[FFmpeg-devel] [PATCH] avfilter: add arbitrary audio FIR filter
Muhammad Faiz
mfcc64 at gmail.com
Mon May 8 18:07:27 EEST 2017
On Mon, May 8, 2017 at 6:59 PM, Paul B Mahol <onemda at gmail.com> wrote:
> Signed-off-by: Paul B Mahol <onemda at gmail.com>
> ---
> configure | 2 +
> doc/filters.texi | 23 ++
> libavfilter/Makefile | 1 +
> libavfilter/af_afir.c | 544 +++++++++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> 5 files changed, 571 insertions(+)
> create mode 100644 libavfilter/af_afir.c
>
> diff --git a/configure b/configure
> index 2e1786a..a46c375 100755
> --- a/configure
> +++ b/configure
> @@ -3081,6 +3081,8 @@ unix_protocol_select="network"
> # filters
> afftfilt_filter_deps="avcodec"
> afftfilt_filter_select="fft"
> +afir_filter_deps="avcodec"
> +afir_filter_select="fft"
> amovie_filter_deps="avcodec avformat"
> aresample_filter_deps="swresample"
> ass_filter_deps="libass"
> diff --git a/doc/filters.texi b/doc/filters.texi
> index f431274..0efce9a 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -878,6 +878,29 @@ afftfilt="1-clip((b/nb)*b,0,1)"
> @end example
> @end itemize
>
> + at section afir
> +
> +Apply an Arbitary Frequency Impulse Response filter.
> +
> +This filter uses second stream as FIR coefficients.
> +If second stream holds single channel, it will be used
> +for all input channels in first stream, otherwise
> +number of channels in second stream must be same as
> +number of channels in first stream.
> +
> +It accepts the following parameters:
> +
> + at table @option
> + at item dry
> +Set dry gain. This sets input gain.
> +
> + at item wet
> +Set wet gain. This sets final output gain.
> +
> + at item length
> +Set Impulse Response filter length. Default is 1, which means whole IR is processed.
> + at end table
> +
> @anchor{aformat}
> @section aformat
>
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 0f99086..de5f992 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -37,6 +37,7 @@ OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o
> OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
> OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
> OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o window_func.o
> +OBJS-$(CONFIG_AFIR_FILTER) += af_afir.o
> OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
> OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o
> OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o
> diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c
> new file mode 100644
> index 0000000..bc1b6a4
> --- /dev/null
> +++ b/libavfilter/af_afir.c
> @@ -0,0 +1,544 @@
> +/*
> + * Copyright (c) 2017 Paul B Mahol
> + *
> + * 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
> + */
> +
> +/**
> + * @file
> + * An arbitrary audio FIR filter
> + */
> +
> +#include "libavutil/audio_fifo.h"
> +#include "libavutil/common.h"
> +#include "libavutil/opt.h"
> +#include "libavcodec/avfft.h"
> +
> +#include "audio.h"
> +#include "avfilter.h"
> +#include "formats.h"
> +#include "internal.h"
> +
> +#define MAX_IR_DURATION 30
> +
> +typedef struct AudioFIRContext {
> + const AVClass *class;
> +
> + float wet_gain;
> + float dry_gain;
> + float length;
> +
> + float gain;
> +
> + int eof_coeffs;
> + int have_coeffs;
> + int nb_coeffs;
> + int nb_taps;
> + int part_size;
> + int part_index;
> + int block_length;
> + int nb_partitions;
> + int nb_channels;
> + int ir_length;
> + int fft_length;
> + int nb_coef_channels;
> + int one2many;
> + int nb_samples;
> + int want_skip;
> + int need_padding;
> +
> + RDFTContext **rdft, **irdft;
> + float **sum;
> + float **block;
> + FFTComplex **coeff;
> +
> + AVAudioFifo *fifo[2];
