On Mon, May 1, 2017 at 3:30 PM, Paul B Mahol <one...@gmail.com> wrote:
> Signed-off-by: Paul B Mahol <one...@gmail.com>
> ---
> configure | 2 +
> doc/filters.texi | 10 ++
> libavfilter/Makefile | 1 +
> libavfilter/af_afirfilter.c | 409
> ++++++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> 5 files changed, 423 insertions(+)
> create mode 100644 libavfilter/af_afirfilter.c
>
> diff --git a/configure b/configure
> index b3cb5b0..7fc7af4 100755
> --- a/configure
> +++ b/configure
> @@ -3078,6 +3078,8 @@ unix_protocol_select="network"
> # filters
> afftfilt_filter_deps="avcodec"
> afftfilt_filter_select="fft"
> +afirfilter_filter_deps="avcodec"
> +afirfilter_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 119e747..ea343d1 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -878,6 +878,16 @@ afftfilt="1-clip((b/nb)*b,0,1)"
> @end example
> @end itemize
>
> +@section afirfilter
> +
> +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.
> +
> @anchor{aformat}
> @section aformat
>
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 66c36e4..1a0f24b 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -38,6 +38,7 @@ 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-$(CONFIG_AFIRFILTER_FILTER) += af_afirfilter.o
> OBJS-$(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_afirfilter.c b/libavfilter/af_afirfilter.c
> new file mode 100644
> index 0000000..ef2488a
> --- /dev/null
> +++ b/libavfilter/af_afirfilter.c
> @@ -0,0 +1,409 @@
> +/*
> + * 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/avassert.h"
> +#include "libavutil/channel_layout.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"
> +
> +typedef struct FIRContext {
> + const AVClass *class;
> +
> + int n;
> + int eof_coeffs;
> + int have_coeffs;
> + int nb_taps;
> + int fft_length;
> + int nb_channels;
> + int one2many;
> +
> + FFTContext *fft, *ifft;
> + FFTComplex **fft_data;
> + FFTComplex **fft_coef;
Probably you may use rdft for performance reason.
> +
> + AVAudioFifo *fifo[2];
> + AVFrame *in[2];
> + AVFrame *buffer;
> + int64_t pts;
> + int hop_size;
> + int start, end;
> +} FIRContext;
> +
> +static int fir_filter(FIRContext *s, AVFilterLink *outlink)
> +{
> + AVFilterContext *ctx = outlink->src;
> + int start = s->start, end = s->end;
> + int ret = 0, n, ch, j, k;
> + int nb_samples;
> + AVFrame *out;
> +
> + nb_samples = FFMIN(s->fft_length, av_audio_fifo_size(s->fifo[0]));
> +
> + s->in[0] = ff_get_audio_buffer(ctx->inputs[0], nb_samples);
> + if (!s->in[0])
> + return AVERROR(ENOMEM);
> +
> + av_audio_fifo_peek(s->fifo[0], (void **)s->in[0]->extended_data,
> nb_samples);
> +
> + for (ch = 0; ch < outlink->channels; ch++) {
> + const float *src = (float *)s->in[0]->extended_data[ch];
> + float *buf = (float *)s->buffer->extended_data[ch];
> + FFTComplex *fft_data = s->fft_data[ch];
> + FFTComplex *fft_coef = s->fft_coef[ch];
> +
> + memset(fft_data, 0, sizeof(*fft_data) * s->fft_length);
> + for (n = 0; n < nb_samples; n++) {
> + fft_data[n].re = src[n];
> + fft_data[n].im = 0;
> + }
> +
> + av_fft_permute(s->fft, fft_data);
> + av_fft_calc(s->fft, fft_data);
> +
> + fft_data[0].re *= fft_coef[0].re;
> + fft_data[0].im *= fft_coef[0].im;
> + for (n = 1; n < s->fft_length; n++) {
> + const float re = fft_data[n].re;
> + const float im = fft_data[n].im;
> +
> + fft_data[n].re = re * fft_coef[n].re - im * fft_coef[n].im;
> + fft_data[n].im = re * fft_coef[n].im + im * fft_coef[n].re;
> + }
> +
> + av_fft_permute(s->ifft, fft_data);
> + av_fft_calc(s->ifft, fft_data);
> +
> + start = s->start;
> + end = s->end;
> + k = end;
> +
> + for (n = 0, j = start; j < k && n < s->fft_length; n++, j++) {
> + buf[j] = fft_data[n].re;
> + }
> +
> + for (; n < s->fft_length; n++, j++) {
> + buf[j] = fft_data[n].re;
> + }
> +
> + start += s->hop_size;
> + end = j;
> + }
> +
> + s->start = start;
> + s->end = end;
> +
> + if (start >= nb_samples) {
> + float *dst, *buf;
> +
> + start -= nb_samples;
> + end -= nb_samples;
> +
> + s->start = start;
> + s->end = end;
> +
> + out = ff_get_audio_buffer(outlink, nb_samples);
> + if (!out)
> + return AVERROR(ENOMEM);
> +
> + out->pts = s->pts;
> + s->pts += nb_samples;
Is pts handled correctly here? Seem it is not derived from input pts.
> +
> + for (ch = 0; ch < s->nb_channels; ch++) {
> + dst = (float *)out->extended_data[ch];
> + buf = (float *)s->buffer->extended_data[ch];
> +
> + for (n = 0; n < nb_samples; n++)
> + dst[n] = buf[n];
> + memmove(buf, buf + nb_samples, nb_samples * 4);
> + }
> +
> + ret = ff_filter_frame(outlink, out);
> + }
> +
> + av_audio_fifo_drain(s->fifo[0], FFMIN(nb_samples, s->hop_size));
> + av_frame_free(&s->in[0]);
> +
> + return ret;
> +}
> +
> +static int convert_coeffs(AVFilterContext *ctx)
> +{
> + FIRContext *s = ctx->priv;
> + int ch, n;
> +
> + s->nb_taps = av_audio_fifo_size(s->fifo[1]);
> + if (s->nb_taps > 32768) {
> + av_log(ctx, AV_LOG_ERROR, "Too big number of taps: %d > 32768.\n",
> s->nb_taps);
> + return AVERROR(EINVAL);
> + }
> +
> + for (n = 1; (1 << n) < s->nb_taps; n++);
> + s->n = n + 2;
> + s->fft_length = 1 << s->n;
> +
> + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
> + s->fft_data[ch] = av_calloc(s->fft_length, sizeof(**s->fft_data));
> + if (!s->fft_data[ch])
> + return AVERROR(ENOMEM);
> + }
> +
> + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
> + s->fft_coef[ch] = av_calloc(s->fft_length, sizeof(**s->fft_coef));
> + if (!s->fft_coef[ch])
> + return AVERROR(ENOMEM);
> + }
> +
> + s->hop_size = s->fft_length - s->nb_taps + 1;
> + if (s->hop_size <= 0) {
> + av_log(ctx, AV_LOG_ERROR, "Too big number of taps.\n");
> + return AVERROR(EINVAL);
> + }
> +
> + s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->fft_length * 2);
> + if (!s->buffer)
> + return AVERROR(ENOMEM);
> +
> + s->fft = av_fft_init(s->n, 0);
> + s->ifft = av_fft_init(s->n, 1);
> + if (!s->fft || !s->ifft)
> + return AVERROR(ENOMEM);
> +
> + s->in[1] = ff_get_audio_buffer(ctx->inputs[1], s->nb_taps);
> + if (!s->in[1])
> + 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++) {
> + FFTComplex *fft_coef = s->fft_coef[ch];
> + const float *re = (const float
> *)s->in[1]->extended_data[!s->one2many * ch];
> + const float scale = 1.f / s->fft_length;
> + const int offset = (s->fft_length - s->nb_taps);
> +
> + memset(fft_coef, 0, sizeof(*fft_coef) * s->fft_length);
> + for (n = 0; n < s->nb_taps; n++) {
> + fft_coef[n + offset].re = re[n] * scale;
> + }
> + av_fft_permute(s->fft, fft_coef);
> + av_fft_calc(s->fft, fft_coef);
> + }
> +
> + av_frame_free(&s->in[1]);
> + s->have_coeffs = 1;
> +
> + return 0;
> +}
> +
> +static int read_coeffs(AVFilterLink *link, AVFrame *frame)
> +{
> + AVFilterContext *ctx = link->dst;
> + FIRContext *s = ctx->priv;
> +
> + av_audio_fifo_write(s->fifo[1], (void **)frame->extended_data,
> + frame->nb_samples);
> + av_frame_free(&frame);
> +
> + return 0;
> +}
> +
> +static int filter_frame(AVFilterLink *link, AVFrame *frame)
> +{
> + AVFilterContext *ctx = link->dst;
> + FIRContext *s = ctx->priv;
> + AVFilterLink *outlink = ctx->outputs[0];
> + int ret = 0;
> +
> + av_audio_fifo_write(s->fifo[0], (void **)frame->extended_data,
> + frame->nb_samples);
> + av_frame_free(&frame);
> +
> + if (!s->have_coeffs && s->eof_coeffs) {
> + ret = convert_coeffs(ctx);
> + if (ret < 0)
> + return ret;
> + }
> +
> + if (s->have_coeffs) {
> + while (av_audio_fifo_size(s->fifo[0]) >= s->fft_length) {
> + ret = fir_filter(s, outlink);
> + if (ret < 0)
> + break;
> + }
> + }
> + return ret;
> +}
> +
> +static int request_frame(AVFilterLink *outlink)
> +{
> + AVFilterContext *ctx = outlink->src;
> + FIRContext *s = ctx->priv;
> + int ret;
> +
> + if (!s->eof_coeffs) {
> + ret = ff_request_frame(ctx->inputs[1]);
> + if (ret == AVERROR_EOF) {
> + s->eof_coeffs = 1;
> + ret = 0;
> + }
> + return ret;
> + }
> + ret = ff_request_frame(ctx->inputs[0]);
> + if (ret == AVERROR_EOF && s->have_coeffs) {
> + while (av_audio_fifo_size(s->fifo[0]) > 0) {
> + ret = fir_filter(s, outlink);
> + if (ret < 0)
> + return ret;
> + }
> + ret = AVERROR_EOF;
> + }
> + return ret;
> +}
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> + AVFilterFormats *formats;
