[FFmpeg-devel] [PATCH] avfilter: add arbitrary audio FIR filter
Paul B Mahol
onemda at gmail.com
Mon May 8 19:06:55 EEST 2017
On 5/8/17, Muhammad Faiz <mfcc64 at gmail.com> wrote:
> 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.
Could be later added, but for low latency, one uses NUPOLS or first
partition is done in time domain.
Using small N drastically reduces speed.
>
>
>> + 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.
Its needed. Various IRs gives different peak values.
The calculation is not perfect but it helps.
>
> Otherwise LGTM.
>
> Thank's.
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