diff options
| -rw-r--r-- | libavresample/resample.c | 163 | ||||
| -rw-r--r-- | libavresample/resample_template.c | 102 | ||||
| -rw-r--r-- | libavresample/utils.c | 22 |
3 files changed, 199 insertions, 88 deletions
diff --git a/libavresample/resample.c b/libavresample/resample.c index 7316e4ec60..1c3d13ae0a 100644 --- a/libavresample/resample.c +++ b/libavresample/resample.c @@ -24,34 +24,10 @@ #include "internal.h" #include "audio_data.h" -#ifdef CONFIG_RESAMPLE_FLT -/* float template */ -#define FILTER_SHIFT 0 -#define FELEM float -#define FELEM2 float -#define FELEML float -#elifdef CONFIG_RESAMPLE_S32 -/* s32 template */ -#define FILTER_SHIFT 30 -#define FELEM int32_t -#define FELEM2 int64_t -#define FELEML int64_t -#define FELEM_MAX INT32_MAX -#define FELEM_MIN INT32_MIN -#else -/* s16 template */ -#define FILTER_SHIFT 15 -#define FELEM int16_t -#define FELEM2 int32_t -#define FELEML int64_t -#define FELEM_MAX INT16_MAX -#define FELEM_MIN INT16_MIN -#endif - struct ResampleContext { AVAudioResampleContext *avr; AudioData *buffer; - FELEM *filter_bank; + uint8_t *filter_bank; int filter_length; int ideal_dst_incr; int dst_incr; @@ -65,8 +41,32 @@ struct ResampleContext { enum AVResampleFilterType filter_type; int kaiser_beta; double factor; + void (*set_filter)(void *filter, double *tab, int phase, int tap_count); + void (*resample_one)(struct ResampleContext *c, int no_filter, void *dst0, + int dst_index, const void *src0, int src_size, + int index, int frac); }; + +/* double template */ +#define CONFIG_RESAMPLE_DBL +#include "resample_template.c" +#undef CONFIG_RESAMPLE_DBL + +/* float template */ +#define CONFIG_RESAMPLE_FLT +#include "resample_template.c" +#undef CONFIG_RESAMPLE_FLT + +/* s32 template */ +#define CONFIG_RESAMPLE_S32 +#include "resample_template.c" +#undef CONFIG_RESAMPLE_S32 + +/* s16 template */ +#include "resample_template.c" + + /** * 0th order modified bessel function of the first kind. */ @@ -98,13 +98,13 @@ static double bessel(double x) * @param kaiser_beta kaiser window beta * @return 0 on success, negative AVERROR code on failure */ -static int build_filter(FELEM *filter, double factor, int tap_count, - int phase_count, int scale, int filter_type, - int kaiser_beta) +static int build_filter(ResampleContext *c) { int ph, i; - double x, y, w; + double x, y, w, factor; double *tab; + int tap_count = c->filter_length; + int phase_count = 1 << c->phase_shift; const int center = (tap_count - 1) / 2; tab = av_malloc(tap_count * sizeof(*tab)); @@ -112,8 +112,7 @@ static int build_filter(FELEM *filter, double factor, int tap_count, return AVERROR(ENOMEM); /* if upsampling, only need to interpolate, no filter */ - if (factor > 1.0) - factor = 1.0; + factor = FFMIN(c->factor, 1.0); for (ph = 0; ph < phase_count; ph++) { double norm = 0; @@ -121,7 +120,7 @@ static int build_filter(FELEM *filter, double factor, int tap_count, x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; if (x == 0) y = 1.0; else y = sin(x) / x; - switch (filter_type) { + switch (c->filter_type) { case AV_RESAMPLE_FILTER_TYPE_CUBIC: { const float d = -0.5; //first order derivative = -0.5 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); @@ -137,23 +136,18 @@ static int build_filter(FELEM *filter, double factor, int tap_count, break; case AV_RESAMPLE_FILTER_TYPE_KAISER: w = 2.0 * x / (factor * tap_count * M_PI); - y *= bessel(kaiser_beta * sqrt(FFMAX(1 - w * w, 0))); + y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0))); break; } tab[i] = y; norm += y; } - /* normalize so that an uniform color remains the same */ - for (i = 0; i < tap_count; i++) { -#ifdef CONFIG_RESAMPLE_FLT - filter[ph * tap_count + i] = tab[i] / norm; -#else - filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), - FELEM_MIN, FELEM_MAX); -#endif - } + for (i = 0; i < tap_count; i++) + tab[i] = tab[i] / norm; + + c->set_filter(c->filter_bank, tab, ph, tap_count); } av_free(tab); @@ -167,9 +161,12 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr) int in_rate = avr->in_sample_rate; double factor = FFMIN(out_rate * avr->cutoff / in_rate, 1.0); int phase_count = 1 << avr->phase_shift; + int felem_size; - /* TODO: add support for s32 and float internal formats */ - if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) { + if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P && + avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P && + avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP && + avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) { av_log(avr, AV_LOG_ERROR, "Unsupported internal format for " "resampling: %s\n", av_get_sample_fmt_name(avr->internal_sample_fmt)); @@ -188,17 +185,37 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr) c->filter_type = avr->filter_type; c->kaiser_beta = avr->kaiser_beta; - c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * sizeof(FELEM)); + switch (avr->internal_sample_fmt) { + case AV_SAMPLE_FMT_DBLP: + c->resample_one = resample_one_dbl; + c->set_filter = set_filter_dbl; + break; + case AV_SAMPLE_FMT_FLTP: + c->resample_one = resample_one_flt; + c->set_filter = set_filter_flt; + break; + case AV_SAMPLE_FMT_S32P: + c->resample_one = resample_one_s32; + c->set_filter = set_filter_s32; + break; + case AV_SAMPLE_FMT_S16P: + c->resample_one = resample_one_s16; + c->set_filter = set_filter_s16; + break; + } + + felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt); + c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size); if (!c->filter_bank) goto error; - if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, - 1 << FILTER_SHIFT, c->filter_type, c->kaiser_beta) < 0) + if (build_filter(c) < 0) goto error; - memcpy(&c->filter_bank[c->filter_length * phase_count + 1], - c->filter_bank, (c->filter_length - 1) * sizeof(FELEM)); - c->filter_bank[c->filter_length * phase_count] = c->filter_bank[c->filter_length - 1]; + memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size], + c->filter_bank, (c->filter_length - 1) * felem_size); + memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size], + &c->filter_bank[(c->filter_length - 1) * felem_size], felem_size); c->compensation_distance = 0; if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate, @@ -312,10 +329,10 @@ reinit_fail: return ret; } -static int resample(ResampleContext *c, int16_t *dst, const int16_t *src, +static int resample(ResampleContext *c, void *dst, const void *src, int *consumed, int src_size, int dst_size, int update_ctx) { - int dst_index, i; + int dst_index; int index = c->index; int frac = c->frac; int dst_incr_frac = c->dst_incr % c->src_incr; @@ -335,7 +352,7 @@ static int resample(ResampleContext *c, int16_t *dst, const int16_t *src, if (dst) { for(dst_index = 0; dst_index < dst_size; dst_index++) { - dst[dst_index] = src[index2 >> 32]; + c->resample_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0); index2 += incr; } } else { @@ -346,42 +363,14 @@ static int resample(ResampleContext *c, int16_t *dst, const int16_t *src, frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr; } else { for (dst_index = 0; dst_index < dst_size; dst_index++) { - FELEM *filter = c->filter_bank + - c->filter_length * (index & c->phase_mask); int sample_index = index >> c->phase_shift; - if (!dst && (sample_index + c->filter_length > src_size || - -sample_index >= src_size)) + if (sample_index + c->filter_length > src_size || + -sample_index >= src_size) break; - if (dst) { - FELEM2 val = 0; - - if (sample_index < 0) { - for (i = 0; i < c->filter_length; i++) - val += src[FFABS(sample_index + i) % src_size] * - (FELEM2)filter[i]; - } else if (sample_index + c->filter_length > src_size) { - break; - } else if (c->linear) { - FELEM2 v2 = 0; - for (i = 0; i < c->filter_length; i++) { - val += src[abs(sample_index + i)] * (FELEM2)filter[i]; - v2 += src[abs(sample_index + i)] * (FELEM2)filter[i + c->filter_length]; - } - val += (v2 - val) * (FELEML)frac / c->src_incr; - } else { - for (i = 0; i < c->filter_length; i++) - val += src[sample_index + i] * (FELEM2)filter[i]; - } - -#ifdef CONFIG_RESAMPLE_FLT - dst[dst_index] = av_clip_int16(lrintf(val)); -#else - val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT; - dst[dst_index] = av_clip_int16(val); -#endif - } + if (dst) + c->resample_one(c, 0, dst, dst_index, src, src_size, index, frac); frac += dst_incr_frac; index += dst_incr; @@ -452,8 +441,8 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src, /* resample each channel plane */ for (ch = 0; ch < c->buffer->channels; ch++) { - out_samples = resample(c, (int16_t *)dst->data[ch], - (const int16_t *)c->buffer->data[ch], consumed, + out_samples = resample(c, (void *)dst->data[ch], + (const void *)c->buffer->data[ch], consumed, c->buffer->nb_samples, dst->allocated_samples, ch + 1 == c->buffer->channels); } diff --git a/libavresample/resample_template.c b/libavresample/resample_template.c new file mode 100644 index 0000000000..5b0fbecdf5 --- /dev/null +++ b/libavresample/resample_template.c @@ -0,0 +1,102 @@ +/* + * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> + * + * This file is part of Libav. + * + * Libav is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * Libav 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 + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#if defined(CONFIG_RESAMPLE_DBL) +#define SET_TYPE(func) func ## _dbl +#define FELEM double +#define FELEM2 double +#define FELEML double +#define OUT(d, v) d = v +#define DBL_TO_FELEM(d, v) d = v +#elif defined(CONFIG_RESAMPLE_FLT) +#define SET_TYPE(func) func ## _flt +#define FELEM float +#define FELEM2 float +#define FELEML float +#define OUT(d, v) d = v +#define DBL_TO_FELEM(d, v) d = v +#elif defined(CONFIG_RESAMPLE_S32) +#define SET_TYPE(func) func ## _s32 +#define FELEM int32_t +#define FELEM2 int64_t +#define FELEML int64_t +#define OUT(d, v) d = av_clipl_int32((v + (1 << 29)) >> 30) +#define DBL_TO_FELEM(d, v) d = av_clipl_int32(llrint(v * (1 << 30))); +#else +#define SET_TYPE(func) func ## _s16 +#define FELEM int16_t +#define FELEM2 int32_t +#define FELEML int64_t +#define OUT(d, v) d = av_clip_int16((v + (1 << 14)) >> 15) +#define DBL_TO_FELEM(d, v) d = av_clip_int16(lrint(v * (1 << 15))) +#endif + +static void SET_TYPE(resample_one)(ResampleContext *c, int no_filter, + void *dst0, int dst_index, const void *src0, + int src_size, int index, int frac) +{ + FELEM *dst = dst0; + const FELEM *src = src0; + + if (no_filter) { + dst[dst_index] = src[index]; + } else { + int i; + int sample_index = index >> c->phase_shift; + FELEM2 val = 0; + FELEM *filter = ((FELEM *)c->filter_bank) + + c->filter_length * (index & c->phase_mask); + + if (sample_index < 0) { + for (i = 0; i < c->filter_length; i++) + val += src[FFABS(sample_index + i) % src_size] * + (FELEM2)filter[i]; + } else if (c->linear) { + FELEM2 v2 = 0; + for (i = 0; i < c->filter_length; i++) { + val += src[abs(sample_index + i)] * (FELEM2)filter[i]; + v2 += src[abs(sample_index + i)] * (FELEM2)filter[i + c->filter_length]; + } + val += (v2 - val) * (FELEML)frac / c->src_incr; + } else { + for (i = 0; i < c->filter_length; i++) + val += src[sample_index + i] * (FELEM2)filter[i]; + } + + OUT(dst[dst_index], val); + } +} + +static void SET_TYPE(set_filter)(void *filter0, double *tab, int phase, + int tap_count) +{ + int i; + FELEM *filter = ((FELEM *)filter0) + phase * tap_count; + for (i = 0; i < tap_count; i++) { + DBL_TO_FELEM(filter[i], tab[i]); + } +} + +#undef SET_TYPE +#undef FELEM +#undef FELEM2 +#undef FELEML +#undef OUT +#undef DBL_TO_FELEM diff --git a/libavresample/utils.c b/libavresample/utils.c index ac1c36e17c..1aca566282 100644 --- a/libavresample/utils.c +++ b/libavresample/utils.c @@ -64,10 +64,30 @@ int avresample_open(AVAudioResampleContext *avr) enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt); int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt), av_get_bytes_per_sample(out_fmt)); - if (avr->resample_needed || max_bps <= 2) { + if (max_bps <= 2) { avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P; } else if (avr->mixing_needed) { avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP; + } else { + if (max_bps <= 4) { + if (in_fmt == AV_SAMPLE_FMT_S32P || + out_fmt == AV_SAMPLE_FMT_S32P) { + if (in_fmt == AV_SAMPLE_FMT_FLTP || + out_fmt == AV_SAMPLE_FMT_FLTP) { + /* if one is s32 and the other is flt, use dbl */ + avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP; + } else { + /* if one is s32 and the other is s32, s16, or u8, use s32 */ + avr->internal_sample_fmt = AV_SAMPLE_FMT_S32P; + } + } else { + /* if one is flt and the other is flt, s16 or u8, use flt */ + avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP; + } + } 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)); |