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

Muhammad Faiz mfcc64 at gmail.com
Wed May 10 10:12:23 EEST 2017


On Wed, May 10, 2017 at 1:55 AM, Paul B Mahol <git at videolan.org> wrote:
> ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Thu Jan 26 17:03:08 2017 +0100| [49bbfb9d13936ee8bb7fee9983ca3710dc683a2e] | committer: Paul B Mahol
>
> avfilter: add arbitrary audio FIR filter
>
> Signed-off-by: Paul B Mahol <onemda at gmail.com>
>
>> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=49bbfb9d13936ee8bb7fee9983ca3710dc683a2e
> ---
>
>  configure                      |   3 +
>  doc/filters.texi               |  43 ++++
>  libavfilter/Makefile           |   1 +
>  libavfilter/af_afir.c          | 535 +++++++++++++++++++++++++++++++++++++++++
>  libavfilter/af_afir.h          |  83 +++++++
>  libavfilter/allfilters.c       |   1 +
>  libavfilter/version.h          |   2 +-
>  libavfilter/x86/Makefile       |   2 +
>  libavfilter/x86/af_afir.asm    |  60 +++++
>  libavfilter/x86/af_afir_init.c |  35 +++
>  10 files changed, 764 insertions(+), 1 deletion(-)
>
> diff --git a/configure b/configure
> index e797567780..5ae5227868 100755
> --- a/configure
> +++ b/configure
> @@ -3083,6 +3083,8 @@ 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"
> @@ -6476,6 +6478,7 @@ enabled zlib && add_cppflags -DZLIB_CONST
>
>  # conditional library dependencies, in linking order
>  enabled afftfilt_filter     && prepend avfilter_deps "avcodec"
> +enabled afir_filter         && prepend avfilter_deps "avcodec"
>  enabled amovie_filter       && prepend avfilter_deps "avformat avcodec"
>  enabled aresample_filter    && prepend avfilter_deps "swresample"
>  enabled atempo_filter       && prepend avfilter_deps "avcodec"
> diff --git a/doc/filters.texi b/doc/filters.texi
> index 3739fbcc04..c54f5f2dcd 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -878,6 +878,49 @@ afftfilt="1-clip((b/nb)*b,0,1)"
>  @end example
>  @end itemize
>
> + at section afir
> +
> +Apply an arbitrary Frequency Impulse Response filter.
> +
> +This filter is designed for applying long FIR filters,
> +up to 30 seconds long.
> +
> +It can be used as component for digital crossover filters,
> +room equalization, cross talk cancellation, wavefield synthesis,
> +auralization, ambiophonics and ambisonics.
> +
> +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 item again
> +Enable applying gain measured from power of IR.
> + at end table
> +
> + at subsection Examples
> +
> + at itemize
> + at item
> +Apply reverb to stream using mono IR file as second input, complete command using ffmpeg:
> + at example
> +ffmpeg -i input.wav -i middle_tunnel_1way_mono.wav -lavfi afir output.wav
> + at end example
> + at end itemize
> +
>  @anchor{aformat}
>  @section aformat
>
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 0f990866e8..de5f992795 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 0000000000..d85c70710e
> --- /dev/null
> +++ b/libavfilter/af_afir.c
> @@ -0,0 +1,535 @@
> +/*
> + * 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/float_dsp.h"
> +#include "libavutil/opt.h"
> +#include "libavcodec/avfft.h"
> +
> +#include "audio.h"
> +#include "avfilter.h"
> +#include "formats.h"
> +#include "internal.h"
> +#include "af_afir.h"
> +
> +static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
> +{
> +    int n;
> +
> +    for (n = 0; n < len; n++) {
> +        const float cre = c[2 * n    ];
> +        const float cim = c[2 * n + 1];
> +        const float tre = t[2 * n    ];
> +        const float tim = t[2 * n + 1];
> +
> +        sum[2 * n    ] += tre * cre - tim * cim;
> +        sum[2 * n + 1] += tre * cim + tim * cre;
> +    }
> +
> +    sum[2 * n] += t[2 * n] * c[2 * n];
> +}
> +
> +static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
> +{
> +    AudioFIRContext *s = ctx->priv;
> +    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_size;
> +    memset(block, 0, sizeof(*block) * s->fft_length);
> +
> +    s->fdsp->vector_fmul_scalar(block + s->part_size, src, s->dry_gain, s->nb_samples);
> +    emms_c();
> +
> +    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->coeff_size;
> +        const FFTComplex *coeff = s->coeff[ch * !s->one2many] + coffset;
> +
> +        block = s->block[ch] + j * s->block_size;
> +        s->fcmul_add(sum, block, (const float *)coeff, s->part_size);
> +
> +        if (j == 0)
> +            j = s->nb_partitions;
> +        j--;
> +    }
> +
> +    sum[1] = sum[2 * s->part_size];
> +    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];
> +        s->fdsp->vector_fmul_scalar(ptr, dst, s->gain * s->wet_gain, out->nb_samples);
