diff options
Diffstat (limited to 'libavresample/resample.c')
| -rw-r--r-- | libavresample/resample.c | 163 |
1 files changed, 76 insertions, 87 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); } |