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lavr: resampling: add support for s32p, fltp, and dblp internal sample formats
Based partially on implementation by Michael Niedermayer <michaelni@gmx.at> in libswresample in FFmpeg. See commits:7f1ae79d3824ab1abfb6
This commit is contained in:
Justin Ruggles
parent
372647aed0
commit
6410397600
@ -24,34 +24,10 @@
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#include "internal.h"
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#include "audio_data.h"
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#ifdef CONFIG_RESAMPLE_FLT
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/* float template */
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#define FILTER_SHIFT 0
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#define FELEM float
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#define FELEM2 float
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#define FELEML float
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#elifdef CONFIG_RESAMPLE_S32
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/* s32 template */
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#define FILTER_SHIFT 30
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#define FELEM int32_t
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#define FELEM2 int64_t
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#define FELEML int64_t
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#define FELEM_MAX INT32_MAX
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#define FELEM_MIN INT32_MIN
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#else
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/* s16 template */
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#define FILTER_SHIFT 15
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#define FELEM int16_t
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#define FELEM2 int32_t
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#define FELEML int64_t
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#define FELEM_MAX INT16_MAX
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#define FELEM_MIN INT16_MIN
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#endif
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struct ResampleContext {
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AVAudioResampleContext *avr;
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AudioData *buffer;
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FELEM *filter_bank;
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uint8_t *filter_bank;
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int filter_length;
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int ideal_dst_incr;
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int dst_incr;
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@ -65,8 +41,32 @@ struct ResampleContext {
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enum AVResampleFilterType filter_type;
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int kaiser_beta;
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double factor;
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void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
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void (*resample_one)(struct ResampleContext *c, int no_filter, void *dst0,
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int dst_index, const void *src0, int src_size,
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int index, int frac);
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};
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/* double template */
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#define CONFIG_RESAMPLE_DBL
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#include "resample_template.c"
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#undef CONFIG_RESAMPLE_DBL
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/* float template */
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#define CONFIG_RESAMPLE_FLT
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#include "resample_template.c"
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#undef CONFIG_RESAMPLE_FLT
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/* s32 template */
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#define CONFIG_RESAMPLE_S32
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#include "resample_template.c"
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#undef CONFIG_RESAMPLE_S32
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/* s16 template */
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#include "resample_template.c"
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/**
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* 0th order modified bessel function of the first kind.
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*/
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@ -98,13 +98,13 @@ static double bessel(double x)
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* @param kaiser_beta kaiser window beta
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* @return 0 on success, negative AVERROR code on failure
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*/
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static int build_filter(FELEM *filter, double factor, int tap_count,
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int phase_count, int scale, int filter_type,
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int kaiser_beta)
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static int build_filter(ResampleContext *c)
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{
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int ph, i;
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double x, y, w;
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double x, y, w, factor;
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double *tab;
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int tap_count = c->filter_length;
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int phase_count = 1 << c->phase_shift;
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const int center = (tap_count - 1) / 2;
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tab = av_malloc(tap_count * sizeof(*tab));
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@ -112,8 +112,7 @@ static int build_filter(FELEM *filter, double factor, int tap_count,
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return AVERROR(ENOMEM);
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/* if upsampling, only need to interpolate, no filter */
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if (factor > 1.0)
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factor = 1.0;
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factor = FFMIN(c->factor, 1.0);
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for (ph = 0; ph < phase_count; ph++) {
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double norm = 0;
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@ -121,7 +120,7 @@ static int build_filter(FELEM *filter, double factor, int tap_count,
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x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
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if (x == 0) y = 1.0;
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else y = sin(x) / x;
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switch (filter_type) {
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switch (c->filter_type) {
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case AV_RESAMPLE_FILTER_TYPE_CUBIC: {
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const float d = -0.5; //first order derivative = -0.5
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x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
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@ -137,23 +136,18 @@ static int build_filter(FELEM *filter, double factor, int tap_count,
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break;
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case AV_RESAMPLE_FILTER_TYPE_KAISER:
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w = 2.0 * x / (factor * tap_count * M_PI);
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y *= bessel(kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
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y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
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break;
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}
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tab[i] = y;
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norm += y;
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}
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/* normalize so that an uniform color remains the same */
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for (i = 0; i < tap_count; i++) {
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#ifdef CONFIG_RESAMPLE_FLT
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filter[ph * tap_count + i] = tab[i] / norm;
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#else
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filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm),
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FELEM_MIN, FELEM_MAX);
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#endif
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}
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for (i = 0; i < tap_count; i++)
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tab[i] = tab[i] / norm;
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c->set_filter(c->filter_bank, tab, ph, tap_count);
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}
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av_free(tab);
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@ -167,9 +161,12 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
