| /* |
| * 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; |
| 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 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); |
| } |