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- /*
- * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg 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.
- *
- * FFmpeg 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 FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
- #include "libavutil/common.h"
- #include "libavutil/libm.h"
- #include "libavutil/log.h"
- #include "internal.h"
- #include "resample.h"
- #include "audio_data.h"
-
-
- /* 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. */
- static double bessel(double x)
- {
- double v = 1;
- double lastv = 0;
- double t = 1;
- int i;
-
- x = x * x / 4;
- for (i = 1; v != lastv; i++) {
- lastv = v;
- t *= x / (i * i);
- v += t;
- }
- return v;
- }
-
- /* Build a polyphase filterbank. */
- static int build_filter(ResampleContext *c, double factor)
- {
- int ph, i;
- double x, y, w;
- 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));
- if (!tab)
- return AVERROR(ENOMEM);
-
- for (ph = 0; ph < phase_count; ph++) {
- double norm = 0;
- for (i = 0; i < tap_count; i++) {
- x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
- if (x == 0) y = 1.0;
- else y = sin(x) / x;
- 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);
- if (x < 1.0) y = 1 - 3 * x*x + 2 * x*x*x + d * ( -x*x + x*x*x);
- else y = d * (-4 + 8 * x - 5 * x*x + x*x*x);
- break;
- }
- case AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL:
- w = 2.0 * x / (factor * tap_count) + M_PI;
- y *= 0.3635819 - 0.4891775 * cos( w) +
- 0.1365995 * cos(2 * w) -
- 0.0106411 * cos(3 * w);
- break;
- case AV_RESAMPLE_FILTER_TYPE_KAISER:
- w = 2.0 * x / (factor * tap_count * M_PI);
- 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++)
- tab[i] = tab[i] / norm;
-
- c->set_filter(c->filter_bank, tab, ph, tap_count);
- }
-
- av_free(tab);
- return 0;
- }
-
- ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
- {
- ResampleContext *c;
- int out_rate = avr->out_sample_rate;
- 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;
-
- 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));
- return NULL;
- }
- c = av_mallocz(sizeof(*c));
- if (!c)
- return NULL;
-
- c->avr = avr;
- c->phase_shift = avr->phase_shift;
- c->phase_mask = phase_count - 1;
- c->linear = avr->linear_interp;
- c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);
- c->filter_type = avr->filter_type;
- c->kaiser_beta = avr->kaiser_beta;
-
- switch (avr->internal_sample_fmt) {
- case AV_SAMPLE_FMT_DBLP:
- c->resample_one = c->linear ? resample_linear_dbl : resample_one_dbl;
- c->resample_nearest = resample_nearest_dbl;
- c->set_filter = set_filter_dbl;
- break;
- case AV_SAMPLE_FMT_FLTP:
- c->resample_one = c->linear ? resample_linear_flt : resample_one_flt;
- c->resample_nearest = resample_nearest_flt;
- c->set_filter = set_filter_flt;
- break;
- case AV_SAMPLE_FMT_S32P:
- c->resample_one = c->linear ? resample_linear_s32 : resample_one_s32;
- c->resample_nearest = resample_nearest_s32;
- c->set_filter = set_filter_s32;
- break;
- case AV_SAMPLE_FMT_S16P:
- c->resample_one = c->linear ? resample_linear_s16 : resample_one_s16;
- c->resample_nearest = resample_nearest_s16;
- c->set_filter = set_filter_s16;
- break;
- }
-
- if (ARCH_AARCH64)
- ff_audio_resample_init_aarch64(c, avr->internal_sample_fmt);
- if (ARCH_ARM)
- ff_audio_resample_init_arm(c, avr->internal_sample_fmt);
-
- 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, factor) < 0)
- goto error;
-
- 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,
- in_rate * (int64_t)phase_count, INT32_MAX / 2))
- goto error;
- c->ideal_dst_incr = c->dst_incr;
-
- c->padding_size = (c->filter_length - 1) / 2;
- c->initial_padding_filled = 0;
- c->index = 0;
- c->frac = 0;
-
- /* allocate internal buffer */
- c->buffer = ff_audio_data_alloc(avr->resample_channels, c->padding_size,
- avr->internal_sample_fmt,
- "resample buffer");
- if (!c->buffer)
- goto error;
- c->buffer->nb_samples = c->padding_size;
- c->initial_padding_samples = c->padding_size;
-
- av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
- av_get_sample_fmt_name(avr->internal_sample_fmt),
- avr->in_sample_rate, avr->out_sample_rate);
-
- return c;
-
- error:
- ff_audio_data_free(&c->buffer);
- av_free(c->filter_bank);
- av_free(c);
- return NULL;
- }
-
- void ff_audio_resample_free(ResampleContext **c)
- {
- if (!*c)
- return;
- ff_audio_data_free(&(*c)->buffer);
- av_free((*c)->filter_bank);
- av_freep(c);
- }
-
- int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
- int compensation_distance)
- {
- ResampleContext *c;
-
- if (compensation_distance < 0)
- return AVERROR(EINVAL);
- if (!compensation_distance && sample_delta)
- return AVERROR(EINVAL);
-
- if (!avr->resample_needed) {
- av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n");
- return AVERROR(EINVAL);
- }
- c = avr->resample;
- c->compensation_distance = compensation_distance;
- if (compensation_distance) {
- c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr *
- (int64_t)sample_delta / compensation_distance;
- } else {
- c->dst_incr = c->ideal_dst_incr;
- }
-
- return 0;
- }
-
- static int resample(ResampleContext *c, void *dst, const void *src,