> + AVFrame *in[2];
> + AVFrame *buffer;
> + int64_t pts;
> + int index;
> +} AudioFIRContext;
> +
> +static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
> +{
> + AudioFIRContext *s = ctx->priv;
> + const FFTComplex *coeff = s->coeff[ch * !s->one2many];
> + const float *src = (const float *)s->in[0]->extended_data[ch];
> + int index1 = (s->index + 1) % 3;
> + int index2 = (s->index + 2) % 3;
> + float *sum = s->sum[ch];
> + AVFrame *out = arg;
> + float *block;
> + float *dst;
> + int n, i, j;
> +
> + memset(sum, 0, sizeof(*sum) * s->fft_length);
> + block = s->block[ch] + s->part_index * s->block_length;
> + memset(block, 0, sizeof(*block) * s->fft_length);
> + for (n = 0; n < s->nb_samples; n++) {
> + block[s->part_size + n] = src[n] * s->dry_gain;
> + }
> +
> + av_rdft_calc(s->rdft[ch], block);
> + block[2 * s->part_size] = block[1];
> + block[1] = 0;
> +
> + j = s->part_index;
> +
> + for (i = 0; i < s->nb_partitions; i++) {
> + const int coffset = i * (s->part_size + 1);
> +
> + block = s->block[ch] + j * s->block_length;
> + for (n = 0; n < s->part_size; n++) {
> + const float cre = coeff[coffset + n].re;
> + const float cim = coeff[coffset + n].im;
> + const float tre = block[2 * n ];
> + const float tim = block[2 * n + 1];
> +
> + sum[2 * n ] += tre * cre - tim * cim;
> + sum[2 * n + 1] += tre * cim + tim * cre;
> + }
> + sum[2 * n] += block[2 * n] * coeff[coffset + n].re;
> +
> + if (j == 0)
> + j = s->nb_partitions;
> + j--;
> + }
> +
> + sum[1] = sum[2 * n];
> + av_rdft_calc(s->irdft[ch], sum);
> +
> + dst = (float *)s->buffer->extended_data[ch] + index1 * s->part_size;
> + for (n = 0; n < s->part_size; n++) {
> + dst[n] += sum[n];
> + }
> +
> + dst = (float *)s->buffer->extended_data[ch] + index2 * s->part_size;
> +
> + memcpy(dst, sum + s->part_size, s->part_size * sizeof(*dst));
> +
> + dst = (float *)s->buffer->extended_data[ch] + s->index * s->part_size;
> +
> + if (out) {
> + float *ptr = (float *)out->extended_data[ch];
> + for (n = 0; n < out->nb_samples; n++) {
> + ptr[n] = dst[n] * s->gain * s->wet_gain;
> + }
> + }
> +
> + return 0;
> +}
> +
> +static int fir_frame(AudioFIRContext *s, AVFilterLink *outlink)
> +{
> + AVFilterContext *ctx = outlink->src;
> + AVFrame *out = NULL;
> + int ret;
> +
> + s->nb_samples = FFMIN(s->part_size, av_audio_fifo_size(s->fifo[0]));
> +
> + if (!s->want_skip) {
> + out = ff_get_audio_buffer(outlink, s->nb_samples);
> + if (!out)
> + return AVERROR(ENOMEM);
> + }
> +
> + s->in[0] = ff_get_audio_buffer(ctx->inputs[0], s->nb_samples);
> + if (!s->in[0]) {
> + av_frame_free(&out);
> + return AVERROR(ENOMEM);
> + }
> +
> + av_audio_fifo_peek(s->fifo[0], (void **)s->in[0]->extended_data, s->nb_samples);
> +
> + ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels);
> +
> + s->part_index = (s->part_index + 1) % s->nb_partitions;
> +
> + av_audio_fifo_drain(s->fifo[0], s->nb_samples);
> +
> + if (!s->want_skip) {
> + out->pts = s->pts;
> + if (s->pts != AV_NOPTS_VALUE)
> + s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
> + }
> +
> + s->index++;
> + if (s->index == 3)
> + s->index = 0;
> +
> + av_frame_free(&s->in[0]);
> +
> + if (s->want_skip == 1) {
> + s->want_skip = 0;
> + ret = 0;
> + } else {
> + ret = ff_filter_frame(outlink, out);
> + }
> +
> + return ret;
> +}
> +
> +static int convert_coeffs(AVFilterContext *ctx)
> +{
> + AudioFIRContext *s = ctx->priv;
> + int i, ch, n, N;
> + float power = 0;
> +
> + s->nb_taps = av_audio_fifo_size(s->fifo[1]);
> +
> + for (n = 4; (1 << n) < s->nb_taps; n++);
> + N = FFMIN(n, 16);
It is nice to allow user set maximum N e.g. for low latency app, user
can set low N with higher nb_partitions.
