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Diffstat (limited to 'ffmpeg1/libavresample/resample.c')
| -rw-r--r-- | ffmpeg1/libavresample/resample.c | 469 |
1 files changed, 0 insertions, 469 deletions
diff --git a/ffmpeg1/libavresample/resample.c b/ffmpeg1/libavresample/resample.c deleted file mode 100644 index 69c9bab..0000000 --- a/ffmpeg1/libavresample/resample.c +++ /dev/null @@ -1,469 +0,0 @@ -/* - * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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 - */ - -#include "libavutil/common.h" -#include "libavutil/libm.h" -#include "libavutil/log.h" -#include "internal.h" -#include "resample.h" -#include "audio_data.h" - -struct ResampleContext { - AVAudioResampleContext *avr; - AudioData *buffer; - uint8_t *filter_bank; - int filter_length; - int ideal_dst_incr; - int dst_incr; - int index; - int frac; - int src_incr; - int compensation_distance; - int phase_shift; - int phase_mask; - int linear; - 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. */ -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) -{ - int ph, i; - 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)); - if (!tab) - return AVERROR(ENOMEM); - - /* if upsampling, only need to interpolate, no filter */ - factor = FFMIN(c->factor, 1.0); - - 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->factor = factor; - 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 = 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) < 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->index = -phase_count * ((c->filter_length - 1) / 2); - c->frac = 0; - - /* allocate internal buffer */ - c->buffer = ff_audio_data_alloc(avr->resample_channels, 0, - avr->internal_sample_fmt, - "resample buffer"); - if (!c->buffer) - goto error; - - 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; - AudioData *fifo_buf = NULL; - int ret = 0; - - if (compensation_distance < 0) - return AVERROR(EINVAL); - if (!compensation_distance && sample_delta) - return AVERROR(EINVAL); - - if (!avr->resample_needed) { -#if FF_API_RESAMPLE_CLOSE_OPEN - /* if resampling was not enabled previously, re-initialize the - AVAudioResampleContext and force resampling */ - int fifo_samples; - int restore_matrix = 0; - double matrix[AVRESAMPLE_MAX_CHANNELS * AVRESAMPLE_MAX_CHANNELS] = { 0 }; - - /* buffer any remaining samples in the output FIFO before closing */ - fifo_samples = av_audio_fifo_size(avr->out_fifo); - if (fifo_samples > 0) { - fifo_buf = ff_audio_data_alloc(avr->out_channels, fifo_samples, - avr->out_sample_fmt, NULL); - if (!fifo_buf) - return AVERROR(EINVAL); - ret = ff_audio_data_read_from_fifo(avr->out_fifo, fifo_buf, - fifo_samples); - if (ret < 0) - goto reinit_fail; - } - /* save the channel mixing matrix */ - if (avr->am) { - ret = avresample_get_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS); - if (ret < 0) - goto reinit_fail; - restore_matrix = 1; - } - - /* close the AVAudioResampleContext */ - avresample_close(avr); - - avr->force_resampling = 1; - - /* restore the channel mixing matrix */ - if (restore_matrix) { - ret = avresample_set_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS); - if (ret < 0) - goto reinit_fail; - } - - /* re-open the AVAudioResampleContext */ - ret = avresample_open(avr); - if (ret < 0) - goto reinit_fail; - - /* restore buffered samples to the output FIFO */ - if (fifo_samples > 0) { - ret = ff_audio_data_add_to_fifo(avr->out_fifo, fifo_buf, 0, - fifo_samples); - if (ret < 0) - goto reinit_fail; - ff_audio_data_free(&fifo_buf); - } -#else - av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n"); - return AVERROR(EINVAL); -#endif - } - 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; - -reinit_fail: - ff_audio_data_free(&fifo_buf); - return ret; -} - -static int resample(ResampleContext *c, void *dst, const void *src, - int *consumed, int src_size, int dst_size, int update_ctx) -{ - int dst_index; - 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 (compensation_distance == 0 && c->filter_length == 1 && - c->phase_shift == 0) { - int64_t index2 = ((int64_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_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0); - 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 || - -sample_index >= src_size) - break; - - if (dst) - c->resample_one(c, 0, dst, dst_index, src, src_size, 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 = FFMAX(index, 0) >> c->phase_shift; - - if (update_ctx) { - if (index >= 0) - 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); - - 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) { - /* no remaining samples to flush */ - return 0; - } else { - /* TODO: pad buffer to flush completely */ - } - - /* 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); - 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); - } - 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); - - av_dlog(c->avr, "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) -{ - if (!avr->resample_needed || !avr->resample) - return 0; - - return avr->resample->buffer->nb_samples; -} |