> +        emms_c();
> +    }
> +
> +    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]);
> +    if (s->nb_taps <= 0)
> +        return AVERROR(EINVAL);
> +
> +    for (n = 4; (1 << n) < s->nb_taps; n++);
> +    N = FFMIN(n, 16);
> +    s->ir_length = 1 << n;
> +    s->fft_length = (1 << (N + 1)) + 1;
> +    s->part_size = 1 << (N - 1);
> +    s->block_size = FFALIGN(s->fft_length, 32);
> +    s->coeff_size = FFALIGN(s->part_size + 1, 32);
> +    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_partitions * s->coeff_size, 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_size, 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];
> +
> +        power += s->fdsp->scalarproduct_float(time, time, s->nb_taps);
> +
> +        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->coeff_size;
> +            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);
> +            memcpy(block + boffset, time + toffset, size * sizeof(*block));
> +
> +            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 = s->again ? 1.f / sqrtf(power / ctx->inputs[1]->channels) : 1.f;
> +    av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
> +    av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->nb_partitions);
> +    av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->part_size);
> +    av_log(ctx, AV_LOG_DEBUG, "ir_length: %d\n", s->ir_length);
> +
> +    s->have_coeffs = 1;
> +
> +    return 0;
> +}
> +
> +static int read_ir(AVFilterLink *link, AVFrame *frame)
> +{
> +    AVFilterContext *ctx = link->dst;
> +    AudioFIRContext *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 (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;
> +    AudioFIRContext *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;
> +    AudioFIRContext *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) {
> +        if (s->need_padding) {
> +            AVFrame *silence = ff_get_audio_buffer(outlink, s->part_size);
> +
> +            if (!silence)
> +                return AVERROR(ENOMEM);
> +            av_audio_fifo_write(s->fifo[0], (void **)silence->extended_data,
> +                        silence->nb_samples);
> +            av_frame_free(&silence);
> +            s->need_padding = 0;
> +        }
> +
> +        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;
> +    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;
> +    AudioFIRContext *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->want_skip = 1;
> +    s->need_padding = 1;
> +    s->pts = AV_NOPTS_VALUE;
> +
> +    return 0;
> +}
> +
> +static av_cold void uninit(AVFilterContext *ctx)
> +{
> +    AudioFIRContext *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]);
> +
> +    av_freep(&s->fdsp);
> +}
> +
> +static av_cold int init(AVFilterContext *ctx)
> +{
> +    AudioFIRContext *s = ctx->priv;
> +
> +    s->fcmul_add = fcmul_add_c;
> +
> +    s->fdsp = avpriv_float_dsp_alloc(0);
> +    if (!s->fdsp)
> +        return AVERROR(ENOMEM);
> +
> +    if (ARCH_X86)
> +        ff_afir_init_x86(s);
> +
> +    return 0;
> +}
> +
> +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(AudioFIRContext, 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 },
> +    { "length", "set IR length",    OFFSET(length),   AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
> +    { "again",  "enable auto gain", OFFSET(again),    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(AudioFIRContext),
> +    .priv_class    = &afir_class,
> +    .query_formats = query_formats,
> +    .init          = init,
> +    .uninit        = uninit,
> +    .inputs        = afir_inputs,
> +    .outputs       = afir_outputs,
> +    .flags         = AVFILTER_FLAG_SLICE_THREADS,
> +};
> diff --git a/libavfilter/af_afir.h b/libavfilter/af_afir.h
> new file mode 100644
> index 0000000000..7414f5438e
> --- /dev/null
> +++ b/libavfilter/af_afir.h
> @@ -0,0 +1,83 @@
> +/*
> + * 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
> + */
> +
> +#ifndef  AVFILTER_AFIR_H
> +#define  AVFILTER_AFIR_H
> +
> +#include "libavutil/audio_fifo.h"
> +#include "libavutil/common.h"
> +#include "libavutil/float_dsp.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;
> +    int again;
> +
> +    float gain;
> +
> +    int eof_coeffs;
> +    int have_coeffs;
> +    int nb_coeffs;
> +    int nb_taps;
> +    int part_size;
> +    int part_index;
> +    int coeff_size;
> +    int block_size;
> +    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;
> +
> +    AVFloatDSPContext *fdsp;
> +    void (*fcmul_add)(float *sum, const float *t, const float *c,
> +                      ptrdiff_t len);