- int *consumed, int src_size, int dst_size, int update_ctx,
- int nearest_neighbour)
- {
- int dst_index;
- unsigned int index = c->index;
- int frac = c->frac;
- int dst_incr_frac = c->dst_incr % c->src_incr;
- int dst_incr = c->dst_incr / c->src_incr;
- int compensation_distance = c->compensation_distance;
-
- if (!dst != !src)
- return AVERROR(EINVAL);
-
- if (nearest_neighbour) {
- uint64_t index2 = ((uint64_t)index) << 32;
- int64_t incr = (1LL << 32) * c->dst_incr / c->src_incr;
- dst_size = FFMIN(dst_size,
- (src_size-1-index) * (int64_t)c->src_incr /
- c->dst_incr);
-
- if (dst) {
- for(dst_index = 0; dst_index < dst_size; dst_index++) {
- c->resample_nearest(dst, dst_index, src, index2 >> 32);
- index2 += incr;
- }
- } else {
- dst_index = dst_size;
- }
- index += dst_index * dst_incr;
- index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
- frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
- } else {
- for (dst_index = 0; dst_index < dst_size; dst_index++) {
- int sample_index = index >> c->phase_shift;
-
- if (sample_index + c->filter_length > src_size)
- break;
-
- if (dst)
- c->resample_one(c, dst, dst_index, src, index, frac);
-
- frac += dst_incr_frac;
- index += dst_incr;
- if (frac >= c->src_incr) {
- frac -= c->src_incr;
- index++;
- }
- if (dst_index + 1 == compensation_distance) {
- compensation_distance = 0;
- dst_incr_frac = c->ideal_dst_incr % c->src_incr;
- dst_incr = c->ideal_dst_incr / c->src_incr;
- }
- }
- }
- if (consumed)
- *consumed = index >> c->phase_shift;
-
- if (update_ctx) {
- index &= c->phase_mask;
-
- if (compensation_distance) {
- compensation_distance -= dst_index;
- if (compensation_distance <= 0)
- return AVERROR_BUG;
- }
- c->frac = frac;
- c->index = index;
- c->dst_incr = dst_incr_frac + c->src_incr*dst_incr;
- c->compensation_distance = compensation_distance;
- }
-
- return dst_index;
- }
-
- int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
- {
- int ch, in_samples, in_leftover, consumed = 0, out_samples = 0;
- int ret = AVERROR(EINVAL);
- int nearest_neighbour = (c->compensation_distance == 0 &&
- c->filter_length == 1 &&
- c->phase_shift == 0);
-
- in_samples = src ? src->nb_samples : 0;
- in_leftover = c->buffer->nb_samples;
-
- /* add input samples to the internal buffer */
- if (src) {
- ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);
- if (ret < 0)
- return ret;
- } else if (in_leftover <= c->final_padding_samples) {
- /* no remaining samples to flush */
- return 0;
- }
-
- if (!c->initial_padding_filled) {
- int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
- int i;
-
- if (src && c->buffer->nb_samples < 2 * c->padding_size)
- return 0;
-
- for (i = 0; i < c->padding_size; i++)
- for (ch = 0; ch < c->buffer->channels; ch++) {
- if (c->buffer->nb_samples > 2 * c->padding_size - i) {
- memcpy(c->buffer->data[ch] + bps * i,
- c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps);
- } else {
- memset(c->buffer->data[ch] + bps * i, 0, bps);
- }
- }
- c->initial_padding_filled = 1;
- }
-
- if (!src && !c->final_padding_filled) {
- int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
- int i;
-
- ret = ff_audio_data_realloc(c->buffer,
- FFMAX(in_samples, in_leftover) +
- c->padding_size);
- if (ret < 0) {
- av_log(c->avr, AV_LOG_ERROR, "Error reallocating resampling buffer\n");
- return AVERROR(ENOMEM);
- }
-
- for (i = 0; i < c->padding_size; i++)
- for (ch = 0; ch < c->buffer->channels; ch++) {
- if (in_leftover > i) {
- memcpy(c->buffer->data[ch] + bps * (in_leftover + i),
- c->buffer->data[ch] + bps * (in_leftover - i - 1),
- bps);
- } else {
- memset(c->buffer->data[ch] + bps * (in_leftover + i),
- 0, bps);
- }
- }
- c->buffer->nb_samples += c->padding_size;
- c->final_padding_samples = c->padding_size;
- c->final_padding_filled = 1;
- }
-
-
- /* calculate output size and reallocate output buffer if needed */
- /* TODO: try to calculate this without the dummy resample() run */
- if (!dst->read_only && dst->allow_realloc) {
- out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,
- INT_MAX, 0, nearest_neighbour);
- ret = ff_audio_data_realloc(dst, out_samples);
- if (ret < 0) {
- av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");
- return ret;
- }
- }
-
- /* resample each channel plane */
- for (ch = 0; ch < c->buffer->channels; ch++) {
- 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, nearest_neighbour);
- }
- if (out_samples < 0) {
- av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");
- return out_samples;
- }
-
- /* drain consumed samples from the internal buffer */
- ff_audio_data_drain(c->buffer, consumed);
- c->initial_padding_samples = FFMAX(c->initial_padding_samples - consumed, 0);
-
- av_log(c->avr, AV_LOG_TRACE, "resampled %d in + %d leftover to %d out + %d leftover\n",
- in_samples, in_leftover, out_samples, c->buffer->nb_samples);
-
- dst->nb_samples = out_samples;
- return 0;
- }
-
- int avresample_get_delay(AVAudioResampleContext *avr)
- {
- ResampleContext *c = avr->resample;
-
- if (!avr->resample_needed || !avr->resample)
- return 0;
-
- return FFMAX(c->buffer->nb_samples - c->padding_size, 0);
- }
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