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int in_rate = avr->in_sample_rate;
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double factor = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);
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int phase_count = 1 << avr->phase_shift;
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int felem_size;
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/* TODO: add support for s32 and float internal formats */
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if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) {
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if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
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avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&
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avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&
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avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {
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av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
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"resampling: %s\n",
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av_get_sample_fmt_name(avr->internal_sample_fmt));
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@ -188,17 +185,37 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
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c->filter_type = avr->filter_type;
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c->kaiser_beta = avr->kaiser_beta;
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c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * sizeof(FELEM));
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switch (avr->internal_sample_fmt) {
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case AV_SAMPLE_FMT_DBLP:
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c->resample_one = resample_one_dbl;
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c->set_filter = set_filter_dbl;
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break;
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case AV_SAMPLE_FMT_FLTP:
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c->resample_one = resample_one_flt;
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c->set_filter = set_filter_flt;
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break;
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case AV_SAMPLE_FMT_S32P:
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c->resample_one = resample_one_s32;
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c->set_filter = set_filter_s32;
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break;
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case AV_SAMPLE_FMT_S16P:
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c->resample_one = resample_one_s16;
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c->set_filter = set_filter_s16;
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break;
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}
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felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
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c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);
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if (!c->filter_bank)
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goto error;
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if (build_filter(c->filter_bank, factor, c->filter_length, phase_count,
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1 << FILTER_SHIFT, c->filter_type, c->kaiser_beta) < 0)
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if (build_filter(c) < 0)
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goto error;
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memcpy(&c->filter_bank[c->filter_length * phase_count + 1],
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c->filter_bank, (c->filter_length - 1) * sizeof(FELEM));
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c->filter_bank[c->filter_length * phase_count] = c->filter_bank[c->filter_length - 1];
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memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
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c->filter_bank, (c->filter_length - 1) * felem_size);
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memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],
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&c->filter_bank[(c->filter_length - 1) * felem_size], felem_size);
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c->compensation_distance = 0;
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if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
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@ -312,10 +329,10 @@ reinit_fail:
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return ret;
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}
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static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,
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static int resample(ResampleContext *c, void *dst, const void *src,
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int *consumed, int src_size, int dst_size, int update_ctx)
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{
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int dst_index, i;
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int dst_index;
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int index = c->index;
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int frac = c->frac;
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int dst_incr_frac = c->dst_incr % c->src_incr;
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@ -335,7 +352,7 @@ static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,
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if (dst) {
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for(dst_index = 0; dst_index < dst_size; dst_index++) {
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dst[dst_index] = src[index2 >> 32];
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c->resample_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0);
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index2 += incr;
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}
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} else {
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@ -346,42 +363,14 @@ static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,
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frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
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} else {
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for (dst_index = 0; dst_index < dst_size; dst_index++) {
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FELEM *filter = c->filter_bank +
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c->filter_length * (index & c->phase_mask);
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int sample_index = index >> c->phase_shift;
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if (!dst && (sample_index + c->filter_length > src_size ||
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-sample_index >= src_size))
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if (sample_index + c->filter_length > src_size ||
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-sample_index >= src_size)
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break;
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if (dst) {
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FELEM2 val = 0;
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if (sample_index < 0) {
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for (i = 0; i < c->filter_length; i++)
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val += src[FFABS(sample_index + i) % src_size] *
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(FELEM2)filter[i];
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} else if (sample_index + c->filter_length > src_size) {
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break;
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} else if (c->linear) {
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FELEM2 v2 = 0;
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for (i = 0; i < c->filter_length; i++) {
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val += src[abs(sample_index + i)] * (FELEM2)filter[i];
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v2 += src[abs(sample_index + i)] * (FELEM2)filter[i + c->filter_length];
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}
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val += (v2 - val) * (FELEML)frac / c->src_incr;
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} else {
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for (i = 0; i < c->filter_length; i++)
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val += src[sample_index + i] * (FELEM2)filter[i];
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}
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#ifdef CONFIG_RESAMPLE_FLT
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dst[dst_index] = av_clip_int16(lrintf(val));
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#else
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val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
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dst[dst_index] = av_clip_int16(val);
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#endif
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}
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if (dst)
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c->resample_one(c, 0, dst, dst_index, src, src_size, index, frac);
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frac += dst_incr_frac;
