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. > + 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. Otherwise LGTM. Thank's. _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel