[FFmpeg-devel] [PATCH] avfilter: add arbitrary audio FIR filter

Muhammad Faiz mfcc64 at gmail.com
Sat May 6 03:21:33 EEST 2017


On Sat, May 6, 2017 at 2:30 AM, Paul B Mahol <onemda at gmail.com> wrote:
> Signed-off-by: Paul B Mahol <onemda at gmail.com>
> ---
>  configure                |   2 +
>  doc/filters.texi         |  10 +
>  libavfilter/Makefile     |   1 +
>  libavfilter/af_afir.c    | 484 +++++++++++++++++++++++++++++++++++++++++++++++
>  libavfilter/allfilters.c |   1 +
>  5 files changed, 498 insertions(+)
>  create mode 100644 libavfilter/af_afir.c
>
> diff --git a/configure b/configure
> index b3cb5b0..0d83c6a 100755
> --- a/configure
> +++ b/configure
> @@ -3078,6 +3078,8 @@ 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 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
>
> + at 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

Seems that you forgot to update the documentation.

>
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 66c36e4..c797eb5 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-$(CONFIG_AFIR_FILTER)                   += af_afir.o
>  OBJS-$(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..9411c9b
> --- /dev/null
> +++ b/libavfilter/af_afir.c
> @@ -0,0 +1,484 @@
> +/*
> + * 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 20
> +
> +typedef struct FIRContext {
> +    const AVClass *class;
> +
> +    float wet_gain;
> +    float dry_gain;
> +    int auto_gain;
> +
> +    float gain;
> +
> +    int eof_coeffs;
> +    int have_coeffs;
> +    int nb_coeffs;
> +    int nb_taps;
> +    int part_size;
> +    int nb_partitions;
> +    int fft_length;
> +    int nb_channels;
> +    int nb_coef_channels;
> +    int one2many;
> +    int nb_samples;
> +
> +    RDFTContext **rdft, **irdft;
> +    float **sum;
> +    float **block;
> +    FFTComplex **coeff;
> +
> +    AVAudioFifo *fifo[2];
> +    AVFrame *in[2];
> +    AVFrame *buffer;
> +    int64_t pts;
> +} FIRContext;
> +
> +static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
> +{
> +    FIRContext *s = ctx->priv;
> +    AVFrame *out = arg;
> +    const FFTComplex *coeff = s->coeff[ch * !s->one2many];
> +    const float *src = (const float *)s->in[0]->extended_data[ch];
> +    float *dst = (float *)out->extended_data[ch];
> +    float *buf = (float *)s->buffer->extended_data[ch];
> +    float *sum = s->sum[ch];
> +    float *block = s->block[ch];
> +    int n, i;
> +
> +    memset(sum, 0, sizeof(*sum) * 2 * (s->part_size + 1));
> +    memset(block, 0, sizeof(*block) * 2 * (s->part_size + 1));
> +    for (n = 0; n < s->nb_samples; n++) {
> +        block[n] = src[n] * s->dry_gain;
> +    }
> +
> +    av_rdft_calc(s->rdft[ch], block);
> +    block[s->part_size / 2] = block[1];
> +    block[1] = 0;
> +
> +    for (i = 0; i < s->nb_partitions; i++) {
> +        const int coffset = i * (s->part_size + 1);
> +
> +        for (n = 0; n <= s->part_size; n++) {
> +            const float re = block[2 * n    ];
> +            const float im = block[2 * n + 1];
> +            const float cre = coeff[coffset + n].re;
> +            const float cim = coeff[coffset + n].im;
> +
> +            sum[2 * n    ] += re * cre - im * cim;
> +            sum[2 * n + 1] += re * cim + im * cre;
> +        }
> +    }
> +
> +    sum[1] = sum[n];
> +    av_rdft_calc(s->irdft[ch], sum);
> +
> +    for (n = 0; n < out->nb_samples; n++) {
> +        float sample;
> +
> +        sample = sum[out->nb_samples + n];
> +        dst[n] = sample * s->wet_gain * s->gain;
> +        buf[n] = sum[n];
> +    }
> +
> +    return 0;
> +}
> +
> +static int fir_frame(FIRContext *s, AVFilterLink *outlink)
> +{
> +    AVFilterContext *ctx = outlink->src;
> +    AVFrame *out;
> +
> +    s->nb_samples = FFMIN(s->part_size, av_audio_fifo_size(s->fifo[0]));
> +
> +    out = ff_get_audio_buffer(outlink, s->nb_samples < s->part_size / 2 ? s->nb_samples : s->part_size / 2);
> +    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);