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index += dst_incr;
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@ -452,8 +441,8 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
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/* resample each channel plane */
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for (ch = 0; ch < c->buffer->channels; ch++) {
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out_samples = resample(c, (int16_t *)dst->data[ch],
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(const int16_t *)c->buffer->data[ch], consumed,
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out_samples = resample(c, (void *)dst->data[ch],
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(const void *)c->buffer->data[ch], consumed,
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c->buffer->nb_samples, dst->allocated_samples,
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ch + 1 == c->buffer->channels);
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}
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102
libavresample/resample_template.c
Normal file
102
libavresample/resample_template.c
Normal file
@ -0,0 +1,102 @@
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/*
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* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#if defined(CONFIG_RESAMPLE_DBL)
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#define SET_TYPE(func) func ## _dbl
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#define FELEM double
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#define FELEM2 double
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#define FELEML double
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#define OUT(d, v) d = v
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#define DBL_TO_FELEM(d, v) d = v
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#elif defined(CONFIG_RESAMPLE_FLT)
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#define SET_TYPE(func) func ## _flt
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#define FELEM float
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#define FELEM2 float
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#define FELEML float
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#define OUT(d, v) d = v
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#define DBL_TO_FELEM(d, v) d = v
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#elif defined(CONFIG_RESAMPLE_S32)
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#define SET_TYPE(func) func ## _s32
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#define FELEM int32_t
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#define FELEM2 int64_t
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#define FELEML int64_t
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#define OUT(d, v) d = av_clipl_int32((v + (1 << 29)) >> 30)
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#define DBL_TO_FELEM(d, v) d = av_clipl_int32(llrint(v * (1 << 30)));
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#else
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#define SET_TYPE(func) func ## _s16
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#define FELEM int16_t
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#define FELEM2 int32_t
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#define FELEML int64_t
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#define OUT(d, v) d = av_clip_int16((v + (1 << 14)) >> 15)
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#define DBL_TO_FELEM(d, v) d = av_clip_int16(lrint(v * (1 << 15)))
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#endif
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static void SET_TYPE(resample_one)(ResampleContext *c, int no_filter,
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void *dst0, int dst_index, const void *src0,
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int src_size, int index, int frac)
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{
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FELEM *dst = dst0;
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const FELEM *src = src0;
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if (no_filter) {
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dst[dst_index] = src[index];
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} else {
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int i;
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int sample_index = index >> c->phase_shift;
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FELEM2 val = 0;
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FELEM *filter = ((FELEM *)c->filter_bank) +
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c->filter_length * (index & c->phase_mask);
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if (sample_index < 0) {
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for (i = 0; i < c->filter_length; i++)
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val += src[FFABS(sample_index + i) % src_size] *
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(FELEM2)filter[i];
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} else if (c->linear) {
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FELEM2 v2 = 0;
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for (i = 0; i < c->filter_length; i++) {
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val += src[abs(sample_index + i)] * (FELEM2)filter[i];
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v2 += src[abs(sample_index + i)] * (FELEM2)filter[i + c->filter_length];
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}
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val += (v2 - val) * (FELEML)frac / c->src_incr;
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} else {
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for (i = 0; i < c->filter_length; i++)
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val += src[sample_index + i] * (FELEM2)filter[i];
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}
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OUT(dst[dst_index], val);
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}
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}
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static void SET_TYPE(set_filter)(void *filter0, double *tab, int phase,
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int tap_count)
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{
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int i;
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FELEM *filter = ((FELEM *)filter0) + phase * tap_count;
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for (i = 0; i < tap_count; i++) {
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DBL_TO_FELEM(filter[i], tab[i]);
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}
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}
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#undef SET_TYPE
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#undef FELEM
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#undef FELEM2
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#undef FELEML
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#undef OUT
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#undef DBL_TO_FELEM
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@ -64,10 +64,30 @@ int avresample_open(AVAudioResampleContext *avr)
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enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
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int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt),
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av_get_bytes_per_sample(out_fmt));
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if (avr->resample_needed || max_bps <= 2) {
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if (max_bps <= 2) {
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avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
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} else if (avr->mixing_needed) {
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avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
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} else {
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if (max_bps <= 4) {
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if (in_fmt == AV_SAMPLE_FMT_S32P ||
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out_fmt == AV_SAMPLE_FMT_S32P) {
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if (in_fmt == AV_SAMPLE_FMT_FLTP ||
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out_fmt == AV_SAMPLE_FMT_FLTP) {
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/* if one is s32 and the other is flt, use dbl */
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avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
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} else {
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/* if one is s32 and the other is s32, s16, or u8, use s32 */
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avr->internal_sample_fmt = AV_SAMPLE_FMT_S32P;
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}
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} else {
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/* if one is flt and the other is flt, s16 or u8, use flt */
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avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
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}
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} else {
|
||||
/* if either is dbl, use dbl */
|
||||
avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
|
||||
}
|
||||
}
|
||||
av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n",
|
||||
av_get_sample_fmt_name(avr->internal_sample_fmt));
|
||||
|
||||
Loading…
Reference in New Issue
Block a user