[FFmpeg-devel] [PATCH] avfilter: add arbitrary audio FIR filter
Paul B Mahol
onemda at gmail.com
Tue May 9 13:16:29 EEST 2017
On 5/9/17, Muhammad Faiz <mfcc64 at gmail.com> wrote:
> On Tue, May 9, 2017 at 5:03 AM, Paul B Mahol <onemda at gmail.com> wrote:
>> On 5/8/17, Muhammad Faiz <mfcc64 at gmail.com> wrote:
>>> On Mon, May 8, 2017 at 11:06 PM, Paul B Mahol <onemda at gmail.com> wrote:
>>>> 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);
>
> sqrtf(power/ctx->inputs[1]->channels)
done.
>
>
>>>>>
>>>>> 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.
>>>
>>> OK. So, make it optional again (e.g using auto option).
>>
>> I don't see need for it, without it its always worse.
>
> Is it bad to preserve the actual frequency response.
> I mean here s->gain = 1.0f;
> not s->gain = 1.0f / s->part_size;
Added back.
Gonna apply soon.
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