> +
> +    av_audio_fifo_drain(s->fifo[0], out->nb_samples);
> +
> +    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);
> +
> +    av_frame_free(&s->in[0]);
> +
> +    return ff_filter_frame(outlink, out);
> +}
> +
> +static int convert_coeffs(AVFilterContext *ctx)
> +{
> +    FIRContext *s = ctx->priv;
> +    int max_nb_taps, i, ch, n, N;
> +    float power = 0;
> +
> +    s->nb_taps = av_audio_fifo_size(s->fifo[1]);
> +    max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate;
> +    if (s->nb_taps > max_nb_taps) {
> +        av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", s->nb_taps, max_nb_taps);
> +        return AVERROR(EINVAL);
> +    }
> +
> +    for (n = 1; (1 << n) < s->nb_taps; n++);
> +    N = FFMIN(n, 16);
> +    s->fft_length = 1 << n;
> +    s->part_size = 1 << (N - 1);
> +    s->nb_partitions = (s->fft_length + s->part_size - 1) / s->part_size;
> +    s->nb_coeffs = s->fft_length + s->nb_partitions;
> +
> +    for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
> +        s->sum[ch] = av_calloc(2 * (s->part_size + 1), 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(2 * (s->part_size + 1), sizeof(**s->block));
> +        if (!s->block[ch])
> +            return AVERROR(ENOMEM);
> +    }
> +
> +    s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size);
> +    if (!s->buffer)
> +        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);
> +
> +    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++) {
> +        const float *re = (const float *)s->in[1]->extended_data[!s->one2many * ch];
> +        float *block = s->block[ch];
> +        FFTComplex *coeff = s->coeff[ch];
> +
> +        for (i = 0; i < s->nb_partitions; i++) {
> +            const int offset = i * s->part_size;
> +            const int coffset = i * (s->part_size + 1);
> +            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) * (2 * (s->part_size + 1)));
> +            for (n = 0; n < size; n++) {
> +                block[n] = re[n + offset];
> +                power += block[n] * block[n];
> +            }
> +
> +            av_rdft_calc(s->rdft[0], block);
> +
> +            coeff[coffset].re = block[0];
> +            coeff[coffset].im = 0;
> +            for (n = 1; n < s->part_size; n++) {
> +                coeff[coffset + n].re = block[2 * n];
> +                coeff[coffset + n].im = block[2 * n + 1];
> +            }
> +            coeff[coffset + n].re = block[1];
> +            coeff[coffset + n].im = 0;
> +        }
> +    }
> +    power /= ctx->inputs[1]->channels;
> +
> +    av_frame_free(&s->in[1]);
> +    s->gain = (1.f / (1 << N)) / (s->auto_gain ? sqrtf(power) : sqrtf(s->part_size));
> +    av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", 1 << N);
> +    av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
> +    av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", s->fft_length);
> +    av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->nb_partitions);
> +
> +    s->have_coeffs = 1;
> +
> +    return 0;
> +}
> +
> +static int read_ir(AVFilterLink *link, AVFrame *frame)
> +{
> +    AVFilterContext *ctx = link->dst;
> +    FIRContext *s = ctx->priv;
> +    int nb_taps, max_nb_taps;
> +
> +    av_audio_fifo_write(s->fifo[1], (void **)frame->extended_data,
> +                        frame->nb_samples);
> +    av_frame_free(&frame);
> +
> +    nb_taps = av_audio_fifo_size(s->fifo[1]);
> +    max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate;
> +    if (s->nb_taps > max_nb_taps) {
> +        av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
> +        return AVERROR(EINVAL);
> +    }
> +
> +    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);
> +    if (s->pts == AV_NOPTS_VALUE)
> +        s->pts = frame->pts;
> +
> +    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->part_size) {