> + AVFilterChannelLayouts *layouts = NULL;
> + static const enum AVSampleFormat sample_fmts[] = {
> + AV_SAMPLE_FMT_FLTP,
> + AV_SAMPLE_FMT_NONE
> + };
> + int ret, i;
> +
> + layouts = ff_all_channel_counts();
> + if ((ret = ff_channel_layouts_ref(layouts,
> &ctx->outputs[0]->in_channel_layouts)) < 0)
> + return ret;
> +
> + for (i = 0; i < 2; i++) {
> + layouts = ff_all_channel_counts();
> + if ((ret = ff_channel_layouts_ref(layouts,
> &ctx->inputs[i]->out_channel_layouts)) < 0)
> + return ret;
> + }
> +
> + formats = ff_make_format_list(sample_fmts);
> + if ((ret = ff_set_common_formats(ctx, formats)) < 0)
> + return ret;
> +
> + formats = ff_all_samplerates();
> + return ff_set_common_samplerates(ctx, formats);
> +}
> +
> +static int config_output(AVFilterLink *outlink)
> +{
> + AVFilterContext *ctx = outlink->src;
> + FIRContext *s = ctx->priv;
> +
> + if (ctx->inputs[0]->channels != ctx->inputs[1]->channels &&
> + ctx->inputs[1]->channels != 1) {
> + av_log(ctx, AV_LOG_ERROR,
> + "Second input must have same number of channels as first
> input or "
> + "exactly 1 channel.\n");
> + return AVERROR(EINVAL);
> + }
> +
> + s->one2many = ctx->inputs[1]->channels == 1;
> + outlink->sample_rate = ctx->inputs[0]->sample_rate;
> + outlink->time_base = ctx->inputs[0]->time_base;
> + outlink->channel_layout = ctx->inputs[0]->channel_layout;
> + outlink->channels = ctx->inputs[0]->channels;
> +
> + s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format,
> ctx->inputs[0]->channels, 1024);
> + s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format,
> ctx->inputs[1]->channels, 1024);
> + if (!s->fifo[0] || !s->fifo[1])
> + return AVERROR(ENOMEM);
> +
> + s->fft_data = av_calloc(outlink->channels, sizeof(*s->fft_data));
> + s->fft_coef = av_calloc(ctx->inputs[1]->channels, sizeof(*s->fft_coef));
> + if (!s->fft_data || !s->fft_coef)
> + return AVERROR(ENOMEM);
> + s->nb_channels = outlink->channels;
> +
> + return 0;
> +}
> +
> +static av_cold void uninit(AVFilterContext *ctx)
> +{
> + FIRContext *s = ctx->priv;
> + int ch;
> +
> + for (ch = 0; ch < s->nb_channels; ch++) {
> + if (s->fft_data)
> + av_freep(&s->fft_data[ch]);
> + }
> + av_freep(&s->fft_data);
> +
> + for (ch = 0; ch < s->nb_channels; ch++) {
> + if (s->fft_coef)
> + av_freep(&s->fft_coef[ch]);
> + }
> + av_freep(&s->fft_coef);
> +
> + av_fft_end(s->fft);
> + av_fft_end(s->ifft);
> +
> + av_frame_free(&s->in[0]);
> + av_frame_free(&s->in[1]);
> +
> + av_audio_fifo_free(s->fifo[0]);
> + av_audio_fifo_free(s->fifo[1]);
> +}
> +
> +static const AVFilterPad afirfilter_inputs[] = {
> + {
> + .name = "main",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .filter_frame = filter_frame,
> + },{
> + .name = "coefficients",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .filter_frame = read_coeffs,
> + },
> + { NULL }
> +};
> +
> +static const AVFilterPad afirfilter_outputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .config_props = config_output,
> + .request_frame = request_frame,
> + },
> + { NULL }
> +};
> +
> +AVFilter ff_af_afirfilter = {
> + .name = "afirfilter",
> + .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response
> filter with supplied coefficients in 2nd stream."),
> + .priv_size = sizeof(FIRContext),
> + .query_formats = query_formats,
> + .uninit = uninit,
> + .inputs = afirfilter_inputs,
> + .outputs = afirfilter_outputs,
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index 8fb87eb..8bfe1ae 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -50,6 +50,7 @@ static void register_all(void)
> REGISTER_FILTER(AEVAL, aeval, af);
> REGISTER_FILTER(AFADE, afade, af);
> REGISTER_FILTER(AFFTFILT, afftfilt, af);
> + REGISTER_FILTER(AFIRFILTER, afirfilter, af);
> REGISTER_FILTER(AFORMAT, aformat, af);
> REGISTER_FILTER(AGATE, agate, af);
> REGISTER_FILTER(AINTERLEAVE, ainterleave, af);
> --
> 2.9.3
>
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
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