On Mon, May 8, 2017 at 11:06 PM, Paul B Mahol <one...@gmail.com> wrote:
> On 5/8/17, Muhammad Faiz <mfc...@gmail.com> wrote:
>> On Mon, May 8, 2017 at 6:59 PM, Paul B Mahol <one...@gmail.com> wrote:
>>> Signed-off-by: Paul B Mahol <one...@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
>>>
>>> +@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:
>>> +
>>> +@table @option
>>> +@item dry
>>> +Set dry gain. This sets input gain.
>>> +
>>> +@item wet
>>> +Set wet gain. This sets final output gain.
>>> +
>>> +@item length
>>> +Set Impulse Response filter length. Default is 1, which means whole IR is
>>> processed.
>>> +@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.
OK. So, make it optional again (e.g using auto option).
Thank's.
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