| Message ID | 20170508115948.3597-1-onemda@gmail.com |
|---|---|
| State | Superseded |
| Headers | show |
On Mon, May 8, 2017 at 6:59 PM, Paul B Mahol <onemda@gmail.com> wrote: > Signed-off-by: Paul B Mahol <onemda@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.
On 5/8/17, Muhammad Faiz <mfcc64@gmail.com> wrote: > On Mon, May 8, 2017 at 6:59 PM, Paul B Mahol <onemda@gmail.com> wrote: >> Signed-off-by: Paul B Mahol <onemda@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. Could be later added, but for low latency, one uses NUPOLS or first partition is done in time domain. Using small N drastically reduces speed. > > >> + 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. Its needed. Various IRs gives different peak values. The calculation is not perfect but it helps. > > Otherwise LGTM. > > Thank's. > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel >
On Mon, May 8, 2017 at 11:06 PM, Paul B Mahol <onemda@gmail.com> wrote: > On 5/8/17, Muhammad Faiz <mfcc64@gmail.com> wrote: >> On Mon, May 8, 2017 at 6:59 PM, Paul B Mahol <onemda@gmail.com> wrote: >>> Signed-off-by: Paul B Mahol <onemda@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. > > Could be later added, but for low latency, one uses NUPOLS or first > partition is done in time domain. > Using small N drastically reduces speed. > >> >> >>> + 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. > > Its needed. Various IRs gives different peak values. > The calculation is not perfect but it helps. OK. So, make it optional again (e.g using auto option). Thank's.
On 5/8/17, Muhammad Faiz <mfcc64@gmail.com> wrote: > On Mon, May 8, 2017 at 11:06 PM, Paul B Mahol <onemda@gmail.com> wrote: >> On 5/8/17, Muhammad Faiz <mfcc64@gmail.com> wrote: >>> On Mon, May 8, 2017 at 6:59 PM, Paul B Mahol <onemda@gmail.com> wrote: >>>> Signed-off-by: Paul B Mahol <onemda@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. >> >> Could be later added, but for low latency, one uses NUPOLS or first >> partition is done in time domain. >> Using small N drastically reduces speed. >> >>> >>> >>>> + 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. >> >> Its needed. Various IRs gives different peak values. >> The calculation is not perfect but it helps. > > OK. So, make it optional again (e.g using auto option). I don't see need for it, without it its always worse. I updated patch with added SIMD for trivial complex multiplication. It is faster (not much) then what gcc generates.
On Tue, May 9, 2017 at 5:03 AM, Paul B Mahol <onemda@gmail.com> wrote: > On 5/8/17, Muhammad Faiz <mfcc64@gmail.com> wrote: >> On Mon, May 8, 2017 at 11:06 PM, Paul B Mahol <onemda@gmail.com> wrote: >>> On 5/8/17, Muhammad Faiz <mfcc64@gmail.com> wrote: >>>> On Mon, May 8, 2017 at 6:59 PM, Paul B Mahol <onemda@gmail.com> wrote: >>>>> Signed-off-by: Paul B Mahol <onemda@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. >>> >>> Could be later added, but for low latency, one uses NUPOLS or first >>> partition is done in time domain. >>> Using small N drastically reduces speed. >>> >>>> >>>> >>>>> + 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); sqrtf(power/ctx->inputs[1]->channels) >>>> >>>> I think s->gain is not required at all. The coeffs are already scaled by >>>> scale. >>> >>> Its needed. Various IRs gives different peak values. >>> The calculation is not perfect but it helps. >> >> OK. So, make it optional again (e.g using auto option). > > I don't see need for it, without it its always worse. Is it bad to preserve the actual frequency response. I mean here s->gain = 1.0f; not s->gain = 1.0f / s->part_size; > > I updated patch with added SIMD for trivial complex multiplication. > > It is faster (not much) then what gcc generates.
On 5/9/17, Muhammad Faiz <mfcc64@gmail.com> wrote: > On Tue, May 9, 2017 at 5:03 AM, Paul B Mahol <onemda@gmail.com> wrote: >> On 5/8/17, Muhammad Faiz <mfcc64@gmail.com> wrote: >>> On Mon, May 8, 2017 at 11:06 PM, Paul B Mahol <onemda@gmail.com> wrote: >>>> On 5/8/17, Muhammad Faiz <mfcc64@gmail.com> wrote: >>>>> On Mon, May 8, 2017 at 6:59 PM, Paul B Mahol <onemda@gmail.com> wrote: >>>>>> Signed-off-by: Paul B Mahol <onemda@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. >>>> >>>> Could be later added, but for low latency, one uses NUPOLS or first >>>> partition is done in time domain. >>>> Using small N drastically reduces speed. >>>> >>>>> >>>>> >>>>>> + 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); > > sqrtf(power/ctx->inputs[1]->channels) done. > > >>>>> >>>>> I think s->gain is not required at all. The coeffs are already scaled >>>>> by >>>>> scale. >>>> >>>> Its needed. Various IRs gives different peak values. >>>> The calculation is not perfect but it helps. >>> >>> OK. So, make it optional again (e.g using auto option). >> >> I don't see need for it, without it its always worse. > > Is it bad to preserve the actual frequency response. > I mean here s->gain = 1.0f; > not s->gain = 1.0f / s->part_size; Added back. Gonna apply soon.
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); + 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); + 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]); +} + +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 }, + { 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, + .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);
Signed-off-by: Paul B Mahol <onemda@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