6, [r0]\n"                         // phaseFraction
766        "   ldr r0, [sp, #" ST_PARAM5 " + 8]\n"     // &inputIndex
767        "   str r7, [r0]\n"                         // inputIndex
768        "   ldr r0, [sp, #" ST_PARAM5 " + 4]\n"     // out
769        "   sub r8, r0\n"                           // curOut - out
770        "   asr r8, #2\n"                           // new outputIndex
771        "   ldr r0, [sp, #" ST_PARAM5 " + 0]\n"     // &outputIndex
772        "   str r8, [r0]\n"                         // save outputIndex
773
774        "   ldmfd   sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, pc}\n"
775    );
776}
777
778#endif  // ASM_ARM_RESAMP1
779
780
781// ----------------------------------------------------------------------------
782
783} // namespace android
784