diff options
Diffstat (limited to 'ffmpeg/libavresample/resample.c')
| -rw-r--r-- | ffmpeg/libavresample/resample.c | 469 |
1 files changed, 469 insertions, 0 deletions
diff --git a/ffmpeg/libavresample/resample.c b/ffmpeg/libavresample/resample.c new file mode 100644 index 0000000..69c9bab --- /dev/null +++ b/ffmpeg/libavresample/resample.c @@ -0,0 +1,469 @@ +/* + * 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; +} |