> +} AudioFIRContext;
> +
> +void ff_afir_init_x86(AudioFIRContext *s);
> +
> +#endif /* AVFILTER_AFIR_H */
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index 8fb87eb81e..555c44250b 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);
> diff --git a/libavfilter/version.h b/libavfilter/version.h
> index fb232c8e8a..ebfa644d1c 100644
> --- a/libavfilter/version.h
> +++ b/libavfilter/version.h
> @@ -30,7 +30,7 @@
>  #include "libavutil/version.h"
>
>  #define LIBAVFILTER_VERSION_MAJOR   6
> -#define LIBAVFILTER_VERSION_MINOR  88
> +#define LIBAVFILTER_VERSION_MINOR  89
>  #define LIBAVFILTER_VERSION_MICRO 100
>
>  #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
> diff --git a/libavfilter/x86/Makefile b/libavfilter/x86/Makefile
> index b6195f84c4..135e75f60f 100644
> --- a/libavfilter/x86/Makefile
> +++ b/libavfilter/x86/Makefile
> @@ -1,3 +1,4 @@
> +OBJS-$(CONFIG_AFIR_FILTER)                   += x86/af_afir_init.o
>  OBJS-$(CONFIG_BLEND_FILTER)                  += x86/vf_blend_init.o
>  OBJS-$(CONFIG_BWDIF_FILTER)                  += x86/vf_bwdif_init.o
>  OBJS-$(CONFIG_COLORSPACE_FILTER)             += x86/colorspacedsp_init.o
> @@ -23,6 +24,7 @@ OBJS-$(CONFIG_VOLUME_FILTER)                 += x86/af_volume_init.o
>  OBJS-$(CONFIG_W3FDIF_FILTER)                 += x86/vf_w3fdif_init.o
>  OBJS-$(CONFIG_YADIF_FILTER)                  += x86/vf_yadif_init.o
>
> +YASM-OBJS-$(CONFIG_AFIR_FILTER)              += x86/af_afir.o
>  YASM-OBJS-$(CONFIG_BLEND_FILTER)             += x86/vf_blend.o
>  YASM-OBJS-$(CONFIG_BWDIF_FILTER)             += x86/vf_bwdif.o
>  YASM-OBJS-$(CONFIG_COLORSPACE_FILTER)        += x86/colorspacedsp.o
> diff --git a/libavfilter/x86/af_afir.asm b/libavfilter/x86/af_afir.asm
> new file mode 100644
> index 0000000000..849d85e70f
> --- /dev/null
> +++ b/libavfilter/x86/af_afir.asm
> @@ -0,0 +1,60 @@
> +;*****************************************************************************
> +;* x86-optimized functions for afir filter
> +;* 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
> +;******************************************************************************
> +
> +%include "libavutil/x86/x86util.asm"
> +
> +SECTION .text
> +
> +;------------------------------------------------------------------------------
> +; void ff_fcmul_add(float *sum, const float *t, const float *c, int len)
> +;------------------------------------------------------------------------------
> +
> +INIT_XMM sse3
> +cglobal fcmul_add, 4,4,6, sum, t, c, len
> +    shl       lend, 3
> +    add       lend, mmsize*2
> +    add         tq, lenq
> +    add         cq, lenq
> +    add       sumq, lenq
> +    neg       lenq
> +ALIGN 16
> +.loop:
> +    movsldup  m0, [tq + lenq]
> +    movsldup  m3, [tq + lenq+mmsize]
> +    movaps    m1, [cq + lenq]
> +    movaps    m4, [cq + lenq+mmsize]
> +    mulps     m0, m1
> +    mulps     m3, m4
> +    shufps    m1, m1, 0xb1
> +    shufps    m4, m4, 0xb1
> +    movshdup  m2, [tq + lenq]
> +    movshdup  m5, [tq + lenq+mmsize]
> +    mulps     m2, m1
> +    mulps     m5, m4
> +    addsubps  m0, m2
> +    addsubps  m3, m5
> +    addps     m0, [sumq + lenq]
> +    addps     m3, [sumq + lenq+mmsize]
> +    movaps    [sumq + lenq], m0
> +    movaps    [sumq + lenq+mmsize], m3
> +    add       lenq, mmsize*2
> +    jl .loop
> +    REP_RET
> diff --git a/libavfilter/x86/af_afir_init.c b/libavfilter/x86/af_afir_init.c
> new file mode 100644
> index 0000000000..6a652b9b83
> --- /dev/null
> +++ b/libavfilter/x86/af_afir_init.c
> @@ -0,0 +1,35 @@
> +/*
> + * 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
> + */
> +
> +#include "config.h"
> +#include "libavutil/attributes.h"
> +#include "libavutil/cpu.h"
> +#include "libavutil/x86/cpu.h"
> +#include "libavfilter/af_afir.h"
> +
> +void ff_fcmul_add_sse3(float *sum, const float *t, const float *c,
> +                       ptrdiff_t len);
> +
> +av_cold void ff_afir_init_x86(AudioFIRContext *s)
> +{
> +    int cpu_flags = av_get_cpu_flags();
> +
> +    if (EXTERNAL_SSE3(cpu_flags)) {
> +        s->fcmul_add = ff_fcmul_add_sse3;
> +    }
> +}
>
> _______________________________________________
> ffmpeg-cvslog mailing list
> ffmpeg-cvslog at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-cvslog

segfault with this filtergraph

aevalsrc        = 'if(n, 0, 1)',
firequalizer    =
    delay       = 0.023:
    fixed       = off:
    wfunc       = nuttall:
    gain        = 'if(between(f, 1000, 5000), -INF, 0)',
atrim = end_sample = 2048 [ir];

aevalsrc='0.5*sin(3000*t*t)':d=10.3 [data];

[data][ir]
afir


More information about the ffmpeg-devel mailing list