> +            ret = fir_frame(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_frame(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->sum = av_calloc(outlink->channels, sizeof(*s->sum));
> +    s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff));
> +    s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block));
> +    s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft));
> +    s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft));
> +    if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft)
> +        return AVERROR(ENOMEM);
> +
> +    s->nb_channels = outlink->channels;
> +    s->nb_coef_channels = ctx->inputs[1]->channels;
> +    s->pts = AV_NOPTS_VALUE;
> +
> +    return 0;
> +}
> +
> +static av_cold void uninit(AVFilterContext *ctx)
> +{
> +    FIRContext *s = ctx->priv;
> +    int ch;
> +
> +    if (s->sum) {
> +        for (ch = 0; ch < s->nb_channels; ch++) {
> +            av_freep(&s->sum[ch]);
> +        }
> +    }
> +    av_freep(&s->sum);
> +
> +    if (s->coeff) {
> +        for (ch = 0; ch < s->nb_coef_channels; ch++) {
> +            av_freep(&s->coeff[ch]);
> +        }
> +    }
> +    av_freep(&s->coeff);
> +
> +    if (s->block) {
> +        for (ch = 0; ch < s->nb_channels; ch++) {
> +            av_freep(&s->block[ch]);
> +        }
> +    }
> +    av_freep(&s->block);
> +
> +    if (s->rdft) {
> +        for (ch = 0; ch < s->nb_channels; ch++) {
> +            av_rdft_end(s->rdft[ch]);
> +        }
> +    }
> +    av_freep(&s->rdft);
> +
> +    if (s->irdft) {
> +        for (ch = 0; ch < s->nb_channels; ch++) {
> +            av_rdft_end(s->irdft[ch]);
> +        }
> +    }
> +    av_freep(&s->irdft);
> +
> +    av_frame_free(&s->in[0]);
> +    av_frame_free(&s->in[1]);
> +    av_frame_free(&s->buffer);
> +
> +    av_audio_fifo_free(s->fifo[0]);
> +    av_audio_fifo_free(s->fifo[1]);
> +}
> +
> +static const AVFilterPad afir_inputs[] = {
> +    {
> +        .name           = "main",
> +        .type           = AVMEDIA_TYPE_AUDIO,
> +        .filter_frame   = filter_frame,
> +    },{
> +        .name           = "ir",
> +        .type           = AVMEDIA_TYPE_AUDIO,
> +        .filter_frame   = read_ir,
> +    },
> +    { NULL }
> +};
> +
> +static const AVFilterPad afir_outputs[] = {
> +    {
> +        .name          = "default",
> +        .type          = AVMEDIA_TYPE_AUDIO,
> +        .config_props  = config_output,
> +        .request_frame = request_frame,
> +    },
> +    { NULL }
> +};
> +
> +#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
> +#define OFFSET(x) offsetof(FIRContext, x)
> +
> +static const AVOption afir_options[] = {
> +    { "dry",  "set dry gain",     OFFSET(dry_gain),  AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
> +    { "wet",  "set wet gain",     OFFSET(wet_gain),  AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
> +    { "auto", "enable auto-gain", OFFSET(auto_gain), AV_OPT_TYPE_BOOL,  {.i64=1}, 0, 1, AF },
> +    { NULL }
> +};
> +
> +AVFILTER_DEFINE_CLASS(afir);
> +
> +AVFilter ff_af_afir = {
> +    .name          = "afir",
> +    .description   = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
> +    .priv_size     = sizeof(FIRContext),
> +    .priv_class    = &afir_class,
> +    .query_formats = query_formats,
> +    .uninit        = uninit,
> +    .inputs        = afir_inputs,
> +    .outputs       = afir_outputs,
> +    .flags         = AVFILTER_FLAG_SLICE_THREADS,
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index 8fb87eb..555c442 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(AFIR,           afir,           af);
>      REGISTER_FILTER(AFORMAT,        aformat,        af);
>      REGISTER_FILTER(AGATE,          agate,          af);
>      REGISTER_FILTER(AINTERLEAVE,    ainterleave,    af);

Seems that the partitioned convolution code here doesn't work. I can't
help here.
IMHO, you should stuck to traditional convolution code.

Thank's


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