On Wed, May 10, 2017 at 1:55 AM, Paul B Mahol <g...@videolan.org> wrote:
> ffmpeg | branch: master | Paul B Mahol <one...@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 <one...@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
>
> +@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:
> +
> +@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.
> +
> +@item again
> +Enable applying gain measured from power of IR.
> +@end table
> +
> +@subsection Examples
> +
> +@itemize
> +@item
> +Apply reverb to stream using mono IR file as second input, complete command
> using ffmpeg:
> +@example
> +ffmpeg -i input.wav -i middle_tunnel_1way_mono.wav -lavfi afir output.wav
> +@end example
> +@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;
> + }
> +}
>
> _______________________________________________
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> ffmpeg-cvs...@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
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