> + s->ir_length = 1 << n;
> + s->fft_length = (1 << (N + 1)) + 1;
> + s->part_size = 1 << (N - 1);
> + s->block_length = FFALIGN(s->fft_length, 16);
> + s->nb_partitions = (s->nb_taps + s->part_size - 1) / s->part_size;
> + s->nb_coeffs = s->ir_length + s->nb_partitions;
> +
> + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
> + s->sum[ch] = av_calloc(s->fft_length, sizeof(**s->sum));
> + if (!s->sum[ch])
> + return AVERROR(ENOMEM);
> + }
> +
> + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
> + s->coeff[ch] = av_calloc(s->nb_coeffs, sizeof(**s->coeff));
> + if (!s->coeff[ch])
> + return AVERROR(ENOMEM);
> + }
> +
> + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
> + s->block[ch] = av_calloc(s->nb_partitions * s->block_length, sizeof(**s->block));
> + if (!s->block[ch])
> + return AVERROR(ENOMEM);
> + }
> +
> + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
> + s->rdft[ch] = av_rdft_init(N, DFT_R2C);
> + s->irdft[ch] = av_rdft_init(N, IDFT_C2R);
> + if (!s->rdft[ch] || !s->irdft[ch])
> + return AVERROR(ENOMEM);
> + }
> +
> + s->in[1] = ff_get_audio_buffer(ctx->inputs[1], s->nb_taps);
> + if (!s->in[1])
> + return AVERROR(ENOMEM);
> +
> + s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size * 3);
> + if (!s->buffer)
> + return AVERROR(ENOMEM);
> +
> + av_audio_fifo_read(s->fifo[1], (void **)s->in[1]->extended_data, s->nb_taps);
> +
> + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
> + float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
> + float *block = s->block[ch];
> + FFTComplex *coeff = s->coeff[ch];
> +
> + for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
> + time[i] = 0;
> +
> + for (i = 0; i < s->nb_partitions; i++) {
> + const float scale = 1.f / s->part_size;
> + const int toffset = i * s->part_size;
> + const int coffset = i * (s->part_size + 1);
> + const int boffset = s->part_size;
> + const int remaining = s->nb_taps - (i * s->part_size);
> + const int size = remaining >= s->part_size ? s->part_size : remaining;
> +
> + memset(block, 0, sizeof(*block) * s->fft_length);
> + for (n = 0; n < size; n++) {
> + power += time[n + toffset] * time[n + toffset];
> + block[n + boffset] = time[n + toffset];
> + }
> +
> + av_rdft_calc(s->rdft[0], block);
> +
> + coeff[coffset].re = block[0] * scale;
> + coeff[coffset].im = 0;
> + for (n = 1; n < s->part_size; n++) {
> + coeff[coffset + n].re = block[2 * n] * scale;
> + coeff[coffset + n].im = block[2 * n + 1] * scale;
> + }
> + coeff[coffset + s->part_size].re = block[1] * scale;
> + coeff[coffset + s->part_size].im = 0;
> + }
> + }
> +
> + av_frame_free(&s->in[1]);
> + s->gain = 1.f / sqrtf(power);
I think s->gain is not required at all. The coeffs are already scaled by scale.
Otherwise LGTM.
Thank's.
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